[cisco-voip] upgrading CUCM from 6.1.2 to 6.1.3a
Robert Knapp
robert.knapp at spanlink.com
Thu Jun 25 12:22:39 EDT 2009
Having "issues" with upgrade.
the CLI displays error: syslogd: /var/log/active/platform/log/authneticatefile.log : No such file or directory
I am upgrading via the a browser:
the log says such items as
exception ontaperestore
and
raise Exception, ("exception caught during ontape restore [%s]" % msg)|<LVL::Debug>
06/25/2009 12:14:38 CCMInstall|Internal Error, File:instMain.c:1403, Function: handlePhase(), Failed to exec
I have reboot the pub and sub, any suggestions?
Thanks,
Robert Knapp
________________________________________
From: cisco-voip-bounces at puck.nether.net [cisco-voip-bounces at puck.nether.net] On Behalf Of cisco-voip-request at puck.nether.net [cisco-voip-request at puck.nether.net]
Sent: Thursday, June 25, 2009 12:00 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 68, Issue 23
Send cisco-voip mailing list submissions to
cisco-voip at puck.nether.net
To subscribe or unsubscribe via the World Wide Web, visit
https://puck.nether.net/mailman/listinfo/cisco-voip
or, via email, send a message with subject or body 'help' to
cisco-voip-request at puck.nether.net
You can reach the person managing the list at
cisco-voip-owner at puck.nether.net
When replying, please edit your Subject line so it is more specific
than "Re: Contents of cisco-voip digest..."
Today's Topics:
1. Has Anyone Run A Primary Unity Server on a Physical Server
and the Failover on a Virtual Server? (Miller, Steve)
2. create a software CFB that got deleted (Tim Frazee)
3. Call Manager 5.1.1a install file? (Erick Bergquist)
4. Re: Has Anyone Run A Primary Unity Server on a Physical
Server and the Failover on a Virtual Server? (Paul)
5. Re: Call Manager 5.1.1a install file? (Jason Burns)
6. Re: Has Anyone Run A Primary Unity Server on aPhysical Server
and the Failover on a Virtual Server? (Jason Aarons (US))
7. Re: FW: CME web access disable (Nick Matthews)
8. Re: FW: CME web access disable (Ahmed Elnagar)
9. PRI Protocol NAT1, NAT2, custom? (Jeff Ruttman)
10. Re: PRI Protocol NAT1, NAT2, custom? (Matt Slaga (US))
11. Re: PRI Protocol NAT1, NAT2, custom? (Jeff Ruttman)
12. Re: PRI Protocol NAT1, NAT2, custom? (Matt Slaga (US))
13. Re: create a software CFB that got deleted (Peter Slow)
14. Re: destination-pattern "T" question (Dew Swen)
15. Does Unity Connection 7.1.2a support SIP RFC 2833 for DTMF
(Jason Aarons (US))
16. Re: Multicast MoH Delay (Tony Underwood)
17. Re: Multicast MoH Delay (Daniel)
18. SIP Route Pattern (Jake Doe)
19. TAPS and UCCX 5.0.2 ? (Jason Aarons (US))
20. Re: Does Unity Connection 7.1.2a support SIP RFC 2833 for
DTMF (Adam Frankel)
21. Re: Call Manager 5.1.1a install file? (Erick Bergquist)
22. T.37 Fax Redundancy (ciscozest)
23. Re: Multicast MoH Delay (Tony Underwood)
24. One User Cannot Be Dialed By Name (Miller, Steve)
25. Re: One User Cannot Be Dialed By Name (Cristobal Priego)
26. Re: TAPS and UCCX 5.0.2 ? (Dustin S Fowler)
27. Re: Does Unity Connection 7.1.2a support SIP RFC 2833 for
DTMF (Mark Holloway)
28. Show Saved Enterprise Data in CAD (ROJAS, Mario)
29. Re: destination-pattern "T" question (Mehmet Turunc)
30. Re: Show Saved Enterprise Data in CAD (Beck, Christopher)
31. Slow to connect calls (Jeff Ruttman)
32. VoicemailQueuing (Voice Noob)
33. Re: Slow to connect calls (Ian MacKinnon)
34. TAC confirms incorrect filename on CCO (lelio at uoguelph.ca)
----------------------------------------------------------------------
Message: 1
Date: Wed, 24 Jun 2009 13:20:31 -0400
From: "Miller, Steve" <MillerS at DicksteinShapiro.COM>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Has Anyone Run A Primary Unity Server on a
Physical Server and the Failover on a Virtual Server?
Message-ID: <418329B7ED67E64BBD2BAA97078D70D001C62105 at DCEX2.DSMO.COM>
Content-Type: text/plain; charset="us-ascii"
We are looking to run Unity 7.0.2 on Server 2003. I am looking at a
number of different scenarios, but I wanted to get a feel for whether
this idea was totally crazy or just a little crazy.
Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com
--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s)
named above. This communication may contain material protected by attorney-client, work product, or other
privileges. If you are not the intended recipient or person responsible for delivering this confidential
communication to the intended recipient, you have received this communication in error, and any review, use,
dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files
is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created,
received, or sent on its network. If you have received this confidential communication in error, please notify the
sender immediately by reply e-mail message and permanently delete the original message.
To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
==============================================================================
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/e2cda55c/attachment-0001.html>
------------------------------
Message: 2
Date: Wed, 24 Jun 2009 12:43:35 -0500
From: Tim Frazee <tfrazee at gmail.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] create a software CFB that got deleted
Message-ID:
<30ce418a0906241043w2acc572fi4c612471267405b7 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
hey all,
Running CUCM 7.0(2a) and somehow my software CFB (the one that runs on the
server, from the IPVM service) got deleted. How do I recreate the CFB on the
server?
Adding a software CFB is not a choice in the drop down menu when I try to
just rebuild it.
Any ideas?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/60a2a4e8/attachment-0001.html>
------------------------------
Message: 3
Date: Wed, 24 Jun 2009 13:06:07 -0500
From: Erick Bergquist <erickbee at gmail.com>
To: cisco-voip mailinglist <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Call Manager 5.1.1a install file?
Message-ID:
<f4445faf0906241106q6f3c2ddbjb7f4f78be34357ce at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Does anyone recall what the install file / version is for CUCM 5.1.1a?
is it 3000-4?
Thanks, Erick
------------------------------
Message: 4
Date: Wed, 24 Jun 2009 10:24:20 -0700 (PDT)
From: Paul <asobihoudai at yahoo.com>
To: "Miller, Steve" <MillerS at DicksteinShapiro.COM>,
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Has Anyone Run A Primary Unity Server on a
Physical Server and the Failover on a Virtual Server?
Message-ID: <851685.56437.qm at web111314.mail.gq1.yahoo.com>
Content-Type: text/plain; charset="us-ascii"
First of all, is it supported?
If it's not supported and you're planning on using it in a production environment, then yes you are crazy.
________________________________
From: "Miller, Steve" <MillerS at DicksteinShapiro.COM>
To: cisco-voip at puck.nether.net
Sent: Wednesday, June 24, 2009 1:20:31 PM
Subject: [cisco-voip] Has Anyone Run A Primary Unity Server on a Physical Server and the Failover on a Virtual Server?
We are looking to
run Unity 7.0.2 on Server 2003. I am looking at a number of different
scenarios, but I wanted to get a feel for whether this idea was totally crazy or
just a little crazy.
