From Scott.Voll at wesd.org  Tue Jun 10 15:33:14 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Jun 10 17:34:24 2003
Subject: [cisco-voip] Does this list get used??
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DD54@venus.wesd.org>

I have not seen any activity on this list in a week or so. Is there
anyone else subscribed?

 

We are looking at moving to Cisco IP Telephony.  We are also looking
into installing it our selves.  Has anyone done the same?  

 

Scott Voll

Network Analyst, CCNA 

Willamette ESD

scott.voll@wesd.org <mailto:scott.voll@wesd.org> 

 

 

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From jared at puck.nether.net  Tue Jun 10 20:09:27 2003
From: jared at puck.nether.net (Jared Mauch)
Date: Tue Jun 10 19:09:30 2003
Subject: [cisco-voip] Does this list get used??
In-Reply-To: <A85EF8C2C10DA6499807D44B7FDFEB4F17DD54@venus.wesd.org>
References: <A85EF8C2C10DA6499807D44B7FDFEB4F17DD54@venus.wesd.org>
Message-ID: <20030610230927.GD10766@puck.nether.net>

On Tue, Jun 10, 2003 at 02:33:14PM -0700, Voll, Scott wrote:
> I have not seen any activity on this list in a week or so. Is there
> anyone else subscribed?

	Yeah.

> We are looking at moving to Cisco IP Telephony.  We are also looking
> into installing it our selves.  Has anyone done the same?  

	I'm doing this with Cisco hardware, (phones, routers w/ NM-2V)
running SIP based images with Asterisk.

	- Jared

-- 
Jared Mauch  | pgp key available via finger from jared@puck.nether.net
clue++;      | http://puck.nether.net/~jared/  My statements are only mine.
From listuser at peternixon.net  Wed Jun 11 11:24:43 2003
From: listuser at peternixon.net (Peter Nixon)
Date: Wed Jun 11 03:43:48 2003
Subject: [cisco-voip] Does this list get used??
In-Reply-To: <A85EF8C2C10DA6499807D44B7FDFEB4F17DD54@venus.wesd.org>
References: <A85EF8C2C10DA6499807D44B7FDFEB4F17DD54@venus.wesd.org>
Message-ID: <200306111024.43309.listuser@peternixon.net>

On Wed June 11 2003 00:33, Voll, Scott wrote:
> I have not seen any activity on this list in a week or so. Is there
> anyone else subscribed?

I also subscribed a week or so ago and have not seen any traffic.

> We are looking at moving to Cisco IP Telephony.  We are also looking
> into installing it our selves.  Has anyone done the same?

I run a large Cisco based VoIP termination network across several contries.

-- 

Peter Nixon
http://www.peternixon.net/
PGP Key: http://www.peternixon.net/public.asc

From joshua.ej.smith at usa.net  Wed Jun 11 08:03:49 2003
From: joshua.ej.smith at usa.net (Joshua Sahala)
Date: Wed Jun 11 07:04:54 2003
Subject: [[cisco-voip] Does this list get used??]
Message-ID: <820HFkLDx7968S08.1055329429@uwdvg008.cms.usa.net>



"Voll, Scott" <Scott.Voll@wesd.org> wrote:
> --------------------------------------------- 
>	Attachment:  
>	MIME Type: multipart/alternative 
> --------------------------------------------- 

tsk tsk  ;)

> I have not seen any activity on this list in a week or so. Is there
> anyone else subscribed?
> 
>  

yep

> 
> We are looking at moving to Cisco IP Telephony.  We are also looking
> into installing it our selves.  Has anyone done the same?  
> 
>  

depends on what you want to do (and how much you want||can spend).
i am using primarily h.323 for a multinational carrier voip network - 
sip may work better (and will probably provide more bells & whistles)
for your needs

/joshua

> 
> Scott Voll
> 
> Network Analyst, CCNA 
> 
> Willamette ESD
> 
> scott.voll@wesd.org <mailto:scott.voll@wesd.org> 
> 
>  
> 
>  
> 
> 

> --------------------------------------------- 
>	Attachment: text/plain; charset="us-ascii" 
>	MIME Type: text/plain 
> --------------------------------------------- 


