<div>Is the below the ok I should be getting?</div>
<div> </div>
<div> </div>
<div>They did send this with the first debug</div>
<div> </div>
<div>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">sip:6782282221@sip.talkinip.net</a>>;tag=32DA608-109A<br>To: <<a href="mailto:sip%3A18774675464@64.154.41.200">sip:18774675464@64.154.41.200</a>><br>
Call-ID: <a href="mailto:9F060E11-C23511DE-8027C992-790F56B7@173.14.220.57">9F060E11-C23511DE-8027C992-790F56B7@173.14.220.57</a><br>CSeq: 102 CANCEL<br>Content-Length: 0</div>
<div><br>*Oct 27 13:44:12.828: //922/009B1B501B00/SIP/Info/sipSPICheckResponse: non-INVITE response with no RSEQ - do not disable IS_REL1XX<br>*Oct 27 13:44:12.828: //922/009B1B501B00/SIP/Info/sipSPIIcpifUpdate: CallState: 3 Playout: 0 DiscTime:5333362 ConnTime 0<br>
*Oct 27 13:44:12.836: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>*Oct 27 13:44:12.840: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1<br>
*Oct 27 13:44:12.840: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000<br>*Oct 27 13:44:12.840: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog<br>*Oct 27 13:44:12.840: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</div>
<div> </div>
<div>This with the 2nd debug</div>
<div> </div>
<div>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>
Received:<br>SIP/2.0 487 Request Terminated<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>Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>Content-Length: 0<br><br></div>
<div class="gmail_quote">On Tue, Oct 27, 2009 at 8:43 AM, 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">In the 183 Session Progress they're not advertising DTMF:<br>
<div class="im"><br>m=audio 45846 RTP/AVP 0<br><br></div>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><font color="#888888"><br><br>-nick<br></font>
<div>
<div></div>
<div class="h5"><br>On Tue, Oct 27, 2009 at 7:36 AM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>> wrote:<br>> Nick<br>><br>> I removed voice-class sip asymmetric payload dtmf and added in the other<br>
> line<br>><br>> Just to state incoming dtmf works but not outbound the ITSP has told me 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 this<br>
><br>> *Oct 27 12:34:09.128: //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 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 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">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,<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">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,<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>> 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: //-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: //-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">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<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: //-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: //-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">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" target="_blank">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<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 GTD<br>> found in inbound container<br>> *Oct 27 12:34:10.836: //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 already<br>> bound, use existing source_media_ip_addr<br>
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:<br>> Media src addr for stream 1 = 173.14.220.57<br>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:<br>> Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1<br>
> *Oct 27 12:34:10.836: //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: //846/8094E28C1800/SIP/Media/sipSPIDoPtimeNegotiation:<br>
> Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for 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: //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 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 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 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 for<br>
> m-line:1 and num-a-lines:0<br>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:<br>> 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,<br>> dest_port=45846<br>> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/State/sipSPIChangeStreamState:<br>> 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<br>> *Oct 27 12:34:10.840:<br>
> //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already<br>> bound, use existing source_media_ip_addr<br>> *Oct 27 12:34:10.840: //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: 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 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 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 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: //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: //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 headers<br>> recvd from app container<br>> *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No<br>
> QSIG Body found in inbound container<br>> *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No<br>> RawMsg Body found in inbound container<br>> *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: 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, 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 = 6,<br>> srcCallID = 846, dstCallID = 845<br>> *Oct 27 12:34:10.844: //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: //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds:<br>> Old streamcallid=846, new streamcallid=846<br>> *Oct 27 12:34:10.844: //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: //846/8094E28C1800/SIP/Media/sipSPIProcessRtpSessions:<br>
> sipSPIProcessRtpSessions<br>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIAddStream: 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 already<br>
> bound, use existing source_media_ip_addr<br>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:<br>> Media src addr for stream 1 = 173.14.220.57<br>> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:<br>
> sipSPIUpdateRtcpSession for m-line 1<br>> *Oct 27 12:34:10.848: //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 = 1<br>
> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:<br>> RTP session already created - update<br>> *Oct 27 12:34:10.848: //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: //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 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 VALUES:<br>> stream_callid=846, current_seq_num=0x11D9<br>
> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load 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: //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 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 Xcap<br>> list<br>> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Modem<br>> Relay & Passthru both disabled<br>
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: 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: 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 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 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: Setting<br>> CAPS_RECEIVED flag<br>> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: 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 VALUES:<br>
> stream_callid=846, current_seq_num=0x11D9<br>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load 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: //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 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 Xcap<br>> list<br>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Modem<br>
> Relay & Passthru both disabled<br>> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: 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: 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 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 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 TCS<br>> received for transfers across trunk - set CAPS2_RECEIVED<br>> *Oct 27 12:34:15.876: //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: 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: 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 for<br>
> sending msg immediately<br>> *Oct 27 12:34:15.880:<br>> //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: switch transport<br>> is 0<br>> *Oct 27 12:34:15.880:<br>> //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: Set to send the<br>
> msg=0x4DF0D994<br>> *Oct 27 12:34:15.880:<br>> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send<br>> 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:<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) 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">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<br>> *Oct 27 12:34:15.900: //-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: //-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">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<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>
><br>> On Mon, Oct 26, 2009 at 7:36 PM, Nick Matthews <<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>> 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">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 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 that<br>>> > the<br>>> > problem for right now is not on the ITSP side but on my side since 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 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">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></div></div></blockquote></div><br>