RE: CME --> SIP Trunk DTMF Issues

Joe,

Could you also post a 'debug ccsip messages' for call setup and dtmf-relay events.

Nick,

Do you think he would want to try in-band relay as well?

-ryan

-----Original Message-----
From: Nick Matthews [mailto:matthn_at_gmail.com]
Sent: Thursday, October 01, 2009 8:27 AM
To: Joe Astorino
Cc: Victor Cappuccio; Ryan West; Cisco certification
Subject: Re: CME --> SIP Trunk DTMF Issues

I would suggest doing 'debug voip rtp session named' as that's the
debug to see RTP events.

As well, you can use the IP traffic export feature to do a packet
capture (very cool), and then look at the capture for rtp payload type
101 to see if they're going out.

-nick

On Thu, Oct 1, 2009 at 8:18 AM, Joe Astorino <jastorino_at_ipexpert.com> wrote:
> Hey Victor,
>
> Thanks for the reply. I ran that debug and I get the following: I see my
> negotiated DTMF-Relay is 6 whatever that means not sure... However when I
> actually am pressing buttons I do not see any messages in the debug. I am
> not sure if the digits are actually being sent or not. Any ideas? BTW cool
> video : )
>
> Bono#
> *Oct 1 2009 05:23:09: //5313/B859EB7A882D/SIP/Call/sipSPIMediaCallInfo:
> Number of Media Streams: 1
> Media Stream : 1
> Negotiated Codec : g711ulaw
> Negotiated Codec Bytes : 160
> Nego. Codec payload : 0 (tx), 0 (rx)
> Negotiated Dtmf-relay : 6
> Dtmf-relay Payload : 101 (tx), 101 (rx)
> Source IP Address (Media): 68.43.145.171
> Source IP Port (Media): 16882
> Destn IP Address (Media): x.x.x.x
> Destn IP Port (Media): 18634
> Orig Destn IP Address:Port (Media): 0.0.0.0:0
>
>
> On Thu, Oct 1, 2009 at 7:59 AM, Victor Cappuccio <vcappuccio_at_gmail.com>wrote:
>
>> Hi,
>>
>> I find use full this debug command deb ccsip call to see the reason and
>> codes why SIP is failing/(or not)
>>
>> I was dealing with this issue some days ago, and I got it working for
>> callcentric.
>> probably this http://anetworkerblog.com/2009/09/20/ could help
>>
>>
>> just my 2 cents.
>>
>>
>>
>> On Wed, Sep 30, 2009 at 8:07 PM, Joe Astorino <jastorino_at_ipexpert.com>wrote:
>>
>>> Thanks for the suggestion Ryan. I gave that a shot, but still no luck :(
>>>
>>> On Wed, Sep 30, 2009 at 8:51 AM, Ryan West <rwest_at_zyedge.com> wrote:
>>>
>>> > Joe,
>>> >
>>> > You might give this a shot too:
>>> >
>>> > dtmf-relay sip-notify rtp-nte on the dial-peer. I've needed this before
>>> > when acting as the gateway between two asterisk boxes.
>>> >
>>> > -ryan
>>> >
>>> > -----Original Message-----
>>> > From: nobody_at_groupstudy.com [mailto:nobody_at_groupstudy.com] On Behalf Of
>>> > Joe Astorino
>>> > Sent: Wednesday, September 30, 2009 12:59 AM
>>> > To: Cisco certification
>>> > Subject: OT: CME --> SIP Trunk DTMF Issues
>>> >
>>> > Hey guys,
>>> >
>>> > I have setup a basic VOIP system at home and am having an issue I have
>>> not
>>> > been able to solve yet. I am not into voice so much yet, just enough to
>>> > get
>>> > things going :P Essentially, I have a 3725 running latest 12.4T code
>>> with
>>> > CME. That router has a SIP trunk to a telephony provider and I have 4
>>> > Cisco
>>> > phones here in the house running SCCP to the CME. All that works great.
>>> I
>>> > can place and receive calls, everybody is happy. The only issue is that