"Walk with me through the Universe,
 And along the way see how all of us are Connected.
 Feast the eyes of your Soul,
 On the Love that abounds.
 In all places at once, seemingly endless,
 Like your own existence."
     - Stephen Hawking -



From nham.kandala at keaneindia.com  Wed Jun 11 11:12:31 2003
From: nham.kandala at keaneindia.com (Nham Kandala)
Date: Wed Jun 11 13:07:25 2003
Subject: [cisco-voip] Does this list get used??
Message-ID: <0861C25BF9E3D511B5C8009027E32A0A03761A@exla01.metamorgs.com>

Skipped content of type multipart/alternative
From wsisk at cisco.com  Wed Jun 11 14:21:29 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Wed Jun 11 13:23:17 2003
Subject: [cisco-voip] Does this list get used??
In-Reply-To: <0861C25BF9E3D511B5C8009027E32A0A03761A@exla01.metamorgs.com>
Message-ID: <ELEMLGDGDBLINAJEIOCLGEBOMEAA.wsisk@cisco.com>

RE: [cisco-voip] Does this list get used??vg200's are End-Of-Sales for Cisco
so you might be hard pressed to find them. /Wes
  -----Original Message-----
  From: cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Nham Kandala
  Sent: Wednesday, June 11, 2003 1:13 PM
  To: 'cisco-voip@puck.nether.net'
  Subject: RE: [cisco-voip] Does this list get used??


  Hi,

  We are thinking to use VoIP gatewaying using Cisco equipment.

  We are in the initial process, Any help/suggestions about the approach is
appreciated.

  We will be using the Nortel PBXs in Branches.
  We want to install VG200 , with NM-HDs as VoIP gateways from Corporate to
Branches.

  In corporate we want to terminate ISDN PRIs in to the central VG200.

  In the Branches, we want to connect Branch VG200s to the PBXs.

  Any better approach recommendations, any suggestions?



  Thank You,
  _________________________________________
  Nham Kandala, 949-450-4693



  -----Original Message-----
  From: Peter Nixon [mailto:listuser@peternixon.net]
  Sent: Wednesday, June 11, 2003 12:25 AM
  To: cisco-voip@puck.nether.net
  Subject: Re: [cisco-voip] Does this list get used??

  On Wed June 11 2003 00:33, Voll, Scott wrote:
  > I have not seen any activity on this list in a week or so. Is there
  > anyone else subscribed?

  I also subscribed a week or so ago and have not seen any traffic.

  > We are looking at moving to Cisco IP Telephony.  We are also looking
  > into installing it our selves.  Has anyone done the same?

  I run a large Cisco based VoIP termination network across several
contries.

  --

  Peter Nixon
  http://www.peternixon.net/
  PGP Key: http://www.peternixon.net/public.asc

  _______________________________________________
  cisco-voip mailing list
  cisco-voip@puck.nether.net
  http://puck.nether.net/mailman/listinfo/cisco-voip
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From nham.kandala at keaneindia.com  Wed Jun 11 11:37:49 2003
From: nham.kandala at keaneindia.com (Nham Kandala)
Date: Wed Jun 11 13:32:59 2003
Subject: [cisco-voip] Does this list get used??
Message-ID: <0861C25BF9E3D511B5C8009027E32A0A03761D@exla01.metamorgs.com>

Skipped content of type multipart/alternative
From nham.kandala at keaneindia.com  Wed Jun 11 12:16:16 2003
From: nham.kandala at keaneindia.com (Nham Kandala)
Date: Wed Jun 11 14:11:38 2003
Subject: [cisco-voip] Does this list get used??
Message-ID: <0861C25BF9E3D511B5C8009027E32A0A037620@exla01.metamorgs.com>

Skipped content of type multipart/alternative
From Scovin at Symmetricom.com  Wed Jun 11 12:34:12 2003
From: Scovin at Symmetricom.com (Steve Covin)
Date: Wed Jun 11 14:35:32 2003
Subject: [cisco-voip] RE: cisco-voip Digest, Vol 4, Issue 1
Message-ID: <27BBC8307A2C994092D90693E15CDA94E637D4@irvexchange.datum.com>

I don't have much to say, I'm the CIO type that does manage the infusion of
technology as well as anyone.
I do have the list of the 300+ patches that Cisco has performed to their
Call Manager.
Trying to get the patch list for the AVAYA system also.
The S8300 AVAYA is what I'm looking to compare.

