Re: CME --> SIP Trunk DTMF Issues

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
> >
> >
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> >
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> >
> >
> >
> >
> >
> >
> >
>
>
> --
> 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
>
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-- 
Victor Cappuccio
CCIE R/S# 20657
CCSI# 30452
www.anetworkerblog.com
www.linkedin.com/in/vcappuccio
Blogs and organic groups at http://www.ccie.net