<div>Thanks for this response.</div>
<div> </div>
<div>I set it as below and restarted the ccm service on both pub and sub. Still didnt work. I also tried to put</div>
<div> </div>
<div>voice-class Sip rel1xx supported 100rel</div>
<div> </div>
<div>under the dial peer and still didnt work but thanks for the response</div>
<div> </div>
<div>Dane<br><br></div>
<div class="gmail_quote">On Tue, Oct 27, 2009 at 8:48 AM, Chris Osborne (US) <span dir="ltr"><<a href="mailto:chris.osborne@us.didata.com">chris.osborne@us.didata.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 lang="EN-US" vlink="purple" link="blue">
<div>
<p>You could also try enabling this under Clusterwide Params (Device-SIP) , by default its turned off.</p>
<p> </p>
<p><img src="cid:image001.png@01CA56E2.3912EE20" width="984" height="43"></p>
<p> </p>
<p> </p>
<p> </p>
<p>I had a similar issue where when we called an at&t own 1800 number it would just ring forever because of how they handle calls from other carriers via sip. This allowed those calls to start working.</p>
<p> </p>
<p>Chris Osborne CCNP</p>
<p>Sr Engineer - US Southeast</p>
<p>Dimension Data </p>
<div>
<div></div>
<div class="h5">
<p> </p>
<p> </p>
<p>-----Original Message-----<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 Behalf Of Nick Matthews<br>
Sent: Tuesday, October 27, 2009 8:44 AM<br>To: Dane Newman<br>Cc: cisco-voip<br>Subject: Re: [cisco-voip] dtmf from cucm to 2821 cube to sip trunk</p>
<p> </p>
<p>In the 183 Session Progress they're not advertising DTMF:</p>
<p> </p>
<p>m=audio 45846 RTP/AVP 0</p>
<p> </p>
<p>There should be a 100 or 101 there. Although, 183 is just ringback.</p>
<p>You would want to pick up on the other side and they should send a 200</p>
<p>OK with a new SDP. If the other side did pick up, you need to tell</p>
<p>the provider that they need to send a 200 OK, because they're not.</p>
<p> </p>
<p> </p>
<p>-nick</p>
<p> </p>
<p>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>> wrote:</p>
<p>> Nick</p>
<p>> </p>
<p>> I removed voice-class sip asymmetric payload dtmf and added in the other</p>
<p>> line</p>
<p>> </p>
<p>> Just to state incoming dtmf works but not outbound the ITSP has told me they</p>
<p>> are using two different sip servers/vendors for processing inbound and</p>
<p>> outbound</p>
<p>> How does this translate into what I should sent the following too?</p>
<p>> </p>
<p>> rtp payload-type nse</p>
<p>> rtp payload-type nte</p>
<p>> </p>
<p>> In the debug trhe following where set</p>
<p>> </p>
<p>> rtp payload-type nse 101</p>
<p>> rtp payload-type nte 100</p>
<p>> </p>
<p>> In the debug of ccsip If I am looking at it correctly I see me sending this</p>
<p>> </p>
<p>> *Oct 27 12:34:09.128: //846/8094E28C1800/SIP/Media/sipSPIAddSDPMediaPayload:</p>
<p>> Preferred method of dtmf relay is: 6, with payload: 100</p>
<p>> *Oct 27 12:34:09.128:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPIAddSDPPayloadAttributes:</p>
<p>> max_event 15</p>
<p>> </p>
<p>> and</p>
<p>> </p>
<p>> </p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload</p>
<p>> from X-cap = 0</p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not present</p>
<p>> in SDP. Disable modem relay</p>
<p>> </p>
<p>> </p>
<p>> Sent:</p>
<p>> INVITE <a href="http://sip:18774675464@64.154.41.200:5060" target="_blank">sip:18774675464@64.154.41.200:5060</a> SIP/2.0</p>
<p>> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A01ECD</p>
<p>> Remote-Party-ID:</p>
<p>> <<a href="mailto:sip%3A6782282221@173.14.220.57" target="_blank">sip:6782282221@173.14.220.57</a>>;party=calling;screen=yes;privacy=off</p>
<p>> From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6</p>
<p>> To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>></p>
<p>> Date: Tue, 27 Oct 2009 12:34:09 GMT</p>
<p>> Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a></p>
<p>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat</p>
<p>> Min-SE: 1800</p>
<p>> Cisco-Guid: 2157240972-3604177326-402682881-167847941</p>
<p>> User-Agent: Cisco-SIPGateway/IOS-12.x</p>
<p>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,</p>
<p>> NOTIFY, INFO, REGISTER</p>
<p>> CSeq: 101 INVITE</p>
<p>> Max-Forwards: 70</p>
<p>> Timestamp: 1256646849</p>
<p>> Contact: <<a href="http://sip:6782282221@173.14.220.57:5060" target="_blank">sip:6782282221@173.14.220.57:5060</a>></p>