Steve Miller
Telecom Engineer
Dickstein
Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370|
Fax (202) 330-5607
MillerS at dicksteinshapiro.com
--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s)
named above. This communication may contain material protected by attorney-client, work product, or other
privileges. If you are not the intended recipient or person responsible for delivering this confidential
communication to the intended recipient, you have received this communication in error, and any review, use,
dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files
is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created,
received, or sent on its network. If you have received this confidential communication in error, please notify the
sender immediately by reply e-mail message and permanently delete the original message.
To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
==============================================================================
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/261ee02f/attachment-0001.html>
------------------------------
Message: 5
Date: Wed, 24 Jun 2009 14:28:32 -0400
From: Jason Burns <burns.jason at gmail.com>
To: Erick Bergquist <erickbee at gmail.com>
Cc: cisco-voip mailinglist <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Call Manager 5.1.1a install file?
Message-ID:
<78d9bfc20906241128u76aa5207pc5bec4e836ce6b0e at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
That would probably be 5.1.1.2000-1 or 2.
On Wed, Jun 24, 2009 at 2:06 PM, Erick Bergquist <erickbee at gmail.com> wrote:
> Does anyone recall what the install file / version is for CUCM 5.1.1a?
> is it 3000-4?
>
> Thanks, Erick
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/b44d4d60/attachment-0001.html>
------------------------------
Message: 6
Date: Wed, 24 Jun 2009 14:39:05 -0400
From: "Jason Aarons (US)" <jason.aarons at us.didata.com>
To: "Paul" <asobihoudai at yahoo.com>, "Miller, Steve"
<MillerS at DicksteinShapiro.COM>, <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Has Anyone Run A Primary Unity Server on
aPhysical Server and the Failover on a Virtual Server?
Message-ID:
<C1FE15183DA37645BC0633BC604E44F00F48FA41 at USNAEXCH.na.didata.local>
Content-Type: text/plain; charset="us-ascii"
Design Guide for Cisco Unity Virtualization
http://www.cisco.com/en/US/docs/voice_ip_comm/unity/virtualization_desig
n/guide/cuvirtualdgx.html
Check the archives for this list, it was discussed last month -jason
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul
Sent: Wednesday, June 24, 2009 1:24 PM
To: Miller, Steve; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Has Anyone Run A Primary Unity Server on
aPhysical Server and the Failover on a Virtual Server?
First of all, is it supported?
If it's not supported and you're planning on using it in a production
environment, then yes you are crazy.
________________________________
From: "Miller, Steve" <MillerS at DicksteinShapiro.COM>
To: cisco-voip at puck.nether.net
Sent: Wednesday, June 24, 2009 1:20:31 PM
Subject: [cisco-voip] Has Anyone Run A Primary Unity Server on a
Physical Server and the Failover on a Virtual Server?
We are looking to run Unity 7.0.2 on Server 2003. I am looking at a
number of different scenarios, but I wanted to get a feel for whether
this idea was totally crazy or just a little crazy.
Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com
--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s)
named above. This communication may contain material protected by
attorney-client, work product, or other
privileges. If you are not the intended recipient or person responsible
for delivering this confidential
communication to the intended recipient, you have received this
communication in error, and any review, use,
dissemination, forwarding, printing, copying, or other distribution of
this e-mail message and any attached files
is strictly prohibited. Dickstein Shapiro reserves the right to monitor
any communication that is created,
received, or sent on its network. If you have received this
confidential communication in error, please notify the
sender immediately by reply e-mail message and permanently delete the
original message.
To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
========================================================================
======
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
-----------------------------------------
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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/d1461450/attachment-0001.html>
------------------------------
Message: 7
Date: Wed, 24 Jun 2009 14:41:17 -0400
From: Nick Matthews <matthnick at gmail.com>
To: Paul <asobihoudai at yahoo.com>
Cc: VOIP Group <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] FW: CME web access disable
Message-ID:
<56c3b48b0906241141s497c4e82l67cd5dbe7203c14c at mail.gmail.com>
Content-Type: text/plain; charset=windows-1252
telephony-service
service phone webAccess 1
Then reset the phone.
Here's the full list of variables and their settings:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_s1ht.html#wp1093090
-nick
On Wed, Jun 24, 2009 at 10:15 AM, Paul<asobihoudai at yahoo.com> wrote:
>
> Block port 80 on every switchport that has an IP phone on it.
>
>
>
>
> ________________________________
> From: Ahmed Elnagar <ahmed_elnagar at hotmail.com>
> To: VOIP Group <cisco-voip at puck.nether.net>
> Sent: Wednesday, June 24, 2009 3:38:19 AM
> Subject: [cisco-voip] FW: CME web access disable
>
>
>
> Hello all;
>
> Anyway know a way to disable phone web access for CME phones?
>
> ________________________________
> Windows Live?: Keep your life in sync. Check it out!
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
------------------------------
Message: 8
Date: Wed, 24 Jun 2009 21:43:09 +0300
From: Ahmed Elnagar <ahmed_elnagar at hotmail.com>
To: <matthnick at gmail.com>, <asobihoudai at yahoo.com>
Cc: VOIP Group <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] FW: CME web access disable
Message-ID: <BLU106-W13192812FAE56DC190EA5C87370 at phx.gbl>
Content-Type: text/plain; charset="windows-1256"
Nick...you are great thanks a lot really :)
Thanks,
Ahmed Elnagar
> Date: Wed, 24 Jun 2009 14:41:17 -0400
> Subject: Re: [cisco-voip] FW: CME web access disable
> From: matthnick at gmail.com
> To: asobihoudai at yahoo.com
> CC: ahmed_elnagar at hotmail.com; cisco-voip at puck.nether.net
>
> telephony-service
> service phone webAccess 1
>
>
> Then reset the phone.
>
>
> Here's the full list of variables and their settings:
> http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/command/reference/cme_s1ht.html#wp1093090
>
>
> -nick
>
> On Wed, Jun 24, 2009 at 10:15 AM, Paul<asobihoudai at yahoo.com> wrote:
> >
> > Block port 80 on every switchport that has an IP phone on it.
> >
> >
> >
> >
> > ________________________________
> > From: Ahmed Elnagar <ahmed_elnagar at hotmail.com>
> > To: VOIP Group <cisco-voip at puck.nether.net>
> > Sent: Wednesday, June 24, 2009 3:38:19 AM
> > Subject: [cisco-voip] FW: CME web access disable
> >
> >
> >
> > Hello all;
> >
> > Anyway know a way to disable phone web access for CME phones?
> >
> > ________________________________
> > Windows Live?: Keep your life in sync. Check it out!
> >
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
_________________________________________________________________
Show them the way! Add maps and directions to your party invites.
http://www.microsoft.com/windows/windowslive/products/events.aspx
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/c6216cad/attachment-0001.html>
------------------------------
Message: 9
Date: Wed, 24 Jun 2009 14:16:42 -0500
From: "Jeff Ruttman" <ruttmanj at carewisc.org>
To: "cisco-voip" <cisco-voip at puck.nether.net>
Subject: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
Message-ID:
<07365C3161D8D8419EE51C3834C02205B84D47 at ma1-exc01.ec2802.elderc.org>
Content-Type: text/plain; charset="us-ascii"
Greetings,
We're putting in a Verizon PRI at one of our offices. They're asking
what Protocol we want, NAT1, NAT2, or custom. I believe this has to do
with caller ID, and that "NAT" stands for National. Any insight into
what I should choose?
