thorium*CLI> thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> INVITE sip:7323680452@172.16.4.4:5060 SIP/2.0 Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bKl2eqme1068fgqb4p32g0.1 From: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 To: CSeq: 933 INVITE Contact: Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 Max-Forwards: 68 Content-Type: application/sdp Content-Length: 270 Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Accept-Language: en; q=0.0 Allow: INVITE, BYE, ACK, CANCEL, PRACK, INFO Content-Disposition: session; handling=required v=0 o=Sonus_UAC 2371 10093 IN IP4 135.25.29.70 s=SIP Media Capabilities c=IN IP4 135.25.29.70 t=0 0 m=audio 17530 RTP/AVP 2 18 0 96 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv <-------------> --- (14 headers 12 lines) --- Sending to 135.25.29.135 : 5060 (no NAT) Using INVITE request as basis request - SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 Found peer 'first_gateway' Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 96 [2007-10-05 18:08:05] DEBUG[26964]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 135.25.29.70:17530 Found description format G726-32 for ID 2 Found description format G729 for ID 18 Found description format PCMU for ID 0 Found description format telephone-event for ID 96 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x904 (ulaw|g726|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 135.25.29.70:17530 Looking for 7323680452 in pri_from_als (domain 172.16.4.4) list_route: hop: thorium*CLI> <--- Transmitting (no NAT) to 135.25.29.135:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bKl2eqme1068fgqb4p32g0.1;received=135.25.29.135 From: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 To: Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 933 INVITE User-Agent: Thorium Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [7323680452@pri_from_als:1] Macro("SIP/172.16.4.4-0892ada0", "stdexten|0452|SIP/garrish") in new stack -- Executing [s@macro-stdexten:1] LDAPget("SIP/172.16.4.4-0892ada0", "CIDNAME=name/||") in new stack -- LDAPget: varname=CIDNAME, config-section=name, keys=|| == Parsing '/etc/asterisk/ldap.conf': Found > LDAPget: ldap://www.post.att.com/o=att,c=us?name?subtree?(|(telephoneNumber=+%s)(telephoneNumber=+1%s)(mobileTelephoneNumber=+1%s)) -- LDAPget: ldap://www.post.att.com/o=att,c=us?name?subtree?(|(telephoneNumber=+)(telephoneNumber=+1)(mobileTelephoneNumber=+1)) -- LDAPget: bind to www.post.att.com anonymously -- LDAPget: Value not found in directory. [2007-10-05 18:08:05] DEBUG[26992]: app_macro.c:337 _macro_exec: Executed application: LDAPget -- Executing [s@macro-stdexten:2] GotoIf("SIP/172.16.4.4-0892ada0", "1?4:3") in new stack -- Goto (macro-stdexten,s,4) [2007-10-05 18:08:05] DEBUG[26992]: app_macro.c:337 _macro_exec: Executed application: GoToIf -- Executing [s@macro-stdexten:4] Dial("SIP/172.16.4.4-0892ada0", "SIP/garrish|30") in new stack [2007-10-05 18:08:05] DEBUG[26992]: chan_sip.c:3210 update_call_counter: Call to peer 'garrish' is 1 out of 10 Audio is at 172.16.4.4 port 24758 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.4.22:5060: INVITE sip:garrish@172.16.4.22:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK4a046d30;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: Contact: Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 CSeq: 102 INVITE User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "OUT_OF_AREA" ;privacy=off;screen=no ate: Fri, 05 Oct 2007 22:08:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 280 v=0 o=root 26891 26891 IN IP4 172.16.4.4 s=session c=IN IP4 172.16.4.4 t=0 0 m=audio 24758 RTP/AVP 18 0 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called garrish thorium*CLI> <--- SIP read from 172.16.4.22:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK4a046d30;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: ;tag=677CFDC3-DF45507C CSeq: 102 INVITE Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- thorium*CLI> <--- SIP read from 172.16.4.22:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK4a046d30;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: ;tag=677CFDC3-DF45507C CSeq: 102 INVITE Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/garrish-0892fe80 is ringing <--- Transmitting (no NAT) to 135.25.29.135:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bKl2eqme1068fgqb4p32g0.1;received=135.25.29.135 From: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 To: ;tag=as7b7dab74 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 933 INVITE User-Agent: Thorium Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> thorium*CLI> <--- SIP read from 172.16.4.22:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK4a046d30;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: ;tag=677CFDC3-DF45507C CSeq: 102 INVITE Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047 