Steve Covin
Symmetricom Austin
Office: 512-721-4527 
Cell :   512-694-0981


-----Original Message-----
From: cisco-voip-request@puck.nether.net
[mailto:cisco-voip-request@puck.nether.net]
Sent: Wednesday, June 11, 2003 11:01 AM
To: cisco-voip@puck.nether.net
Subject: cisco-voip Digest, Vol 4, Issue 1


Send cisco-voip mailing list submissions to
	cisco-voip@puck.nether.net

To subscribe or unsubscribe via the World Wide Web, visit
	http://puck.nether.net/mailman/listinfo/cisco-voip
or, via email, send a message with subject or body 'help' to
	cisco-voip-request@puck.nether.net

You can reach the person managing the list at
	cisco-voip-owner@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. Does this list get used?? (Voll, Scott)
   2. Re: Does this list get used?? (Jared Mauch)
   3. Re: Does this list get used?? (Peter Nixon)
   4. Re: [[cisco-voip] Does this list get used??] (Joshua Sahala)


----------------------------------------------------------------------

Message: 1
Date: Tue, 10 Jun 2003 14:33:14 -0700
From: "Voll, Scott" <Scott.Voll@wesd.org>
Subject: [cisco-voip] Does this list get used??
To: <cisco-voip@puck.nether.net>
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DD54@venus.wesd.org>
Content-Type: text/plain; charset="us-ascii"

I have not seen any activity on this list in a week or so. Is there
anyone else subscribed?

 

We are looking at moving to Cisco IP Telephony.  We are also looking
into installing it our selves.  Has anyone done the same?  

 

Scott Voll

Network Analyst, CCNA 

Willamette ESD

scott.voll@wesd.org <mailto:scott.voll@wesd.org> 

 

 

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------------------------------

Message: 2
Date: Tue, 10 Jun 2003 19:09:27 -0400
From: Jared Mauch <jared@puck.nether.net>
Subject: Re: [cisco-voip] Does this list get used??
To: "Voll, Scott" <Scott.Voll@wesd.org>
Cc: cisco-voip@puck.nether.net
Message-ID: <20030610230927.GD10766@puck.nether.net>
Content-Type: text/plain; charset=us-ascii

On Tue, Jun 10, 2003 at 02:33:14PM -0700, Voll, Scott wrote:
> I have not seen any activity on this list in a week or so. Is there
> anyone else subscribed?

	Yeah.

> We are looking at moving to Cisco IP Telephony.  We are also looking
> into installing it our selves.  Has anyone done the same?  

	I'm doing this with Cisco hardware, (phones, routers w/ NM-2V)
running SIP based images with Asterisk.

	- Jared

-- 
Jared Mauch  | pgp key available via finger from jared@puck.nether.net
clue++;      | http://puck.nether.net/~jared/  My statements are only mine.

------------------------------

Message: 3
Date: Wed, 11 Jun 2003 10:24:43 +0300
From: Peter Nixon <listuser@peternixon.net>
Subject: Re: [cisco-voip] Does this list get used??
To: <cisco-voip@puck.nether.net>
Message-ID: <200306111024.43309.listuser@peternixon.net>
Content-Type: text/plain;  charset="iso-8859-1"

On Wed June 11 2003 00:33, Voll, Scott wrote:
> I have not seen any activity on this list in a week or so. Is there
> anyone else subscribed?

I also subscribed a week or so ago and have not seen any traffic.

> We are looking at moving to Cisco IP Telephony.  We are also looking
> into installing it our selves.  Has anyone done the same?

I run a large Cisco based VoIP termination network across several contries.