<p>> Expires: 180</p>
<p>> Allow-Events: telephone-event</p>
<p>> Content-Type: application/sdp</p>
<p>> Content-Disposition: session;handling=required</p>
<p>> Content-Length: 250</p>
<p>> v=0</p>
<p>> o=CiscoSystemsSIP-GW-UserAgent 7043 4703 IN IP4 173.14.220.57</p>
<p>> s=SIP Call</p>
<p>> c=IN IP4 173.14.220.57</p>
<p>> t=0 0</p>
<p>> m=audio 16462 RTP/AVP 0 100</p>
<p>> c=IN IP4 173.14.220.57</p>
<p>> a=rtpmap:0 PCMU/8000</p>
<p>> a=rtpmap:100 telephone-event/8000</p>
<p>> a=fmtp:100 0-15</p>
<p>> a=ptime:20</p>
<p>> </p>
<p>> </p>
<p>> Then when I do a search for fmtp again further down I see</p>
<p>> </p>
<p>> Sent:</p>
<p>> INVITE <a href="http://sip:18774675464@64.154.41.200:5060" target="_blank">sip:18774675464@64.154.41.200:5060</a> SIP/2.0</p>
<p>> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE</p>
<p>> Remote-Party-ID:</p>
<p>> <<a href="mailto:sip%3A6782282221@173.14.220.57" target="_blank">sip:6782282221@173.14.220.57</a>>;party=calling;screen=yes;privacy=off</p>
<p>> From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6</p>
<p>> To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>></p>
<p>> Date: Tue, 27 Oct 2009 12:34:09 GMT</p>
<p>> Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a></p>
<p>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat</p>
<p>> Min-SE: 1800</p>
<p>> Cisco-Guid: 2157240972-3604177326-402682881-167847941</p>
<p>> User-Agent: Cisco-SIPGateway/IOS-12.x</p>
<p>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,</p>
<p>> NOTIFY, INFO, REGISTER</p>
<p>> CSeq: 102 INVITE</p>
<p>> Max-Forwards: 70</p>
<p>> Timestamp: 1256646849</p>
<p>> Contact: <<a href="http://sip:6782282221@173.14.220.57:5060" target="_blank">sip:6782282221@173.14.220.57:5060</a>></p>
<p>> Expires: 180</p>
<p>> Allow-Events: telephone-event</p>
<p>> Proxy-Authorization: Digest</p>
<p>> 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</p>
<p>> Content-Type: application/sdp</p>
<p>> Content-Disposition: session;handling=required</p>
<p>> Content-Length: 250</p>
<p>> v=0</p>
<p>> o=CiscoSystemsSIP-GW-UserAgent 7043 4703 IN IP4 173.14.220.57</p>
<p>> s=SIP Call</p>
<p>> c=IN IP4 173.14.220.57</p>
<p>> t=0 0</p>
<p>> m=audio 16462 RTP/AVP 0 100</p>
<p>> c=IN IP4 173.14.220.57</p>
<p>> a=rtpmap:0 PCMU/8000</p>
<p>> a=rtpmap:100 telephone-event/8000</p>
<p>> a=fmtp:100 0-15</p>
<p>> a=ptime:20</p>
<p>> *Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:</p>
<p>> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060</p>
<p>> *Oct 27 12:34:09.332:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:</p>
<p>> ccsip_spi_get_msg_type returned: 2 for event 1</p>
<p>> *Oct 27 12:34:09.332:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:</p>
<p>> context=0x00000000</p>
<p>> *Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:</p>
<p>> Checking Invite Dialog</p>
<p>> *Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>> Received:</p>
<p>> SIP/2.0 100 Trying</p>
<p>> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE</p>
<p>> From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6</p>
<p>> To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>></p>
<p>> Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a></p>
<p>> CSeq: 102 INVITE</p>
<p>> Content-Length: 0</p>
<p>> *Oct 27 12:34:09.332: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:</p>
<p>> INVITE response with no RSEQ - disable IS_REL1XX</p>
<p>> *Oct 27 12:34:09.332: //846/8094E28C1800/SIP/State/sipSPIChangeState:</p>
<p>> 0x4A357FCC : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to</p>
<p>> (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)</p>
<p>> *Oct 27 12:34:10.832: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:</p>
<p>> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060</p>
<p>> *Oct 27 12:34:10.832:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:</p>
<p>> ccsip_spi_get_msg_type returned: 2 for event 1</p>
<p>> *Oct 27 12:34:10.832:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:</p>
<p>> context=0x00000000</p>
<p>> *Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:</p>
<p>> Checking Invite Dialog</p>
<p>> *Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>> Received:</p>
<p>> SIP/2.0 183 Session Progress</p>
<p>> To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>>;tag=3465630735-938664</p>
