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

Thanks for the response Nick.

Still no dice. I see the following in debug ccsip all:

Stream type : voice+dtmf
          Media line : 1
          State : STREAM_ACTIVE (5)
          Callid : 457
          Negotiated Codec : g711ulaw, bytes :160
          Nego. Codec payload : 0 (tx), 0 (rx)
          Negotiated DTMF relay : rtp-nte
          Negotiated NTE payload : 101 (tx), 101 (rx)
          Negotiated CN payload : 0
          Media Srce Addr/Port : 68.43.145.171:18110
          Media Dest Addr/Port : 208.83.69.38:15834

and then a ways down ...

*Sep 29 2009 23:03:43: //457/9C8952A780FC/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): 18110
Destn IP Address (Media): 208.83.69.38
Destn IP Port (Media): 15834
Orig Destn IP Address:Port (Media): 0.0.0.0:0

So, I see the negotiated DTMF-Relay at first days rtp-nte and then "6".
Also, I see the payload type is 101. I tried the other commands you
suggested as well with no luck.

On Wed, Sep 30, 2009 at 1:22 AM, Nick Matthews <matthn_at_gmail.com> wrote:

> 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
> >
> >
> > 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