Content-Type: application/sdp Content-Length: 198 v=0 o=- 1191622086 1191622086 IN IP4 172.16.4.22 s=Polycom IP Phone c=IN IP4 172.16.4.22 t=0 0 m=audio 16404 RTP/AVP 18 96 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:96 telephone-event/8000 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 18 Found RTP audio format 96 [2007-10-05 18:08:07] DEBUG[26964]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 172.16.4.22:16404 Found description format G729 for ID 18 Found description format telephone-event for ID 96 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.4.22:16404 list_route: hop: [2007-10-05 18:08:07] DEBUG[26964]: chan_sip.c:5788 reqprep: Strict routing enforced for session 7c6909a95de7285c5987da7a06946695@172.16.4.4 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.4.22, port 5060 Transmitting (no NAT) to 172.16.4.22:5060: ACK sip:garrish@172.16.4.22:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK21fcec5c;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: ;tag=677CFDC3-DF45507C Contact: Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 CSeq: 102 ACK User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "OUT_OF_AREA" ;privacy=off;screen=no Content-Length: 0 --- -- SIP/garrish-0892fe80 answered SIP/172.16.4.4-0892ada0 Audio is at 172.16.4.4 port 29982 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP thorium*CLI> <--- Reliably Transmitting (no NAT) to 135.25.29.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bKl2eqme1068fgqb4p32g0.1;received=135.25.29.135 From: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 To: ;tag=as7b7dab74 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 933 INVITE User-Agent: Thorium Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 280 v=0 o=root 26891 26891 IN IP4 172.16.4.4 s=session c=IN IP4 172.16.4.4 t=0 0 m=audio 29982 RTP/AVP 18 0 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/172.16.4.4-0892ada0 and SIP/garrish-0892fe80 [2007-10-05 18:08:07] DEBUG[26992]: chan_sip.c:5788 reqprep: Strict routing enforced for session 7c6909a95de7285c5987da7a06946695@172.16.4.4 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.4.22, port 5060 Audio is at 172.16.4.4 port 24758 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.4.22:5060: INVITE sip:garrish@172.16.4.22:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK43217574;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: ;tag=677CFDC3-DF45507C Contact: Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 CSeq: 103 INVITE User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "OUT_OF_AREA" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 260 v=0 o=root 26891 26892 IN IP4 135.25.29.70 s=session c=IN IP4 135.25.29.70 t=0 0 m=audio 17530 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> ACK sip:7323680452@172.16.4.4 SIP/2.0 Via: SIP/2.0/UDP 135.25.29.135:5060;branch=z9hG4bKm2ruq120d0b10c800581.1 From: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 To: ;tag=as7b7dab74 Max-Forwards: 68 CSeq: 933 ACK Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [2007-10-05 18:08:07] DEBUG[26964]: chan_sip.c:5788 reqprep: Strict routing enforced for session SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 set_destination: Parsing for address/port to send to set_destination: set destination to 135.25.29.135, port 5060 Audio is at 172.16.4.4 port 29982 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 135.25.29.135:5060: INVITE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK3f115a9b;rport From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Contact: Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 102 INVITE User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 258 v=0 o=root 26891 26892 IN IP4 172.16.4.22 s=session c=IN IP4 172.16.4.22 t=0 0 m=audio 16404 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK3f115a9b;rport=5060 From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 102 INVITE <-------------> --- (6 headers 0 lines) --- thorium*CLI> <--- SIP read from 172.16.4.22:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK43217574;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: ;tag=677CFDC3-DF45507C CSeq: 103 INVITE Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047 Content-Type: application/sdp Content-Length: 198 v=0 o=- 1191622086 1191622087 IN IP4 172.16.4.22 s=Polycom IP Phone c=IN IP4 172.16.4.22 t=0 0 m=audio 16404 RTP/AVP 18 96 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:96 telephone-event/8000 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 18 Found RTP audio format 96 [2007-10-05 18:08:07] DEBUG[26964]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 172.16.4.22:16404 Found description format G729 for ID 18 Found description format telephone-event for ID 96 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.4.22:16404 [2007-10-05 18:08:07] DEBUG[26964]: chan_sip.c:5788 reqprep: Strict routing enforced for session 7c6909a95de7285c5987da7a06946695@172.16.4.4 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.4.22, port 5060 Transmitting (no NAT) to 172.16.4.22:5060: ACK sip:garrish@172.16.4.22:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK77b02a33;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: ;tag=677CFDC3-DF45507C Contact: Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 CSeq: 103 ACK User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "OUT_OF_AREA" ;privacy=off;screen=no Content-Length: 0 --- thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK3f115a9b;rport=5060 From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 240 ccept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Accept-Language: en; q=0.0 Allow: INVITE, BYE, ACK, CANCEL, PRACK, INFO Content-Disposition: session; handling=required v=0 o=Sonus_UAC 2371 10094 IN IP4 135.25.29.70 s=SIP Media Capabilities c=IN IP4 135.25.29.70 t=0 0 m=audio 17530 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv <-------------> --- (13 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 96 [2007-10-05 18:08:07] DEBUG[26964]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 135.25.29.70:17530 Found description format G729 for ID 18 Found description format telephone-event for ID 96 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 135.25.29.70:17530 list_route: hop: [2007-10-05 18:08:07] DEBUG[26964]: chan_sip.c:5788 reqprep: Strict routing enforced for session SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 set_destination: Parsing for address/port to send to set_destination: set destination to 135.25.29.135, port 5060 Transmitting (no NAT) to 135.25.29.135:5060: ACK sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK57aedc9e;rport From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Contact: Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 102 ACK User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Reliably Transmitting (no NAT) to 135.25.29.1:5060: OPTIONS sip:135.25.29.1 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK07701adc;rport From: "asterisk" ;tag=as13ded0d0 To: Contact: Call-ID: 715caf024f2c37606942467d61c31d5a@172.16.4.4 CSeq: 102 OPTIONS User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Date: Fri, 05 Oct 2007 22:08:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (no NAT) to 135.25.29.1:5060: OPTIONS sip:135.25.29.1 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK07701adc;rport From: "asterisk" ;tag=as13ded0d0 To: Contact: Call-ID: 715caf024f2c37606942467d61c31d5a@172.16.4.4 CSeq: 102 OPTIONS User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Date: Fri, 05 Oct 2007 22:08:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #2 (no NAT) to 135.25.29.1:5060: OPTIONS sip:135.25.29.1 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK07701adc;rport From: "asterisk" ;tag=as13ded0d0 To: Contact: Call-ID: 715caf024f2c37606942467d61c31d5a@172.16.4.4 CSeq: 102 OPTIONS User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Date: Fri, 05 Oct 2007 22:08:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- thorium*CLI> <--- SIP read from 172.16.4.22:5060 ---> INVITE sip:+17323681000@172.16.4.4 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK8955885eD96E383D From: ;tag=677CFDC3-DF45507C To: "OUT_OF_AREA" ;tag=as23d58722 CSeq: 1 INVITE Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 198 v=0 o=- 1191622086 1191622088 IN IP4 172.16.4.22 s=Polycom IP Phone c=IN IP4 172.16.4.22 t=0 0 m=audio 16404 RTP/AVP 18 96 a=sendonly a=rtpmap:18 G729/8000 a=rtpmap:96 telephone-event/8000 <-------------> --- (14 headers 9 lines) --- Sending to 172.16.4.22 : 5060 (no NAT) Found RTP audio format 18 Found RTP audio format 96 [2007-10-05 18:08:10] DEBUG[26964]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 172.16.4.22:16404 Found description format G729 for ID 18 Found description format telephone-event for ID 96 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.4.22:16404 thorium*CLI> <--- Transmitting (no NAT) to 172.16.4.22:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK8955885eD96E383D;received=172.16.4.22 From: ;tag=677CFDC3-DF45507C To: "OUT_OF_AREA" ;tag=as23d58722 Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 CSeq: 1 INVITE User-Agent: Thorium Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 172.16.4.4 port 24758 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP thorium*CLI> <--- Transmitting (no NAT) to 172.16.4.22:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK8955885eD96E383D;received=172.16.4.22 From: ;tag=677CFDC3-DF45507C To: "OUT_OF_AREA" ;tag=as23d58722 Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 CSeq: 1 INVITE User-Agent: Thorium Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 26891 26893 IN IP4 135.25.29.70 s=session c=IN IP4 