<div>The difference I see between the invite and the 183 session progression from the telco is</div>
<div> </div>
<div>invite</div>
<div>a=fmtp:101 0-15</div>
<div> </div>
<div>session progression</div>
<div>a=fmtp:101 0-16</div>
<div> </div>
<div>Could this miss match in supported digits be what is causing all dtmf not to work? How can I make my cisco router support 0-16?</div>
<div> </div>
<div>Dane</div>
<div> </div>
<div><strong>Invite</strong></div>
<div><strong></strong> </div>
<div><strong></strong> </div>
<div>v=0<br>o=CiscoSystemsSIP-GW-UserAgent 2461 126 IN IP4 173.14.220.57<br>s=SIP Call<br>c=IN IP4 173.14.220.57<br>t=0 0<br>m=audio 18770 RTP/AVP 0 101 19<br>c=IN IP4 173.14.220.57<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>a=rtpmap:19 CN/8000<br>a=ptime:20</div>
<div> </div>
<div> </div>
<div> </div>
<div><strong>session progression</strong></div>
<div> </div>
<div> </div>
<div>v=0<br>o=root 5115 5115 IN IP4 64.34.181.47<br>s=session<br>c=IN IP4 64.34.181.47<br>t=0 0<br>m=audio 17646 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>
<br></div>
<div class="gmail_quote">On Tue, Oct 27, 2009 at 2:10 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com">rratliff@cisco.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div style="WORD-WRAP: break-word">Sorry this part is the actual DTMF:
<div class="im">
<div><br></div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div><br></div></div>
<div>The line you quoted is part of the SDP and references both RTP and DTMF.</div>
<div class="im">
<div>m=audio 11680 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -</div>
<div><br></div></div>
<div>The fist line means your RTP is on port 11680 and references the a:rtpmap entries for 0 and 101.</div>
<div>The second line means your RTP is g.711.</div>
<div>The 3rd line is the DTMF with a payload type of 101.</div>
<div>The 4th line means it can accept DTMF 0-16</div>
<div>The last line is pretty self explanatory (silence suppression disabled).</div>
<div><br></div>
<div>This is a very basic interpretation of the SDP info. RFC 2327 is where you want to go to get into the nitty-gritty details.</div>
<div><br><font color="#888888">
<div><span style="TEXT-TRANSFORM: none; TEXT-INDENT: 0px; BORDER-COLLAPSE: separate; FONT: medium Helvetica; WHITE-SPACE: normal; LETTER-SPACING: normal; COLOR: rgb(0,0,0); WORD-SPACING: 0px">
<div>-Ryan</div></span></div></font>
<div>
<div></div>
<div class="h5"><br>
<div>
<div>On Oct 27, 2009, at 2:00 PM, Ryan Ratliff wrote:</div><br>
<div style="WORD-WRAP: break-word">That is RFC2833 DTMF with a payload type of 101.
<div><br></div>
<div>I do know that CUBE cannot do dynamic RFC2833 payload types. It can only send the payloadType defined in the voip dial-peer. So if inbound calls use a different payloadType than outbound calls you will want to update the dial-peers accordingly.</div>
<div><br></div>
<div><br>
<div><span style="TEXT-TRANSFORM: none; TEXT-INDENT: 0px; BORDER-COLLAPSE: separate; FONT: medium Helvetica; WHITE-SPACE: normal; LETTER-SPACING: normal; WORD-SPACING: 0px">
<div>-Ryan</div></span></div><br>
<div>
<div>On Oct 27, 2009, at 12:56 PM, Dane Newman wrote:</div><br>
<div>Well I tried to switch providers just to test it out and now I am getting something back in the 183 but still no dtmf hmm</div>
<div> </div>
<div>I see they are sending me </div>
<div> </div>
<div>m=audio 11680 RTP/AVP 0 101</div>
<div> </div>
<div>How do I interperate that line?</div>
<div> </div>
<div> </div>
<div>Received:<br>SIP/2.0 183 Session Progress<br>Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK749136B;received=173.14.220.57<br>From: <<a href="mailto:sip%3A6782282221@did.voip.les.net" target="_blank">sip:6782282221@did.voip.les.net</a>>;tag=419FE94-8A1<br>
To: <<a href="mailto:sip%3A18774675464@did.voip.les.net" target="_blank">sip:18774675464@did.voip.les.net</a>>;tag=as5677a12c<br>Call-ID: <a href="mailto:AF45B372-C25911DE-80DAC992-790F56B7@173.14.220.57" target="_blank">AF45B372-C25911DE-80DAC992-790F56B7@173.14.220.57</a><br>
CSeq: 101 INVITE<br>User-Agent: LES.NET.VoIP<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <<a href="mailto:sip%3A18774675464@64.34.181.47" target="_blank">sip:18774675464@64.34.181.47</a>><br>
Content-Type: application/sdp<br>Content-Length: 214</div>
<div>v=0<br>o=root 5115 5115 IN IP4 64.34.181.47<br>s=session<br>c=IN IP4 64.34.181.47<br>t=0 0<br>m=audio 11680 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -</div>
<div>*Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Info/sipSPICheckResponse: INVITE response with no RSEQ - disable IS_REL1XX<br>*Oct 27 18:02:12.551: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container<br>
*Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1<br>SIP: Attribute mid, level 1 instance 1 not found.<br>*Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr<br>
*Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.14.220.57<br>*Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1<br>
*Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Info/sipSPIDoPtimeNegotiation: No ptime present or multiple ptime attributes that can't be handled<br>*Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1<br>
*Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved.<br>*Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Info/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF relay option<br>
*Oct 27 18:02:12.555: //1345/0008DE602400/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full named event(NE) match in fmtp list of events.<br>*Oct 27 18:02:12.555: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0<br>