-- 

Peter Nixon
http://www.peternixon.net/
PGP Key: http://www.peternixon.net/public.asc


------------------------------

Message: 4
Date: Wed, 11 Jun 2003 07:03:49 -0400
From: Joshua Sahala <joshua.ej.smith@usa.net>
Subject: Re: [[cisco-voip] Does this list get used??]
To: "Voll, Scott" <Scott.Voll@wesd.org>, <cisco-voip@puck.nether.net>
Message-ID: <820HFkLDx7968S08.1055329429@uwdvg008.cms.usa.net>
Content-Type: text/plain; charset=ISO-8859-1



"Voll, Scott" <Scott.Voll@wesd.org> wrote:
> --------------------------------------------- 
>	Attachment:  
>	MIME Type: multipart/alternative 
> --------------------------------------------- 

tsk tsk  ;)

> I have not seen any activity on this list in a week or so. Is there
> anyone else subscribed?
> 
>  

yep

> 
> We are looking at moving to Cisco IP Telephony.  We are also looking
> into installing it our selves.  Has anyone done the same?  
> 
>  

depends on what you want to do (and how much you want||can spend).
i am using primarily h.323 for a multinational carrier voip network - 
sip may work better (and will probably provide more bells & whistles)
for your needs

/joshua

> 
> Scott Voll
> 
> Network Analyst, CCNA 
> 
> Willamette ESD
> 
> scott.voll@wesd.org <mailto:scott.voll@wesd.org> 
> 
>  
> 
>  
> 
> 

> --------------------------------------------- 
>	Attachment: text/plain; charset="us-ascii" 
>	MIME Type: text/plain 
> --------------------------------------------- 


"Walk with me through the Universe,
 And along the way see how all of us are Connected.
 Feast the eyes of your Soul,
 On the Love that abounds.
 In all places at once, seemingly endless,
 Like your own existence."
     - Stephen Hawking -




------------------------------

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


End of cisco-voip Digest, Vol 4, Issue 1
****************************************
From eugen at romtelecom.net  Mon Jun 16 13:09:37 2003
From: eugen at romtelecom.net (Eugen Pantica)
Date: Mon Jun 16 05:10:07 2003
Subject: [cisco-voip] Calltracker problem
Message-ID: <3EED8951.6020806@romtelecom.net>

I have one AS 5400 for VOIP traffic. I did configure the calltracker, 
because I wanted to obtain the call informations for to accomplish the 
statistical reports.
Look it that is the configuration for calltracker:
!
calltracker enable
calltracker call-record verbose
spe call-record modem quiet
!

When type
sh call calltracker active I obtain:

Jun 16 11:28:12.514: %CALLTRKR-6-CALL_RECORD: 
ct_hndl=236675,service=None, origin=Answer,
category=Modem, DS0 slot/port/ds1/chan=-1/-1/-1/-1, 
called=393478999362,calling=233709172,
resource slot/port=(n/a)/(n/a),userid=(n/a), ip=0.0.0.0, mask=0.0.0.0, 
account id=0,
setup=06/16/2003 11:26:55, conn=0.00, phys=0.00, 
service=0.00,authen=0.00, init-rx/tx b-rate=0/0,
rx/tx chars=0/0, time=10.05, disc subsys=ISDN, disc code=0x66, disc 
text=Recovery on timer expiry

The question is, if know somebody what means the values -1/-1/-1/-1 for 
DS0 and
why not obtain the real ip address for the calls arrived in my gateway 
from Internet?

How can I remediate this issue?

thank you.

Eugen



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From sg at comptek.ru  Mon Jun 16 15:29:42 2003
From: sg at comptek.ru (Sergey Gromov)
Date: Mon Jun 16 06:30:53 2003
Subject: [cisco-voip] Calltracker problem
In-Reply-To: <3EED8951.6020806@romtelecom.net>
Message-ID: <006301c333f2$32db9760$dec4b4d5@sg>

Hi, Eugen 
seems that you are hitting ddts CSCds68856, please look at IOS version
which you curently use, to be fully sure that it in the list of affected
versions 

=====================================
CSCds68856 Bug Details
Headline: wrong channel displayed by calltracker
Product: allModel 
Component: call-mgmtDuplicate of 
Severity: 3
Status: Resolved
First Found-in Version: 12.1(4.4)T03    
First Fixed-in Version: 12.2(0.1), 12.2(0.4), 12.2(0.3)T, 12.2(0.5)T,
12.2(0.7)PI01, 12.2(0.18)S, 12.2(3.4)PB, 12.2(2)B  Version help
Release Notes:

On a AS5x00 running 12.1(5)T, "sh call callt act" shows the DS0 number
incorrectly. "sh users" however displays the correct channel numbers.
There is no work around.
======================================


About "ip address" issue I think that calltracker is designed more to
l2tp/ppp/... enviroment, on CCO I don`t find any doc, which point that
at voice call it shoud show ip address of remote gw. As workaround you
may try to use Router# show call active voice brief command, which show
you remote gw ip adress

Best regards,
Sergey

-----Original Message-----
From: cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] On Behalf Of Eugen Pantica
Sent: Monday, June 16, 2003 1:10 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Calltracker problem


I have one AS 5400 for VOIP traffic. I did configure the calltracker,
because I wanted to obtain the call informations for to accomplish the
statistical reports.
Look it that is the configuration for calltracker:
!
calltracker enable
calltracker call-record verbose
spe call-record modem quiet
!