135.25.29.70 t=0 0 m=audio 17530 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [2007-10-05 18:08:10] DEBUG[26992]: chan_sip.c:5788 reqprep: Strict routing enforced for session SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 set_destination: Parsing for address/port to send to set_destination: set destination to 135.25.29.135, port 5060 Audio is at 172.16.4.4 port 29982 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 135.25.29.135:5060: INVITE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK01289afd;rport From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Contact: Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 103 INVITE User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 256 v=0 o=root 26891 26893 IN IP4 172.16.4.4 s=session c=IN IP4 172.16.4.4 t=0 0 m=audio 29982 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Started music on hold, class 'sip', on SIP/172.16.4.4-0892ada0 thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK01289afd;rport=5060 From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 103 INVITE <-------------> --- (6 headers 0 lines) --- thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK01289afd;rport=5060 From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 103 INVITE Contact: Content-Type: application/sdp Content-Length: 240 ccept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Accept-Language: en; q=0.0 Allow: INVITE, BYE, ACK, CANCEL, PRACK, INFO Content-Disposition: session; handling=required v=0 o=Sonus_UAC 2371 10095 IN IP4 135.25.29.70 s=SIP Media Capabilities c=IN IP4 135.25.29.70 t=0 0 m=audio 17530 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv <-------------> --- (13 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 96 [2007-10-05 18:08:10] DEBUG[26964]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 135.25.29.70:17530 Found description format G729 for ID 18 Found description format telephone-event for ID 96 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 135.25.29.70:17530 [2007-10-05 18:08:10] DEBUG[26964]: chan_sip.c:5788 reqprep: Strict routing enforced for session SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 set_destination: Parsing for address/port to send to set_destination: set destination to 135.25.29.135, port 5060 Transmitting (no NAT) to 135.25.29.135:5060: ACK sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK2f6a035b;rport From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Contact: Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 103 ACK User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- thorium*CLI> <--- SIP read from 172.16.4.22:5060 ---> ACK sip:+17323681000@172.16.4.4 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK832894904D05DAC7 From: ;tag=677CFDC3-DF45507C To: "OUT_OF_AREA" ;tag=as23d58722 CSeq: 1 ACK Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Retransmitting #3 (no NAT) to 135.25.29.1:5060: OPTIONS sip:135.25.29.1 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK07701adc;rport From: "asterisk" ;tag=as13ded0d0 To: Contact: Call-ID: 715caf024f2c37606942467d61c31d5a@172.16.4.4 CSeq: 102 OPTIONS User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Date: Fri, 05 Oct 2007 22:08:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #4 (no NAT) to 135.25.29.1:5060: OPTIONS sip:135.25.29.1 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK07701adc;rport From: "asterisk" ;tag=as13ded0d0 To: Contact: Call-ID: 715caf024f2c37606942467d61c31d5a@172.16.4.4 CSeq: 102 OPTIONS User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Date: Fri, 05 Oct 2007 22:08:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '715caf024f2c37606942467d61c31d5a@172.16.4.4' Method: OPTIONS [2007-10-05 18:08:12] DEBUG[26964]: chan_sip.c:10891 sip_dump_history: ---------- SIP HISTORY for '715caf024f2c37606942467d61c31d5a@172.16.4.4' [2007-10-05 18:08:12] DEBUG[26964]: chan_sip.c:10895 sip_dump_history: * SIP Call [2007-10-05 18:08:12] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 001. TxReqRel OPTIONS / 102 OPTIONS - -UNKNOWN- [2007-10-05 18:08:12] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 002. ReTx 1000 OPTIONS sip:135.25.29.1 SIP/2.0 [2007-10-05 18:08:12] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 003. ReTx 1000 OPTIONS sip:135.25.29.1 SIP/2.0 [2007-10-05 18:08:12] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 004. ReTx 1000 OPTIONS sip:135.25.29.1 SIP/2.0 [2007-10-05 18:08:12] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 005. ReTx 1000 OPTIONS sip:135.25.29.1 SIP/2.0 [2007-10-05 18:08:12] DEBUG[26964]: chan_sip.c:10901 sip_dump_history: ---------- END SIP HISTORY for '715caf024f2c37606942467d61c31d5a@172.16.4.4' thorium*CLI> <--- SIP read from 172.16.4.22:5060 ---> INVITE sip:+17323681000@172.16.4.4 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK85840992CBDD6FE1 From: ;tag=677CFDC3-DF45507C