*Oct 27 18:02:12.555: //1345/0008DE602400/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay<br>*Oct 27 18:02:12.555: //1345/0008DE602400/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0<br>
*Oct 27 18:02:12.555: //1345/0008DE602400/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1<br> payload_type=0, codec_bytes=160, codec=g711ulaw, dtmf_relay=rtp-nte<br> stream_type=voice+dtmf (1), dest_ip_address=64.34.181.47, dest_port=11680<br>
*Oct 27 18:02:12.555: //1345/0008DE602400/SIP/State/sipSPIChangeStreamState: Stream (callid = -1) State changed from (STREAM_DEAD) to (STREAM_ADDING)<br>*Oct 27 18:02:12.555: //1345/0008DE602400/SIP/Media/sipSPIUpdCallWithSdpInfo:<br>
Preferred Codec : g711ulaw, bytes :160<br> Preferred DTMF relay : rtp-nte<br> Preferred NTE payload : 101<br> Early Media : No<br> Delayed Media : No<br> Bridge Done : No<br>
New Media : No<br> DSP DNLD Reqd : No<br><br></div>
<div class="gmail_quote">On Tue, Oct 27, 2009 at 10:47 AM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">The 200 OK that you've pasted is confirming the CANCEL that we sent.<br>You can tell because in the 200 OK: CSeq: 102 CANCEL. You should see<br>
a 200 OK with the CSeq for 101 INVITE.<br><br>I've seen this for certain IVRs/providers - sometimes they don't<br>properly terminate a call with a 200 OK. If you were not sending an<br>SDP in your original INVITE, then you would need the PRACK setting<br>
mentioned. You have two problems, either could fix the problem: They<br>could advertise DTMF in their 183, or they could send you a 200 OK for<br>the call. It is assumed you would get DTMF in the 200 OK. It's<br>common for endpoints that support DTMF to not advertise it in the 183<br>
because you technically shouldn't need DTMF to hear ringback.<br><font color="#888888"><br>-nick<br></font>
<div>
<div></div>
<div><br>On Tue, Oct 27, 2009 at 9:30 AM, Ryan Ratliff <<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>> wrote:<br>> There is no SDP in that 200 OK so I would assume the media info is the same<br>
> as in the 183 Ringing message. You really need your ITSP to tell you what<br>> dtmf method they want you to use on your outbound calls. As Nick said they<br>> don't appear to be advertising any dtmf method at all.<br>
> -Ryan<br>> On Oct 27, 2009, at 8:51 AM, Dane Newman wrote:<br>> Is the below the ok I should be getting?<br>><br>><br>> They did send this with the first debug<br>><br>> Received:<br>> SIP/2.0 200 OK<br>
> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK51214CC<br>> From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=32DA608-109A<br>> To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>><br>
> Call-ID: <a href="mailto:9F060E11-C23511DE-8027C992-790F56B7@173.14.220.57" target="_blank">9F060E11-C23511DE-8027C992-790F56B7@173.14.220.57</a><br>> CSeq: 102 CANCEL<br>> Content-Length: 0<br>> *Oct 27 13:44:12.828: //922/009B1B501B00/SIP/Info/sipSPICheckResponse:<br>
> non-INVITE response with no RSEQ - do not disable IS_REL1XX<br>> *Oct 27 13:44:12.828: //922/009B1B501B00/SIP/Info/sipSPIIcpifUpdate:<br>> CallState: 3 Playout: 0 DiscTime:5333362 ConnTime 0<br>> *Oct 27 13:44:12.836: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:<br>
> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>> *Oct 27 13:44:12.840:<br>> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:<br>> ccsip_spi_get_msg_type returned: 2 for event 1<br>> *Oct 27 13:44:12.840:<br>
> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:<br>> context=0x00000000<br>> *Oct 27 13:44:12.840: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:<br>> Checking Invite Dialog<br>> *Oct 27 13:44:12.840: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
><br>> This with the 2nd debug<br>><br>> Received:<br>> SIP/2.0 200 OK<br>> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>> From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>
> To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>><br>> Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>
> CSeq: 102 CANCEL<br>> Content-Length: 0<br>> *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:<br>> non-INVITE response with no RSEQ - do not disable IS_REL1XX<br>> *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPIIcpifUpdate:<br>
> CallState: 3 Playout: 0 DiscTime:4913670 ConnTime 0<br>> *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:<br>> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>> *Oct 27 12:34:15.912:<br>
> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:<br>> ccsip_spi_get_msg_type returned: 2 for event 1<br>> *Oct 27 12:34:15.912:<br>> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:<br>
> context=0x00000000<br>> *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:<br>> Checking Invite Dialog<br>> *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>> Received:<br>
> SIP/2.0 487 Request Terminated<br>> To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>>;tag=3465630735-938664<br>> From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>
> Contact: <<a href="http://sip:18774675464@64.154.41.200:5060/" target="_blank">sip:18774675464@64.154.41.200:5060</a>><br>> Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>
> CSeq: 102 INVITE<br>> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>> Content-Length: 0<br>><br>> On Tue, Oct 27, 2009 at 8:43 AM, Nick Matthews <<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>> wrote:<br>
>><br>>> In the 183 Session Progress they're not advertising DTMF:<br>>><br>>> m=audio 45846 RTP/AVP 0<br>>><br>>> There should be a 100 or 101 there. Although, 183 is just ringback.<br>
>> You would want to pick up on the other side and they should send a 200<br>>> OK with a new SDP. If the other side did pick up, you need to tell<br>>> the provider that they need to send a 200 OK, because they're not.<br>