On our existing gateways with PRIs, the dropdowns in Call Routing
Information where I could choose "National" we have chosen "Cisco Call
Manager."
Thanks
jeff
CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/f43362a0/attachment-0001.html>
------------------------------
Message: 10
Date: Wed, 24 Jun 2009 15:22:41 -0400
From: "Matt Slaga (US)" <Matt.Slaga at us.didata.com>
To: Jeff Ruttman <ruttmanj at carewisc.org>, cisco-voip
<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
Message-ID:
<5FE225375F6E3F4493474B90BC1B94EFB95160515B at USISPCLEXDB01.na.didata.local>
Content-Type: text/plain; charset="us-ascii"
You will want NI2, NI1 is not an option with Cisco gateways (surprised they are even willing to offer it, it's quite antiquated)..
This is selected through the PRI protocol however, not through Call Routing information.
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jeff Ruttman
Sent: Wednesday, June 24, 2009 3:17 PM
To: cisco-voip
Subject: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
Greetings,
We're putting in a Verizon PRI at one of our offices. They're asking what Protocol we want, NAT1, NAT2, or custom. I believe this has to do with caller ID, and that "NAT" stands for National. Any insight into what I should choose?
On our existing gateways with PRIs, the dropdowns in Call Routing Information where I could choose "National" we have chosen "Cisco Call Manager."
Thanks
jeff
CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. 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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/ed3e73b6/attachment-0001.html>
------------------------------
Message: 11
Date: Wed, 24 Jun 2009 14:30:55 -0500
From: "Jeff Ruttman" <ruttmanj at carewisc.org>
To: "Matt Slaga (US)" <Matt.Slaga at us.didata.com>, "cisco-voip"
<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
Message-ID:
<07365C3161D8D8419EE51C3834C02205B84D49 at ma1-exc01.ec2802.elderc.org>
Content-Type: text/plain; charset="us-ascii"
Thanks Matt. Yes all our GWs with a PRI have NI2 as the PRI Protocol
Type, and that was my first thought, but since NAT2 wasn't a option in
the dropdown, I began looking elsewhere. So when Verizon says NAT2 that
means NI2 in CCM?
Thanks
jeff
________________________________
From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com]
Sent: Wednesday, June 24, 2009 2:23 PM
To: Jeff Ruttman; cisco-voip
Subject: RE: PRI Protocol NAT1, NAT2, custom?
You will want NI2, NI1 is not an option with Cisco gateways (surprised
they are even willing to offer it, it's quite antiquated)..
This is selected through the PRI protocol however, not through Call
Routing information.
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jeff Ruttman
Sent: Wednesday, June 24, 2009 3:17 PM
To: cisco-voip
Subject: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
Greetings,
We're putting in a Verizon PRI at one of our offices. They're asking
what Protocol we want, NAT1, NAT2, or custom. I believe this has to do
with caller ID, and that "NAT" stands for National. Any insight into
what I should choose?
On our existing gateways with PRIs, the dropdowns in Call Routing
Information where I could choose "National" we have chosen "Cisco Call
Manager."
Thanks
jeff
CONFIDENTIALITY NOTICE: The information contained in this email
including attachments is intended for the specific delivery to and use
by the individual(s) to whom it is addressed, and includes information
which should be considered as private and confidential. Any review,
retransmission, dissemination, or taking of any action in reliance upon
this information by anyone other than the intended recipient is
prohibited. If you have received this message in error, please reply to
the sender immediately and delete the original message and any copy of
it from your computer system. 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.
CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/c235301f/attachment-0001.html>
------------------------------
Message: 12
Date: Wed, 24 Jun 2009 15:34:57 -0400
From: "Matt Slaga (US)" <Matt.Slaga at us.didata.com>
To: Jeff Ruttman <ruttmanj at carewisc.org>, cisco-voip
<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
Message-ID:
<5FE225375F6E3F4493474B90BC1B94EFB951605167 at USISPCLEXDB01.na.didata.local>
Content-Type: text/plain; charset="us-ascii"
Yes, NAT1 and NAT2 are short for National-1 and National-2. Cisco (and some others) call it NI which is short for National ISDN.
So, you are right on track that you would select NI-2 for the telco's NAT-2.
From: Jeff Ruttman [mailto:ruttmanj at carewisc.org]
Sent: Wednesday, June 24, 2009 3:31 PM
To: Matt Slaga (US); cisco-voip
Subject: RE: PRI Protocol NAT1, NAT2, custom?
Thanks Matt. Yes all our GWs with a PRI have NI2 as the PRI Protocol Type, and that was my first thought, but since NAT2 wasn't a option in the dropdown, I began looking elsewhere. So when Verizon says NAT2 that means NI2 in CCM?
Thanks
jeff
________________________________
From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com]
Sent: Wednesday, June 24, 2009 2:23 PM
To: Jeff Ruttman; cisco-voip
Subject: RE: PRI Protocol NAT1, NAT2, custom?
You will want NI2, NI1 is not an option with Cisco gateways (surprised they are even willing to offer it, it's quite antiquated)..
This is selected through the PRI protocol however, not through Call Routing information.
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jeff Ruttman
Sent: Wednesday, June 24, 2009 3:17 PM
To: cisco-voip
Subject: [cisco-voip] PRI Protocol NAT1, NAT2, custom?
Greetings,
We're putting in a Verizon PRI at one of our offices. They're asking what Protocol we want, NAT1, NAT2, or custom. I believe this has to do with caller ID, and that "NAT" stands for National. Any insight into what I should choose?
On our existing gateways with PRIs, the dropdowns in Call Routing Information where I could choose "National" we have chosen "Cisco Call Manager."
Thanks
jeff
CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. 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.
CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. 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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/65d7c1e6/attachment-0001.html>
------------------------------
Message: 13
Date: Wed, 24 Jun 2009 15:57:08 -0400
From: Peter Slow <peter.slow at gmail.com>
To: Tim Frazee <tfrazee at gmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] create a software CFB that got deleted
Message-ID:
<53fc16d40906241257r19946bc3qcd6282ad951f7dbd at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Try deactivating the IPVMS service, and then reactivating it. This is
different from restarting the service. use service activation. Let us
know how it goes.
-Peter
On Wed, Jun 24, 2009 at 1:43 PM, Tim Frazee<tfrazee at gmail.com> wrote:
> hey all,
>
> Running CUCM 7.0(2a) and somehow my software CFB (the one that runs on the
> server, from the IPVM service) got deleted. How do I recreate the CFB on the
> server?
>
> Adding a software CFB is not a choice in the drop down menu when I try to
> just rebuild it.
>
> Any ideas?
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
------------------------------
Message: 14
Date: Wed, 24 Jun 2009 23:12:35 +0300
From: Dew Swen <dew.swen at gmail.com>
To: Mehmet Turunc <turunc.mehmet at gmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] destination-pattern "T" question
Message-ID:
<ae5778960906241312j5eb22bf5g91fcc7abceea99d5 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Well, let me tell u.
Matching occurs digit by digit unless en-bloc is not been configured.
The number is "90114989123456"
When it is press to 9, none of the dial peers are matched.
After 0 is pressed dial-peer 90 is matched beacuse of T parameter which
collects all digits. However, dial-peer 90110 still does not match.
