Re: CME --> SIP Trunk DTMF Issues

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