=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2008.10.21 11:48:10 =~=~=~=~=~=~=~=~=~=~=~= asterisk -r Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.4.21.2 currently running on pbxsipgw-01 (pid = 21325) pbxsipgw-01*CLI> Verbosity is at least 12 Core debug is at least 12 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> INVITE sip:16474274213@10.5.1.30 SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;refresher=uac Supported: timer, 100rel To: From: "Comwave" ;tag=3433592901-624862 Call-ID: 4166470-3433592901-624855@msx-02.mgmt.hook2 CSeq: 1 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bKf2852d2e6f7e9c375c5906f455b6e45f Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 179 v=0 o=msx-02 1473 1473 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31592 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 10 lines) --- Sending to 10.2.12.6 : 5060 (no NAT) Using INVITE request as basis request - 4166470-3433592901-624855@msx-02.mgmt.hook2 Found peer 'comwave-nextone' Found RTP audio format 0 Peer audio RTP is at port 10.2.12.7:31592 Found audio description format PCMU for ID 0 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.2.12.7:31592 Looking for 16474274213 in from-nextone (domain 10.5.1.30) list_route: hop: <--- Transmitting (no NAT) to 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bKf2852d2e6f7e9c375c5906f455b6e45f;received=10.2.12.6 From: "Comwave" ;tag=3433592901-624862 To: Call-ID: 4166470-3433592901-624855@msx-02.mgmt.hook2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [16474274213@from-nextone:1] Answer("SIP/10.2.12.6-0902b9f0", "") in new stack Audio is at 10.5.1.30 port 10440 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bKf2852d2e6f7e9c375c5906f455b6e45f;received=10.2.12.6 From: "Comwave" ;tag=3433592901-624862 To: ;tag=as28b1adab Call-ID: 4166470-3433592901-624855@msx-02.mgmt.hook2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 178 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 10440 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Executing [16474274213@from-nextone:2] Playback("SIP/10.2.12.6-0902b9f0", "cw/admin/carriersupport_greeting") in new stack -- Playing 'cw/admin/carriersupport_greeting' (language 'en') <--- SIP read from 10.2.12.6:5060 ---> ACK sip:16474274213@10.5.1.30 SIP/2.0 Max-Forwards: 69 To: ;tag=as28b1adab From: "Comwave" ;tag=3433592901-624862 Call-ID: 4166470-3433592901-624855@msx-02.mgmt.hook2 CSeq: 1 ACK Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bKb169a4971a779a9ccecef2ac4239691a Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (10 headers 0 lines) --- pbxsipgw-01*CLI> -- Executing [16474274213@from-nextone:3] Set("SIP/10.2.12.6-0902b9f0", "CALLERID(all)="CarrierSupport" <416-623-8520>") in new stack -- Executing [16474274213@from-nextone:4] FollowMe("SIP/10.2.12.6-0902b9f0", "carriersupport|a") in new stack -- Playing 'vm-rec-name' (language 'en') -- Playing 'beep' (language 'en') pbxsipgw-01*CLI> -- x=0, open writing: /var/spool/asterisk/followme.1224603623.66770 format: sln, 0x902d678 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> INFO sip:16474274213@10.5.1.30 SIP/2.0 Max-Forwards: 69 To: ;tag=as28b1adab From: "Comwave" ;tag=3433592901-624862 Call-ID: 4166470-3433592901-624855@msx-02.mgmt.hook2 CSeq: 2 INFO Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bK0e316eb8505f5f86500fce40e012e566 Contact: Content-Type: application/dtmf-relay Content-Length: 24 Signal=# Duration=160 <-------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: # <--- Transmitting (no NAT) to 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bK0e316eb8505f5f86500fce40e012e566;received=10.2.12.6 From: "Comwave" ;tag=3433592901-624862 To: ;tag=as28b1adab Call-ID: 4166470-3433592901-624855@msx-02.mgmt.hook2 CSeq: 2 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> -- User ended message by pressing # pbxsipgw-01*CLI> -- Playing 'auth-thankyou' (language 'en') pbxsipgw-01*CLI> -- Playing 'followme/pls-hold-while-try' (language 'en') pbxsipgw-01*CLI> -- Started music on hold, class 'default', on SIP/10.2.12.6-0902b9f0 -- calling 916472044341@trunkld -- calling 914165093362@trunkld -- calling 914167978854@trunkld -- calling 916478831388@trunkld -- Executing [916472044341@trunkld:1] Dial("Local/916472044341@trunkld-b96f,2", "SIP/comwave-nextone/16472044341") in new stack pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 19206 pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP pbxsipgw-01*CLI> Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK79d094e1 From: "CarrierSupport" ;tag=as10448601 To: Contact: Call-ID: 45687311624de16d4ccdf57307f3ec22@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 19206 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Executing [914165093362@trunkld:1] Dial("Local/914165093362@trunkld-a8d9,2", "SIP/comwave-nextone/14165093362") in new stack -- Executing [914167978854@trunkld:1] Dial("Local/914167978854@trunkld-6ba8,2", "SIP/comwave-nextone/14167978854") in new stack pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 13090 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5ff76373 From: "CarrierSupport" ;tag=as31288e0b To: Contact: Call-ID: 29445fb0294636c149057d7d0e7b3c48@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 13090 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> -- Called comwave-nextone/14167978854 Audio is at 10.5.1.30 port 10780 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> Adding non-codec 0x1 (telephone-event) to SDP -- Executing [916478831388@trunkld:1] Dial("Local/916478831388@trunkld-9522,2", "SIP/comwave-nextone/16478831388") in new stack Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK1d1686fb From: "CarrierSupport" ;tag=as60673069 To: Contact: Call-ID: 63decba6508d77665a88f1d60608373c@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 10780 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/14165093362 -- Called comwave-nextone/16472044341 Audio is at 10.5.1.30 port 15856 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK71e2fea3 From: "CarrierSupport" ;tag=as420668c3 To: Contact: Call-ID: 2742f80f78412518301c50c02e0a094c@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 15856 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/16478831388 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5ff76373 From: "CarrierSupport" ;tag=as31288e0b To: Call-ID: 29445fb0294636c149057d7d0e7b3c48@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK79d094e1 From: "CarrierSupport" ;tag=as10448601 To: Call-ID: 45687311624de16d4ccdf57307f3ec22@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (7 headers 0 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK1d1686fb From: "CarrierSupport" ;tag=as60673069 To: Call-ID: 63decba6508d77665a88f1d60608373c@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (7 headers 0 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK71e2fea3 From: "CarrierSupport" ;tag=as420668c3 To: Call-ID: 2742f80f78412518301c50c02e0a094c@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (7 headers 0 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK1d1686fb To: ;tag=3433592915-516865 From: "CarrierSupport" ;tag=as60673069 Call-ID: 63decba6508d77665a88f1d60608373c@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 220 v=0 o=msx-02 2106337557 2106337283 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31702 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31702 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer pbxsipgw-01*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31702 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK71e2fea3 To: ;tag=3433592915-517562 From: "CarrierSupport" ;tag=as420668c3 Call-ID: 2742f80f78412518301c50c02e0a094c@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 220 v=0 o=msx-02 2095875873 2095875611 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31706 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 pbxsipgw-01*CLI> Found RTP audio format 101 pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31706 pbxsipgw-01*CLI> Found audio description format PCMU for ID 0 pbxsipgw-01*CLI> Found audio description format telephone-event for ID 101 pbxsipgw-01*CLI> Got unsupported a:fmtp in SDP offer pbxsipgw-01*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31706 pbxsipgw-01*CLI> -- SIP/comwave-nextone-08fb0ab0 is making progress passing it to Local/914165093362@trunkld-a8d9,2 pbxsipgw-01*CLI> -- Local/914165093362@trunkld-a8d9,1 is making progress passing it to SIP/10.2.12.6-0902b9f0 pbxsipgw-01*CLI> -- SIP/comwave-nextone-0907e4d0 is making progress passing it to Local/916478831388@trunkld-9522,2 pbxsipgw-01*CLI> -- Local/916478831388@trunkld-9522,1 is making progress passing it to SIP/10.2.12.6-0902b9f0 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK79d094e1 To: ;tag=3433592915-593795 From: "CarrierSupport" ;tag=as10448601 Call-ID: 45687311624de16d4ccdf57307f3ec22@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 220 v=0 o=msx-02 2106239876 2106239609 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31698 