>><br>>><br>>> -nick<br>>><br>>> On Tue, Oct 27, 2009 at 7:36 AM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>><br>>> wrote:<br>>> > Nick<br>
>> ><br>>> > I removed voice-class sip asymmetric payload dtmf and added in the<br>>> > other<br>>> > line<br>>> ><br>>> > Just to state incoming dtmf works but not outbound the ITSP has told me<br>
>> > they<br>>> > are using two different sip servers/vendors for processing inbound and<br>>> > outbound<br>>> > How does this translate into what I should sent the following too?<br>>> ><br>
>> > rtp payload-type nse<br>>> > rtp payload-type nte<br>>> ><br>>> > In the debug trhe following where set<br>>> ><br>>> > rtp payload-type nse 101<br>>> > rtp payload-type nte 100<br>
>> ><br>>> > In the debug of ccsip If I am looking at it correctly I see me sending<br>>> > this<br>>> ><br>>> > *Oct 27 12:34:09.128:<br>>> > //846/8094E28C1800/SIP/Media/sipSPIAddSDPMediaPayload:<br>
>> > Preferred method of dtmf relay is: 6, with payload: 100<br>>> > *Oct 27 12:34:09.128:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIAddSDPPayloadAttributes:<br>>> > max_event 15<br>>> ><br>
>> > and<br>>> ><br>>> ><br>>> > *Oct 27 12:34:10.836:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE<br>>> > payload<br>>> > from X-cap = 0<br>
>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not<br>>> > present<br>>> > in SDP. Disable modem relay<br>>> ><br>>> ><br>
>> > Sent:<br>>> > INVITE <a href="http://sip:18774675464@64.154.41.200:5060/" target="_blank">sip:18774675464@64.154.41.200:5060</a> SIP/2.0<br>>> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A01ECD<br>
>> > Remote-Party-ID:<br>>> > <<a href="mailto:sip%3A6782282221@173.14.220.57" target="_blank">sip:6782282221@173.14.220.57</a>>;party=calling;screen=yes;privacy=off<br>>> > From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>
>> > To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>><br>>> > Date: Tue, 27 Oct 2009 12:34:09 GMT<br>>> > Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>
>> > Supported: 100rel,timer,resource-priority,replaces,sdp-anat<br>>> > Min-SE: 1800<br>>> > Cisco-Guid: 2157240972-3604177326-402682881-167847941<br>>> > User-Agent: Cisco-SIPGateway/IOS-12.x<br>
>> > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,<br>>> > SUBSCRIBE,<br>>> > NOTIFY, INFO, REGISTER<br>>> > CSeq: 101 INVITE<br>>> > Max-Forwards: 70<br>>> > Timestamp: 1256646849<br>
>> > Contact: <<a href="http://sip:6782282221@173.14.220.57:5060/" target="_blank">sip:6782282221@173.14.220.57:5060</a>><br>>> > Expires: 180<br>>> > Allow-Events: telephone-event<br>>> > Content-Type: application/sdp<br>
>> > Content-Disposition: session;handling=required<br>>> > Content-Length: 250<br>>> > v=0<br>>> > o=CiscoSystemsSIP-GW-UserAgent 7043 4703 IN IP4 173.14.220.57<br>>> > s=SIP Call<br>
>> > c=IN IP4 173.14.220.57<br>>> > t=0 0<br>>> > m=audio 16462 RTP/AVP 0 100<br>>> > c=IN IP4 173.14.220.57<br>>> > a=rtpmap:0 PCMU/8000<br>>> > a=rtpmap:100 telephone-event/8000<br>
>> > a=fmtp:100 0-15<br>>> > a=ptime:20<br>>> ><br>>> ><br>>> > Then when I do a search for fmtp again further down I see<br>>> ><br>>> > Sent:<br>>> > INVITE <a href="http://sip:18774675464@64.154.41.200:5060/" target="_blank">sip:18774675464@64.154.41.200:5060</a> SIP/2.0<br>
>> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>>> > Remote-Party-ID:<br>>> > <<a href="mailto:sip%3A6782282221@173.14.220.57" target="_blank">sip:6782282221@173.14.220.57</a>>;party=calling;screen=yes;privacy=off<br>
>> > From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>>> > To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>><br>
>> > Date: Tue, 27 Oct 2009 12:34:09 GMT<br>>> > Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>>> > Supported: 100rel,timer,resource-priority,replaces,sdp-anat<br>
>> > Min-SE: 1800<br>>> > Cisco-Guid: 2157240972-3604177326-402682881-167847941<br>>> > User-Agent: Cisco-SIPGateway/IOS-12.x<br>>> > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,<br>
>> > SUBSCRIBE,<br>>> > NOTIFY, INFO, REGISTER<br>>> > CSeq: 102 INVITE<br>>> > Max-Forwards: 70<br>>> > Timestamp: 1256646849<br>>> > Contact: <<a href="http://sip:6782282221@173.14.220.57:5060/" target="_blank">sip:6782282221@173.14.220.57:5060</a>><br>
>> > Expires: 180<br>>> > Allow-Events: telephone-event<br>>> > Proxy-Authorization: Digest<br>>> ><br>>> > username="1648245954",realm="64.154.41.110",uri="<a href="http://sip:18774675464@64.154.41.200:5060/" target="_blank">sip:18774675464@64.154.41.200:5060</a>",response="ab63d4755ff4182631ad2db0f9ed0e44",nonce="12901115532:303fa5d884d6d0a5a0328a838545395b",algorithm=MD5<br>
>> > Content-Type: application/sdp<br>>> > Content-Disposition: session;handling=required<br>>> > Content-Length: 250<br>>> > v=0<br>>> > o=CiscoSystemsSIP-GW-UserAgent 7043 4703 IN IP4 173.14.220.57<br>
>> > s=SIP Call<br>>> > c=IN IP4 173.14.220.57<br>>> > t=0 0<br>>> > m=audio 16462 RTP/AVP 0 100<br>>> > c=IN IP4 173.14.220.57<br>>> > a=rtpmap:0 PCMU/8000<br>>> > a=rtpmap:100 telephone-event/8000<br>
>> > a=fmtp:100 0-15<br>>> > a=ptime:20<br>>> > *Oct 27 12:34:09.332:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:<br>>> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>
>> > *Oct 27 12:34:09.332:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:<br>>> > ccsip_spi_get_msg_type returned: 2 for event 1<br>>> > *Oct 27 12:34:09.332:<br>>> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:<br>
>> > context=0x00000000<br>>> > *Oct 27 12:34:09.332:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:<br>>> > Checking Invite Dialog<br>>> > *Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