<p>> From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6</p>
<p>> Contact: <<a href="http://sip:18774675464@64.154.41.200:5060" target="_blank">sip:18774675464@64.154.41.200:5060</a>></p>
<p>> Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a></p>
<p>> CSeq: 102 INVITE</p>
<p>> Content-Type: application/sdp</p>
<p>> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE</p>
<p>> Content-Length: 146</p>
<p>> v=0</p>
<p>> o=msx71 490 6110 IN IP4 64.154.41.200</p>
<p>> s=sip call</p>
<p>> c=IN IP4 64.154.41.101</p>
<p>> t=0 0</p>
<p>> m=audio 45846 RTP/AVP 0</p>
<p>> a=ptime:20</p>
<p>> a=rtpmap:0 PCMU/8000</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:</p>
<p>> INVITE response with no RSEQ - disable IS_REL1XX</p>
<p>> *Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD</p>
<p>> found in inbound container</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoMediaNegotiation:</p>
<p>> Number of m-lines = 1</p>
<p>> SIP: Attribute mid, level 1 instance 1 not found.</p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already</p>
<p>> bound, use existing source_media_ip_addr</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:</p>
<p>> Media src addr for stream 1 = 173.14.220.57</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:</p>
<p>> Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoPtimeNegotiation:</p>
<p>> One ptime attribute found - value:20</p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec:</p>
<p>> g711ulaw ptime :20, codecbytes: 160</p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec:</p>
<p>> g711ulaw codecbytes :160, ptime: 20</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPIDoPtimeNegotiation:</p>
<p>> Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec</p>
<p>> g711ulaw</p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPICheckDynPayloadUse:</p>
<p>> Dynamic payload(100) could not be reserved.</p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full named</p>
<p>> event(NE) match in fmtp list of events.</p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload</p>
<p>> from X-cap = 0</p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not present</p>
<p>> in SDP. Disable modem relay</p>
<p>> *Oct 27 12:34:10.836:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction</p>
<p>> attribute present or multiple direction attributes that can't be handled for</p>
<p>> m-line:1 and num-a-lines:0</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:</p>
<p>> Codec negotiation successful for media line 1</p>
<p>> payload_type=0, codec_bytes=160, codec=g711ulaw, dtmf_relay=rtp-nte</p>
<p>> stream_type=voice+dtmf (1), dest_ip_address=64.154.41.101,</p>
<p>> dest_port=45846</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/State/sipSPIChangeStreamState:</p>
<p>> Stream (callid = -1) State changed from (STREAM_DEAD) to (STREAM_ADDING)</p>
<p>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:</p>
<p>> Preferred Codec : g711ulaw, bytes :160</p>
<p>> Preferred DTMF relay : rtp-nte</p>
<p>> Preferred NTE payload : 100</p>
<p>> Early Media : No</p>
<p>> Delayed Media : No</p>
<p>> Bridge Done : No</p>
<p>> New Media : No</p>
<p>> DSP DNLD Reqd : No</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already</p>
<p>> bound, use existing source_media_ip_addr</p>
<p>> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:</p>
<p>> Media src addr for stream 1 = 173.14.220.57</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:</p>
<p>> callId 846 peer 845 flags 0x200005 state STATE_RECD_PROCEEDING</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:</p>
<p>> CallID 846, sdp 0x497E29C0 channels 0x4A35926C</p>
<p>> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:</p>
<p>> callId 846 size 240 ptr 0x4A170B28)</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:</p>
<p>> Hndl ptype 0 mline 1</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting</p>
<p>> codec g711ulaw</p>
<p>> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/codec_found:</p>
<p>> Codec to be matched: 5</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO</p>
<p>> CODEC 5</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec:</p>
<p>> g711ulaw codecbytes :160, ptime: 20</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media</p>
<p>> negotiation done:</p>
<p>> stream->negotiated_ptime=20,stream->negotiated_codec_bytes=160, coverted</p>