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31698 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer pbxsipgw-01*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31698 -- SIP/comwave-nextone-08fe60b0 is making progress passing it to Local/916472044341@trunkld-b96f,2 -- Local/916472044341@trunkld-b96f,1 is making progress passing it to SIP/10.2.12.6-0902b9f0 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5ff76373 To: ;tag=3433592915-619608 From: "CarrierSupport" ;tag=as31288e0b Call-ID: 29445fb0294636c149057d7d0e7b3c48@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 220 v=0 o=msx-02 2095780769 2095780508 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31694 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31694 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 pbxsipgw-01*CLI> Got unsupported a:fmtp in SDP offer pbxsipgw-01*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31694 pbxsipgw-01*CLI> -- SIP/comwave-nextone-08ff5080 is making progress passing it to Local/914167978854@trunkld-6ba8,2 pbxsipgw-01*CLI> -- Local/914167978854@trunkld-6ba8,1 is making progress passing it to SIP/10.2.12.6-0902b9f0 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> BYE sip:16474274213@10.5.1.30 SIP/2.0 Max-Forwards: 69 To: ;tag=as28b1adab From: "Comwave" ;tag=3433592901-624862 Call-ID: 4166470-3433592901-624855@msx-02.mgmt.hook2 CSeq: 3 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bK4191628564807a2fd9df6e3b032e6fc9 Contact: Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.2.12.6 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bK4191628564807a2fd9df6e3b032e6fc9;received=10.2.12.6 From: "Comwave" ;tag=3433592901-624862 To: ;tag=as28b1adab Call-ID: 4166470-3433592901-624855@msx-02.mgmt.hook2 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> -- Stopped music on hold on SIP/10.2.12.6-0902b9f0 pbxsipgw-01*CLI> -- calling 916472044341@trunkld pbxsipgw-01*CLI> -- calling 914165093362@trunkld pbxsipgw-01*CLI> -- calling 914167978854@trunkld pbxsipgw-01*CLI> -- calling 916478831388@trunkld pbxsipgw-01*CLI> -- Executing [916472044341@trunkld:1] Dial("Local/916472044341@trunkld-e52d,2", "SIP/comwave-nextone/16472044341") in new stack pbxsipgw-01*CLI> -- Executing [914165093362@trunkld:1] Dial("Local/914165093362@trunkld-c69a,2", "SIP/comwave-nextone/14165093362") in new stack pbxsipgw-01*CLI> -- Executing [914167978854@trunkld:1] Dial("Local/914167978854@trunkld-bc0e,2", "SIP/comwave-nextone/14167978854") in new stack pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '45687311624de16d4ccdf57307f3ec22@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK79d094e1 From: "CarrierSupport" ;tag=as10448601 To: Call-ID: 45687311624de16d4ccdf57307f3ec22@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '45687311624de16d4ccdf57307f3ec22@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '63decba6508d77665a88f1d60608373c@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK1d1686fb From: "CarrierSupport" ;tag=as60673069 To: Call-ID: 63decba6508d77665a88f1d60608373c@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '63decba6508d77665a88f1d60608373c@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '2742f80f78412518301c50c02e0a094c@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '29445fb0294636c149057d7d0e7b3c48@10.5.1.30' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5ff76373 From: "CarrierSupport" ;tag=as31288e0b To: Call-ID: 29445fb0294636c149057d7d0e7b3c48@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '29445fb0294636c149057d7d0e7b3c48@10.5.1.30' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK71e2fea3 From: "CarrierSupport" ;tag=as420668c3 To: Call-ID: 2742f80f78412518301c50c02e0a094c@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '2742f80f78412518301c50c02e0a094c@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> -- Executing [916478831388@trunkld:1] Dial("Local/916478831388@trunkld-a830,2", "SIP/comwave-nextone/16478831388") in new stack pbxsipgw-01*CLI> == Spawn extension (trunkld, 916472044341, 1) exited non-zero on 'Local/916472044341@trunkld-b96f,2' == Spawn extension (trunkld, 914165093362, 1) exited non-zero on 'Local/914165093362@trunkld-a8d9,2' <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK79d094e1 To: ;tag=3433592915-593795 From: "CarrierSupport" ;tag=as10448601 Call-ID: 45687311624de16d4ccdf57307f3ec22@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- calling 916472044341@trunkld pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 19828 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK1d1686fb To: ;tag=3433592915-516865 From: "CarrierSupport" ;tag=as60673069 Call-ID: 63decba6508d77665a88f1d60608373c@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP pbxsipgw-01*CLI> Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5ff76373 To: ;tag=3433592915-619608 From: "CarrierSupport" ;tag=as31288e0b Call-ID: 29445fb0294636c149057d7d0e7b3c48@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> -- calling 914165093362@trunkld pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK450f4167 From: "CarrierSupport" ;tag=as44ab87dd To: Contact: Call-ID: 07094584496afd2b2d49c8d27dd74038@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 19828 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> -- Called comwave-nextone/14165093362 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK450f4167 From: "CarrierSupport" ;tag=as44ab87dd To: Call-ID: 07094584496afd2b2d49c8d27dd74038@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '07094584496afd2b2d49c8d27dd74038@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> == Spawn extension (trunkld, 914165093362, 1) exited non-zero on 'Local/914165093362@trunkld-c69a,2' pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 16108 pbxsipgw-01*CLI> --- (9 headers 0 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK71e2fea3 To: ;tag=3433592915-517562 From: "CarrierSupport" ;tag=as420668c3 Call-ID: 2742f80f78412518301c50c02e0a094c@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP pbxsipgw-01*CLI> Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK7dd1477d From: "CarrierSupport" ;tag=as4d8f4adc To: Contact: Call-ID: 73c41d366c05424f1c21930f4a2b9c5b@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 16108 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> -- Called comwave-nextone/14167978854 pbxsipgw-01*CLI> -- calling 914167978854@trunkld pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 14020 pbxsipgw-01*CLI> == Spawn extension (trunkld, 914167978854, 1) exited non-zero on 'Local/914167978854@trunkld-6ba8,2' == Spawn extension (trunkld, 916478831388, 1) exited non-zero on 'Local/916478831388@trunkld-9522,2' -- Executing [914167978854@trunkld:1] Dial("Local/914167978854@trunkld-8d3f,2", "SIP/comwave-nextone/14167978854") in new stack Audio is at 10.5.1.30 port 16990 -- calling 916478831388@trunkld pbxsipgw-01*CLI> -- calling 916472044341@trunkld pbxsipgw-01*CLI> -- calling 914165093362@trunkld pbxsipgw-01*CLI> -- calling 914167978854@trunkld pbxsipgw-01*CLI> -- calling 916478831388@trunkld pbxsipgw-01*CLI> -- Executing [916478831388@trunkld:1] Dial("Local/916478831388@trunkld-f095,2", "SIP/comwave-nextone/16478831388") in new stack pbxsipgw-01*CLI> -- Executing [916472044341@trunkld:1] Dial("Local/916472044341@trunkld-320e,2", "SIP/comwave-nextone/16472044341") in new stack pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK450f4167 From: "CarrierSupport" ;tag=as44ab87dd To: Call-ID: 07094584496afd2b2d49c8d27dd74038@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK450f4167 To: From: "CarrierSupport" ;tag=as44ab87dd Call-ID: 07094584496afd2b2d49c8d27dd74038@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK7dd1477d From: "CarrierSupport" ;tag=as4d8f4adc To: Call-ID: 73c41d366c05424f1c21930f4a2b9c5b@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Audio is at 10.5.1.30 port 19028 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbxsipgw-01*CLI> -- Executing [916478831388@trunkld:1] Dial("Local/916478831388@trunkld-1676,2", "SIP/comwave-nextone/16478831388") in new stack pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK299bee50 From: "CarrierSupport" ;tag=as24191131 To: Contact: Call-ID: 5493d730526b17682acb49760933b2c1@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 19028 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> -- Called comwave-nextone/16472044341 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK299bee50 From: "CarrierSupport" ;tag=as24191131 To: Call-ID: 5493d730526b17682acb49760933b2c1@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '5493d730526b17682acb49760933b2c1@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> == Spawn extension (trunkld, 916472044341, 1) exited non-zero on 'Local/916472044341@trunkld-320e,2' pbxsipgw-01*CLI> -- calling 916472044341@trunkld pbxsipgw-01*CLI> -- Executing [916472044341@trunkld:1] Dial("Local/916472044341@trunkld-0158,2", "SIP/comwave-nextone/16472044341") in new stack pbxsipgw-01*CLI> -- Executing [914165093362@trunkld:1] Dial("Local/914165093362@trunkld-666b,2", "SIP/comwave-nextone/14165093362") in new stack pbxsipgw-01*CLI> -- Executing [914167978854@trunkld:1] Dial("Local/914167978854@trunkld-8e98,2", "SIP/comwave-nextone/14167978854") in new stack pbxsipgw-01*CLI> -- Executing [916472044341@trunkld:1] Dial("Local/916472044341@trunkld-f0c1,2", "SIP/comwave-nextone/16472044341") in new stack pbxsipgw-01*CLI> -- calling 914165093362@trunkld pbxsipgw-01*CLI> -- Executing [914165093362@trunkld:1] Dial("Local/914165093362@trunkld-6483,2", "SIP/comwave-nextone/14165093362") in new stack pbxsipgw-01*CLI> -- calling 914167978854@trunkld pbxsipgw-01*CLI> -- calling 916478831388@trunkld pbxsipgw-01*CLI> -- Executing [914165093362@trunkld:1] Dial("Local/914165093362@trunkld-ea6e,2", "SIP/comwave-nextone/14165093362") in new stack pbxsipgw-01*CLI> -- Executing [914167978854@trunkld:1] Dial("Local/914167978854@trunkld-510e,2", "SIP/comwave-nextone/14167978854") in new stack pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK7dd1477d From: "CarrierSupport" ;tag=as4d8f4adc To: Call-ID: 73c41d366c05424f1c21930f4a2b9c5b@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '73c41d366c05424f1c21930f4a2b9c5b@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK450f4167 To: ;tag=3433592926-483237 From: "CarrierSupport" ;tag=as44ab87dd Call-ID: 07094584496afd2b2d49c8d27dd74038@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> == Spawn extension (trunkld, 914167978854, 1) exited non-zero on 'Local/914167978854@trunkld-bc0e,2' Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Audio is at 10.5.1.30 port 16962 pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3df800a5 From: "CarrierSupport" ;tag=as1ce32652 To: Contact: Call-ID: 56ee7a1e1a5a221a4a67337d494689f3@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 16962 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> -- Called comwave-nextone/16478831388 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3df800a5 From: "CarrierSupport" ;tag=as1ce32652 To: Call-ID: 56ee7a1e1a5a221a4a67337d494689f3@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '56ee7a1e1a5a221a4a67337d494689f3@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> == Spawn extension (trunkld, 916478831388, 1) exited non-zero on 'Local/916478831388@trunkld-a830,2' pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73149900 From: "CarrierSupport" ;tag=as02c5f8ec To: Contact: Call-ID: 7edf56753422e0ee7dc6967723ed6512@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 14020 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> --- (10 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK450f4167 From: "CarrierSupport" ;tag=as44ab87dd To: ;tag=3433592926-483237 Contact: Call-ID: 07094584496afd2b2d49c8d27dd74038@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '07094584496afd2b2d49c8d27dd74038@10.5.1.30' Method: INVITE pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 10078 pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 17514 pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK418cdd6a From: "CarrierSupport" ;tag=as600ef616 To: Contact: Call-ID: 51b9e4e70d9cd5161beadcae55042dbc@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 17514 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 16466 pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP pbxsipgw-01*CLI> Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK0035d991 From: "CarrierSupport" ;tag=as42787f48 To: Contact: Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 16466 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> -- Called comwave-nextone/16478831388 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK0035d991 From: "CarrierSupport" ;tag=as42787f48 To: Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '7011db9b4fa927df16ce9c4c3441e914@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> == Spawn extension (trunkld, 916478831388, 1) exited non-zero on 'Local/916478831388@trunkld-f095,2' pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK299bee50 From: "CarrierSupport" ;tag=as24191131 To: Call-ID: 5493d730526b17682acb49760933b2c1@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK299bee50 To: From: "CarrierSupport" ;tag=as24191131 Call-ID: 5493d730526b17682acb49760933b2c1@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK299bee50 To: ;tag=3433592926-489140 From: "CarrierSupport" ;tag=as24191131 Call-ID: 5493d730526b17682acb49760933b2c1@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK299bee50 From: "CarrierSupport" ;tag=as24191131 To: ;tag=3433592926-489140 Contact: Call-ID: 5493d730526b17682acb49760933b2c1@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '5493d730526b17682acb49760933b2c1@10.5.1.30' Method: INVITE Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73a66919 From: "CarrierSupport" ;tag=as0859bae5 To: Contact: Call-ID: 3cef9a16483fc7d72a65c1466a6acbe7@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 10078 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/16478831388 pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 16254 pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP pbxsipgw-01*CLI> Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK204a1f64 From: "CarrierSupport" ;tag=as4bad16e2 To: Contact: Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 16254 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> -- Called comwave-nextone/16472044341 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK204a1f64 From: "CarrierSupport" ;tag=as4bad16e2 To: Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK7dd1477d To: From: "CarrierSupport" ;tag=as4d8f4adc Call-ID: 73c41d366c05424f1c21930f4a2b9c5b@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK7dd1477d To: ;tag=3433592926-490452 From: "CarrierSupport" ;tag=as4d8f4adc Call-ID: 73c41d366c05424f1c21930f4a2b9c5b@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK7dd1477d From: "CarrierSupport" ;tag=as4d8f4adc To: ;tag=3433592926-490452 Contact: Call-ID: 73c41d366c05424f1c21930f4a2b9c5b@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '73c41d366c05424f1c21930f4a2b9c5b@10.5.1.30' Method: INVITE pbxsipgw-01*CLI> == Spawn extension (trunkld, 916472044341, 1) exited non-zero on 'Local/916472044341@trunkld-f0c1,2' pbxsipgw-01*CLI> -- Executing [916478831388@trunkld:1] Dial("Local/916478831388@trunkld-03cf,2", "SIP/comwave-nextone/16478831388") in new stack pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK79d094e1 To: ;tag=3433592915-593795 From: "CarrierSupport" ;tag=as10448601 Reason: SIP;cause=487;text="487 Request Terminated" Call-ID: 45687311624de16d4ccdf57307f3ec22@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK79d094e1 From: "CarrierSupport" ;tag=as10448601 To: ;tag=3433592915-593795 Contact: Call-ID: 45687311624de16d4ccdf57307f3ec22@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Really destroying SIP dialog '45687311624de16d4ccdf57307f3ec22@10.5.1.30' Method: INVITE pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 10690 pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 19194 pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP pbxsipgw-01*CLI> Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK004c8a9b From: "CarrierSupport" ;tag=as3bb52046 To: Contact: Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 19194 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbxsipgw-01*CLI> -- Called comwave-nextone/14165093362 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK004c8a9b From: "CarrierSupport" ;tag=as3bb52046 To: Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> == Spawn extension (trunkld, 914165093362, 1) exited non-zero on 'Local/914165093362@trunkld-ea6e,2' pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73a66919 From: "CarrierSupport" ;tag=as0859bae5 To: Call-ID: 3cef9a16483fc7d72a65c1466a6acbe7@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '3cef9a16483fc7d72a65c1466a6acbe7@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> == Spawn extension (trunkld, 916478831388, 1) exited non-zero on 'Local/916478831388@trunkld-1676,2' pbxsipgw-01*CLI> -- calling 916472044341@trunkld pbxsipgw-01*CLI> Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> -- Called comwave-nextone/16472044341 pbxsipgw-01*CLI> -- Executing [916472044341@trunkld:1] Dial("Local/916472044341@trunkld-6bfe,2", "SIP/comwave-nextone/16472044341") in new stack pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3df800a5 From: "CarrierSupport" ;tag=as1ce32652 To: Call-ID: 56ee7a1e1a5a221a4a67337d494689f3@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK142b912f From: "CarrierSupport" ;tag=as293d1d50 To: Contact: Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 16990 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/14167978854 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK142b912f From: "CarrierSupport" ;tag=as293d1d50 To: Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> == Spawn extension (trunkld, 914167978854, 1) exited non-zero on 'Local/914167978854@trunkld-8d3f,2' pbxsipgw-01*CLI> Adding codec 0x2 (gsm) to SDP