If dial-peer 90 does not exist, dial-peer 90110 matches "after all the 9011
digits are pressed, and another digit is pressed".
On the other hand, if en-bloc is enabled, all digits are sent at the same
time. So 9T and 9011T are being processed at the same time. Because being a
longer prefix, dial-peer 90110 matches.
Hope it is clear.
Regards,
*
-
Dew Swen*
On Tue, Jun 23, 2009 at 12:44 PM, Mehmet Turunc <turunc.mehmet at gmail.com>wrote:
> Hi all,
>
> I was studying Cisco Voice over IP (CVOICE) -Kevin Wallace 2009- and didn't
> understand this example, so I'm confused. Probably a newbee issue:)
>
> Router(config)#dial-peer voice 90 pots
> Router(config-dial-peer)#destination-pattern 9T
> Router(config-dial-peer)#port 0/0/0:23
> Router(config-dial-peer)#exit
> Router(config)#dial-peer voice 90110 pots
> Router(config-dial-peer)#destination-pattern 9011T
> Router(config-dial-peer)#port 0/0/1:23
>
> And the explanation:
>
> The following steps describe what occurs during the call in this example.
> 1. A user wants to call the international number 90114989123456 and starts
> to dial.
> 2. Because the first digit received is a 9, the gateway performs dial-peer
> matching.
> 3. Dial-peer 90 is matched, and any further digits are collected by the
> control character
> T that indicates the destination-pattern value is a variable-length dial
> string. (WHY? why doesnt longest prefix match?)
> 4. The user finishes dialing, and the call is routed using dial-peer 90.
> Dial-peer 90110
> will never be considered.
>
>
> For en bloc signaling, the DNIS is used, so the process is as follows:
> 1. A user wants to call the international number 90114989123456 and starts
> to dial.
> 2. Because en bloc signaling is enabled, the gateway continues to collect
> digits until the
> interdigit timeout value is exceeded.
> 3. The user finishes dialing, and the call is routed using dial-peer 90110.
>
> Thanks for the help
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/b4134229/attachment-0001.html>
------------------------------
Message: 15
Date: Wed, 24 Jun 2009 16:44:39 -0400
From: "Jason Aarons (US)" <jason.aarons at us.didata.com>
To: <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Does Unity Connection 7.1.2a support SIP RFC
2833 for DTMF
Message-ID:
<C1FE15183DA37645BC0633BC604E44F00F48FCF1 at USNAEXCH.na.didata.local>
Content-Type: text/plain; charset="us-ascii"
I have a SIP Trunk from Verizon Business running thru ACME Packet box to
CallManager 7.1(2a) which then routes to users voicemail on Unity
Connection 7.1.2a connected via SIP trunk.
If I press Zero or another dtmf key press does Unity support RFC2833 for
DTMF or is a dynamic MPT resource needing to be invoked ?
-----------------------------------------
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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/5fa289ec/attachment-0001.html>
------------------------------
Message: 16
Date: Wed, 24 Jun 2009 12:49:31 -0700
From: Tony Underwood <tony at cambiumdata.com>
To: Daniel <dan.voip at danofive.id.au>, "cisco-voip at puck.nether.net"
<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Multicast MoH Delay
Message-ID:
<0F205F18DCB4724DB15EAF8FF93E0A21129EAEE6FE at P3PW5EX1MB04.EX1.SECURESERVER.NET>
Content-Type: text/plain; charset="us-ascii"
If it's a delay in the route set up then you could try a static igmp join on the far end router.
ip igmp join-group group-address
Tony Underwood CCIE #7112
Sr. Network Engineer
Cambium Data Inc.
5050 So. 111th St.
Omaha, NE 68137
(402) 556-1388
http://www.cambiumdata.com<http://www.cambiumdata.com/>
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel
Sent: Wednesday, June 24, 2009 1:46 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Multicast MoH Delay
Hi All,
I've setup multicast routing for music on hold between our data centre network and building floor subnets. The setup is that the data centre is on a different subnet and gear to the floor subnets so multicast routing needs to be used to get MoH packets to the phones.
This consists of the following traffic flow,
MoH Server > Access Switch > Distribution Switch/router(RP) > Distribution Switch/router > Floor Switches > Phones
There are three routing hops (1) from vlan interface to routed interfaces of distribution switch (2) between Distribution and (3) from routed interface of distributionswitch to phone vlan interface. Only the distribution switches are layer 3.
The first distribution switch is the RP for the group and only that group.The second distribution switch is accepting auto RP only. We are using sparse mode.
Multicast traffic is working, the RP mappings are there, the mroute is there, the problem is that when a call is placed on hold there is a 5 second delay before the music is heard. I assume this to be somewhat because of the join message and the delay of the route or path being setup.
Anyone know of a way to reduce the delay before music is heard? If not I guess its back to the lab.
regards,
Dan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/7a7d2fcc/attachment-0001.html>
------------------------------
Message: 17
Date: Thu, 25 Jun 2009 08:44:12 +1000
From: Daniel <dan.voip at danofive.id.au>
To: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Multicast MoH Delay
Message-ID:
<f861d63a0906241544t965b6c6x53696a9ddd691e7f at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Thanks all for your replies
I would rather use sparse mode to take use of the join messages etc.. so
that the phones join and leave the group. Dense mode I think floods the
network and then prunes interfaces that are not in use, this occurs every 3
minuts or so. It just seems that sparse mode is more precise, I don't have
much experience in this so not sure. Any thoughts / technical reasons on why
to go Sparse, Dense or both?
The RP is on a 6500 with sup720s and MSFC3's, the other distribution switch
is an older 6500 with SUP2s and MSFC2, so c3voip similair to the issue you
had with TAC but the RP is on a different device.
I have added the "ip igmp join-group group-address" command in, this fixes
the issue. My question is with this command, the router will accept and
forward these packets preventing fast switching, if i use the static-group
command the router doesnt accept the packets itself but forwards them thus
allowing fast switching, anyone know of benefits either way? I did
originally have a look at this command but I assumed its use was to always
have the multicast traffic flowing which I didnt think ideal. But it turns
out after actually trying this from Tony's point below it works quite well.
Between the dsitribution switches the mroute is always setup but not from
the distribution switch to the floors, when the phone joins the group the
phones vlan interface is added to the mroute and MoH is heard straight away.
There is no multicast MoH flooding the floor vlans which is what I was
concerned about.
On Thu, Jun 25, 2009 at 5:49 AM, Tony Underwood <tony at cambiumdata.com>wrote:
> If it's a delay in the route set up then you could try a static igmp join
> on the far end router.
>
> *ip igmp join-group **group-address*
>
>
>
> *Tony Underwood CCIE #7112*
>
> Sr. Network Engineer
>
> Cambium Data Inc.
>
> 5050 So. 111th St.
>
> Omaha, NE 68137
>
> (402) 556-1388
>
> http://www.cambiumdata.com
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Daniel
> *Sent:* Wednesday, June 24, 2009 1:46 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Multicast MoH Delay
>
>
>
> Hi All,
>
>
>
> I've setup multicast routing for music on hold between our data centre
> network and building floor subnets. The setup is that the data centre is on
> a different subnet and gear to the floor subnets so multicast routing needs
> to be used to get MoH packets to the phones.