>> > Received:<br>>> > SIP/2.0 100 Trying<br>>> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>>> > From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>
>> > To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>><br>>> > Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>
>> > CSeq: 102 INVITE<br>>> > Content-Length: 0<br>>> > *Oct 27 12:34:09.332: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:<br>>> > INVITE response with no RSEQ - disable IS_REL1XX<br>
>> > *Oct 27 12:34:09.332: //846/8094E28C1800/SIP/State/sipSPIChangeState:<br>>> > 0x4A357FCC : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to<br>>> > (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)<br>
>> > *Oct 27 12:34:10.832:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:<br>>> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>>> > *Oct 27 12:34:10.832:<br>
>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:<br>>> > ccsip_spi_get_msg_type returned: 2 for event 1<br>>> > *Oct 27 12:34:10.832:<br>>> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:<br>
>> > context=0x00000000<br>>> > *Oct 27 12:34:10.836:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:<br>>> > Checking Invite Dialog<br>>> > *Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
>> > Received:<br>>> > SIP/2.0 183 Session Progress<br>>> > To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>>;tag=3465630735-938664<br>
>> > From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>>> > Contact: <<a href="http://sip:18774675464@64.154.41.200:5060/" target="_blank">sip:18774675464@64.154.41.200:5060</a>><br>
>> > Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>>> > CSeq: 102 INVITE<br>>> > Content-Type: application/sdp<br>
>> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>>> > Content-Length: 146<br>>> > v=0<br>>> > o=msx71 490 6110 IN IP4 64.154.41.200<br>>> > s=sip call<br>>> > c=IN IP4 64.154.41.101<br>
>> > t=0 0<br>>> > m=audio 45846 RTP/AVP 0<br>>> > a=ptime:20<br>>> > a=rtpmap:0 PCMU/8000<br>>> > *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:<br>>> > INVITE response with no RSEQ - disable IS_REL1XX<br>
>> > *Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No<br>>> > GTD<br>>> > found in inbound container<br>>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIDoMediaNegotiation:<br>
>> > Number of m-lines = 1<br>>> > SIP: Attribute mid, level 1 instance 1 not found.<br>>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media<br>
>> > already<br>>> > bound, use existing source_media_ip_addr<br>>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:<br>>> > Media src addr for stream 1 = 173.14.220.57<br>
>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:<br>>> > Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1<br>>> > *Oct 27 12:34:10.836:<br>
>> > //846/8094E28C1800/SIP/Info/sipSPIDoPtimeNegotiation:<br>>> > One ptime attribute found - value:20<br>>> > *Oct 27 12:34:10.836:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec:<br>
>> > g711ulaw ptime :20, codecbytes: 160<br>>> > *Oct 27 12:34:10.836:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec:<br>>> > g711ulaw codecbytes :160, ptime: 20<br>
>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Media/sipSPIDoPtimeNegotiation:<br>>> > Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for<br>>> > codec<br>
>> > g711ulaw<br>>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1<br>>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/sipSPICheckDynPayloadUse:<br>
>> > Dynamic payload(100) could not be reserved.<br>>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full<br>>> > named<br>>> > event(NE) match in fmtp list of events.<br>
>> > *Oct 27 12:34:10.836:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE<br>>> > payload<br>>> > from X-cap = 0<br>>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not<br>
>> > present<br>>> > in SDP. Disable modem relay<br>>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction<br>>> > attribute present or multiple direction attributes that can't be handled<br>
>> > for<br>>> > m-line:1 and num-a-lines:0<br>>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:<br>>> > Codec negotiation successful for media line 1<br>
>> > payload_type=0, codec_bytes=160, codec=g711ulaw,<br>>> > dtmf_relay=rtp-nte<br>>> > stream_type=voice+dtmf (1), dest_ip_address=64.154.41.101,<br>>> > dest_port=45846<br>
>> > *Oct 27 12:34:10.836:<br>>> > //846/8094E28C1800/SIP/State/sipSPIChangeStreamState:<br>>> > Stream (callid = -1) State changed from (STREAM_DEAD) to<br>>> > (STREAM_ADDING)<br>>> > *Oct 27 12:34:10.836:<br>
>> > //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:<br>>> > Preferred Codec : g711ulaw, bytes :160<br>>> > Preferred DTMF relay : rtp-nte<br>>> > Preferred NTE payload : 100<br>
>> > Early Media : No<br>>> > Delayed Media : No<br>>> > Bridge Done : No<br>>> > New Media : No<br>>> > DSP DNLD Reqd : No<br>
>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media<br>>> > already<br>>> > bound, use existing source_media_ip_addr<br>>> > *Oct 27 12:34:10.840:<br>
>> > //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:<br>>> > Media src addr for stream 1 = 173.14.220.57<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:<br>
>> > callId 846 peer 845 flags 0x200005 state STATE_RECD_PROCEEDING<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:<br>>> > CallID 846, sdp 0x497E29C0 channels 0x4A35926C<br>
>> > *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:<br>>> > callId 846 size 240 ptr 0x4A170B28)<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:<br>