pbxsipgw-01*CLI> Adding codec 0x8 (alaw) to SDP pbxsipgw-01*CLI> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3df800a5 To: From: "CarrierSupport" ;tag=as1ce32652 Call-ID: 56ee7a1e1a5a221a4a67337d494689f3@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> Audio is at 10.5.1.30 port 12582 pbxsipgw-01*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK641b189f From: "CarrierSupport" ;tag=as53880a2f To: Contact: Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 12582 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/14165093362 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK641b189f From: "CarrierSupport" ;tag=as53880a2f To: Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 12008 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73149900 From: "CarrierSupport" ;tag=as02c5f8ec To: Call-ID: 7edf56753422e0ee7dc6967723ed6512@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '7edf56753422e0ee7dc6967723ed6512@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> == Spawn extension (trunkld, 916472044341, 1) exited non-zero on 'Local/916472044341@trunkld-e52d,2' pbxsipgw-01*CLI> == Spawn extension (trunkld, 914165093362, 1) exited non-zero on 'Local/914165093362@trunkld-666b,2' pbxsipgw-01*CLI> Audio is at 10.5.1.30 port 10980 --- (9 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5ff76373 To: ;tag=3433592915-619608 From: "CarrierSupport" ;tag=as31288e0b Reason: SIP;cause=487;text="487 Request Terminated" Call-ID: 29445fb0294636c149057d7d0e7b3c48@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5ff76373 From: "CarrierSupport" ;tag=as31288e0b To: ;tag=3433592915-619608 Contact: Call-ID: 29445fb0294636c149057d7d0e7b3c48@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '29445fb0294636c149057d7d0e7b3c48@10.5.1.30' Method: INVITE Audio is at 10.5.1.30 port 11340 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK217d85c0 From: "CarrierSupport" ;tag=as754b4586 To: Contact: Call-ID: 4fe5fbf945314180116b012477fee2e3@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 11340 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/14167978854 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Audio is at 10.5.1.30 port 13040 Adding non-codec 0x1 (telephone-event) to SDP -- Called comwave-nextone/14167978854 Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK217d85c0 From: "CarrierSupport" ;tag=as754b4586 To: Call-ID: 4fe5fbf945314180116b012477fee2e3@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '4fe5fbf945314180116b012477fee2e3@10.5.1.30' in 32000 ms (Method: INVITE) Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK22e441c6 From: "CarrierSupport" ;tag=as4b45cb87 To: Contact: Call-ID: 63b4e95c2f3446d80ecd18c71f62857b@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 12008 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- calling 914165093362@trunkld Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK418cdd6a From: "CarrierSupport" ;tag=as600ef616 To: Call-ID: 51b9e4e70d9cd5161beadcae55042dbc@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '51b9e4e70d9cd5161beadcae55042dbc@10.5.1.30' in 32000 ms (Method: INVITE) -- Executing [914165093362@trunkld:1] Dial("Local/914165093362@trunkld-54a3,2", "SIP/comwave-nextone/14165093362") in new stack == Spawn extension (trunkld, 914167978854, 1) exited non-zero on 'Local/914167978854@trunkld-510e,2' <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73149900 From: "CarrierSupport" ;tag=as02c5f8ec To: Call-ID: 7edf56753422e0ee7dc6967723ed6512@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK4f22af6d From: "CarrierSupport" ;tag=as24e4eac8 To: Contact: Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 10980 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/16472044341 Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK4f22af6d From: "CarrierSupport" ;tag=as24e4eac8 To: Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Called comwave-nextone/16472044341 Audio is at 10.5.1.30 port 12568 --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK418cdd6a From: "CarrierSupport" ;tag=as600ef616 To: Call-ID: 51b9e4e70d9cd5161beadcae55042dbc@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3df800a5 To: ;tag=3433592926-496357 From: "CarrierSupport" ;tag=as1ce32652 Call-ID: 56ee7a1e1a5a221a4a67337d494689f3@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3df800a5 From: "CarrierSupport" ;tag=as1ce32652 To: ;tag=3433592926-496357 Contact: Call-ID: 56ee7a1e1a5a221a4a67337d494689f3@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '56ee7a1e1a5a221a4a67337d494689f3@10.5.1.30' Method: INVITE Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2e8e0c49 From: "CarrierSupport" ;tag=as175fe1f0 To: Contact: Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 10690 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- == Spawn extension (trunkld, 914167978854, 1) exited non-zero on 'Local/914167978854@trunkld-8e98,2' Adding codec 0x4 (ulaw) to SDP Scheduling destruction of SIP dialog '557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30' in 32000 ms (Method: INVITE) -- Called comwave-nextone/16478831388 == Spawn extension (trunkld, 916472044341, 1) exited non-zero on 'Local/916472044341@trunkld-0158,2' -- calling 914167978854@trunkld Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP -- Executing [914167978854@trunkld:1] Dial("Local/914167978854@trunkld-5a51,2", "SIP/comwave-nextone/14167978854") in new stack Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK78a0fc04 From: "CarrierSupport" ;tag=as674f18ff To: Contact: Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 13040 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- calling 916478831388@trunkld -- Called comwave-nextone/14165093362 Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK78a0fc04 From: "CarrierSupport" ;tag=as674f18ff To: Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2e8e0c49 From: "CarrierSupport" ;tag=as175fe1f0 To: Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK22e441c6 From: "CarrierSupport" ;tag=as4b45cb87 To: Call-ID: 63b4e95c2f3446d80ecd18c71f62857b@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK71e2fea3 To: ;tag=3433592915-517562 From: "CarrierSupport" ;tag=as420668c3 Reason: SIP;cause=487;text="487 Request Terminated" Call-ID: 2742f80f78412518301c50c02e0a094c@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> Scheduling destruction of SIP dialog '2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30' in 32000 ms (Method: INVITE) -- Executing [916478831388@trunkld:1] Dial("Local/916478831388@trunkld-f0a3,2", "SIP/comwave-nextone/16478831388") in new stack Adding codec 0x4 (ulaw) to SDP Audio is at 10.5.1.30 port 11128 --- (11 headers 0 lines) --- Audio is at 10.5.1.30 port 17136 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK125de900 From: "CarrierSupport" ;tag=as3e9d5d7b To: Contact: Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 17136 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/16478831388 Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK125de900 From: "CarrierSupport" ;tag=as3e9d5d7b To: Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '6c92551b6cecfdad6de12b375b2398bb@10.5.1.30' in 32000 ms (Method: INVITE) == Spawn extension (trunkld, 914165093362, 1) exited non-zero on 'Local/914165093362@trunkld-6483,2' Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP == Spawn extension (trunkld, 916478831388, 1) exited non-zero on 'Local/916478831388@trunkld-03cf,2' Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK71e2fea3 From: "CarrierSupport" ;tag=as420668c3 To: ;tag=3433592915-517562 Contact: Call-ID: 2742f80f78412518301c50c02e0a094c@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '2742f80f78412518301c50c02e0a094c@10.5.1.30' Method: INVITE Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec From: "CarrierSupport" ;tag=as2607b576 To: Contact: Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 11128 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/14167978854 Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec From: "CarrierSupport" ;tag=as2607b576 To: Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '34b965a109bd935d210ec4e8319c9771@10.5.1.30' in 32000 ms (Method: INVITE) == Spawn extension (trunkld, 914167978854, 1) exited non-zero on 'Local/914167978854@trunkld-5a51,2' Scheduling destruction of SIP dialog '63b4e95c2f3446d80ecd18c71f62857b@10.5.1.30' in 32000 ms (Method: INVITE) == Spawn extension (trunkld, 916472044341, 1) exited non-zero on 'Local/916472044341@trunkld-6bfe,2' Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.2.12.6:5060: INVITE sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c From: "CarrierSupport" ;tag=as3614de04 To: Contact: Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:40:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 281 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 12568 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called comwave-nextone/14165093362 Reliably Transmitting (no NAT) to 10.2.12.6:5060: CANCEL sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c