>
>
>
> This consists of the following traffic flow,
>
>
>
> MoH Server > Access Switch > Distribution Switch/router(RP) > Distribution
> Switch/router > Floor Switches > Phones
>
>
>
> There are three routing hops (1) from vlan interface to routed interfaces
> of distribution switch (2) between Distribution and (3) from routed
> interface of distributionswitch to phone vlan interface. Only the
> distribution switches are layer 3.
>
>
>
> The first distribution switch is the RP for the group and only that
> group.The second distribution switch is accepting auto RP only. We are using
> sparse mode.
>
>
>
> Multicast traffic is working, the RP mappings are there, the mroute is
> there, the problem is that when a call is placed on hold there is a 5 second
> delay before the music is heard. I assume this to be somewhat because of the
> join message and the delay of the route or path being setup.
>
>
>
> Anyone know of a way to reduce the delay before music is heard? If not I
> guess its back to the lab.
>
>
>
> regards,
>
>
>
> Dan
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090625/780d8c15/attachment-0001.html>
------------------------------
Message: 18
Date: Wed, 24 Jun 2009 15:00:37 -0700 (PDT)
From: Jake Doe <jd80301 at yahoo.com>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SIP Route Pattern
Message-ID: <873114.16995.qm at web50806.mail.re2.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello.
We just upgraded to CUCM 7.1.2.20000-2 and are trying to add a SIP Route Pattern.? However, we are getting the following error:
Add failed. [25256] International Strip Digits should be empty for devices other than H323 gateways and trunks and MGCP T1/E1 PRI and BRI gateways
Any ideas how to correct this problem?? Also, I just noticed that we are using demo licenses.? Could this be causing the issue above?
Thanks.
JD
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/41e0f470/attachment-0001.html>
------------------------------
Message: 19
Date: Wed, 24 Jun 2009 19:19:13 -0400
From: "Jason Aarons (US)" <jason.aarons at us.didata.com>
To: <cisco-voip at puck.nether.net>
Subject: [cisco-voip] TAPS and UCCX 5.0.2 ?
Message-ID:
<C1FE15183DA37645BC0633BC604E44F00F4D14F7 at USNAEXCH.na.didata.local>
Content-Type: text/plain; charset="us-ascii"
Customer is using CallManager 7.1 with off-box CRS 5.0.2 MCS-7845 server
for TAPS (Tool for Auto-Registered Phones Support). Currently I can have
up to 5 phones connecting running the TAPS .aef script.
What is the UCCX license part number to increase the number of ports for
TAPS?
They currently have 150IVR ports and I assume 5 Agent Licenses? Does
TAPS use Agent Licenses?
Or I suspect I don't need Agent licenses and that in AppAdmin on UCCX
under Trigger and/or Media Termination Dialog Group they might currently
be set to 5 and just need to be increased to 150 to allow 150 sessions
of TAPS?
-----------------------------------------
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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/a7af2704/attachment-0001.html>
------------------------------
Message: 20
Date: Wed, 24 Jun 2009 20:34:25 -0400
From: "Adam Frankel" <afrankel at cisco.com>
To: "'Jason Aarons \(US\)'" <jason.aarons at us.didata.com>,
<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Does Unity Connection 7.1.2a support SIP RFC
2833 for DTMF
Message-ID: <007201c9f52c$b1453a00$13cfae00$@com>
Content-Type: text/plain; charset="us-ascii"
Jason,
I believe it does (don't quote me on that) but one way to tell would be to
check the CCM traces for the Capabilities Response sent by the Unity port
when it registers with CUCM. Check for 257.
Adam
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (US)
Sent: Wednesday, June 24, 2009 4:45 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Does Unity Connection 7.1.2a support SIP RFC 2833 for
DTMF
I have a SIP Trunk from Verizon Business running thru ACME Packet box to
CallManager 7.1(2a) which then routes to users voicemail on Unity Connection
7.1.2a connected via SIP trunk.
If I press Zero or another dtmf key press does Unity support RFC2833 for
DTMF or is a dynamic MPT resource needing to be invoked ?
_____
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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/8c01e4df/attachment-0001.html>
------------------------------
Message: 21
Date: Wed, 24 Jun 2009 19:45:42 -0500
From: Erick Bergquist <erickbee at gmail.com>
To: Jason Burns <burns.jason at gmail.com>
Cc: cisco-voip mailinglist <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Call Manager 5.1.1a install file?
Message-ID:
<f4445faf0906241745q7b616ea6p3c44f01d8374fd6 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Yea, thanks.
On Wed, Jun 24, 2009 at 1:28 PM, Jason Burns<burns.jason at gmail.com> wrote:
> That would probably be 5.1.1.2000-1 or 2.
>
> On Wed, Jun 24, 2009 at 2:06 PM, Erick Bergquist <erickbee at gmail.com> wrote:
>>
>> Does anyone recall what the install file / version is for CUCM 5.1.1a?
>> ?is it 3000-4?
>>
>> Thanks, Erick
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
------------------------------
Message: 22
Date: Thu, 25 Jun 2009 11:19:57 +1000
From: ciscozest <ciscozest at gmail.com>
To: cisco-voip mailinglist <cisco-voip at puck.nether.net>
Subject: [cisco-voip] T.37 Fax Redundancy
Message-ID:
<f99cc3f60906241819l32750916n541060c9184d049e at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
we are planning for on-ramp T.37 store and forward fax and wondering about
the redundancy of this way. Can anyone enlighten me on this? We have an
on-ramp T.37 gateway at site A while the IP fax server is located in
different site. Connectivity is over the WAN.
1. What happen to the active fax session when the WAN link is down? Would
the gateway keep trying to reach the Fax server few times and then give up
and drop the fax content?
2. What happen to the NEW incoming fax session when the WAN link is down?
Would it be stored locally in IOS gateway which run the T.37 protocol?
3. Can I create another dial-peer for the T.37 fax number with higher
preference value and push it to the local fax mahine attached to the FXS
port on the on-ramp gateway?
Thank you
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090625/8c259a1c/attachment-0001.html>
------------------------------
Message: 23
Date: Wed, 24 Jun 2009 18:42:42 -0700
From: Tony Underwood <tony at cambiumdata.com>
To: Daniel <dan.voip at danofive.id.au>, "cisco-voip at puck.nether.net"
<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Multicast MoH Delay
Message-ID:
<0F205F18DCB4724DB15EAF8FF93E0A21129EAEE76A at P3PW5EX1MB04.EX1.SECURESERVER.NET>
Content-Type: text/plain; charset="us-ascii"
I'm the furthest thing from a Multicast expert, but to my knowledge the static join is joining only the router interface to the multicast stream and it forwards the packets continuously. But, when the packets get to the L2 switch, it doesn't have an active IGMP join in it's table so it doesn't forward the traffic out any ports. Then when the phone joins the group it is instantly available at the access layer due to the static join.
So, if anything this tells you that your delay is between the L3 devices and not at the access layer.
Tony Underwood CCIE #7112
Sr. Network Engineer
Cambium Data Inc.
5050 So. 111th St.
Omaha, NE 68137
(402) 556-1388
http://www.cambiumdata.com<http://www.cambiumdata.com/>
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel
Sent: Wednesday, June 24, 2009 5:44 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Multicast MoH Delay
Thanks all for your replies
I would rather use sparse mode to take use of the join messages etc.. so that the phones join and leave the group. Dense mode I think floods the network and then prunes interfaces that are not in use, this occurs every 3 minuts or so. It just seems that sparse mode is more precise, I don't have much experience in this so not sure. Any thoughts / technical reasons on why to go Sparse, Dense or both?
