Re: OT: CME --> SIP Trunk DTMF Issues

You would want to debug ccsip messages and check whether the payload
type you're using is the same. 101 is the default but 100 is a common
variation. rtp payload nte 100 is the dial peer configuration if that
is the case.

Otherwise, you may want to try:

voice service voip
 dtmf-interworking rtp-nte

This is a slight change in the way DTMF is sent out that a few more
PBXs interop with.

Other than that, 'debug voip rtp session named' is what you're looking
for to confirm if you're actually sending the digits out.

-nick

On Wed, Sep 30, 2009 at 12:59 AM, Joe Astorino <jastorino_at_ipexpert.com> wrote:
> 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 expires 3600
> sip-server dns: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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Received on Wed Sep 30 2009 - 01:22:47 ART