To: "OUT_OF_AREA" ;tag=as23d58722 CSeq: 2 INVITE Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 224 v=0 o=- 1191622086 1191622089 IN IP4 172.16.4.22 s=Polycom IP Phone c=IN IP4 172.16.4.22 t=0 0 m=audio 16404 RTP/AVP 18 0 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 10 lines) --- Sending to 172.16.4.22 : 5060 (no NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 172.16.4.22:16404 Found description format G729 for ID 18 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.4.22:16404 <--- Transmitting (no NAT) to 172.16.4.22:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK85840992CBDD6FE1;received=172.16.4.22 From: ;tag=677CFDC3-DF45507C To: "OUT_OF_AREA" ;tag=as23d58722 Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 CSeq: 2 INVITE User-Agent: Thorium Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 172.16.4.4 port 24758 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 172.16.4.22:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK85840992CBDD6FE1;received=172.16.4.22 From: ;tag=677CFDC3-DF45507C To: "OUT_OF_AREA" ;tag=as23d58722 Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 CSeq: 2 INVITE User-Agent: Thorium Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 26891 26894 IN IP4 135.25.29.70 s=session c=IN IP4 135.25.29.70 t=0 0 m=audio 17530 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2007-10-05 18:08:14] DEBUG[26992]: chan_sip.c:5788 reqprep: Strict routing enforced for session SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 set_destination: Parsing for address/port to send to set_destination: set destination to 135.25.29.135, port 5060 Audio is at 172.16.4.4 port 29982 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 135.25.29.135:5060: INVITE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK2d07f837;rport From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Contact: Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 104 INVITE User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 258 v=0 o=root 26891 26894 IN IP4 172.16.4.22 s=session c=IN IP4 172.16.4.22 t=0 0 m=audio 16404 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Stopped music on hold on SIP/172.16.4.4-0892ada0 thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK2d07f837;rport=5060 From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 104 INVITE <-------------> --- (6 headers 0 lines) --- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:5788 reqprep: Strict routing enforced for session SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 set_destination: Parsing for address/port to send to set_destination: set destination to 135.25.29.135, port 5060 Audio is at 172.16.4.4 port 29982 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 135.25.29.135:5060: INVITE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK50ec5ed7;rport From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Contact: Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 105 INVITE User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 258 v=0 o=root 26891 26895 IN IP4 172.16.4.22 s=session c=IN IP4 172.16.4.22 t=0 0 m=audio 16404 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK50ec5ed7;rport=5060 From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 105 INVITE <-------------> --- (6 headers 0 lines) --- thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK50ec5ed7;rport=5060 From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 105 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:5788 reqprep: Strict routing enforced for session SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 set_destination: Parsing for address/port to send to set_destination: set destination to 135.25.29.135, port 5060 Transmitting (no NAT) to 135.25.29.135:5060: ACK sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK50ec5ed7;rport From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Contact: Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 105 ACK User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [2007-10-05 18:08:14] DEBUG[26992]: rtp.c:2957 bridge_native_loop: Got a FRAME_CONTROL (8) frame on channel SIP/172.16.4.4-0892ada0 [2007-10-05 18:08:14] DEBUG[26992]: chan_sip.c:3184 update_call_counter: Call to peer 'garrish' removed from call limit 10 Scheduling destruction of SIP dialog '7c6909a95de7285c5987da7a06946695@172.16.4.4' in 6400 ms (Method: INVITE) == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'SIP/172.16.4.4-0892ada0' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'SIP/172.16.4.4-0892ada0' Scheduling destruction of SIP dialog 'SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32' in 640 ms (Method: ACK) [2007-10-05 18:08:14] DEBUG[26992]: chan_sip.c:5788 reqprep: Strict routing enforced for session SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 set_destination: Parsing for address/port to send to set_destination: set destination to 135.25.29.135, port 5060 Reliably Transmitting (no NAT) to 135.25.29.135:5060: BYE sip:+17323681000@135.25.29.