Asterisk 1.4.21, 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 currently running on pbx-gr (pid = 28709) pbx-gr*CLI> Verbosity is at least 3 pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> INVITE sip:80625@pbx-gr.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1139161174 Max-Forwards: 70 From: "Yehavi (Test)" ;tag=1c1139159289 To: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Remote-Party-ID: ;party=called Remote-Party-ID: "Yehavi (Test)" ;party=calling;privacy=off;screen=no;screen-ind=0 User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 255 v=0 o=AudiocodesGW 1139153857 1139153771 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 0 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (15 headers 12 lines) --- pbx-gr*CLI> Sending to 132.64.4.44 : 5060 (no NAT) pbx-gr*CLI> Using INVITE request as basis request - 113915894411200062657@132.64.4.44 pbx-gr*CLI> <--- Reliably Transmitting (no NAT) to 132.64.4.44:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1139161174;received=132.64.4.44 From: "Yehavi (Test)" ;tag=1c1139159289 To: ;tag=as501956df Call-ID: 113915894411200062657@132.64.4.44 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="33389aa3" Content-Length: 0 <------------> pbx-gr*CLI> Scheduling destruction of SIP dialog '113915894411200062657@132.64.4.44' in 32000 ms (Method: INVITE) pbx-gr*CLI> Found user '80634' pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> ACK sip:80625@pbx-gr.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1139161174 Max-Forwards: 70 From: "Yehavi (Test)" ;tag=1c1139159289 To: ;tag=as501956df Call-ID: 113915894411200062657@132.64.4.44 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (12 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> INVITE sip:80625@pbx-gr.cc.huji.ac.il;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1139199782 Max-Forwards: 70 From: "Yehavi (Test)" ;tag=1c1139159289 To: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 2 INVITE Proxy-Authorization: Digest username="80634",realm="cc.huji.ac.il",nonce="33389aa3",uri="sip:80625@pbx-gr.cc.huji.ac.il",algorithm=MD5,response="27cda1df93aafe002d10027bdb552642" Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Remote-Party-ID: ;party=called Remote-Party-ID: "Yehavi (Test)" ;party=calling;privacy=off;screen=no;screen-ind=0 User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 255 v=0 o=AudiocodesGW 1139153857 1139153771 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 0 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (16 headers 12 lines) --- pbx-gr*CLI> Sending to 132.64.4.44 : 5060 (no NAT) pbx-gr*CLI> Using INVITE request as basis request - 113915894411200062657@132.64.4.44 pbx-gr*CLI> Found user '80634' pbx-gr*CLI> Found RTP audio format 8 pbx-gr*CLI> Found RTP audio format 0 pbx-gr*CLI> Found RTP audio format 101 pbx-gr*CLI> [Jul 21 13:07:26] DEBUG[28720]: chan_sip.c:5232 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.44:6000 pbx-gr*CLI> Found audio description format pcma for ID 8 pbx-gr*CLI> Found audio description format pcmu for ID 0 pbx-gr*CLI> Found audio description format telephone-event for ID 101 pbx-gr*CLI> Capabilities: us - 0xc040e (gsm|ulaw|alaw|ilbc|h261|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) pbx-gr*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbx-gr*CLI> Peer audio RTP is at port 132.64.4.44:6000 pbx-gr*CLI> [Jul 21 13:07:26] DEBUG[28720]: chan_sip.c:3296 update_call_counter: Call from peer '80634' is 1 out of 5 pbx-gr*CLI> Looking for 80625 in huji-remote-ms (domain pbx-gr.cc.huji.ac.il) pbx-gr*CLI> list_route: hop: pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.44:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1139199782;received=132.64.4.44 From: "Yehavi (Test)" ;tag=1c1139159289 To: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> -- Executing [80625@huji-remote-ms:1] NoOp("SIP/80634-b7e371e0", "") in new stack -- Executing [80625@huji-remote-ms:2] Set("SIP/80634-b7e371e0", "_To=80625") in new stack -- Executing [80625@huji-remote-ms:3] GotoIf("SIP/80634-b7e371e0", "0?4:7") in new stack pbx-gr*CLI> -- Goto (huji-remote-ms,80625,7) [Jul 21 13:07:26] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. -- Executing [80625@huji-remote-ms:7] Set("SIP/80634-b7e371e0", "_From=80634") in new stack [Jul 21 13:07:26] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80634-b7e371e0' AND host = 'dynamic' -- Executing [80625@huji-remote-ms:8] Set("SIP/80634-b7e371e0", "_FromName=System group") in new stack -- Executing [80625@huji-remote-ms:9] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-8") in new stack -- Executing [80625@huji-remote-ms:10] GotoIf("SIP/80634-b7e371e0", "0?11:14") in new stack pbx-gr*CLI> -- Goto (huji-remote-ms,80625,14) -- Executing [80625@huji-remote-ms:14] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-9") in new stack -- Executing [80625@huji-remote-ms:15] GotoIf("SIP/80634-b7e371e0", "1?16:17") in new stack -- Goto (huji-remote-ms,80625,16) -- Executing [80625@huji-remote-ms:16] Set("SIP/80634-b7e371e0", "DB(80625/LastCaller)=80634") in new stack pbx-gr*CLI> [Jul 21 13:07:26] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:26] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80634-b7e371e0' pbx-gr*CLI> -- Executing [80625@huji-remote-ms:17] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-10") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:18] GotoIf("SIP/80634-b7e371e0", "1?19:20") in new stack pbx-gr*CLI> -- Goto (huji-remote-ms,80625,19) pbx-gr*CLI> -- Executing [80625@huji-remote-ms:19] Set("SIP/80634-b7e371e0", "DB(80634/LastCalled)=80625") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:20] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-11") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:21] Set("SIP/80634-b7e371e0", "DB(80625/CallingID)="System group" <80634>") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:22] MYSQL("SIP/80634-b7e371e0", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:23] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT additional_numbers from shared_lines where orig_number='80625'") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:24] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 aEXTEN") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:25] NoOp("SIP/80634-b7e371e0", "") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:26] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:27] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT waiting_call_enabled from sip_users where name='80625'") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:28] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 WaitingCallEnabled") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:29] NoOp("SIP/80634-b7e371e0", "NO") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:30] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:31] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT callerid from sip_users where name='80625'") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:32] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 CalledName") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:33] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:34] GotoIf("SIP/80634-b7e371e0", "0?35:41") in new stack pbx-gr*CLI> -- Goto (huji-remote-ms,80625,41) pbx-gr*CLI> -- Executing [80625@huji-remote-ms:41] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-12") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:42] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT regserver from sip_users where name='80625'") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:43] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 RegServer") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:44] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:45] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT fullcontact from sip_users where name='80625'") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:46] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 FullContact") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:47] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:48] MYSQL("SIP/80634-b7e371e0", "Disconnect 19") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:49] Set("SIP/80634-b7e371e0", "CONNECTEDLINE(all)="Test <80625>" <80625>") in new stack pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.44:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1139199782;received=132.64.4.44 From: "Yehavi (Test)" ;tag=1c1139159289 To: ;tag=as4c014dea Call-ID: 113915894411200062657@132.64.4.44 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Remote-Party-ID: "Test" ;party=called;privacy=off;screen=yes <------------> pbx-gr*CLI> -- Executing [80625@huji-remote-ms:50] GotoIf("SIP/80634-b7e371e0", "0?51:56") in new stack pbx-gr*CLI> -- Goto (huji-remote-ms,80625,56) pbx-gr*CLI> -- Executing [80625@huji-remote-ms:56] GotoIf("SIP/80634-b7e371e0", "0?57:61") in new stack pbx-gr*CLI> -- Goto (huji-remote-ms,80625,61) pbx-gr*CLI> -- Executing [80625@huji-remote-ms:61] NoOp("SIP/80634-b7e371e0", "Finish if-if-huji-local-14-15") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:62] Set("SIP/80634-b7e371e0", "_PICKUPMARK=80625") in new stack pbx-gr*CLI> -- Executing [80625@huji-remote-ms:63] GotoIf("SIP/80634-b7e371e0", "0?64:67") in new stack pbx-gr*CLI> -- Goto (huji-remote-ms,80625,67) pbx-gr*CLI> -- Executing [80625@huji-remote-ms:67] Dial("SIP/80634-b7e371e0", "SIP/80625|20|") in new stack pbx-gr*CLI> [Jul 21 13:07:26] DEBUG[2976]: chan_sip.c:3296 update_call_counter: Call to peer '80625' is 1 out of 5 pbx-gr*CLI> Audio is at 132.64.9.163 port 4872 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.249:5060: INVITE sip:80625@132.64.4.249 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3b091759;rport From: "System group" ;tag=as46d4a29e To: Contact: Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 Jul 2008 10:07:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "System group" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 261 v=0 o=root 28709 28709 IN IP4 132.64.4.44 s=session c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> -- Called 80625 pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.44:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1139199782;received=132.64.4.44 From: "Yehavi (Test)" ;tag=1c1139159289 To: ;tag=as4c014dea Call-ID: 113915894411200062657@132.64.4.44 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Remote-Party-ID: "Test" ;party=called;privacy=off;screen=yes <------------> pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3b091759;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D CSeq: 102 INVITE Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> pbx-gr*CLI> --- (9 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.201.12:5060 ---> SUBSCRIBE sip:80608@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.201.12;branch=z9hG4bK7146bf645AD865F7 From: "83098" ;tag=A1B5C76A-1332529B To: CSeq: 1 SUBSCRIBE Call-ID: ae70bdb6-6d1a3038-fc5912d1@132.64.201.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> pbx-gr*CLI> --- (14 headers 0 lines) --- pbx-gr*CLI> Creating new subscription pbx-gr*CLI> Sending to 132.64.201.12 : 5060 (no NAT) pbx-gr*CLI> Found peer '83098' pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.201.12:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.201.12;branch=z9hG4bK7146bf645AD865F7;received=132.64.201.12 From: "83098" ;tag=A1B5C76A-1332529B To: ;tag=as57ee27c9 Call-ID: ae70bdb6-6d1a3038-fc5912d1@132.64.201.12 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="24b9f929" Content-Length: 0 <------------> pbx-gr*CLI> Scheduling destruction of SIP dialog 'ae70bdb6-6d1a3038-fc5912d1@132.64.201.12' in 32000 ms (Method: SUBSCRIBE) pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3b091759;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D CSeq: 102 INVITE Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> [Jul 21 13:07:27] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:27] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80625-08bf9468' AND host = 'dynamic' pbx-gr*CLI> -- SIP/80625-08bf9468 is ringing pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.44:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1139199782;received=132.64.4.44 From: "Yehavi (Test)" ;tag=1c1139159289 To: ;tag=as4c014dea Call-ID: 113915894411200062657@132.64.4.44 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> [Jul 21 13:07:27] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:27] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80625-08bf9468' pbx-gr*CLI> <--- SIP read from 132.64.201.12:5060 ---> SUBSCRIBE sip:80608@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.201.12;branch=z9hG4bK9b7c67d59314DFCC From: "83098" ;tag=A1B5C76A-1332529B To: CSeq: 2 SUBSCRIBE Call-ID: ae70bdb6-6d1a3038-fc5912d1@132.64.201.