<p>> ptime=20 stream->mline_index=1, media_ndx=1</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:</p>
<p>> Adding codec 5 ptype 0 time 20, bytes 160 as channel 0 mline 1 ss 1</p>
<p>> <a href="http://64.154.41.101:45846/" target="_blank">64.154.41.101:45846</a></p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to</p>
<p>> channel- AFTER CODEC FILTERING:</p>
<p>> ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = 5</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to</p>
<p>> channel- AFTER CODEC FILTERING:</p>
<p>> ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = -1</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:</p>
<p>> callId 846 flags 0x100 state STATE_RECD_PROCEEDING</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:</p>
<p>> Report initial call media</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags</p>
<p>> 0x400018, ccb->pld.flags_ipip 0x200005</p>
<p>> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:</p>
<p>> callId 846 size 240 ptr 0x4DEC000C)</p>
<p>> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/ccsip_update_srtp_caps:</p>
<p>> 5030: Posting Remote SRTP caps to other callleg.</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: do</p>
<p>> cc_api_caps_ind()</p>
<p>> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:</p>
<p>> Stream type : voice+dtmf</p>
<p>> Media line : 1</p>
<p>> State : STREAM_ADDING (2)</p>
<p>> Stream address type : 1</p>
<p>> Callid : 846</p>
<p>> Negotiated Codec : g711ulaw, bytes :160</p>
<p>> Nego. Codec payload : 0 (tx), 0 (rx)</p>
<p>> Negotiated DTMF relay : rtp-nte</p>
<p>> Negotiated NTE payload : 100 (tx), 100 (rx)</p>
<p>> Negotiated CN payload : 0</p>
<p>> Media Srce Addr/Port : [173.14.220.57]:16462</p>
<p>> Media Dest Addr/Port : [64.154.41.101]:45846</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers</p>
<p>> recvd from app container</p>
<p>> *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No</p>
<p>> QSIG Body found in inbound container</p>
<p>> *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No</p>
<p>> RawMsg Body found in inbound container</p>
<p>> *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No</p>
<p>> Data to form The Raw Message</p>
<p>> *Oct 27 12:34:10.840:</p>
<p>> //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress:</p>
<p>> ccsip_api_call_cut_progress returned: SIP_SUCCESS</p>
<p>> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/State/sipSPIChangeState:</p>
<p>> 0x4A357FCC : State change from (STATE_RECD_PROCEEDING,</p>
<p>> SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_NONE)</p>
<p>> *Oct 27 12:34:10.844:</p>
<p>> //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress: Transaction</p>
<p>> Complete. Lock on Facilities released.</p>
<p>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge: confID = 6,</p>
<p>> srcCallID = 846, dstCallID = 845</p>
<p>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds:</p>
<p>> Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 846/845</p>
<p>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds:</p>
<p>> Old streamcallid=846, new streamcallid=846</p>
<p>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_gw_set_sipspi_mode:</p>
<p>> Setting SPI mode to SIP-H323</p>
<p>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge:</p>
<p>> xcoder_attached = 0, xmitFunc = 1131891908, ccb xmitFunc = 1131891908</p>
<p>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIProcessRtpSessions:</p>
<p>> sipSPIProcessRtpSessions</p>
<p>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIAddStream: Adding</p>
<p>> stream 1 of type voice+dtmf (callid 846) to the VOIP RTP library</p>
<p>> *Oct 27 12:34:10.844:</p>
<p>> //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already</p>
<p>> bound, use existing source_media_ip_addr</p>
<p>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:</p>
<p>> Media src addr for stream 1 = 173.14.220.57</p>
<p>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:</p>
<p>> sipSPIUpdateRtcpSession for m-line 1</p>
<p>> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:</p>
<p>> rtcp_session info</p>
<p>> laddr = 173.14.220.57, lport = 16462, raddr = 64.154.41.101,</p>
<p>> rport=45846, do_rtcp=TRUE</p>
<p>> src_callid = 846, dest_callid = 845, stream type = voice+dtmf,</p>
<p>> stream direction = SENDRECV</p>
