Outpulse digits from a voice gateway when connected to a SIP trunk?

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<p>We have a AS5350XM acting as a CUBE to provide the ingress gateway in a CVP deployment connected to AT&T IP Toll Free Service (SIP trunk). I am doing interoperability testing with AT&T and the AT&T side is set up for transfer connect (using *8 followed by a 8YY number or a POTS number). I am trying to make this work.</p>

<p>I have WireShark spanning the port back to AT&T so I can see the SIP messages from the CUBE to the carrier, and vice versa.</p>

<p>When CVP is told to perform a transfer connect the digits to outpulse are sent as a SIP INFO message to the ingress gateway. debug ccsip all on the CUBE shows these messages arriving and the CUBE replying with a 200 OK.</p>

<p>debug voice ccapi all shows that something is happening on the ingress gateway as a result of this SIP INFO message  - for example:</p>

<p>Jan 31 23:00:23.187: //-1/xxxxxxxxxxxx/SIP/Info/sip_info_parse_dtmf: Parsed digit=*, duration=100ms, retval=1<br />

Jan 31 23:00:23.187: //1625/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:<br />

   Consume mask is not set. Relaying Digit * to dstCallId 0x658<br />

Jan 31 23:00:23.187: //1625/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:<br />

   Check DTMF relay digit begin for 3way conf</p>

<p>but the upstream side to AT&T does not show any SIP INFO messages.</p>

<p>I know this works with a TDM trunk, and the gateway has certain wav files installed in the flash to make the appropriate tones and play them, going back to the carrier. But in my case, with a SIP trunk, I don't want it to do that. I want it to construct the appropriate SIP INFO message and send it to the SIP provider.</p>

<p>Something like ....</p>

<p>INFO sip:[email protected]:5060 SIP/2.0<br />

To: <sip:[email protected]>;tag=38AD31FC-4C3<br />

From: <sip:[email protected]>;tag=ds5308bd17<br />

Call-ID: [email protected]<br />

CSeq: 1 INFO<br />

Content-Length: 26<br />

Contact: <sip:a.b.c.d:5060;transport=udp><br />

Content-Type: application/dtmf-relay</p>

<p>Signal=* <br />


<p>Do you know if this is supported by IOS 12.4(22)T?</p>

<p>If so, can you tell me what additional debug I can apply to see more information (I have ccsip all and voice ccapi all)?</p>

<p>Regards,<br />


I have this problem too.
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I was doing some more research and I was reading an Avaya document on integration to the AT&T IP Toll Free service, and it describes how transfer connect works by sending a SIP 302 message (Moved Temporarily) back to the carrier.

When I search Cisco.com for 302 on a voice gateway I found

SIP Gateway Enhancements - http://www.cisco.com/en/US/docs/ios/12_2/12_2z/12_2zj/feature/guide/ftsipenh.html#wp1027434')">http://www.cisco.com/en/US/docs/ios/12_2/12_2z/12_2zj/feature/guide/ftsipenh.html#wp1027434

which talks about the redirect ip2ip (voice service) and it seems that this is what I would need, but how to make this work when the INFO messages are coming one at a time. I seem to be missing a piece.



Nicholas Matthews Sun, 02/01/2009 - 10:52

Generally SIP INFO messages are not a widely support DTMF method.  I would work on having CVP send RFC 2833 DTMF (rtp-nte).  I think you will find much greater support with that.


I believe that Cisco Gateways will interpret INFO messages correctly, but we do not support sending INFO.  In CUBE, since it is a stateful SIP stack, it is very possible that we will not forward those through.


I would see if you could use either in-band DTMF with CVP or RFC 2833.




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