From: "CarrierSupport" ;tag=as3614de04 To: Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30' in 32000 ms (Method: INVITE) == Spawn extension (trunkld, 914165093362, 1) exited non-zero on 'Local/914165093362@trunkld-54a3,2' == Spawn extension (trunkld, 916478831388, 1) exited non-zero on 'Local/916478831388@trunkld-f0a3,2' Really destroying SIP dialog '4166470-3433592901-624855@msx-02.mgmt.hook2' Method: BYE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73a66919 From: "CarrierSupport" ;tag=as0859bae5 To: Call-ID: 3cef9a16483fc7d72a65c1466a6acbe7@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK0035d991 From: "CarrierSupport" ;tag=as42787f48 To: Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK1d1686fb To: ;tag=3433592915-516865 From: "CarrierSupport" ;tag=as60673069 Reason: SIP;cause=487;text="487 Request Terminated" Call-ID: 63decba6508d77665a88f1d60608373c@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14165093362@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK1d1686fb From: "CarrierSupport" ;tag=as60673069 To: ;tag=3433592915-516865 Contact: Call-ID: 63decba6508d77665a88f1d60608373c@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '63decba6508d77665a88f1d60608373c@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK204a1f64 From: "CarrierSupport" ;tag=as4bad16e2 To: Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK0035d991 To: From: "CarrierSupport" ;tag=as42787f48 Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK004c8a9b From: "CarrierSupport" ;tag=as3bb52046 To: Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73149900 To: From: "CarrierSupport" ;tag=as02c5f8ec Call-ID: 7edf56753422e0ee7dc6967723ed6512@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK217d85c0 From: "CarrierSupport" ;tag=as754b4586 To: Call-ID: 4fe5fbf945314180116b012477fee2e3@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK204a1f64 To: From: "CarrierSupport" ;tag=as4bad16e2 Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK142b912f From: "CarrierSupport" ;tag=as293d1d50 To: Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73149900 To: ;tag=3433592926-506595 From: "CarrierSupport" ;tag=as02c5f8ec Call-ID: 7edf56753422e0ee7dc6967723ed6512@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73149900 From: "CarrierSupport" ;tag=as02c5f8ec To: ;tag=3433592926-506595 Contact: Call-ID: 7edf56753422e0ee7dc6967723ed6512@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '7edf56753422e0ee7dc6967723ed6512@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK004c8a9b To: From: "CarrierSupport" ;tag=as3bb52046 Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK22e441c6 From: "CarrierSupport" ;tag=as4b45cb87 To: Call-ID: 63b4e95c2f3446d80ecd18c71f62857b@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK641b189f From: "CarrierSupport" ;tag=as53880a2f To: Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2e8e0c49 From: "CarrierSupport" ;tag=as175fe1f0 To: Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK217d85c0 To: From: "CarrierSupport" ;tag=as754b4586 Call-ID: 4fe5fbf945314180116b012477fee2e3@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73a66919 To: From: "CarrierSupport" ;tag=as0859bae5 Call-ID: 3cef9a16483fc7d72a65c1466a6acbe7@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK4f22af6d From: "CarrierSupport" ;tag=as24e4eac8 To: Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK125de900 From: "CarrierSupport" ;tag=as3e9d5d7b To: Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2e8e0c49 To: From: "CarrierSupport" ;tag=as175fe1f0 Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK142b912f To: From: "CarrierSupport" ;tag=as293d1d50 Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK22e441c6 To: From: "CarrierSupport" ;tag=as4b45cb87 Call-ID: 63b4e95c2f3446d80ecd18c71f62857b@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK78a0fc04 From: "CarrierSupport" ;tag=as674f18ff To: Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK641b189f To: From: "CarrierSupport" ;tag=as53880a2f Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK418cdd6a To: From: "CarrierSupport" ;tag=as600ef616 Call-ID: 51b9e4e70d9cd5161beadcae55042dbc@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec From: "CarrierSupport" ;tag=as2607b576 To: Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK125de900 To: From: "CarrierSupport" ;tag=as3e9d5d7b Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK4f22af6d To: From: "CarrierSupport" ;tag=as24e4eac8 Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c From: "CarrierSupport" ;tag=as3614de04 To: Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK78a0fc04 To: From: "CarrierSupport" ;tag=as674f18ff Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK22e441c6 To: ;tag=3433592926-521893 From: "CarrierSupport" ;tag=as4b45cb87 Call-ID: 63b4e95c2f3446d80ecd18c71f62857b@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16472044341@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK22e441c6 From: "CarrierSupport" ;tag=as4b45cb87 To: ;tag=3433592926-521893 Contact: Call-ID: 63b4e95c2f3446d80ecd18c71f62857b@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '63b4e95c2f3446d80ecd18c71f62857b@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73a66919 To: ;tag=3433592926-522504 From: "CarrierSupport" ;tag=as0859bae5 Call-ID: 3cef9a16483fc7d72a65c1466a6acbe7@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16478831388@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK73a66919 From: "CarrierSupport" ;tag=as0859bae5 To: ;tag=3433592926-522504 Contact: Call-ID: 3cef9a16483fc7d72a65c1466a6acbe7@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '3cef9a16483fc7d72a65c1466a6acbe7@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK418cdd6a To: ;tag=3433592926-523088 From: "CarrierSupport" ;tag=as600ef616 Call-ID: 51b9e4e70d9cd5161beadcae55042dbc@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14167978854@10.2.12.6 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK418cdd6a From: "CarrierSupport" ;tag=as600ef616 To: ;tag=3433592926-523088 Contact: Call-ID: 51b9e4e70d9cd5161beadcae55042dbc@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '51b9e4e70d9cd5161beadcae55042dbc@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK0035d991 To: ;tag=3433592927-944083 From: "CarrierSupport" ;tag=as42787f48 Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 150673697 150673430 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31866 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31866 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31866 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK641b189f To: ;tag=3433592928-95454 From: "CarrierSupport" ;tag=as53880a2f Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161513836 161513542 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31890 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31890 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31890 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK0035d991 To: ;tag=3433592927-944083 From: "CarrierSupport" ;tag=as42787f48 Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 150673697 150673430 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31866 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31866 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31866 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK125de900 To: ;tag=3433592928-269002 From: "CarrierSupport" ;tag=as3e9d5d7b Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151203065 151202804 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31894 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31894 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31894 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK641b189f To: ;tag=3433592928-95454 From: "CarrierSupport" ;tag=as53880a2f Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161513836 161513542 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31890 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31890 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31890 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK142b912f To: ;tag=3433592928-348238 From: "CarrierSupport" ;tag=as293d1d50 Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151300448 151300181 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31898 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31898 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31898 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK4f22af6d To: ;tag=3433592928-348881 From: "CarrierSupport" ;tag=as24e4eac8 Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161662973 161662706 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31902 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 pbxsipgw-01*CLI> Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31902 Found audio description format PCMU for ID 0 pbxsipgw-01*CLI> Found audio description format telephone-event for ID 101 pbxsipgw-01*CLI> Got unsupported a:fmtp in SDP offer pbxsipgw-01*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31902 