The RP is on a 6500 with sup720s and MSFC3's, the other distribution switch is an older 6500 with SUP2s and MSFC2, so c3voip similair to the issue you had with TAC but the RP is on a different device.
I have added the "ip igmp join-group group-address" command in, this fixes the issue. My question is with this command, the router will accept and forward these packets preventing fast switching, if i use the static-group command the router doesnt accept the packets itself but forwards them thus allowing fast switching, anyone know of benefits either way? I did originally have a look at this command but I assumed its use was to always have the multicast traffic flowing which I didnt think ideal. But it turns out after actually trying this from Tony's point below it works quite well. Between the dsitribution switches the mroute is always setup but not from the distribution switch to the floors, when the phone joins the group the phones vlan interface is added to the mroute and MoH is heard straight away. There is no multicast MoH flooding the floor vlans which is what I was concerned about.
On Thu, Jun 25, 2009 at 5:49 AM, Tony Underwood <tony at cambiumdata.com<mailto:tony at cambiumdata.com>> wrote:
If it's a delay in the route set up then you could try a static igmp join on the far end router.
ip igmp join-group group-address
Tony Underwood CCIE #7112
Sr. Network Engineer
Cambium Data Inc.
5050 So. 111th St.
Omaha, NE 68137
(402) 556-1388
http://www.cambiumdata.com<http://www.cambiumdata.com/>
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 Daniel
Sent: Wednesday, June 24, 2009 1:46 AM
To: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: [cisco-voip] Multicast MoH Delay
Hi All,
I've setup multicast routing for music on hold between our data centre network and building floor subnets. The setup is that the data centre is on a different subnet and gear to the floor subnets so multicast routing needs to be used to get MoH packets to the phones.
This consists of the following traffic flow,
MoH Server > Access Switch > Distribution Switch/router(RP) > Distribution Switch/router > Floor Switches > Phones
There are three routing hops (1) from vlan interface to routed interfaces of distribution switch (2) between Distribution and (3) from routed interface of distributionswitch to phone vlan interface. Only the distribution switches are layer 3.
The first distribution switch is the RP for the group and only that group.The second distribution switch is accepting auto RP only. We are using sparse mode.
Multicast traffic is working, the RP mappings are there, the mroute is there, the problem is that when a call is placed on hold there is a 5 second delay before the music is heard. I assume this to be somewhat because of the join message and the delay of the route or path being setup.
Anyone know of a way to reduce the delay before music is heard? If not I guess its back to the lab.
regards,
Dan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/c91ab828/attachment-0001.html>
------------------------------
Message: 24
Date: Wed, 24 Jun 2009 22:56:15 -0400
From: "Miller, Steve" <MillerS at DicksteinShapiro.COM>
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] One User Cannot Be Dialed By Name
Message-ID: <418329B7ED67E64BBD2BAA97078D70D001C62118 at DCEX2.DSMO.COM>
Content-Type: text/plain; charset="us-ascii"
Any ideas why this happens? This one person cannot be dialed by name.
I've checked his name and tried to do it myself, but cannot.
Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com
--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s)
named above. This communication may contain material protected by attorney-client, work product, or other
privileges. If you are not the intended recipient or person responsible for delivering this confidential
communication to the intended recipient, you have received this communication in error, and any review, use,
dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files
is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created,
received, or sent on its network. If you have received this confidential communication in error, please notify the
sender immediately by reply e-mail message and permanently delete the original message.
To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
==============================================================================
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/7581ab28/attachment-0001.html>
------------------------------
Message: 25
Date: Wed, 24 Jun 2009 20:00:39 -0700
From: Cristobal Priego <cristobalpriego at gmail.com>
To: "Miller, Steve" <MillerS at DicksteinShapiro.COM>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] One User Cannot Be Dialed By Name
Message-ID: <3E4765E5-CCD9-485C-8FC9-949F63DB0E93 at gmail.com>
Content-Type: text/plain; charset="us-ascii"; Format="flowed";
DelSp="yes"
Please make sure that the user has a recorded name and that the option
for list in directory is checked
Sent from my iPhone
On Jun 24, 2009, at 7:56 PM, "Miller, Steve" <MillerS at DicksteinShapiro.COM
> wrote:
> Any ideas why this happens? This one person cannot be dialed by
> name. I've checked his name and tried to do it myself, but cannot.
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370| Fax (202) 330-5607
> MillerS at dicksteinshapiro.com
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are
> intended solely for the use of the addressee(s)
> named above. This communication may contain material protected by
> attorney-client, work product, or other
> privileges. If you are not the intended recipient or person
> responsible for delivering this confidential
> communication to the intended recipient, you have received this
> communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or other distribution
> of this e-mail message and any attached files
> is strictly prohibited. Dickstein Shapiro reserves the right to
> monitor any communication that is created,
> received, or sent on its network. If you have received this
> confidential communication in error, please notify the
> sender immediately by reply e-mail message and permanently delete
> the original message.
>
> To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
>
> ===
> ===
> ===
> =====================================================================
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/55e82cab/attachment-0001.html>
------------------------------
Message: 26
Date: Wed, 24 Jun 2009 23:44:08 -0400
From: Dustin S Fowler <dustin.s.fowler at gmail.com>
To: "Jason Aarons (US)" <jason.aarons at us.didata.com>
Cc: "cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] TAPS and UCCX 5.0.2 ?
Message-ID:
<f2d16ce0906242044r1ce0dc65jd1eb8c3d31375e05 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Jason,
Your setting would need to be changed in UCCX App admin. There are a few
spots but you should look for the 'sessions'. I would starts at the trigger
and work your way to the Media Termination Group.
Let us know how it goes.
Dustin Fowler
On Wed, Jun 24, 2009 at 7:19 PM, Jason Aarons (US) <
jason.aarons at us.didata.com> wrote:
> Customer is using CallManager 7.1 with off-box CRS 5.0.2 MCS-7845 server
> for TAPS (Tool for Auto-Registered Phones Support). Currently I can have up
> to 5 phones connecting running the TAPS .aef script.
>
> What is the UCCX license part number to increase the number of ports for
> TAPS?
>
> They currently have 150IVR ports and I assume 5 Agent Licenses? Does TAPS
> use Agent Licenses?
>
> Or I suspect I don't need Agent licenses and that in AppAdmin on UCCX under
> Trigger and/or Media Termination Dialog Group they might currently be set to
> 5 and just need to be increased to 150 to allow 150 sessions of TAPS?
>
> ------------------------------
>
> *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. *
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/cd4f94a3/attachment-0001.html>
------------------------------
Message: 27
Date: Wed, 24 Jun 2009 21:07:31 -0700
From: Mark Holloway <mh at markholloway.com>
To: Jason Aarons (US) <jason.aarons at us.didata.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Does Unity Connection 7.1.2a support SIP RFC
2833 for DTMF
Message-ID: <F6250C96-D4DD-4FAD-BDCC-E64DF7B9C442 at markholloway.com>
Content-Type: text/plain; charset="us-ascii"; Format="flowed";
DelSp="yes"
Unity Connections should already support RFC 2833. The Acme Session
Director realms can be configured for Transparent DTMF (RFC 2833 to
RFC 2833, or SIP Info to SIP Info), RFC 2833 to SIP Info, SIP Info to
RFC 2833, or dual-mode where one realm is set to Transparent and the
other realm (facing UC) sends both event types. This is common where
one SIP server requires one DTMF type then redirects the call to
another SIP server that uses another DTMF type.