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK7a31bf04;rport From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 106 BYE User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- thorium*CLI> <--- SIP read from 135.25.29.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.4.4:5060;received=172.16.4.4;branch=z9hG4bK7a31bf04;rport=5060 From: ;tag=as7b7dab74 To: "OUT_OF_AREA" ;tag=SDpcljc01-1191622085-4500241192072109-11 Call-ID: SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 CSeq: 106 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- thorium*CLI> <--- SIP read from 172.16.4.22:5060 ---> ACK sip:+17323681000@172.16.4.4 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.22:5060;branch=z9hG4bK190efe447C3660B From: ;tag=677CFDC3-DF45507C To: "OUT_OF_AREA" ;tag=as23d58722 CSeq: 2 ACK Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:5788 reqprep: Strict routing enforced for session 7c6909a95de7285c5987da7a06946695@172.16.4.4 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.4.22, port 5060 Reliably Transmitting (no NAT) to 172.16.4.22:5060: BYE sip:garrish@172.16.4.22:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK390342c6;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: ;tag=677CFDC3-DF45507C Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 CSeq: 104 BYE User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "OUT_OF_AREA" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '7c6909a95de7285c5987da7a06946695@172.16.4.4' in 6400 ms (Method: ACK) thorium*CLI> <--- SIP read from 172.16.4.22:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK390342c6;rport From: "OUT_OF_AREA" ;tag=as23d58722 To: ;tag=677CFDC3-DF45507C CSeq: 104 BYE Call-ID: 7c6909a95de7285c5987da7a06946695@172.16.4.4 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/2.2.0.0047 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '7c6909a95de7285c5987da7a06946695@172.16.4.4' Method: ACK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10891 sip_dump_history: ---------- SIP HISTORY for '7c6909a95de7285c5987da7a06946695@172.16.4.4' [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10895 sip_dump_history: * SIP Call [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 001. NewChan Channel SIP/garrish-0892fe80 - from 7c6909a95de7285c5987da7a069 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 002. TxReqRel INVITE / 102 INVITE - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 003. Rx SIP/2.0 / 102 INVITE / 100 Trying [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 004. Rx SIP/2.0 / 102 INVITE / 180 Ringing [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 005. Rx SIP/2.0 / 102 INVITE / 200 OK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 006. TxReq ACK / 102 ACK - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 007. ReInv Re-invite sent [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 008. TxReqRel INVITE / 103 INVITE - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 009. Rx SIP/2.0 / 103 INVITE / 200 OK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 010. TxReq ACK / 103 ACK - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 011. Rx INVITE / 1 INVITE / sip:+17323681000@172.16.4.4 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 012. Hold INVITE [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 013. ReInv Re-invite received [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 014. TxResp SIP/2.0 / 1 INVITE - 100 Trying [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 015. TxResp SIP/2.0 / 1 INVITE - 200 OK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 016. Rx ACK / 1 ACK / sip:+17323681000@172.16.4.4 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 017. Rx INVITE / 2 INVITE / sip:+17323681000@172.16.4.4 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 018. Unhold INVITE [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 019. ReInv Re-invite received [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 020. TxResp SIP/2.0 / 2 INVITE - 100 Trying [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 021. TxResp SIP/2.0 / 2 INVITE - 200 OK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 022. Hangup Cause Normal Clearing [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 023. SchedDestroy 6400 ms [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 024. CancelDestroy [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 025. Rx ACK / 2 ACK / sip:+17323681000@172.16.4.4 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 026. TxReqRel BYE / 104 BYE - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 027. SchedDestroy 6400 ms [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 028. Rx SIP/2.0 / 104 BYE / 200 