<div style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">Hi all,</font></div>
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<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">Here is some aditional information from the ISP side. they are using a Genband MSX release 4.3m6 and we have the following options for DTMF transmission:</font></p>
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<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">SIP NOTIFY</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">SIP INFO</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">RFC2833 (payload type 101)</font></p>
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<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">When we send DTMF to the CCM with SIP NOTIFY we see a 403 Forbidden comming back from the CCM.</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas"> </font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">One other thing i noticed is that when we place a inbound call to the CCM --> IVR (using RFC2833) the 200OK doesen't contain a media type 101 (rfc2833), and if i'm correct the CCM was configured for RFC2833 at that point. If the SBC doesen't receive the correct media type in the 200OK it assumes that there is no support for RFC2833 and it will fallback to inband DTMF tones in the audio stream.</font></p>
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<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">Could this have something to do with the DTMF transmission between the CCM and unity express? Is the DTMF transmitted in the siggnaling between the CCM and the unity express? Or is it using RTP payload like RFC2833 does?</font></p>
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<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">Here also a brief description of our setup:</font></p>
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<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">[ISDN30/DMS100]--------------------[Cisco AS5850]------------------[SIP proxy]--------------------[Genband SBC]-----------------[CCM]</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas"> ISDN30 SIP SIP SIP</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas"> </font></p>
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<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">Thanks in advance,</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas"> </font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas">Jan Hazenberg</font></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoPlainText"><font face="Consolas"> </font></p><br><br>
<div class="gmail_quote">2010/7/15 Nicholas Samios <span dir="ltr"><<a href="mailto:nsamios@staff.iinet.net.au">nsamios@staff.iinet.net.au</a>></span><br>
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<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">What codec are you using ? i.e. Region settings, etc.</span></p>
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<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><i><span style="COLOR: #1f497d; FONT-SIZE: 11pt">>> If you are not using a cube, you might need an enhanced IOS software MTP termination point which will allow the capturing of DTMF packets inband and process them out of band. These can run as software MTPs on the gateway (cannot use the >>UCM built in >>software MTPs for this purpose).</span></i></p>
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<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">That’s incorrect. CUCM’s inbuilt MTPs can be used to convert DTMF i.e. take OOB H245 and make it RFC2833, etc.</span></p>
<p class="MsoNormal"><a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/media.html#wp1054848" target="_blank">http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/media.html#wp1054848</a> </p>
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<p class="MsoNormal"><b><span style="FONT-SIZE: 10pt">From:</span></b><span style="FONT-SIZE: 10pt"> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Paul van den IJssel<br>
<b>Sent:</b> Thursday, July 15, 2010 4:21 PM
<div class="im"><br><b>To:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br></div>
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<div></div>
<div class="h5"><b>Subject:</b> Re: [cisco-voip] SIP Trunk from ISP DTMF issue</div></div></span>
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<div class="h5">
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<p class="MsoNormal"><span>My ISP is able to set the DTMF to</span>:</p></div>
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<p style="MARGIN: 0in 0in 0pt"><span>- SIP INFO;</span></p>
<p style="MARGIN: 0in 0in 0pt"><span>- SIP NOTIFY;</span></p>
<p style="MARGIN: 0in 0in 0pt"><span>- RFC2833 (payload type 101);</span></p>
<p style="MARGIN: 0in 0in 0pt"> </p>
<p style="MARGIN: 0in 0in 0pt"><span>If my ISP sets it to RFC2833 the CUCM should accept it right?</span></p>
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<p class="MsoNormal">2010/7/13 Matt Slaga (US) <<a href="mailto:Matt.Slaga@us.didata.com" target="_blank">Matt.Slaga@us.didata.com</a>></p>
<p class="MsoNormal">If you are not using a cube, you might need an enhanced IOS software MTP termination point which will allow the capturing of DTMF packets inband and process them out of band. These can run as software MTPs on the gateway (cannot use the UCM built in software MTPs for this purpose).<br>
<br>________________________________________<br>From: <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Nick Matthews [<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>]<br>
Sent: Tuesday, July 13, 2010 11:38 AM<br>To: Mark Holloway<br>Cc: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a>; Paul van den IJssel<br>Subject: Re: [cisco-voip] SIP Trunk from ISP DTMF issue</p>
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<p style="MARGIN-BOTTOM: 12pt" class="MsoNormal"><br>Also check the ccn subsystem sip on the CUE - make sure you're<br>configured for the DTMF method you're using everywhere else. Plus use<br>a CUBE as stated.<br>
<br>-nick<br><br>On Tue, Jul 13, 2010 at 11:17 AM, Mark Holloway <<a href="mailto:mh@markholloway.com" target="_blank">mh@markholloway.com</a>> wrote:<br>> The Genband is probably doing the same thing that CUBE would be doing. What<br>
> are the DTMF requirements from your provider? Have you confirmed that DTMF<br>> is being passed to the provider? A Wireshark capture on the public side of<br>> the S3 would verify DTMF is being passed.<br>> On Jul 13, 2010, at 7:58 AM, Bill wrote:<br>
><br>> You should not trunk directly to CUCM from your ITSP. You need to go through<br>> a CUBE router. Terminate the ITSP on the CUBE and then do a SIP trunk to<br>> your CUBE router.<br>><br>> ________________________________<br>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] On<br>
> Behalf Of Paul van den IJssel<br>> Sent: Tuesday, July 13, 2010 9:54 AM<br>> To: <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>> Subject: [cisco-voip] SIP Trunk from ISP DTMF issue<br>
><br>> Hi all,<br>><br>> Currently I'm running a acceptance test on a SIP Trunk from our ISP.<br>> Everything is working fine except the DTMF.<br>><br>> Setup:<br>> [GenBand SBC] ---- [SIP Trunk into CUCM 7.1] ---- [Cisco Unity Express 7.0]<br>
><br>> The SIP Trunk as well as the CUE are in the same region supporting only<br>> G.711. On the SIP Trunk I've tried all different kind of DTMF configurations<br>> (No Preference, RFC 2833 and OOB). We've tried to force the GenBand SBC to<br>
> only use RFC 2833 as well as the SIP Trunk. But this didn't work, not even<br>> after we added a MRGL with MTP's.<br>><br>> Is there some sort of best practive to implement DTMF over a SIP Trunk? Is<br>
> there a way I can do some debugging on the CUCM/CUE?<br>><br>> Kind regards,<br>><br>> Paul van den IJssel<br>> Digacom<br>> _______________________________________________<br>> cisco-voip mailing list<br>
> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
><br>><br>> _______________________________________________<br>> cisco-voip mailing list<br>> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
><br>><br><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a></p>
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