<p>> media_ip_addr = 64.154.41.101, vrf tableid = 0 media_addr_type = 1</p>
<p>> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:</p>
<p>> RTP session already created - update</p>
<p>> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession:</p>
<p>> stun is disabled for stream:4A1709F8</p>
<p>> *Oct 27 12:34:10.848:</p>
<p>> //846/8094E28C1800/SIP/Media/sipSPIGetNewLocalMediaDirection:</p>
<p>> New Remote Media Direction = SENDRECV</p>
<p>> Present Local Media Direction = SENDRECV</p>
<p>> New Local Media Direction = SENDRECV</p>
<p>> retVal = 0</p>
<p>> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/State/sipSPIChangeStreamState:</p>
<p>> Stream (callid = 846) State changed from (STREAM_ADDING) to</p>
<p>> (STREAM_ACTIVE)</p>
<p>> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Info/ccsip_bridge: really can't</p>
<p>> find peer_stream for</p>
<p>> dtmf-relay interworking</p>
<p>> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Entry</p>
<p>> *Oct 27 12:34:11.140:</p>
<p>> //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT</p>
<p>> VALUES: stream_callid=846, current_seq_num=0x23ED</p>
<p>> *Oct 27 12:34:11.140:</p>
<p>> //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES:</p>
<p>> stream_callid=846, current_seq_num=0x11D9</p>
<p>> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP</p>
<p>> with negotiated codec: g711ulaw, Bytes=160</p>
<p>> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Set</p>
<p>> forking flag to 0x0</p>
<p>> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode:</p>
<p>> Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx payload =</p>
<p>> 100, tx payload = 100</p>
<p>> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:</p>
<p>> Preferred (or the one that came from DSM) modem relay=0, from CLI config=0</p>
<p>> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:</p>
<p>> Disabling Modem Relay...</p>
<p>> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:</p>
<p>> Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap</p>
<p>> list</p>
<p>> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Modem</p>
<p>> Relay & Passthru both disabled</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: nse</p>
<p>> payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0,</p>
<p>> sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 1</p>
<p>> Active Streams</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:</p>
<p>> Adding stream type (voice+dtmf) from media</p>
<p>> line 1 codec g711ulaw</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:</p>
<p>> caps.stream_count=1,caps.stream[0].stream_type=0x3,</p>
<p>> caps.stream_list.xmitFunc=</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:</p>
<p>> voip_rtp_xmit, caps.stream_list.context=</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:</p>
<p>> 0x497E0B60 (gccb)</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP</p>
<p>> with codec : g711ulaw, Bytes=160, payload = 0</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:</p>
<p>> ccsip_caps_ind: ccb->pld.flags_ipip = 0x200405</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No video</p>
<p>> caps detected in the caps posted by peer leg</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Setting</p>
<p>> CAPS_RECEIVED flag</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Calling</p>
<p>> cc_api_caps_ack()</p>
<p>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ack: Set</p>
<p>> forking flag to 0x0</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Entry</p>
<p>> *Oct 27 12:34:11.168:</p>
<p>> //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT</p>
<p>> VALUES: stream_callid=846, current_seq_num=0x11D9</p>
<p>> *Oct 27 12:34:11.168:</p>
<p>> //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES:</p>
<p>> stream_callid=846, current_seq_num=0x11D9</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP</p>
<p>> with negotiated codec: g711ulaw, Bytes=160</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Set</p>
<p>> forking flag to 0x0</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode:</p>
<p>> Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx payload =</p>
<p>> 100, tx payload = 100</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:</p>
<p>> Preferred (or the one that came from DSM) modem relay=0, from CLI config=0</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:</p>