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2e8e0c49 To: ;tag=3433592928-444292 From: "CarrierSupport" ;tag=as175fe1f0 Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151103704 151103444 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31886 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31886 Found audio description format PCMU for ID 0 pbxsipgw-01*CLI> Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31886 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK0035d991 To: ;tag=3433592927-944083 From: "CarrierSupport" ;tag=as42787f48 Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 150673697 150673430 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31866 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31866 Found audio description format PCMU for ID 0 pbxsipgw-01*CLI> Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31866 pbxsipgw-01*CLI> list_route: hop: pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16478831388@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK41746726 From: "CarrierSupport" ;tag=as42787f48 To: ;tag=3433592927-944083 Contact: Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:16478831388@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK21cdf9b4 From: "CarrierSupport" ;tag=as42787f48 To: ;tag=3433592927-944083 Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '7011db9b4fa927df16ce9c4c3441e914@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK21cdf9b4 To: ;tag=3433592927-944083 From: "CarrierSupport" ;tag=as42787f48 Call-ID: 7011db9b4fa927df16ce9c4c3441e914@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- Really destroying SIP dialog '7011db9b4fa927df16ce9c4c3441e914@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK125de900 To: ;tag=3433592928-269002 From: "CarrierSupport" ;tag=as3e9d5d7b Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151203065 151202804 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31894 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31894 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31894 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK142b912f To: ;tag=3433592928-348238 From: "CarrierSupport" ;tag=as293d1d50 Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151300448 151300181 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31898 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31898 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31898 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK641b189f To: ;tag=3433592928-95454 From: "CarrierSupport" ;tag=as53880a2f Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161513836 161513542 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31890 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31890 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31890 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14165093362@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK15d7152c From: "CarrierSupport" ;tag=as53880a2f To: ;tag=3433592928-95454 Contact: Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:14165093362@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3b48a014 From: "CarrierSupport" ;tag=as53880a2f To: ;tag=3433592928-95454 Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK4f22af6d To: ;tag=3433592928-348881 From: "CarrierSupport" ;tag=as24e4eac8 Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161662973 161662706 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31902 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31902 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31902 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3b48a014 To: ;tag=3433592928-95454 From: "CarrierSupport" ;tag=as53880a2f Call-ID: 42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- pbxsipgw-01*CLI> Really destroying SIP dialog '42bad5fb2689feda3ec8e03c6ea4a377@10.5.1.30' Method: INVITE pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31910 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31910 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK125de900 To: ;tag=3433592928-269002 From: "CarrierSupport" ;tag=as3e9d5d7b Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151203065 151202804 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31894 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31894 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31894 pbxsipgw-01*CLI> list_route: hop: pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to pbxsipgw-01*CLI> set_destination: set destination to 10.2.12.6, port 5060 pbxsipgw-01*CLI> Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16478831388@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK1a9207bc From: "CarrierSupport" ;tag=as3e9d5d7b To: ;tag=3433592928-269002 Contact: Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to pbxsipgw-01*CLI> set_destination: set destination to 10.2.12.6, port 5060 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:16478831388@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK228d6fa6 From: "CarrierSupport" ;tag=as3e9d5d7b To: ;tag=3433592928-269002 Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '6c92551b6cecfdad6de12b375b2398bb@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK228d6fa6 To: ;tag=3433592928-269002 From: "CarrierSupport" ;tag=as3e9d5d7b Call-ID: 6c92551b6cecfdad6de12b375b2398bb@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- Really destroying SIP dialog '6c92551b6cecfdad6de12b375b2398bb@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK4f22af6d To: ;tag=3433592928-348881 From: "CarrierSupport" ;tag=as24e4eac8 Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161662973 161662706 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31902 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31902 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31902 pbxsipgw-01*CLI> list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16472044341@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2148d714 From: "CarrierSupport" ;tag=as24e4eac8 To: ;tag=3433592928-348881 Contact: Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:16472044341@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK55fbc50e From: "CarrierSupport" ;tag=as24e4eac8 To: ;tag=3433592928-348881 Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK55fbc50e To: ;tag=3433592928-348881 From: "CarrierSupport" ;tag=as24e4eac8 Call-ID: 557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- Really destroying SIP dialog '557cecc777f7534d4b7f5d7c5f98d1ef@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK142b912f To: ;tag=3433592928-348238 From: "CarrierSupport" ;tag=as293d1d50 Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151300448 151300181 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31898 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 pbxsipgw-01*CLI> Found RTP audio format 101 pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31898 pbxsipgw-01*CLI> Found audio description format PCMU for ID 0 pbxsipgw-01*CLI> Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31898 list_route: hop: set_destination: Parsing for address/port to send to pbxsipgw-01*CLI> set_destination: set destination to 10.2.12.6, port 5060 Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14167978854@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK48e4e30a From: "CarrierSupport" ;tag=as293d1d50 To: ;tag=3433592928-348238 Contact: Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:14167978854@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2f4f10a9 From: "CarrierSupport" ;tag=as293d1d50 To: ;tag=3433592928-348238 Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2f4f10a9 To: ;tag=3433592928-348238 From: "CarrierSupport" ;tag=as293d1d50 Call-ID: 564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- Really destroying SIP dialog '564dd85c65250ee11c0a4c6003d7be7c@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK217d85c0 To: ;tag=3433592928-935447 From: "CarrierSupport" ;tag=as754b4586 Call-ID: 4fe5fbf945314180116b012477fee2e3@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 150868877 150868616 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31878 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31878 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31878 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK004c8a9b To: ;tag=3433592928-969337 From: "CarrierSupport" ;tag=as3bb52046 Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161165417 161165150 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31874 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31874 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31874 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK004c8a9b To: ;tag=3433592928-969337 From: "CarrierSupport" ;tag=as3bb52046 Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161165417 161165150 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31874 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31874 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31874 