On Jun 24, 2009, at 1:44 PM, Jason Aarons (US) wrote:
> I have a SIP Trunk from Verizon Business running thru ACME Packet
> box to CallManager 7.1(2a) which then routes to users voicemail on
> Unity Connection 7.1.2a connected via SIP trunk.
>
> If I press Zero or another dtmf key press does Unity support RFC2833
> for DTMF or is a dynamic MPT resource needing to be invoked ?
>
>
>
> 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.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090624/a0742c5d/attachment-0001.html>
------------------------------
Message: 28
Date: Thu, 25 Jun 2009 02:59:33 -0300
From: "ROJAS, Mario" <Mario.ROJAS at LA.LOGICALIS.COM>
To: <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Show Saved Enterprise Data in CAD
Message-ID:
<1081D526E13AB442851D3AE541A2F112CBBB8C at SNLAR-EXCH01.LA.LOGICALIS.COM>
Content-Type: text/plain; charset="iso-8859-1"
Hello,
We are working on a proposal of a Unified CCX 7, and the customer wants the following to happen:
1) The agent answers a call and has fields in his desktop client to classify the call (like, New Customer, VIP, etc). I know this can be done with customizable Enterprise Data.
2) The next time the agent answers the call coming from the same telephone number (or whatever method of identifying the called, like a customer ID), the agent desktop shows the last variable saved. Like, in step 1, the first call was classified as a New Customer. The next time the a call from the same number goes in, the agent can see how the previous call was treated.
Is that possible? I have configured custom Enterprise Data fields, and I can save information on them, but they don't show up the next time the call comes in.
Best regards,
MARIO ROJAS GUERRERO
Systems Engineer
LOGICALIS
Los Sauces 325 - San Isidro
Lima 27 - Per?
Tel/Fax: +51-1 611-9682
Mov:+51-1 980300124
<mailto:mario.rojas at la.logicalis.com> mario.rojas at la.logicalis.com
<http://www.la.logicalis.com> www.la.logicalis.com
<http://www.logicalisnow.com/> www.logicalisnow.com
Por favor, piense en el medioambiente antes de imprimir este email.
La presente informaci?n se env?a ?nicamente para el destinatario, y contiene informaci?n de car?cter CONFIDENCIAL o PRIVLEGIADA.
La modificaci?n, retransmisi?n, difusi?n, copia u otro uso de esta informaci?n por cualquier medio, por personas distintas al destinatario, est?n estrictamente prohibidas.
Please, think about the environment before printing this email.
The present information is sent solely for the adressee, and contains information of CONFIDENTIAL or PRIVILEGED nature. The modification, broadcasting, diffusion, copy or another use of this information by any means, of people different from the adressee, are strictly prohibited.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090625/bbfc10e9/attachment-0001.html>
------------------------------
Message: 29
Date: Thu, 25 Jun 2009 11:17:58 +0300
From: Mehmet Turunc <turunc.mehmet at gmail.com>
To: Dew Swen <dew.swen at gmail.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] destination-pattern "T" question
Message-ID:
<7d505d120906250117n274614c7j16d6148da9a1102b at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Thanx for the reply Dew. I understand the general idea of your response. But
I couldn't understand some points.
When I open "debug dial-peer voice" debugging, and after that starting to
dial digits, digit by digit matching happens. So, does it mean "by default
digit by digit analysis happens"?
For enabling en-bloc signaling which command should i use? I couldn't find
more specific details.
On Wed, Jun 24, 2009 at 11:12 PM, Dew Swen <dew.swen at gmail.com> wrote:
> Well, let me tell u.
>
> Matching occurs digit by digit unless en-bloc is not been configured.
>
> The number is "90114989123456"
>
> When it is press to 9, none of the dial peers are matched.
>
> After 0 is pressed dial-peer 90 is matched beacuse of T parameter which
> collects all digits. However, dial-peer 90110 still does not match.
>
> If dial-peer 90 does not exist, dial-peer 90110 matches "after all the 9011
> digits are pressed, and another digit is pressed".
>
>
> On the other hand, if en-bloc is enabled, all digits are sent at the same
> time. So 9T and 9011T are being processed at the same time. Because being a
> longer prefix, dial-peer 90110 matches.
>
> Hope it is clear.
>
> Regards,
> *
> -
> Dew Swen*
>
>
> On Tue, Jun 23, 2009 at 12:44 PM, Mehmet Turunc <turunc.mehmet at gmail.com>wrote:
>
>> Hi all,
>>
>> I was studying Cisco Voice over IP (CVOICE) -Kevin Wallace 2009- and
>> didn't understand this example, so I'm confused. Probably a newbee issue:)
>>
>> Router(config)#dial-peer voice 90 pots
>> Router(config-dial-peer)#destination-pattern 9T
>> Router(config-dial-peer)#port 0/0/0:23
>> Router(config-dial-peer)#exit
>> Router(config)#dial-peer voice 90110 pots
>> Router(config-dial-peer)#destination-pattern 9011T
>> Router(config-dial-peer)#port 0/0/1:23
>>
>> And the explanation:
>>
>> The following steps describe what occurs during the call in this example.
>> 1. A user wants to call the international number 90114989123456 and starts
>> to dial.
>> 2. Because the first digit received is a 9, the gateway performs dial-peer
>> matching.
>> 3. Dial-peer 90 is matched, and any further digits are collected by the
>> control character
>> T that indicates the destination-pattern value is a variable-length dial
>> string. (WHY? why doesnt longest prefix match?)
>> 4. The user finishes dialing, and the call is routed using dial-peer 90.
>> Dial-peer 90110
>> will never be considered.
>>
>>
>> For en bloc signaling, the DNIS is used, so the process is as follows:
>> 1. A user wants to call the international number 90114989123456 and starts
>> to dial.
>> 2. Because en bloc signaling is enabled, the gateway continues to collect
>> digits until the
>> interdigit timeout value is exceeded.
>> 3. The user finishes dialing, and the call is routed using dial-peer
>> 90110.
>>
>> Thanks for the help
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090625/f548cac5/attachment-0001.html>
------------------------------
Message: 30
Date: Thu, 25 Jun 2009 08:37:29 -0500
From: "Beck, Christopher" <CBeck at usg.com>
To: "ROJAS, Mario" <Mario.ROJAS at LA.LOGICALIS.COM>,
"cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Show Saved Enterprise Data in CAD
Message-ID:
<AF8E5A5BA2DF074E8B46A3B4A8BC265513806156 at CERO-MB-01.USG.NET>
Content-Type: text/plain; charset="iso-8859-1"
Others may correct me for this, but I believe you are going to need to integrate a database into this mix to store those variables. Thus, you can use ANI to do a lookup and read these values into your variables prior to presenting the call to an agent.
Chris Beck
IT Lead - Voice Technologies
USG Corporation
312-436-4541 (office)
312-730-5524 (Mobile)
312-672-4541 (FAX)
cbeck at usg.com
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of ROJAS, Mario
Sent: Thursday, June 25, 2009 1:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Show Saved Enterprise Data in CAD
Hello,
We are working on a proposal of a Unified CCX 7, and the customer wants the following to happen:
1) The agent answers a call and has fields in his desktop client to classify the call (like, New Customer, VIP, etc). I know this can be done with customizable Enterprise Data.