<p>> Disabling Modem Relay...</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:</p>
<p>> Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap</p>
<p>> list</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Modem</p>
<p>> Relay & Passthru both disabled</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: nse</p>
<p>> payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0,</p>
<p>> sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 1</p>
<p>> Active Streams</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:</p>
<p>> Adding stream type (voice+dtmf) from media</p>
<p>> line 1 codec g711ulaw</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:</p>
<p>> caps.stream_count=1,caps.stream[0].stream_type=0x3,</p>
<p>> caps.stream_list.xmitFunc=</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:</p>
<p>> voip_rtp_xmit, caps.stream_list.context=</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:</p>
<p>> 0x497E0B60 (gccb)</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP</p>
<p>> with codec : g711ulaw, Bytes=160, payload = 0</p>
<p>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:</p>
<p>> ccsip_caps_ind: ccb->pld.flags_ipip = 0x200425</p>
<p>> *Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No video</p>
<p>> caps detected in the caps posted by peer leg</p>
<p>> *Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Second TCS</p>
<p>> received for transfers across trunk - set CAPS2_RECEIVED</p>
<p>> *Oct 27 12:34:15.876: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession:</p>
<p>> stun is disabled for stream:4A1709F8</p>
<p>> *Oct 27 12:34:15.876: //846/8094E28C1800/SIP/Info/ccsip_call_statistics:</p>
<p>> Stats are not supported for IPIP call.</p>
<p>> *Oct 27 12:34:15.876: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued</p>
<p>> event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT</p>
<p>> *Oct 27 12:34:15.880:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:</p>
<p>> ccsip_spi_get_msg_type returned: 3 for event 7</p>
<p>> *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Info/sipSPISendCancel:</p>
<p>> Associated container=0x4E310C1C to Cancel</p>
<p>> *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Transport/sipSPISendCancel:</p>
<p>> Sending CANCEL to the transport layer</p>
<p>> *Oct 27 12:34:15.880:</p>
<p>> //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage: msg=0x4DF0D994,</p>
<p>> addr=64.154.41.200, port=5060, sentBy_port=0, is_req=1, transport=1,</p>
<p>> switch=0, callBack=0x419703BC</p>
<p>> *Oct 27 12:34:15.880:</p>
<p>> //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage: Proceedable for</p>
<p>> sending msg immediately</p>
<p>> *Oct 27 12:34:15.880:</p>
<p>> //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: switch transport</p>
<p>> is 0</p>
<p>> *Oct 27 12:34:15.880:</p>
<p>> //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: Set to send the</p>
<p>> msg=0x4DF0D994</p>
<p>> *Oct 27 12:34:15.880:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send</p>
<p>> for msg=0x4DF0D994, addr=64.154.41.200, port=5060, connId=2 for UDP</p>
<p>> *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Info/sentCancelDisconnecting:</p>
<p>> Sent Cancel Request, starting CancelWaitResponseTimer</p>
<p>> *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/State/sipSPIChangeState:</p>
<p>> 0x4A357FCC : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE) to</p>
<p>> (STATE_DISCONNECTING, SUBSTATE_NONE)</p>
<p>> *Oct 27 12:34:15.888: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>> Sent:</p>
<p>> CANCEL <a href="http://sip:18774675464@64.154.41.200:5060" target="_blank">sip:18774675464@64.154.41.200:5060</a> SIP/2.0</p>
<p>> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE</p>
<p>> From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6</p>
<p>> To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>></p>
<p>> Date: Tue, 27 Oct 2009 12:34:09 GMT</p>
<p>> Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a></p>
<p>> CSeq: 102 CANCEL</p>
<p>> Max-Forwards: 70</p>
<p>> Timestamp: 1256646855</p>
<p>> Reason: Q.850;cause=16</p>
<p>> Content-Length: 0</p>
<p>> *Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:</p>
<p>> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060</p>