>> > Hndl ptype 0 mline 1<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:<br>>> > Selecting<br>>> > codec g711ulaw<br>>> > *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/codec_found:<br>
>> > Codec to be matched: 5<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD<br>>> > AUDIO<br>>> > CODEC 5<br>>> > *Oct 27 12:34:10.840:<br>
>> > //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec:<br>>> > g711ulaw codecbytes :160, ptime: 20<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media<br>
>> > negotiation done:<br>>> > stream->negotiated_ptime=20,stream->negotiated_codec_bytes=160, coverted<br>>> > ptime=20 stream->mline_index=1, media_ndx=1<br>>> > *Oct 27 12:34:10.840:<br>
>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:<br>>> > Adding codec 5 ptype 0 time 20, bytes 160 as channel 0 mline 1 ss 1<br>>> > <a href="http://64.154.41.101:45846/" target="_blank">64.154.41.101:45846</a><br>
>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy<br>>> > sdp to<br>>> > channel- AFTER CODEC FILTERING:<br>>> > ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = 5<br>
>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy<br>>> > sdp to<br>>> > channel- AFTER CODEC FILTERING:<br>>> > ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = -1<br>
>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:<br>>> > callId 846 flags 0x100 state STATE_RECD_PROCEEDING<br>>> > *Oct 27 12:34:10.840:<br>
>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:<br>>> > Report initial call media<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags<br>
>> > 0x400018, ccb->pld.flags_ipip 0x200005<br>>> > *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:<br>>> > callId 846 size 240 ptr 0x4DEC000C)<br>>> > *Oct 27 12:34:10.840:<br>
>> > //846/8094E28C1800/SIP/Info/ccsip_update_srtp_caps:<br>>> > 5030: Posting Remote SRTP caps to other callleg.<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: do<br>
>> > cc_api_caps_ind()<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:<br>>> > Stream type : voice+dtmf<br>>> > Media line : 1<br>
>> > State : STREAM_ADDING (2)<br>>> > Stream address type : 1<br>>> > Callid : 846<br>>> > Negotiated Codec : g711ulaw, bytes :160<br>
>> > Nego. Codec payload : 0 (tx), 0 (rx)<br>>> > Negotiated DTMF relay : rtp-nte<br>>> > Negotiated NTE payload : 100 (tx), 100 (rx)<br>>> > Negotiated CN payload : 0<br>
>> > Media Srce Addr/Port : [173.14.220.57]:16462<br>>> > Media Dest Addr/Port : [64.154.41.101]:45846<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI<br>
>> > headers<br>>> > recvd from app container<br>>> > *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG:<br>>> > No<br>>> > QSIG Body found in inbound container<br>
>> > *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931:<br>>> > No<br>>> > RawMsg Body found in inbound container<br>>> > *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg:<br>
>> > No<br>>> > Data to form The Raw Message<br>>> > *Oct 27 12:34:10.840:<br>>> > //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress:<br>>> > ccsip_api_call_cut_progress returned: SIP_SUCCESS<br>
>> > *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/State/sipSPIChangeState:<br>>> > 0x4A357FCC : State change from (STATE_RECD_PROCEEDING,<br>>> > SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING,<br>
>> > SUBSTATE_NONE)<br>>> > *Oct 27 12:34:10.844:<br>>> > //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress: Transaction<br>>> > Complete. Lock on Facilities released.<br>>> > *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge: confID =<br>
>> > 6,<br>>> > srcCallID = 846, dstCallID = 845<br>>> > *Oct 27 12:34:10.844:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds:<br>>> > Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 846/845<br>
>> > *Oct 27 12:34:10.844:<br>>> > //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds:<br>>> > Old streamcallid=846, new streamcallid=846<br>>> > *Oct 27 12:34:10.844:<br>>> > //846/8094E28C1800/SIP/Info/ccsip_gw_set_sipspi_mode:<br>
>> > Setting SPI mode to SIP-H323<br>>> > *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge:<br>>> > xcoder_attached = 0, xmitFunc = 1131891908, ccb xmitFunc = 1131891908<br>>> > *Oct 27 12:34:10.844:<br>
>> > //846/8094E28C1800/SIP/Media/sipSPIProcessRtpSessions:<br>>> > sipSPIProcessRtpSessions<br>>> > *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIAddStream:<br>>> > Adding<br>
>> > stream 1 of type voice+dtmf (callid 846) to the VOIP RTP library<br>>> > *Oct 27 12:34:10.844:<br>>> > //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media<br>>> > already<br>
>> > bound, use existing source_media_ip_addr<br>>> > *Oct 27 12:34:10.844:<br>>> > //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:<br>>> > Media src addr for stream 1 = 173.14.220.57<br>
>> > *Oct 27 12:34:10.844:<br>>> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:<br>>> > sipSPIUpdateRtcpSession for m-line 1<br>>> > *Oct 27 12:34:10.848:<br>>> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:<br>
>> > rtcp_session info<br>>> > laddr = 173.14.220.57, lport = 16462, raddr = 64.154.41.101,<br>>> > rport=45846, do_rtcp=TRUE<br>>> > src_callid = 846, dest_callid = 845, stream type = voice+dtmf,<br>
>> > stream direction = SENDRECV<br>>> > media_ip_addr = 64.154.41.101, vrf tableid = 0 media_addr_type =<br>>> > 1<br>>> > *Oct 27 12:34:10.848:<br>>> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:<br>
>> > RTP session already created - update<br>>> > *Oct 27 12:34:10.848:<br>>> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession:<br>>> > stun is disabled for stream:4A1709F8<br>>> > *Oct 27 12:34:10.848:<br>