OK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10901 sip_dump_history: ---------- END SIP HISTORY for '7c6909a95de7285c5987da7a06946695@172.16.4.4' Really destroying SIP dialog 'SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32' Method: ACK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10891 sip_dump_history: ---------- SIP HISTORY for 'SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32' [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10895 sip_dump_history: * SIP Call [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 001. Rx INVITE / 933 INVITE / sip:7323680452@172.16.4.4:5060 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 002. NewChan Channel SIP/172.16.4.4-0892ada0 - from SDpcljc01-2f69b698b1dc66 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 003. TxResp SIP/2.0 / 933 INVITE - 100 Trying [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 004. TxResp SIP/2.0 / 933 INVITE - 180 Ringing [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 005. TxRespRel SIP/2.0 / 933 INVITE - 200 OK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 006. Rx ACK / 933 ACK / sip:7323680452@172.16.4.4 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 007. ReInv Re-invite sent [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 008. TxReqRel INVITE / 102 INVITE - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 009. Rx SIP/2.0 / 102 INVITE / 100 Trying [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 010. Rx SIP/2.0 / 102 INVITE / 200 OK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 011. TxReq ACK / 102 ACK - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 012. ReInv Re-invite sent [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 013. TxReqRel INVITE / 103 INVITE - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 014. Rx SIP/2.0 / 103 INVITE / 100 Trying [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 015. Rx SIP/2.0 / 103 INVITE / 200 OK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 016. TxReq ACK / 103 ACK - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 017. ReInv Re-invite sent [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 018. TxReqRel INVITE / 104 INVITE - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 019. Rx SIP/2.0 / 104 INVITE / 100 Trying [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 020. ReInv Re-invite sent [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 021. TxReqRel INVITE / 105 INVITE - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 022. Rx SIP/2.0 / 105 INVITE / 100 Trying [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 023. Rx SIP/2.0 / 105 INVITE / 491 Request Pending [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 024. TxReq ACK / 105 ACK - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 025. Hangup Cause Normal Clearing [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 026. SchedDestroy 640 ms [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 027. TxReqRel BYE / 106 BYE - -UNKNOWN- [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 028. RTCPaudio Quality:ssrc=921414746;themssrc=1642994549;lp=0;rxjitter=0.0000 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 029. Rx SIP/2.0 / 106 BYE / 200 OK [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10898 sip_dump_history: 030. AutoDestroy SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32 [2007-10-05 18:08:14] DEBUG[26964]: chan_sip.c:10901 sip_dump_history: ---------- END SIP HISTORY for 'SDpcljc01-2f69b698b1dc669e29040acb63685562-7c6qh32' thorium*CLI> thorium*CLI> thorium*CLI> thorium*CLI> thorium*CLI> Reliably Transmitting (no NAT) to 135.25.29.1:5060: OPTIONS sip:135.25.29.1 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK4adaaec3;rport From: "asterisk" ;tag=as31df674c To: Contact: Call-ID: 7b0a233922d1cc271f236a8f01ab49b4@172.16.4.4 CSeq: 102 OPTIONS User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Date: Fri, 05 Oct 2007 22:08:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (no NAT) to 135.25.29.1:5060: OPTIONS sip:135.25.29.1 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK4adaaec3;rport From: "asterisk" ;tag=as31df674c To: Contact: Call-ID: 7b0a233922d1cc271f236a8f01ab49b4@172.16.4.4 CSeq: 102 OPTIONS User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Date: Fri, 05 Oct 2007 22:08:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #2 (no NAT) to 135.25.29.1:5060: OPTIONS sip:135.25.29.1 SIP/2.0 Via: SIP/2.0/UDP 172.16.4.4:5060;branch=z9hG4bK4adaaec3;rport From: "asterisk" ;tag=as31df674c To: Contact: Call-ID: 7b0a233922d1cc271f236a8f01ab49b4@172.16.4.4 CSeq: 102 OPTIONS User-Agent: Thorium Asterisk PBX Max-Forwards: 70 Date: Fri, 05 Oct 2007 22:08:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- thorium*CLI> exit Executing last minute cleanups Asterisk cleanly ending (0). Asterisk ending (0). [root@thorium asterisk]#