<p>> *Oct 27 12:34:15.900:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:</p>
<p>> ccsip_spi_get_msg_type returned: 2 for event 1</p>
<p>> *Oct 27 12:34:15.900:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:</p>
<p>> context=0x00000000</p>
<p>> *Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:</p>
<p>> Checking Invite Dialog</p>
<p>> *Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>> Received:</p>
<p>> SIP/2.0 200 OK</p>
<p>> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE</p>
<p>> From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net" target="_blank">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6</p>
<p>> To: <<a href="mailto:sip%3A18774675464@64.154.41.200" target="_blank">sip:18774675464@64.154.41.200</a>></p>
<p>> Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57" target="_blank">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a></p>
<p>> CSeq: 102 CANCEL</p>
<p>> Content-Length: 0</p>
<p>> *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:</p>
<p>> non-INVITE response with no RSEQ - do not disable IS_REL1XX</p>
<p>> *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPIIcpifUpdate:</p>
<p>> CallState: 3 Playout: 0 DiscTime:4913670 ConnTime 0</p>
<p>> *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:</p>
<p>> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060</p>
<p>> *Oct 27 12:34:15.912:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:</p>
<p>> ccsip_spi_get_msg_type returned: 2 for event 1</p>
<p>> *Oct 27 12:34:15.912:</p>
<p>> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:</p>
<p>> context=0x00000000</p>
<p>> *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:</p>
<p>> Checking Invite Dialog</p>
<p>> *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>> </p>
<p>> On Mon, Oct 26, 2009 at 7:36 PM, Nick Matthews <<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>> wrote:</p>
<p>>> </p>
<p>>> You would want to check the SDP of 200 OK the provider sends for your</p>
<p>>> outgoing call. It will list the payload type for the dtmf in the</p>
<p>>> format a=fmtp 101 1-16, or something similar. You want to find out</p>
<p>>> what payload type they are advertising (or if they are at all). It</p>
<p>>> would be worth checking the incoming INVITE from them to see what</p>
<p>>> they're using when they send the first SDP.</p>
<p>>> </p>
<p>>> On that note, I would also remove the asymmetric payload command - to</p>
<p>>> my knowledge it doesn't do what you're expecting it to. You may want</p>
<p>>> to try this command:</p>
<p>>> voice-class sip dtmf-relay force rtp-nte</p>
<p>>> </p>
<p>>> </p>
<p>>> -nick</p>
<p>>> </p>
<p>>> 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>></p>
<p>>> wrote:</p>
<p>>> > Hello,</p>
<p>>> ></p>
<p>>> > I am having an issue with dtmf working outbound. Inbound dtmf works</p>
<p>>> > fine.</p>
<p>>> > It took some playing around with it. At first it didnt work till the</p>
<p>>> > payload was ajusted. I am now trying to get outbound dtmf working</p>
<p>>> > properly.</p>
<p>>> ></p>
<p>>> > On my 2821 I debugged voip rtp session named-events and then made a call</p>
<p>>> > to</p>
<p>>> > a 1800 number and hit digits. I didn't see any dtmf output on the</p>
<p>>> > router</p>
<p>>> > nothing showed up in the debug. Does this mean I can safely asume that</p>
<p>>> > the</p>
<p>>> > problem for right now is not on the ITSP side but on my side since dtmf</p>
<p>>> > is</p>
<p>>> > not being sent down the sip trunk?</p>
<p>>> ></p>
<p>>> > I have my cuc 7.x configured to talk to my 2821 via h323. The</p>
<p>>> > configuration</p>
<p>>> > of the cisco 2821 is shown below. Does anyone have any ideas what I can</p>
<p>>> > do</p>
<p>>> > so dtmf digits process properly outbound?</p>
<p>>> ></p>
<p>>> > The settings in my cuc 7.x to add the gateway h323 are</p>
<p>>> ></p>
<p>>> > h323 cucm gateway configuratration</p>
<p>>> > Signaling Port 1720</p>
<p>>> > media termination point required yes</p>
<p>>> > retry video call as auto yes</p>
<p>>> > wait for far end h.245 terminal capability set yes</p>
<p>>> > transmit utf-8 calling party name no</p>
<p>>> > h.235 pass through allowed no</p>
<p>>> > significant digits all</p>
<p>>> > redirect number IT deliver - inbound no</p>
<p>>> > enable inbound faststart yes</p>
<p>>> > display IE deliver no</p>
<p>>> > redirect nunmber IT deliver - outbound no</p>