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Accept: application/xpidf+xml,text/xml+msrtc.pidf Authorization: Digest username="83098", realm="cc.huji.ac.il", nonce="24b9f929", uri="sip:80608@132.64.9.163", response="b32e226e1a8f1231178cf5ff364aa220", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> pbx-gr*CLI> --- (15 headers 0 lines) --- pbx-gr*CLI> Creating new subscription pbx-gr*CLI> Sending to 132.64.201.12 : 5060 (no NAT) pbx-gr*CLI> Found peer '83098' pbx-gr*CLI> Looking for 80608 in huji-local (domain 132.64.9.163) pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.201.12:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 132.64.201.12;branch=z9hG4bK9b7c67d59314DFCC;received=132.64.201.12 From: "83098" ;tag=A1B5C76A-1332529B To: ;tag=as57ee27c9 Call-ID: ae70bdb6-6d1a3038-fc5912d1@132.64.201.12 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> pbx-gr*CLI> Really destroying SIP dialog 'ae70bdb6-6d1a3038-fc5912d1@132.64.201.12' Method: SUBSCRIBE pbx-gr*CLI> Really destroying SIP dialog '40377dbd3a883cce34e045b607a4a6f6@132.64.9.163' Method: NOTIFY pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3b091759;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D CSeq: 102 INVITE Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Type: application/sdp Content-Length: 199 v=0 o=- 1216617720 1216617720 IN IP4 132.64.4.249 s=Polycom IP Phone c=IN IP4 132.64.4.249 t=0 0 m=audio 2242 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <-------------> pbx-gr*CLI> --- (11 headers 9 lines) --- pbx-gr*CLI> Found RTP audio format 8 pbx-gr*CLI> Found RTP audio format 101 pbx-gr*CLI> [Jul 21 13:07:28] DEBUG[28720]: chan_sip.c:5232 process_sdp: Peer doesn't provide T.38 UDPTL pbx-gr*CLI> Peer audio RTP is at port 132.64.4.249:2242 pbx-gr*CLI> Found audio description format PCMA for ID 8 pbx-gr*CLI> Found audio description format telephone-event for ID 101 pbx-gr*CLI> Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) pbx-gr*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbx-gr*CLI> Peer audio RTP is at port 132.64.4.249:2242 pbx-gr*CLI> list_route: hop: pbx-gr*CLI> [Jul 21 13:07:28] DEBUG[28720]: chan_sip.c:5971 reqprep: Strict routing enforced for session 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 set_destination: Parsing for address/port to send to pbx-gr*CLI> set_destination: set destination to 132.64.4.249, port 5060 pbx-gr*CLI> Transmitting (no NAT) to 132.64.4.249:5060: ACK sip:80625@132.64.4.249 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK26d68fc7;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D Contact: Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-gr*CLI> -- SIP/80625-08bf9468 answered SIP/80634-b7e371e0 pbx-gr*CLI> Audio is at 132.64.9.163 port 8154 pbx-gr*CLI> Adding codec 0x8 (alaw) to SDP pbx-gr*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx-gr*CLI> <--- Reliably Transmitting (no NAT) to 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1139199782;received=132.64.4.44 From: "Yehavi (Test)" ;tag=1c1139159289 To: ;tag=as4c014dea Call-ID: 113915894411200062657@132.64.4.44 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 239 v=0 o=root 28709 28709 IN IP4 132.64.4.249 s=session c=IN IP4 132.64.4.249 t=0 0 m=audio 2242 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> pbx-gr*CLI> [Jul 21 13:07:28] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. -- Native bridging SIP/80634-b7e371e0 and SIP/80625-08bf9468 [Jul 21 13:07:28] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80625-08bf9468' AND host = 'dynamic' pbx-gr*CLI> [Jul 21 13:07:28] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:28] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80625-08bf9468' pbx-gr*CLI> [Jul 21 13:07:28] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:28] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80634-b7e371e0' AND host = 'dynamic' pbx-gr*CLI> [Jul 21 13:07:28] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:28] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80634-b7e371e0' pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> ACK sip:80625@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1140872909 Max-Forwards: 70 From: "Yehavi (Test)" ;tag=1c1139159289 To: ;tag=as4c014dea Call-ID: 113915894411200062657@132.64.4.44 CSeq: 2 ACK Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (12 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.135:5060 ---> SUBSCRIBE sip:80610@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.135;branch=z9hG4bK1a49bb4a6E9EBD91 From: "84989" ;tag=ED172FE5-432D8D98 To: CSeq: 1 SUBSCRIBE Call-ID: c31535f4-2337b0e3-7909bc8e@132.64.4.135 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.2.0084 Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> pbx-gr*CLI> --- (14 headers 0 lines) --- pbx-gr*CLI> Creating new subscription pbx-gr*CLI> Sending