>> > //846/8094E28C1800/SIP/Media/sipSPIGetNewLocalMediaDirection:<br>>> > New Remote Media Direction = SENDRECV<br>>> > Present Local Media Direction = SENDRECV<br>>> > New Local Media Direction = SENDRECV<br>
>> > retVal = 0<br>>> > *Oct 27 12:34:10.848:<br>>> > //846/8094E28C1800/SIP/State/sipSPIChangeStreamState:<br>>> > Stream (callid = 846) State changed from (STREAM_ADDING) to<br>
>> > (STREAM_ACTIVE)<br>>> > *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Info/ccsip_bridge: really<br>>> > can't<br>>> > find peer_stream for<br>>> > dtmf-relay interworking<br>
>> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Entry<br>>> > *Oct 27 12:34:11.140:<br>>> > //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT<br>>> > VALUES: stream_callid=846, current_seq_num=0x23ED<br>
>> > *Oct 27 12:34:11.140:<br>>> > //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW<br>>> > VALUES:<br>>> > stream_callid=846, current_seq_num=0x11D9<br>>> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load<br>
>> > DSP<br>>> > with negotiated codec: g711ulaw, Bytes=160<br>>> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Set<br>>> > forking flag to 0x0<br>>> > *Oct 27 12:34:11.140:<br>
>> > //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode:<br>>> > Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx payload =<br>>> > 100, tx payload = 100<br>>> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>
>> > Preferred (or the one that came from DSM) modem relay=0, from CLI<br>>> > config=0<br>>> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>>> > Disabling Modem Relay...<br>
>> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>>> > Negotiation already Done. Set negotiated Modem caps and generate SDP<br>>> > Xcap<br>>> > list<br>>> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>
>> > Modem<br>>> > Relay & Passthru both disabled<br>>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>>> > nse<br>>> > payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0,<br>
>> > sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32<br>>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>>> > 1<br>>> > Active Streams<br>
>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>>> > Adding stream type (voice+dtmf) from media<br>>> > line 1 codec g711ulaw<br>>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>
>> > caps.stream_count=1,caps.stream[0].stream_type=0x3,<br>>> > caps.stream_list.xmitFunc=<br>>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>>> > voip_rtp_xmit, caps.stream_list.context=<br>
>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>>> > 0x497E0B60 (gccb)<br>>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load<br>>> > DSP<br>
>> > with codec : g711ulaw, Bytes=160, payload = 0<br>>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:<br>>> > ccsip_caps_ind: ccb->pld.flags_ipip = 0x200405<br>>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No<br>
>> > video<br>>> > caps detected in the caps posted by peer leg<br>>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:<br>>> > Setting<br>>> > CAPS_RECEIVED flag<br>
>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:<br>>> > Calling<br>>> > cc_api_caps_ack()<br>>> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ack: Set<br>
>> > forking flag to 0x0<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Entry<br>>> > *Oct 27 12:34:11.168:<br>>> > //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT<br>
>> > VALUES: stream_callid=846, current_seq_num=0x11D9<br>>> > *Oct 27 12:34:11.168:<br>>> > //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW<br>>> > VALUES:<br>>> > stream_callid=846, current_seq_num=0x11D9<br>
>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load<br>>> > DSP<br>>> > with negotiated codec: g711ulaw, Bytes=160<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Set<br>
>> > forking flag to 0x0<br>>> > *Oct 27 12:34:11.168:<br>>> > //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode:<br>>> > Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx payload =<br>
>> > 100, tx payload = 100<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>>> > Preferred (or the one that came from DSM) modem relay=0, from CLI<br>>> > config=0<br>
>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>>> > Disabling Modem Relay...<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>>> > Negotiation already Done. Set negotiated Modem caps and generate SDP<br>
>> > Xcap<br>>> > list<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>>> > Modem<br>>> > Relay & Passthru both disabled<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:<br>
>> > nse<br>>> > payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0,<br>>> > sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>
>> > 1<br>>> > Active Streams<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>>> > Adding stream type (voice+dtmf) from media<br>>> > line 1 codec g711ulaw<br>
>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>>> > caps.stream_count=1,caps.stream[0].stream_type=0x3,<br>>> > caps.stream_list.xmitFunc=<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>
>> > voip_rtp_xmit, caps.stream_list.context=<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:<br>>> > 0x497E0B60 (gccb)<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load<br>
>> > DSP<br>>> > with codec : g711ulaw, Bytes=160, payload = 0<br>>> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:<br>>> > ccsip_caps_ind: ccb->pld.flags_ipip = 0x200425<br>
>> > *Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No<br>>> > video<br>>> > caps detected in the caps posted by peer leg<br>>> > *Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Second<br>