When type 
sh call calltracker active I obtain:

Jun 16 11:28:12.514: %CALLTRKR-6-CALL_RECORD:
ct_hndl=236675,service=None, origin=Answer,
category=Modem, DS0 slot/port/ds1/chan=-1/-1/-1/-1,
called=393478999362,calling=233709172,
resource slot/port=(n/a)/(n/a),userid=(n/a), ip=0.0.0.0, mask=0.0.0.0,
account id=0,
setup=06/16/2003 11:26:55, conn=0.00, phys=0.00,
service=0.00,authen=0.00, init-rx/tx b-rate=0/0,
rx/tx chars=0/0, time=10.05, disc subsys=ISDN, disc code=0x66, disc
text=Recovery on timer expiry

The question is, if know somebody what means the values -1/-1/-1/-1 for
DS0 and 
why not obtain the real ip address for the calls arrived in my gateway
from Internet?

How can I remediate this issue? 

thank you.

Eugen

From Scott.Voll at wesd.org  Mon Jun 16 17:31:43 2003
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Jun 16 19:33:01 2003
Subject: [cisco-voip] Call Manager --Automated Caller
Message-ID: <A85EF8C2C10DA6499807D44B7FDFEB4F17DD5C@venus.wesd.org>

Hey everyone--

Does anyone have an Automated caller connected to Call Manager via TAPI or JTAPI?  And what is it?  How well does it work?  Have you had any problems with it?  How easy was it to put in and use? 


Thanks in advance

Scott Voll
Network Analyst, CCNA 
Willamette ESD
scott.voll@wesd.org
?


From diego at ifxnw.com.uy  Wed Jun 18 15:09:07 2003
From: diego at ifxnw.com.uy (Diego Costa)
Date: Wed Jun 18 13:13:48 2003
Subject: [cisco-voip] Question about Codec complexity
Message-ID: <IOEDKEMHIJODBNKPFAKPIEMDCDAA.diego@ifxnw.com.uy>

People, somebody knows if the codec-complexity works on 1700's platforms?? I
think that only works on 2600+ but I'm not sure, any experience or comment
would appreciated.

Regards
Diego.

From hweiss at platformexperts.com  Thu Jun 19 16:11:10 2003
From: hweiss at platformexperts.com (Chana Weiss)
Date: Thu Jun 19 15:12:58 2003
Subject: [cisco-voip] Troubleshooting commands
Message-ID: <DA67D9BBF53D9C448713DF4ADF1FD6B0E58D@apmail.atarpx.callingcard.com>

Hi,
 
Im looking for good troubleshooting commands or, tools... Im new to this voip stuff.  any help would be greatly appreciated.
 
Thank you.
 
-Chana

-----Original Message-----
From: cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
Sent: Monday, June 16, 2003 7:32 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Call Manager --Automated Caller



Hey everyone--

Does anyone have an Automated caller connected to Call Manager via TAPI or JTAPI?  And what is it?  How well does it work?  Have you had any problems with it?  How easy was it to put in and use?


Thanks in advance

Scott Voll
Network Analyst, CCNA
Willamette ESD
scott.voll@wesd.org
 


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

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From belen.pugliese at tecoar.com.ar  Thu Jun 19 17:53:59 2003
From: belen.pugliese at tecoar.com.ar (=?iso-8859-1?Q?Bel=E9n_Pugliese?=)
Date: Thu Jun 19 15:56:22 2003
Subject: [cisco-voip] HELP !!
References: <DA67D9BBF53D9C448713DF4ADF1FD6B0E58D@apmail.atarpx.callingcard.com>
Message-ID: <00ea01c3369c$8c684e70$160101c9@Admin>

Hi everyone,
I am Belen, from Argentina. I am new in this forum.
I need your help. I must configure an AS5350 with SS7 in the E1 port
(controller E1). Does anyone have a sample configuration file or something
about configuring SS7 on Cisco's AS5350?