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31914 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31914 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK204a1f64 To: ;tag=3433592929-143274 From: "CarrierSupport" ;tag=as4bad16e2 Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161064448 161064180 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31870 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31870 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31870 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK78a0fc04 To: ;tag=3433592929-192782 From: "CarrierSupport" ;tag=as674f18ff Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161772669 161772403 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31906 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31906 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31906 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK004c8a9b To: ;tag=3433592928-969337 From: "CarrierSupport" ;tag=as3bb52046 Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161165417 161165150 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31874 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 pbxsipgw-01*CLI> Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31874 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31874 list_route: hop: pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to pbxsipgw-01*CLI> set_destination: set destination to 10.2.12.6, port 5060 pbxsipgw-01*CLI> Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14165093362@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2f9e5a24 From: "CarrierSupport" ;tag=as3bb52046 To: ;tag=3433592928-969337 Contact: Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to pbxsipgw-01*CLI> set_destination: set destination to 10.2.12.6, port 5060 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:14165093362@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5d47c5a6 From: "CarrierSupport" ;tag=as3bb52046 To: ;tag=3433592928-969337 Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5d47c5a6 To: ;tag=3433592928-969337 From: "CarrierSupport" ;tag=as3bb52046 Call-ID: 37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- Really destroying SIP dialog '37e435ea70b321e24d9b1f0a140ea5c8@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK78a0fc04 To: ;tag=3433592929-192782 From: "CarrierSupport" ;tag=as674f18ff Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161772669 161772403 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31906 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 pbxsipgw-01*CLI> Found RTP audio format 101 pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31906 pbxsipgw-01*CLI> Found audio description format PCMU for ID 0 pbxsipgw-01*CLI> Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31906 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK217d85c0 To: ;tag=3433592928-935447 From: "CarrierSupport" ;tag=as754b4586 Call-ID: 4fe5fbf945314180116b012477fee2e3@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 150868877 150868616 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31878 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31878 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31878 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14167978854@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK579bc3da From: "CarrierSupport" ;tag=as754b4586 To: ;tag=3433592928-935447 Contact: Call-ID: 4fe5fbf945314180116b012477fee2e3@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:14167978854@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK43aab7d5 From: "CarrierSupport" ;tag=as754b4586 To: ;tag=3433592928-935447 Call-ID: 4fe5fbf945314180116b012477fee2e3@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '4fe5fbf945314180116b012477fee2e3@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK43aab7d5 To: ;tag=3433592928-935447 From: "CarrierSupport" ;tag=as754b4586 Call-ID: 4fe5fbf945314180116b012477fee2e3@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- Really destroying SIP dialog '4fe5fbf945314180116b012477fee2e3@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK78a0fc04 To: ;tag=3433592929-192782 From: "CarrierSupport" ;tag=as674f18ff Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161772669 161772403 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31906 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31906 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 pbxsipgw-01*CLI> Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31906 pbxsipgw-01*CLI> list_route: hop: pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to pbxsipgw-01*CLI> set_destination: set destination to 10.2.12.6, port 5060 pbxsipgw-01*CLI> Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:14165093362@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK0c2f0fed From: "CarrierSupport" ;tag=as674f18ff To: ;tag=3433592929-192782 Contact: Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to pbxsipgw-01*CLI> set_destination: set destination to 10.2.12.6, port 5060 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:14165093362@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5d5835ec From: "CarrierSupport" ;tag=as674f18ff To: ;tag=3433592929-192782 Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK5d5835ec To: ;tag=3433592929-192782 From: "CarrierSupport" ;tag=as674f18ff Call-ID: 4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4c42f2cf17ffdd3b28bdf5ff7e364a09@10.5.1.30' Method: INVITE pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2e8e0c49 To: ;tag=3433592928-444292 From: "CarrierSupport" ;tag=as175fe1f0 Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151103704 151103444 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31886 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31886 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31886 <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK2e8e0c49 To: ;tag=3433592928-444292 From: "CarrierSupport" ;tag=as175fe1f0 Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151103704 151103444 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31886 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31886 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31886 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16478831388@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3465b8bb From: "CarrierSupport" ;tag=as175fe1f0 To: ;tag=3433592928-444292 Contact: Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.2.12.6, port 5060 Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:16478831388@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3d3395d0 From: "CarrierSupport" ;tag=as175fe1f0 To: ;tag=3433592928-444292 Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK3d3395d0 To: ;tag=3433592928-444292 From: "CarrierSupport" ;tag=as175fe1f0 Call-ID: 2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- Really destroying SIP dialog '2c5ef5c001de2e52368107d06f6db4bf@10.5.1.30' Method: INVITE <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK204a1f64 To: ;tag=3433592929-143274 From: "CarrierSupport" ;tag=as4bad16e2 Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161064448 161064180 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31870 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.2.12.7:31870 Found audio description format PCMU for ID 0 pbxsipgw-01*CLI> Found audio description format telephone-event for ID 101 pbxsipgw-01*CLI> Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31870 pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK204a1f64 To: ;tag=3433592929-143274 From: "CarrierSupport" ;tag=as4bad16e2 Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161064448 161064180 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31870 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 pbxsipgw-01*CLI> Peer audio RTP is at port 10.2.12.7:31870 pbxsipgw-01*CLI> Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.12.7:31870 pbxsipgw-01*CLI> list_route: hop: pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to pbxsipgw-01*CLI> set_destination: set destination to 10.2.12.6, port 5060 pbxsipgw-01*CLI> Transmitting (no NAT) to 10.2.12.6:5060: ACK sip:16472044341@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK436da504 From: "CarrierSupport" ;tag=as4bad16e2 To: ;tag=3433592929-143274 Contact: Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> set_destination: Parsing for address/port to send to pbxsipgw-01*CLI> set_destination: set destination to 10.2.12.6, port 5060 pbxsipgw-01*CLI> Reliably Transmitting (no NAT) to 10.2.12.6:5060: BYE sip:16472044341@10.2.12.6:5060 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK756f1b17 From: "CarrierSupport" ;tag=as4bad16e2 To: ;tag=3433592929-143274 Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbxsipgw-01*CLI> Scheduling destruction of SIP dialog '233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30' in 32000 ms (Method: INVITE) pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK756f1b17 To: ;tag=3433592929-143274 