Re: CME --> SIP Trunk DTMF Issues

From: Joe Astorino <jastorino_at_ipexpert.com>
Date: Thu, 1 Oct 2009 09:07:31 -0400

Thanks Nick!

I do see the following when I actually press buttons. This would be an
example of me pressing the "5" digit hence the Evt: 5. I see the packet
type is indeed 101 and that it is sending (supposedly) I guess maybe a
packet capture will be next

*Oct 1 2009 06:12:42: s=DSP d=VoIP payload 0x65 ssrc 0x535A4E81
sequence 0x7408 timestamp 0x273D800
*Oct 1 2009 06:12:42: Pt:101 Evt:5 Pkt:04 00 00 <Snd>>>

On Thu, Oct 1, 2009 at 8:27 AM, Nick Matthews <matthn_at_gmail.com> wrote:

> 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
> >
> >
> >
> >
> >
> >
> >
> >
>

-- 
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
Received on Thu Oct 01 2009 - 09:07:31 ART

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