>> > TCS<br>>> > received for transfers across trunk - set CAPS2_RECEIVED<br>>> > *Oct 27 12:34:15.876:<br>>> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession:<br>>> > stun is disabled for stream:4A1709F8<br>
>> > *Oct 27 12:34:15.876: //846/8094E28C1800/SIP/Info/ccsip_call_statistics:<br>>> > Stats are not supported for IPIP call.<br>>> > *Oct 27 12:34:15.876: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo:<br>
>> > Queued<br>>> > event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT<br>>> > *Oct 27 12:34:15.880:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:<br>>> > ccsip_spi_get_msg_type returned: 3 for event 7<br>
>> > *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Info/sipSPISendCancel:<br>>> > Associated container=0x4E310C1C to Cancel<br>>> > *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Transport/sipSPISendCancel:<br>
>> > Sending CANCEL to the transport layer<br>>> > *Oct 27 12:34:15.880:<br>>> > //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage:<br>>> > msg=0x4DF0D994,<br>>> > addr=64.154.41.200, port=5060, sentBy_port=0, is_req=1, transport=1,<br>
>> > switch=0, callBack=0x419703BC<br>>> > *Oct 27 12:34:15.880:<br>>> > //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage: Proceedable<br>>> > for<br>>> > sending msg immediately<br>
>> > *Oct 27 12:34:15.880:<br>>> > //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: switch<br>>> > transport<br>>> > is 0<br>>> > *Oct 27 12:34:15.880:<br>>> > //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: Set to send<br>
>> > the<br>>> > msg=0x4DF0D994<br>>> > *Oct 27 12:34:15.880:<br>>> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting<br>>> > send<br>>> > for msg=0x4DF0D994, addr=64.154.41.200, port=5060, connId=2 for UDP<br>
>> > *Oct 27 12:34:15.880:<br>>> > //846/8094E28C1800/SIP/Info/sentCancelDisconnecting:<br>>> > Sent Cancel Request, starting CancelWaitResponseTimer<br>>> > *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/State/sipSPIChangeState:<br>
>> > 0x4A357FCC : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE)<br>>> > to<br>>> > (STATE_DISCONNECTING, SUBSTATE_NONE)<br>>> > *Oct 27 12:34:15.888: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
>> > Sent:<br>>> > CANCEL <a href="http://sip:18774675464@64.154.41.200:5060/" target="_blank">sip:18774675464@64.154.41.200:5060</a> SIP/2.0<br>>> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>
>> > From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>>> > To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>><br>
>> > Date: Tue, 27 Oct 2009 12:34:09 GMT<br>>> > Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>>> > CSeq: 102 CANCEL<br>
>> > Max-Forwards: 70<br>>> > Timestamp: 1256646855<br>>> > Reason: Q.850;cause=16<br>>> > Content-Length: 0<br>>> > *Oct 27 12:34:15.900:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:<br>
>> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>>> > *Oct 27 12:34:15.900:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:<br>>> > ccsip_spi_get_msg_type returned: 2 for event 1<br>
>> > *Oct 27 12:34:15.900:<br>>> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:<br>>> > context=0x00000000<br>>> > *Oct 27 12:34:15.900:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:<br>
>> > Checking Invite Dialog<br>>> > *Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>>> > Received:<br>>> > SIP/2.0 200 OK<br>>> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>
>> > From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>>> > To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>><br>
>> > Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>>> > CSeq: 102 CANCEL<br>>> > Content-Length: 0<br>
>> > *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:<br>>> > non-INVITE response with no RSEQ - do not disable IS_REL1XX<br>>> > *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPIIcpifUpdate:<br>
>> > CallState: 3 Playout: 0 DiscTime:4913670 ConnTime 0<br>>> > *Oct 27 12:34:15.912:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:<br>>> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>
>> > *Oct 27 12:34:15.912:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:<br>>> > ccsip_spi_get_msg_type returned: 2 for event 1<br>>> > *Oct 27 12:34:15.912:<br>>> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:<br>
>> > context=0x00000000<br>>> > *Oct 27 12:34:15.912:<br>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:<br>>> > Checking Invite Dialog<br>>> > *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
>> ><br>>> > On Mon, Oct 26, 2009 at 7:36 PM, Nick Matthews <<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>><br>>> > wrote:<br>>> >><br>>> >> You would want to check the SDP of 200 OK the provider sends for your<br>
>> >> outgoing call. It will list the payload type for the dtmf in the<br>>> >> format a=fmtp 101 1-16, or something similar. You want to find out<br>>> >> what payload type they are advertising (or if they are at all). It<br>
>> >> would be worth checking the incoming INVITE from them to see what<br>>> >> they're using when they send the first SDP.<br>>> >><br>>> >> On that note, I would also remove the asymmetric payload command - to<br>
>> >> my knowledge it doesn't do what you're expecting it to. You may want<br>>> >> to try this command:<br>>> >> voice-class sip dtmf-relay force rtp-nte<br>>> >><br>
>> >><br>>> >> -nick<br>>> >><br>>> >> On Mon, Oct 26, 2009 at 5:16 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>><br>
>> >> wrote:<br>>> >> > Hello,<br>>> >> ><br>>> >> > I am having an issue with dtmf working outbound. Inbound dtmf works<br>>> >> > fine.<br>>> >> > It took some playing around with it. At first it didnt work till the<br>
