<div>Nick</div>
<div> </div>
<div>I removed voice-class sip asymmetric payload dtmf and added in the other line</div>
<div> </div>
<div>Just to state incoming dtmf works but not outbound the ITSP has told me they are using two different sip servers/vendors for processing inbound and outbound</div>
<div><br>How does this translate into what I should sent the following too?</div>
<div> </div>
<div>rtp payload-type nse <br>rtp payload-type nte </div>
<div> </div>
<div>In the debug trhe following where set</div>
<div> </div>
<div>rtp payload-type nse 101<br> rtp payload-type nte 100</div>
<div> </div>
<div><strong>In the debug of ccsip If I am looking at it correctly I see me sending this</strong></div>
<div><strong></strong> </div>
<div>*Oct 27 12:34:09.128: //846/8094E28C1800/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 100<br>*Oct 27 12:34:09.128: //846/8094E28C1800/SIP/Info/sipSPIAddSDPPayloadAttributes:<br>
max_event 15</div>
<div> </div>
<div><strong>and</strong></div>
<div> </div>
<div> </div>
<div>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay</div>
<div><strong></strong> </div>
<div> </div>
<div>Sent:<br>INVITE <a href="http://sip:18774675464@64.154.41.200:5060">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: <<a href="mailto:sip%3A6782282221@173.14.220.57">sip:6782282221@173.14.220.57</a>>;party=calling;screen=yes;privacy=off<br>
From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>To: <<a href="mailto:sip%3A18774675464@64.154.41.200">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">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, SUBSCRIBE, 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">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</div>
<div>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</div>
<div> </div>
<div> </div>
<div><strong>Then when I do a search for fmtp again further down I see</strong> </div>
<div> </div>
<div>Sent:<br>INVITE <a href="http://sip:18774675464@64.154.41.200:5060">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: <<a href="mailto:sip%3A6782282221@173.14.220.57">sip:6782282221@173.14.220.57</a>>;party=calling;screen=yes;privacy=off<br>
From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>To: <<a href="mailto:sip%3A18774675464@64.154.41.200">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">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, SUBSCRIBE, 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">sip:6782282221@173.14.220.57:5060</a>><br>
Expires: 180<br>Allow-Events: telephone-event<br>Proxy-Authorization: Digest username="1648245954",realm="64.154.41.110",uri="<a href="http://sip:18774675464@64.154.41.200:5060">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</div>
<div>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</div>
<div>*Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>*Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1<br>
*Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000<br>*Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: 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">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>To: <<a href="mailto:sip%3A18774675464@64.154.41.200">sip:18774675464@64.154.41.200</a>><br>
Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>CSeq: 102 INVITE<br>Content-Length: 0</div>
<div><br>*Oct 27 12:34:09.332: //846/8094E28C1800/SIP/Info/sipSPICheckResponse: INVITE response with no RSEQ - disable IS_REL1XX<br>*Oct 27 12:34:09.332: //846/8094E28C1800/SIP/State/sipSPIChangeState: 0x4A357FCC : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)<br>
*Oct 27 12:34:10.832: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>*Oct 27 12:34:10.832: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1<br>
*Oct 27 12:34:10.832: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000<br>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: 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">sip:18774675464@64.154.41.200</a>>;tag=3465630735-938664<br>From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>
Contact: <<a href="http://sip:18774675464@64.154.41.200:5060">sip:18774675464@64.154.41.200:5060</a>><br>Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57">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</div>
<div>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</div>
<div>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPICheckResponse: INVITE response with no RSEQ - disable IS_REL1XX<br>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1<br>SIP: Attribute mid, level 1 instance 1 not found.<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.14.220.57<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:20<br>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :20, codecbytes: 160<br>
*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec g711ulaw<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(100) could not be reserved.<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full named event(NE) match in fmtp list of events.<br>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay<br>*Oct 27 12:34:10.836: //846/8094E28C1800/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 12:34:10.836: //846/8094E28C1800/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=64.154.41.101, dest_port=45846<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/State/sipSPIChangeStreamState: Stream (callid = -1) State changed from (STREAM_DEAD) to (STREAM_ADDING)<br>*Oct 27 12:34:10.836: //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</div>
<div>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.14.220.57<br>
*Oct 27 12:34:10.840: //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: //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: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:<br>
Hndl ptype 0 mline 1<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting 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: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 5</div>
<div>*Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=20,stream->negotiated_codec_bytes=160, coverted ptime=20 stream->mline_index=1, media_ndx=1<br>
*Oct 27 12:34:10.840: //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 <a href="http://64.154.41.101:45846">64.154.41.101:45846</a><br>
*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = 5</div>
<div>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = -1</div>
<div>*Oct 27 12:34:10.840: //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: //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:<br>
Report initial call media<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags 0x400018, ccb->pld.flags_ipip 0x200005</div>
<div>*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: //846/8094E28C1800/SIP/Info/ccsip_update_srtp_caps: 5030: Posting Remote SRTP caps to other callleg.<br>
*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: do cc_api_caps_ind()<br>*Oct 27 12:34:10.840: //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</div>
<div>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers recvd from app container<br>*Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No QSIG Body found in inbound container<br>
*Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No RawMsg Body found in inbound container<br>*Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No Data to form The Raw Message</div>
<div>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress: ccsip_api_call_cut_progress returned: SIP_SUCCESS<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/State/sipSPIChangeState: 0x4A357FCC : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_NONE)<br>
*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress: Transaction Complete. Lock on Facilities released.<br>*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge: confID = 6, srcCallID = 846, dstCallID = 845<br>
*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds: Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 846/845<br>*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=846, new streamcallid=846<br>
*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-H323<br>*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge: xcoder_attached = 0, xmitFunc = 1131891908, ccb xmitFunc = 1131891908<br>
*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions<br>*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 846) to the VOIP RTP library<br>