>>> > when I dial outside lines, my DTMF totally does not work AT ALL. The
>>> SIP
>>> > trunk terminates to an asterisk box, and I have been told I need to be
>>> > using
>>> > standards based DTMF-Relay, which from what I can tell I am. Not sure
>>> if
>>> > this is a bug, or maybe if anybody else out there has seen this or has
>>> any
>>> > more information? Running a debug I can see that the DTMF is negotiated
>>> to
>>> > rtp-nte which from what I understand is the standard method. Thanks for
>>> > any
>>> > leads...
>>> >
>>> > relevant configurations...
>>> >
>>> > Bono#sh ver | i 12.4
>>> > Cisco IOS Software, 3700 Software (C3725-ADVENTERPRISEK9-M), Version
>>> > 12.4(15)T10, RELEASE SOFTWARE (fc3)
>>> >
>>> > voice service voip
>>> > allow-connections sip to sip
>>> > no supplementary-service sip moved-temporarily
>>> > no supplementary-service sip refer
>>> > sip
>>> > registrar server expires max 3600 min 3600
>>> > localhost dns:<my hostname>
>>> > !
>>> > !
>>> > voice class codec 1
>>> > codec preference 1 g711ulaw
>>> >
>>> > dial-peer voice 3 voip
>>> > description Outbound To SIP Trunk 10-Digits
>>> > translation-profile outgoing prepend1
>>> > destination-pattern [2-9]..[2-9]......
>>> > voice-class codec 1
>>> > voice-class sip dtmf-relay force rtp-nte
>>> > session protocol sipv2
>>> > session target sip-server
>>> > dtmf-relay rtp-nte sip-notify
>>> > clid strip name
>>> > no vad
>>> >
>>> > sip-ua
>>> > retry invite 2
>>> > retry register 10
>>> > retry options 1
>>> > timers connect 100
>>> > registrar dns:sip.myISP.net <http://sip.myisp.net/> expires 3600
>>> > sip-server dns:sip.MyISP.net <http://sip.myisp.net/>
>>> > host-registrar
>>> >
>>> >
>>> > --
>>> > Regards,
>>> >
>>> > Joe Astorino - CCIE #24347 R&S
>>> > Technical Instructor - IPexpert, Inc.
>>> > Cell: +1.586.212.6107
>>> > Fax: +1.810.454.0130
>>> > Mailto: jastorino_at_ipexpert.com
>>> >
>>> >
>>> > Blogs and organic groups at http://www.ccie.net
>>> >
>>> > _______________________________________________________________________
>>> > Subscription information may be found at:
>>> > http://www.groupstudy.com/list/CCIELab.html
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>>
>>>
>>> --
>>> Regards,
>>>
>>> Joe Astorino - CCIE #24347 R&S
>>> Technical Instructor - IPexpert, Inc.
>>> Cell: +1.586.212.6107
>>> Fax: +1.810.454.0130
>>> Mailto: jastorino_at_ipexpert.com
>>>
>>>
>>> Blogs and organic groups at http://www.ccie.net
>>>
>>> _______________________________________________________________________
>>> Subscription information may be found at:
>>> http://www.groupstudy.com/list/CCIELab.html
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>
>>
>> --
>> Victor Cappuccio
>> CCIE R/S# 20657
>> CCSI# 30452
>> www.anetworkerblog.com
>> www.linkedin.com/in/vcappuccio
>>
>
>
>
> --
> Regards,
>
> Joe Astorino - CCIE #24347 R&S
> Technical Instructor - IPexpert, Inc.
> Cell: +1.586.212.6107
> Fax: +1.810.454.0130
> Mailto: jastorino_at_ipexpert.com
>
>
> Blogs and organic groups at http://www.ccie.net
>
> _______________________________________________________________________
> Subscription information may be found at:
> http://www.groupstudy.com/list/CCIELab.html

Blogs and organic groups at http://www.ccie.net
Received on Thu Oct 01 2009 - 09:23:15 ART