Best Regards,
Belen Pugliese.


From mbartsch at unix911.ath.cx  Thu Jun 19 21:57:00 2003
From: mbartsch at unix911.ath.cx (Marcelo Bartsch)
Date: Thu Jun 19 20:58:29 2003
Subject: [cisco-voip] Troubleshooting commands
In-Reply-To: <DA67D9BBF53D9C448713DF4ADF1FD6B0E58D@apmail.atarpx.callingcard.com>
References: 
	 <DA67D9BBF53D9C448713DF4ADF1FD6B0E58D@apmail.atarpx.callingcard.com>
Message-ID: <1056070619.30158.0.camel@unix911.ath.cx>

On Thu, 2003-06-19 at 15:11, Chana Weiss wrote:
Chana,
about commands, which help me a lot was:

debug dialpeer
debug isdn q931 (if you had isdn)
debug voice translations

with those 3 you can get a lot of info on how the VOIP GW is working.


> Hi,
>  
> Im looking for good troubleshooting commands or, tools... Im new to this voip stuff.  any help would be greatly appreciated.
>  
> Thank you.
>  
> -Chana
> 
> -----Original Message-----
> From: cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> Sent: Monday, June 16, 2003 7:32 PM
> To: cisco-voip@puck.nether.net
> Subject: [cisco-voip] Call Manager --Automated Caller
> 
> 
> 
> Hey everyone--
> 
> Does anyone have an Automated caller connected to Call Manager via TAPI or JTAPI?  And what is it?  How well does it work?  Have you had any problems with it?  How easy was it to put in and use?
> 
> 
> Thanks in advance
> 
> Scott Voll
> Network Analyst, CCNA
> Willamette ESD
> scott.voll@wesd.org
>  
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> http://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> ______________________________________________________________________
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> http://puck.nether.net/mailman/listinfo/cisco-voip
-- 
Marcelo Bartsch <mbartsch@unix911.ath.cx>

From ian.mac at kinnon.org  Fri Jun 20 09:04:46 2003
From: ian.mac at kinnon.org (Ian MacKinnon)
Date: Fri Jun 20 03:06:24 2003
Subject: [cisco-voip] Re: HELP !!
In-Reply-To: <00ea01c3369c$8c684e70$160101c9@Admin> 
References: <DA67D9BBF53D9C448713DF4ADF1FD6B0E58D@apmail.atarpx.callingcard.com>
	<00ea01c3369c$8c684e70$160101c9@Admin>
Message-ID: <20030620070446.1167.qmail@srv2.phosts.com>

Try 
http://www.cisco.com/univercd/cc/td/doc/product/access/sc/rel7/soln/das22/
Lots of stuff about SS7 on Access Servers 

I assume you are using MGC or something as I believe the 53xx do not support 
SS& directly, but support Q.931 over Ethernet 

You then use nfas_group command to link these 

Bel?n Pugliese writes: 

> Hi everyone,
> I am Belen, from Argentina. I am new in this forum.
> I need your help. I must configure an AS5350 with SS7 in the E1 port
> (controller E1). Does anyone have a sample configuration file or something
> about configuring SS7 on Cisco's AS5350? 
> 
> Best Regards,
> Belen Pugliese. 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> http://puck.nether.net/mailman/listinfo/cisco-voip
 
From joshua.ej.smith at usa.net  Fri Jun 20 11:43:26 2003
From: joshua.ej.smith at usa.net (Joshua Sahala)
Date: Fri Jun 20 10:45:00 2003
Subject: [Re: [cisco-voip] Troubleshooting commands]
Message-ID: <670HFTorA6240S03.1056120206@uwdvg003.cms.usa.net>

some other useful commands:

sho dial-peer voice ?
sho call hist v ?
debug voip ccapi inout (warning - this can generate a lot of output)
debug voice ccapi inout (lots of info)
debug [voip|voice] ?