*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr<br>*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.14.220.57<br>
*Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1<br>*Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info<br> laddr = 173.14.220.57, lport = 16462, raddr = 64.154.41.101, rport=45846, do_rtcp=TRUE<br>
src_callid = 846, dest_callid = 845, stream type = voice+dtmf, stream direction = SENDRECV<br> media_ip_addr = 64.154.41.101, vrf tableid = 0 media_addr_type = 1<br>*Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update<br>
*Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:4A1709F8<br>*Oct 27 12:34:10.848: //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</div>
<div>*Oct 27 12:34:10.848: //846/8094E28C1800/SIP/State/sipSPIChangeStreamState: Stream (callid = 846) State changed from (STREAM_ADDING) to (STREAM_ACTIVE)<br>*Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Info/ccsip_bridge: really can't 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: //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=846, current_seq_num=0x23ED<br>
*Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=846, current_seq_num=0x11D9<br>*Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g711ulaw, Bytes=160<br>
*Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0<br>*Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx payload = 100, tx payload = 100<br>
*Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=0, from CLI config=0<br>*Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...<br>
*Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap list<br>*Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled<br>
*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: nse payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0, sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32<br>*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 1 Active Streams<br>
*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 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, caps.stream_list.xmitFunc=<br>
*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=<br>*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 0x497E0B60 (gccb)<br>*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP with codec : g711ulaw, Bytes=160, payload = 0<br>
*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->pld.flags_ipip = 0x200405<br>*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No video caps detected in the caps posted by peer leg<br>
*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Setting CAPS_RECEIVED flag<br>*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Calling cc_api_caps_ack()<br>*Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ack: Set 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: //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=846, current_seq_num=0x11D9<br>
*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=846, current_seq_num=0x11D9<br>*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g711ulaw, Bytes=160<br>
*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0<br>*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx payload = 100, tx payload = 100<br>
*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=0, from CLI config=0<br>*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...<br>
*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap list<br>*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled<br>
*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: nse payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0, sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32<br>*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 1 Active Streams<br>
*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 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, caps.stream_list.xmitFunc=<br>
*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=<br>*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 0x497E0B60 (gccb)<br>*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP with codec : g711ulaw, Bytes=160, payload = 0<br>
*Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->pld.flags_ipip = 0x200425<br>*Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No video caps detected in the caps posted by peer leg<br>
*Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Second TCS received for transfers across trunk - set CAPS2_RECEIVED<br>*Oct 27 12:34:15.876: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:4A1709F8<br>
*Oct 27 12:34:15.876: //846/8094E28C1800/SIP/Info/ccsip_call_statistics: Stats are not supported for IPIP call.<br>*Oct 27 12:34:15.876: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT<br>
*Oct 27 12:34:15.880: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 7<br>*Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Info/sipSPISendCancel: Associated container=0x4E310C1C to Cancel<br>
*Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Transport/sipSPISendCancel: Sending CANCEL to the transport layer<br>*Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage: msg=0x4DF0D994, addr=64.154.41.200, port=5060, sentBy_port=0, is_req=1, transport=1, switch=0, callBack=0x419703BC<br>
*Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately<br>*Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0<br>
*Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x4DF0D994<br>*Oct 27 12:34:15.880: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4DF0D994, addr=64.154.41.200, port=5060, connId=2 for UDP<br>
*Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Info/sentCancelDisconnecting: Sent Cancel Request, starting CancelWaitResponseTimer<br>*Oct 27 12:34:15.880: //846/8094E28C1800/SIP/State/sipSPIChangeState: 0x4A357FCC : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE) to (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">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">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>To: <<a href="mailto:sip%3A18774675464@64.154.41.200">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">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</div>
<div><br>*Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>*Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1<br>
*Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000<br>*Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: 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">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>To: <<a href="mailto:sip%3A18774675464@64.154.41.200">sip:18774675464@64.154.41.200</a>><br>
Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>CSeq: 102 CANCEL<br>Content-Length: 0</div>
<div><br>*Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPICheckResponse: non-INVITE response with no RSEQ - do not disable IS_REL1XX<br>*Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPIIcpifUpdate: CallState: 3 Playout: 0 DiscTime:4913670 ConnTime 0<br>
*Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>*Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1<br>
*Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000<br>*Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog<br>*Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
<br></div>
<div class="gmail_quote">On Mon, Oct 26, 2009 at 7:36 PM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com">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">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>
<div>
<div></div>
<div class="h5"><br>On Mon, Oct 26, 2009 at 5:16 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>> wrote:<br>> Hello,<br>><br>> I am having an issue with dtmf working outbound. Inbound dtmf works 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 call to<br>
> a 1800 number and hit digits. I didn't see any dtmf output on the router<br>> nothing showed up in the debug. Does this mean I can safely asume that the<br>> problem for right now is not on the ITSP side but on my side since dtmf 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 configuration<br>> of the cisco 2821 is shown below. Does anyone have any ideas what I can 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 <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></div></div>
> _______________________________________________<br>> cisco-voip mailing list<br>> <a href="mailto:cisco-voip@puck.nether.net">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></blockquote></div><br>