2) The next time the agent answers the call coming from the same telephone number (or whatever method of identifying the called, like a customer ID), the agent desktop shows the last variable saved. Like, in step 1, the first call was classified as a New Customer. The next time the a call from the same number goes in, the agent can see how the previous call was treated.
Is that possible? I have configured custom Enterprise Data fields, and I can save information on them, but they don't show up the next time the call comes in.
Best regards,
MARIO ROJAS GUERRERO
Systems Engineer
LOGICALIS
Los Sauces 325 - San Isidro
Lima 27 - Per?
Tel/Fax: +51-1 611-9682
Mov:+51-1 980300124
mario.rojas at la.logicalis.com<mailto:mario.rojas at la.logicalis.com>
www.la.logicalis.com<http://www.la.logicalis.com>
www.logicalisnow.com<http://www.logicalisnow.com/>
Por favor, piense en el medioambiente antes de imprimir este email.
La presente informaci?n se env?a ?nicamente para el destinatario, y contiene informaci?n de car?cter CONFIDENCIAL o PRIVLEGIADA.
La modificaci?n, retransmisi?n, difusi?n, copia u otro uso de esta informaci?n por cualquier medio, por personas distintas al destinatario, est?n estrictamente prohibidas.
Please, think about the environment before printing this email.
The present information is sent solely for the adressee, and contains information of CONFIDENTIAL or PRIVILEGED nature. The modification, broadcasting, diffusion, copy or another use of this information by any means, of people different from the adressee, are strictly prohibited.
__________ Information from ESET Smart Security, version of virus signature database 4187 (20090625) __________
The message was checked by ESET Smart Security.
http://www.eset.com
Confidentiality Notice: This email is intended for the sole use of the intended recipient(s) and may contain confidential, proprietary or privileged information. If you are not the intended recipient, you are notified that any use, review, dissemination, copying or action taken based on this message or its attachments, if any, is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy or delete all copies of the original message and any attachments. Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090625/17ab549e/attachment-0001.html>
------------------------------
Message: 31
Date: Thu, 25 Jun 2009 08:47:04 -0500
From: "Jeff Ruttman" <ruttmanj at carewisc.org>
To: "cisco-voip" <cisco-voip at puck.nether.net>
Subject: [cisco-voip] Slow to connect calls
Message-ID:
<07365C3161D8D8419EE51C3834C02205B84D50 at ma1-exc01.ec2802.elderc.org>
Content-Type: text/plain; charset="us-ascii"
Greetings,
Some of our sites have DID trunk ports and POTS lines, and we have MGCP
controlled GWs with FXS and FXO configured. We also have for these
sites H.323 GWs--which frankly I'm not sure why or what they do.
Anyway, at one of those sites, it takes a count of 15 or more for an
outgoing call to connect. I know some delay is expected with that
setup, but that's quite a bit longer than at our comparable sites.
Is that length of delay still within expectations? Or is there
something perhaps I can do to speed that up?
Thanks
jeff
CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090625/c6fcce38/attachment-0001.html>
------------------------------
Message: 32
Date: Thu, 25 Jun 2009 08:57:39 -0500
From: "Voice Noob" <voicenoob at gmail.com>
To: <cisco-voip at puck.nether.net>,
<ask-icd-ivr-support at external.cisco.com>
Subject: [cisco-voip] VoicemailQueuing
Message-ID: <005401c9f59c$ed698a70$c83c9f50$@com>
Content-Type: text/plain; charset="us-ascii"
I am using the voicemail.aef and voicemailqueing.aef from this website.
http://www.uccx.net/media/g/scriptexamples-5x/default.aspx?PageIndex=2
I have everything working well but have a few questions and hope someone can
help me out. On the queuing aspect of the call when it gets presented to the
agent they will press 2 and dial the original caller number. How can I setup
up some type of logic so that if the remote party does not answer or the
agent gets a voicemail box of the caller that the agent can hang-up and have
the call wait a period of time and then get sent back to the queue to the
agents. I guess I am looking for some type of interaction with the CAD
software or even a DTMF entree to tell UCCX that the call was not handled
and needs to be called again at a different time.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090625/e0d5dbb8/attachment-0001.html>
------------------------------
Message: 33
Date: Thu, 25 Jun 2009 15:13:39 +0100
From: Ian MacKinnon <Ian.Mackinnon at lumison.net>
To: Jeff Ruttman <ruttmanj at carewisc.org>, cisco-voip
<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Slow to connect calls
Message-ID:
<B5F945E48C137C49A98BC86DD35939C012D9445D6A at nbg01-exch-01.entlstaff.domain.lumison.net>
Content-Type: text/plain; charset="us-ascii"
Hi Jeff,
That sounds like a dial plan problem ie it is waiting for another digit, and then timing out.
Can you dial the number before hitting dial on the phone so it is all present as opposed to lifting the handset and dialling each digit in turn?
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jeff Ruttman
Sent: 25 June 2009 14:47
To: cisco-voip
Subject: [cisco-voip] Slow to connect calls
Greetings,
Some of our sites have DID trunk ports and POTS lines, and we have MGCP controlled GWs with FXS and FXO configured. We also have for these sites H.323 GWs--which frankly I'm not sure why or what they do.
Anyway, at one of those sites, it takes a count of 15 or more for an outgoing call to connect. I know some delay is expected with that setup, but that's quite a bit longer than at our comparable sites.
Is that length of delay still within expectations? Or is there something perhaps I can do to speed that up?
Thanks
jeff
CONFIDENTIALITY NOTICE: The information contained in this email including attachments is intended for the specific delivery to and use by the individual(s) to whom it is addressed, and includes information which should be considered as private and confidential. Any review, retransmission, dissemination, or taking of any action in reliance upon this information by anyone other than the intended recipient is prohibited. If you have received this message in error, please reply to the sender immediately and delete the original message and any copy of it from your computer system. Thank you.
________________________________
--
This email and any files transmitted with it are confidential and intended
solely for the use of the individual or entity to whom they are addressed.
If you have received this email in error please notify the sender. Any
offers or quotation of service are subject to formal specification.
Errors and omissions excepted. Please note that any views or opinions
presented in this email are solely those of the author and do not
necessarily represent those of Lumison.
Finally, the recipient should check this email and any attachments for the
presence of viruses. Lumison accept no liability for any
damage caused by any virus transmitted by this email.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20090625/7d4d1dc8/attachment-0001.html>
------------------------------
Message: 34
Date: Thu, 25 Jun 2009 11:17:01 -0400 (EDT)
From: lelio at uoguelph.ca
To: cisco-voip voyp list <cisco-voip at puck.nether.net>
Subject: [cisco-voip] TAC confirms incorrect filename on CCO
Message-ID: <B60138B2-E84E-4814-9A28-4C7F94F0089E at uoguelph.ca>
Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes
For what it's worth, the TAC has confirmed the 7.1(2) CUC filename is
incorrect on CCO.
I mentioned this in an earlier post.
Lelio Fulgenzi, Senior Analyst
Computing & Communications
University of Guelph
519-824-4120 x56354
...sent from my iPod - please pardon my fat fingers ;)
[XKJ2000]
------------------------------
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
End of cisco-voip Digest, Vol 68, Issue 23
******************************************
More information about the cisco-voip
mailing list