useful links:
http://www.cisco.com/en/US/tech/tk652/tk701/tech_problem_troubleshooting_list.html
http://www.cisco.com/en/US/tech/tk652/tk701/tech_tech_notes_list.html

hth

/joshua

Marcelo Bartsch <mbartsch@unix911.ath.cx> wrote:
> On Thu, 2003-06-19 at 15:11, Chana Weiss wrote:
> Chana,
> about commands, which help me a lot was:
> 
> debug dialpeer
> debug isdn q931 (if you had isdn)
> debug voice translations
> 
> with those 3 you can get a lot of info on how the VOIP GW is working.
> 
> 
> > Hi,
> >  
> > Im looking for good troubleshooting commands or, tools... Im new to this
voip stuff.  any help would be greatly appreciated.
> >  
> > Thank you.
> >  
> > -Chana
> > 
> > -----Original Message-----
> > From: cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Voll, Scott
> > Sent: Monday, June 16, 2003 7:32 PM
> > To: cisco-voip@puck.nether.net
> > Subject: [cisco-voip] Call Manager --Automated Caller
> > 
> > 
> > 
> > Hey everyone--
> > 
> > Does anyone have an Automated caller connected to Call Manager via TAPI or
JTAPI?  And what is it?  How well does it work?  Have you had any problems
with it?  How easy was it to put in and use?
> > 
> > 
> > Thanks in advance
> > 
> > Scott Voll
> > Network Analyst, CCNA
> > Willamette ESD
> > scott.voll@wesd.org
> >  
> > 
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > http://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > 
> > ______________________________________________________________________
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip@puck.nether.net
> > http://puck.nether.net/mailman/listinfo/cisco-voip
> -- 
> Marcelo Bartsch <mbartsch@unix911.ath.cx>
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> http://puck.nether.net/mailman/listinfo/cisco-voip
> 



"Walk with me through the Universe,
 And along the way see how all of us are Connected.
 Feast the eyes of your Soul,
 On the Love that abounds.
 In all places at once, seemingly endless,
 Like your own existence."
     - Stephen Hawking -



From vances at motivity.ca  Fri Jun 20 13:42:49 2003
From: vances at motivity.ca (Vance Shipley)
Date: Fri Jun 20 12:44:22 2003
Subject: [Re: [cisco-voip] Troubleshooting commands]
In-Reply-To: <670HFTorA6240S03.1056120206@uwdvg003.cms.usa.net>
References: <670HFTorA6240S03.1056120206@uwdvg003.cms.usa.net>
Message-ID: <20030620164249.GZ599@frogman.motivity.ca>

}  some other useful commands:
  
debug h225 asn1

}  sho dial-peer voice ?
}  sho call hist v ?
}  debug voip ccapi inout (warning - this can generate a lot of output)
}  debug voice ccapi inout (lots of info)
}  debug [voip|voice] ?
From pgrace at fierymoon.com  Fri Jun 20 21:12:25 2003
From: pgrace at fierymoon.com (Peter Grace)
Date: Fri Jun 20 20:13:49 2003
Subject: [cisco-voip] 7960 rebooting cycle
Message-ID: <002001c33789$cb3c74d0$daea5c42@fierymoon.com>

 
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

	<Deep breath -- Long story coming!>

	I received a 7960 today from a company off of the
internet.  When I received the unit, it had firmware SIP
2.2 on it.  Following the suggestions of a few people
related to the Asterisk project, I upgraded the phone to
4.4.  I started getting some weird behavior...  When a call
would come through to the 7960, if you did not answer the
call on the 7960 and the remote party hung up, the 7960
would immediately reboot.  I explained this to some people,
and they recommended that I upgrade to 5.1, which I did. 
Now, when the phone boots, it gets through the TFTP
transfer, gets all addresses necessary, and then it appears
that as soon as it's about to send SIP REGISTER's, the
phone reboots.  I cannot break the phone out of this loop,
short of removing the power.  When the unit is plugged back
in, it continues rebooting endlessly.

A trusted associate of mine said that this problem has been
reported before, and the only way to fix it was to
downgrade to a lower firmware, and then upgrade again.  As
I've upgraded to 5.1, I can no longer downgrade.  Or can I?

If ANYBODY has any ideas, I'd really appreciate it.  Right
about now, I'm feeling like I've got a $300 paperweight on
the desk.