>> >> > payload was ajusted. I am now trying to get outbound dtmf working<br>>> >> > properly.<br>>> >> ><br>>> >> > On my 2821 I debugged voip rtp session named-events and then made a<br>
>> >> > call<br>>> >> > to<br>>> >> > a 1800 number and hit digits. I didn't see any dtmf output on the<br>>> >> > router<br>>> >> > nothing showed up in the debug. Does this mean I can safely asume<br>
>> >> > that<br>>> >> > the<br>>> >> > problem for right now is not on the ITSP side but on my side since<br>>> >> > dtmf<br>>> >> > is<br>>> >> > not being sent down the sip trunk?<br>
>> >> ><br>>> >> > I have my cuc 7.x configured to talk to my 2821 via h323. The<br>>> >> > configuration<br>>> >> > of the cisco 2821 is shown below. Does anyone have any ideas what I<br>
>> >> > can<br>>> >> > do<br>>> >> > so dtmf digits process properly outbound?<br>>> >> ><br>>> >> > The settings in my cuc 7.x to add the gateway h323 are<br>
>> >> ><br>>> >> > h323 cucm gateway configuratration<br>>> >> > Signaling Port 1720<br>>> >> > media termination point required yes<br>>> >> > retry video call as auto yes<br>
>> >> > wait for far end h.245 terminal capability set yes<br>>> >> > transmit utf-8 calling party name no<br>>> >> > h.235 pass through allowed no<br>>> >> > significant digits all<br>
>> >> > redirect number IT deliver - inbound no<br>>> >> > enable inbound faststart yes<br>>> >> > display IE deliver no<br>>> >> > redirect nunmber IT deliver - outbound no<br>
>> >> > enable outbound faststart yes<br>>> >> ><br>>> >> ><br>>> >> > voice service voip<br>>> >> > allow-connections h323 to h323<br>>> >> > allow-connections h323 to sip<br>
>> >> > allow-connections sip to h323<br>>> >> > allow-connections sip to sip<br>>> >> > fax protocol pass-through g711ulaw<br>>> >> > h323<br>>> >> > emptycapability<br>
>> >> > h225 id-passthru<br>>> >> > h245 passthru tcsnonstd-passthru<br>>> >> > sip<br>>> >> ><br>>> >> ><br>>> >> > voice class h323 50<br>
>> >> > h225 timeout tcp establish 3<br>>> >> > !<br>>> >> > !<br>>> >> > !<br>>> >> > !<br>>> >> > !<br>>> >> > !<br>
>> >> > !<br>>> >> > !<br>>> >> > !<br>>> >> > !<br>>> >> > !<br>>> >> > voice translation-rule 1<br>>> >> > rule 1 /.*/ /190/<br>
>> >> > !<br>>> >> > voice translation-rule 2<br>>> >> > rule 1 /.*/ /1&/<br>>> >> > !<br>>> >> > !<br>>> >> > voice translation-profile aa<br>
>> >> > translate called 1<br>>> >> > !<br>>> >> > voice translation-profile addone<br>>> >> > translate called 2<br>>> >> > !<br>>> >> > !<br>
>> >> > voice-card 0<br>>> >> > dspfarm<br>>> >> > dsp services dspfarm<br>>> >> > !<br>>> >> > !<br>>> >> > sccp local GigabitEthernet0/1<br>
>> >> > sccp ccm 10.1.80.11 identifier 2 version 7.0<br>>> >> > sccp ccm 10.1.80.10 identifier 1 version 7.0<br>>> >> > sccp<br>>> >> > !<br>>> >> > sccp ccm group 1<br>
>> >> > associate ccm 1 priority 1<br>>> >> > associate ccm 2 priority 2<br>>> >> > associate profile 1 register 2821transcode<br>>> >> > !<br>>> >> > dspfarm profile 1 transcode<br>
>> >> > codec g711ulaw<br>>> >> > codec g711alaw<br>>> >> > codec g729ar8<br>>> >> > codec g729abr8<br>>> >> > codec g729r8<br>>> >> > maximum sessions 4<br>
>> >> > associate application SCCP<br>>> >> > !<br>>> >> > !<br>>> >> > dial-peer voice 100 voip<br>>> >> > description AA Publisher<br>>> >> > preference 1<br>
>> >> > destination-pattern 1..<br>>> >> > voice-class h323 50<br>>> >> > session target ipv4:10.1.80.10<br>>> >> > dtmf-relay h245-alphanumeric<br>>> >> > codec g711ulaw<br>
>> >> > no vad<br>>> >> > !<br>>> >> > dial-peer voice 1000 voip<br>>> >> > description incoming Call<br>>> >> > translation-profile incoming aa<br>
>> >> > preference 1<br>>> >> > rtp payload-type nse 101<br>>> >> > rtp payload-type nte 100<br>>> >> > incoming called-number 6782282221<br>>> >> > dtmf-relay rtp-nte<br>
>> >> > codec g711ulaw<br>>> >> > ip qos dscp cs5 media<br>>> >> > ip qos dscp cs5 signaling<br>>> >> > no vad<br>>> >> > !<br>>> >> > dial-peer voice 101 voip<br>
>> >> > description AA Subscriber<br>>> >> > preference 2<br>>> >> > destination-pattern 1..<br>>> >> > voice-class h323 50<br>>> >> > session target ipv4:10.1.80.11<br>
>> >> > dtmf-relay h245-alphanumeric<br>>> >> > codec g711ulaw<br>>> >> > no vad<br>>> >> > !<br>>> >> > dial-peer voice 2000 voip<br>>> >> > description outbound<br>
>> >> > translation-profile outgoing addone<br>>> >> > preference 1<br>>> >> > destination-pattern .T<br>>> >> > rtp payload-type nse 101<br>>> >> > rtp payload-type nte 100<br>
>> >> > voice-class sip asymmetric payload dtmf<br>>> >> > session protocol sipv2<br>>> >> > session target ipv4:64.154.41.200<br>>> >> > dtmf-relay rtp-nte<br>
>> >> > codec g711ulaw<br>>> >> > no vad<br>>> >> > !<br>>> >> > !<br>>> >> > sip-ua<br>>> >> > credentials username ***** password 7 ***** realm <a href="http://sip.talkinip.net/" target="_blank">sip.talkinip.net</a><br>
>> >> > authentication username ***** password 7 *****<br>>> >> > authentication username ***** password 7 ***** realm<br>>> >> > <a href="http://sip.talkinip.net/" target="_blank">sip.talkinip.net</a><br>
>> >> > set pstn-cause 3 sip-status 486<br>>> >> > set pstn-cause 34 sip-status 486<br>>> >> > set pstn-cause 47 sip-status 486<br>>> >> > registrar dns:<a href="http://sip.talkinip.net/" target="_blank">sip.talkinip.net</a> expires 60<br>
>> >> > sip-server dns:<a href="http://sip.talkinip.net:5060/" target="_blank">sip.talkinip.net:5060</a><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>><br>> _______________________________________________<br>> cisco-voip mailing list<br>> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
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