From: "CarrierSupport" ;tag=as4bad16e2 Call-ID: 233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30 CSeq: 103 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (9 headers 0 lines) --- Really destroying SIP dialog '233b974e1ae6a15e0dc1acc71fba1e9b@10.5.1.30' Method: INVITE Really destroying SIP dialog '34b965a109bd935d210ec4e8319c9771@10.5.1.30' Method: INVITE pbxsipgw-01*CLI> Really destroying SIP dialog '0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30' Method: INVITE pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> BYE sip:4166238520@10.5.1.30 SIP/2.0 Max-Forwards: 69 To: "CarrierSupport" ;tag=as2607b576 From: ;tag=3433592928-729313 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 2 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bKff0acfd186f7e5561168527c3d1823b2 Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (10 headers 0 lines) --- pbxsipgw-01*CLI> Sending to 10.2.12.6 : 5060 (no NAT) pbxsipgw-01*CLI> <--- Transmitting (no NAT) to 10.2.12.6:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bKff0acfd186f7e5561168527c3d1823b2;received=10.2.12.6 From: ;tag=3433592928-729313 To: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> BYE sip:4166238520@10.5.1.30 SIP/2.0 Max-Forwards: 69 To: "CarrierSupport" ;tag=as3614de04 From: ;tag=3433592929-42881 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 2 BYE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bK809328acc48f1ec3477f438d8d2810ab Contact: Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (10 headers 0 lines) --- pbxsipgw-01*CLI> Sending to 10.2.12.6 : 5060 (no NAT) pbxsipgw-01*CLI> <--- Transmitting (no NAT) to 10.2.12.6:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.2.12.6:5060;branch=z9hG4bK809328acc48f1ec3477f438d8d2810ab;received=10.2.12.6 From: ;tag=3433592929-42881 To: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK6c2685ec To: ;tag=3433592928-729313 From: "CarrierSupport" ;tag=as2607b576 Call-ID: 34b965a109bd935d210ec4e8319c9771@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 151397530 151397265 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31910 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.2.12.6:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.30:5060;branch=z9hG4bK14daf44c To: ;tag=3433592929-42881 From: "CarrierSupport" ;tag=as3614de04 Call-ID: 0213ae6b0fd55b332f8f79ce1bc698d6@10.5.1.30 CSeq: 102 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 218 v=0 o=msx-02 161900016 161899750 IN IP4 10.2.12.6 s=sip call c=IN IP4 10.2.12.7 t=0 0 m=audio 31914 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (11 headers 11 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INVITE sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK7243c910 From: "5555558994" ;tag=as3952447e To: Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 102 INVITE User-Agent: Comwave PBX Max-Forwards: 70 Date: Tue, 21 Oct 2008 15:50:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 176 v=0 o=root 1473 1473 IN IP4 10.5.1.10 s=session c=IN IP4 10.5.1.10 t=0 0 m=audio 59618 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> pbxsipgw-01*CLI> --- (14 headers 10 lines) --- pbxsipgw-01*CLI> Sending to 10.5.1.10 : 5060 (no NAT) pbxsipgw-01*CLI> Using INVITE request as basis request - 02735922350951b6333bf86a1e7d132d@10.5.1.10 pbxsipgw-01*CLI> Found peer 'pbxtel' pbxsipgw-01*CLI> Found RTP audio format 0 pbxsipgw-01*CLI> Peer audio RTP is at port 10.5.1.10:59618 pbxsipgw-01*CLI> Found audio description format PCMU for ID 0 pbxsipgw-01*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) pbxsipgw-01*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 10.5.1.10:59618 Looking for 4168833780 in from-pbxtel (domain 10.5.1.30) pbxsipgw-01*CLI> list_route: hop: pbxsipgw-01*CLI> <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK7243c910;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> -- Executing [4168833780@from-pbxtel:1] Dial("SIP/10.5.1.10-0902b9f0", "Zap/g0/4168833780|90|Tt") in new stack pbxsipgw-01*CLI> -- Called g0/4168833780 pbxsipgw-01*CLI> -- Zap/3-1 answered SIP/10.5.1.10-0902b9f0 Audio is at 10.5.1.30 port 16266 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK7243c910;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 178 v=0 o=root 21325 21325 IN IP4 10.5.1.30 s=session c=IN IP4 10.5.1.30 t=0 0 m=audio 16266 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> ACK sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK584d38c7 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 102 ACK User-Agent: Comwave PBX Max-Forwards: 70 Content-Length: 0 <-------------> pbxsipgw-01*CLI> --- (10 headers 0 lines) --- pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK4a594a2a From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 103 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=0 Duration=120 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 0 <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK4a594a2a;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 103 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK3e0c3c39 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 104 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=1 Duration=100 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 1 pbxsipgw-01*CLI> <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK3e0c3c39;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 104 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK5c226c43 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 105 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=1 Duration=140 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 1 <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK5c226c43;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 105 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK08ed92e2 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 106 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=0 Duration=160 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 0 <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK08ed92e2;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 106 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK2dad3165 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 107 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=2 Duration=120 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 2 <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK2dad3165;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 107 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK70def0f4 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 108 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=2 Duration=160 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 2 <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK70def0f4;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 108 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK758bc606 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 109 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=2 Duration=140 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- pbxsipgw-01*CLI> Receiving INFO! pbxsipgw-01*CLI> * DTMF-relay event received: 2 pbxsipgw-01*CLI> <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK758bc606;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 109 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK3ccf082d From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 110 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 23 Signal=4 Duration=80 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 4 <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK3ccf082d;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 110 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK2c21b6c1 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 111 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 23 Signal=1 Duration=80 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- pbxsipgw-01*CLI> Receiving INFO! pbxsipgw-01*CLI> * DTMF-relay event received: 1 pbxsipgw-01*CLI> <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK2c21b6c1;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 111 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> <--- SIP read from 10.5.1.10:5060 ---> INFO sip:4168833780@10.5.1.30 SIP/2.0 Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK682052e1 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Contact: Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 112 INFO User-Agent: Comwave PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 23 Signal=8 Duration=80 <-------------> pbxsipgw-01*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 8 <--- Transmitting (no NAT) to 10.5.1.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.5.1.10:5060;branch=z9hG4bK682052e1;received=10.5.1.10 From: "5555558994" ;tag=as3952447e To: ;tag=as2100df95 Call-ID: 02735922350951b6333bf86a1e7d132d@10.5.1.10 CSeq: 112 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbxsipgw-01*CLI> [root@pbxsipgw-01 asterisk]#