Thanks,
Peter

- --
pgrace@fierymoon.com

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---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.487 / Virus Database: 286 - Release Date: 6/1/2003

From diego.costa at ifxnw.com.uy  Wed Jun 18 15:07:28 2003
From: diego.costa at ifxnw.com.uy (Diego Costa)
Date: Sun Jun 22 11:15:34 2003
Subject: [cisco-voip] Question about Codec complexity
Message-ID: <IOEDKEMHIJODBNKPFAKPOEMCCDAA.diego.costa@ifxnw.com.uy>

People, somebody knows if the codec-complexity works on 1700's platforms?? I
think that only works on 2600+ but I'm not sure, any experience or comment
would appreciated.

Regards
Diego.

From wsisk at cisco.com  Sun Jun 22 14:01:16 2003
From: wsisk at cisco.com (Wes Sisk)
Date: Sun Jun 22 13:02:52 2003
Subject: [cisco-voip] Question about Codec complexity
In-Reply-To: <IOEDKEMHIJODBNKPFAKPOEMCCDAA.diego.costa@ifxnw.com.uy>
Message-ID: <ELEMLGDGDBLINAJEIOCLAEBDMFAA.wsisk@cisco.com>

This might help:

http://www.cisco.com/en/US/partner/products/hw/routers/ps221/prod_configurat
ion_guide09186a0080106d1c.html#xtocid5

/Wes

> -----Original Message-----
> From: cisco-voip-bounces@puck.nether.net
> [mailto:cisco-voip-bounces@puck.nether.net]On Behalf Of Diego Costa
> Sent: Wednesday, June 18, 2003 1:07 PM
> To: Cisco-Voip
> Subject: [cisco-voip] Question about Codec complexity
>
>
> People, somebody knows if the codec-complexity works on 1700's
> platforms?? I
> think that only works on 2600+ but I'm not sure, any experience or comment
> would appreciated.
>
> Regards
> Diego.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> http://puck.nether.net/mailman/listinfo/cisco-voip

From odalys at fiu.edu  Mon Jun 23 02:02:05 2003
From: odalys at fiu.edu (odalys@fiu.edu)
Date: Mon Jun 23 01:03:44 2003
Subject: [cisco-voip] Odalys Diaz/UTS/FIU is out of the office.
Message-ID: <OF69013EC2.572F9DE0-ON85256D4E.001BA86D-85256D4E.001BA86E@fiu.edu>

I will be out of the office starting  06/20/2003 and will not return until
06/30/2003.

I will have no access to voicemail or e-mail.


From eric at fnords.org  Wed Jun 25 00:09:56 2003
From: eric at fnords.org (Eric Wieling)
Date: Wed Jun 25 00:11:36 2003
Subject: [cisco-voip] ATA-186 Dial Plan
Message-ID: <1056514195.4229.170.camel@vulcan.fnords.org>

I set the Dial Plan on my ATA-186 to: 9.......|^1...

As I understand the docs this means that if the ATA-186 sees a 9
followed by any seven digits send the call to the SIP gateway, if the
dialed number doesn't start with a 9, then check if the first digit is
anything except a 1, followed by three digits and send the call to the
SIP server.  With the above dial plan I can dial 9123 and the call gets
immediately sent to my SIP gateway.  I'm using Version: v2.15 ata18x
(Build 030313a)

Does anyone have any pointers to what I'm confuzzled about?


-- 
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)

From diego at ifxnw.com.uy  Mon Jun 30 19:17:46 2003
From: diego at ifxnw.com.uy (Diego Costa)
Date: Mon Jun 30 17:20:28 2003
Subject: [cisco-voip] Problemas with GW
Message-ID: <NHBBJDNFILCHLGHOFOFJEECFCDAA.diego@ifxnw.com.uy>

People,

     I have the next scenario:

 ------          ------
| GK 1 |        | GK 2 |
 ------          ------
    |               |
    |               |
    |               |
 ------          ------
| GW 1 |        | GW 2 |
 ------          ------

I have a dial-peer voip in the gw1 and the corresponding pots in the gw2,
when I make a call from gw1 to gw2, the gw2 send an ARQ to the gk and it's
respond with the corresponding ARJ ... it's really strange because in the
pots y have added "no register e164".

Anybody knows what can be happening?

Regards,
DC.