<p>>> > enable outbound faststart yes</p>
<p>>> ></p>
<p>>> ></p>
<p>>> > voice service voip</p>
<p>>> > allow-connections h323 to h323</p>
<p>>> > allow-connections h323 to sip</p>
<p>>> > allow-connections sip to h323</p>
<p>>> > allow-connections sip to sip</p>
<p>>> > fax protocol pass-through g711ulaw</p>
<p>>> > h323</p>
<p>>> > emptycapability</p>
<p>>> > h225 id-passthru</p>
<p>>> > h245 passthru tcsnonstd-passthru</p>
<p>>> > sip</p>
<p>>> ></p>
<p>>> ></p>
<p>>> > voice class h323 50</p>
<p>>> > h225 timeout tcp establish 3</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > voice translation-rule 1</p>
<p>>> > rule 1 /.*/ /190/</p>
<p>>> > !</p>
<p>>> > voice translation-rule 2</p>
<p>>> > rule 1 /.*/ /1&/</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > voice translation-profile aa</p>
<p>>> > translate called 1</p>
<p>>> > !</p>
<p>>> > voice translation-profile addone</p>
<p>>> > translate called 2</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > voice-card 0</p>
<p>>> > dspfarm</p>
<p>>> > dsp services dspfarm</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > sccp local GigabitEthernet0/1</p>
<p>>> > sccp ccm 10.1.80.11 identifier 2 version 7.0</p>
<p>>> > sccp ccm 10.1.80.10 identifier 1 version 7.0</p>
<p>>> > sccp</p>
<p>>> > !</p>
<p>>> > sccp ccm group 1</p>
<p>>> > associate ccm 1 priority 1</p>
<p>>> > associate ccm 2 priority 2</p>
<p>>> > associate profile 1 register 2821transcode</p>
<p>>> > !</p>
<p>>> > dspfarm profile 1 transcode</p>
<p>>> > codec g711ulaw</p>
<p>>> > codec g711alaw</p>
<p>>> > codec g729ar8</p>
<p>>> > codec g729abr8</p>
<p>>> > codec g729r8</p>
<p>>> > maximum sessions 4</p>
<p>>> > associate application SCCP</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > dial-peer voice 100 voip</p>
<p>>> > description AA Publisher</p>
<p>>> > preference 1</p>
<p>>> > destination-pattern 1..</p>
<p>>> > voice-class h323 50</p>
<p>>> > session target ipv4:10.1.80.10</p>
<p>>> > dtmf-relay h245-alphanumeric</p>
<p>>> > codec g711ulaw</p>
<p>>> > no vad</p>
<p>>> > !</p>
<p>>> > dial-peer voice 1000 voip</p>
<p>>> > description incoming Call</p>
<p>>> > translation-profile incoming aa</p>
<p>>> > preference 1</p>
<p>>> > rtp payload-type nse 101</p>
<p>>> > rtp payload-type nte 100</p>
<p>>> > incoming called-number 6782282221</p>
<p>>> > dtmf-relay rtp-nte</p>
<p>>> > codec g711ulaw</p>
<p>>> > ip qos dscp cs5 media</p>
<p>>> > ip qos dscp cs5 signaling</p>
<p>>> > no vad</p>
<p>>> > !</p>
<p>>> > dial-peer voice 101 voip</p>
<p>>> > description AA Subscriber</p>
<p>>> > preference 2</p>
<p>>> > destination-pattern 1..</p>
<p>>> > voice-class h323 50</p>
<p>>> > session target ipv4:10.1.80.11</p>
<p>>> > dtmf-relay h245-alphanumeric</p>
<p>>> > codec g711ulaw</p>
<p>>> > no vad</p>
<p>>> > !</p>
<p>>> > dial-peer voice 2000 voip</p>
<p>>> > description outbound</p>
<p>>> > translation-profile outgoing addone</p>
<p>>> > preference 1</p>
<p>>> > destination-pattern .T</p>
<p>>> > rtp payload-type nse 101</p>
<p>>> > rtp payload-type nte 100</p>
<p>>> > voice-class sip asymmetric payload dtmf</p>
<p>>> > session protocol sipv2</p>
<p>>> > session target ipv4:64.154.41.200</p>
<p>>> > dtmf-relay rtp-nte</p>
<p>>> > codec g711ulaw</p>
<p>>> > no vad</p>
<p>>> > !</p>
<p>>> > !</p>
<p>>> > sip-ua</p>
<p>>> > credentials username ***** password 7 ***** realm <a href="http://sip.talkinip.net/" target="_blank">sip.talkinip.net</a></p>
<p>>> > authentication username ***** password 7 *****</p>
<p>>> > authentication username ***** password 7 ***** realm</p>
<p>>> > <a href="http://sip.talkinip.net/" target="_blank">sip.talkinip.net</a></p>
<p>>> > set pstn-cause 3 sip-status 486</p>
<p>>> > set pstn-cause 34 sip-status 486</p>
<p>>> > set pstn-cause 47 sip-status 486</p>
<p>>> > registrar dns:<a href="http://sip.talkinip.net/" target="_blank">sip.talkinip.net</a> expires 60</p>
<p>>> > sip-server dns:<a href="http://sip.talkinip.net:5060/" target="_blank">sip.talkinip.net:5060</a></p>
<p>>> > _______________________________________________</p>
<p>>> > cisco-voip mailing list</p>
<p>>> > <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a></p>
<p>>> > <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a></p>
<p>>> ></p>
<p>>> ></p>
<p>> </p>
<p>> </p>
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