to 132.64.4.135 : 5060 (no NAT) Found peer '84989' <--- Transmitting (no NAT) to 132.64.4.135:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.135;branch=z9hG4bK1a49bb4a6E9EBD91;received=132.64.4.135 From: "84989" ;tag=ED172FE5-432D8D98 To: ;tag=as01fbb077 Call-ID: c31535f4-2337b0e3-7909bc8e@132.64.4.135 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="67b218f8" Content-Length: 0 <------------> pbx-gr*CLI> Scheduling destruction of SIP dialog 'c31535f4-2337b0e3-7909bc8e@132.64.4.135' in 32000 ms (Method: SUBSCRIBE) pbx-gr*CLI> <--- SIP read from 132.64.4.135:5060 ---> SUBSCRIBE sip:80610@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.135;branch=z9hG4bKe71bf3176917C392 From: "84989" ;tag=ED172FE5-432D8D98 To: CSeq: 2 SUBSCRIBE Call-ID: c31535f4-2337b0e3-7909bc8e@132.64.4.135 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.2.0084 Accept: application/xpidf+xml,text/xml+msrtc.pidf Authorization: Digest username="84989", realm="cc.huji.ac.il", nonce="67b218f8", uri="sip:80610@132.64.9.163", response="a286df8625e1c09bcfd160ecc05eb888", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> pbx-gr*CLI> --- (15 headers 0 lines) --- pbx-gr*CLI> Creating new subscription pbx-gr*CLI> Sending to 132.64.4.135 : 5060 (no NAT) pbx-gr*CLI> Found peer '84989' pbx-gr*CLI> Looking for 80610 in huji-local (domain 132.64.9.163) pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.135:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 132.64.4.135;branch=z9hG4bKe71bf3176917C392;received=132.64.4.135 From: "84989" ;tag=ED172FE5-432D8D98 To: ;tag=as01fbb077 Call-ID: c31535f4-2337b0e3-7909bc8e@132.64.4.135 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> pbx-gr*CLI> Really destroying SIP dialog 'c31535f4-2337b0e3-7909bc8e@132.64.4.135' Method: SUBSCRIBE pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> INVITE sip:80634@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.249;branch=z9hG4bKf34ccc6bFFF0AFD6 From: ;tag=352ED078-8CE3133D To: "System group" ;tag=as46d4a29e CSeq: 1 INVITE Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 199 v=0 o=- 1216617720 1216617721 IN IP4 132.64.4.249 s=Polycom IP Phone c=IN IP4 132.64.4.249 t=0 0 m=audio 2242 RTP/AVP 8 101 a=sendonly a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <-------------> pbx-gr*CLI> --- (14 headers 9 lines) --- pbx-gr*CLI> Sending to 132.64.4.249 : 5060 (no NAT) pbx-gr*CLI> Found RTP audio format 8 pbx-gr*CLI> Found RTP audio format 101 pbx-gr*CLI> [Jul 21 13:07:29] DEBUG[28720]: chan_sip.c:5232 process_sdp: Peer doesn't provide T.38 UDPTL pbx-gr*CLI> Peer audio RTP is at port 132.64.4.249:2242 pbx-gr*CLI> Found audio description format PCMA for ID 8 pbx-gr*CLI> Found audio description format telephone-event for ID 101 pbx-gr*CLI> Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) pbx-gr*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbx-gr*CLI> Peer audio RTP is at port 132.64.4.249:2242 pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.249:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.249;branch=z9hG4bKf34ccc6bFFF0AFD6;received=132.64.4.249 From: ;tag=352ED078-8CE3133D To: "System group" ;tag=as46d4a29e Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> Audio is at 132.64.9.163 port 4872 pbx-gr*CLI> Adding codec 0x8 (alaw) to SDP pbx-gr*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx-gr*CLI> <--- Reliably Transmitting (no NAT) to 132.64.4.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.249;branch=z9hG4bKf34ccc6bFFF0AFD6;received=132.64.4.249 From: ;tag=352ED078-8CE3133D To: "System group" ;tag=as46d4a29e Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 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: 237 v=0 o=root 28709 28710 IN IP4 132.64.4.44 s=session c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> pbx-gr*CLI> [Jul 21 13:07:29] DEBUG[2976]: chan_sip.c:5971 reqprep: Strict routing enforced for session 113915894411200062657@132.64.4.44 set_destination: Parsing for address/port to send to pbx-gr*CLI> set_destination: set destination to 132.64.4.44, port 5060 pbx-gr*CLI> Audio is at 132.64.9.163 port 8154 pbx-gr*CLI> Adding codec 0x8 (alaw) to SDP pbx-gr*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK7ab04589;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 102 INVITE User-Agent: 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: 239 v=0 o=root 28709 28710 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> -- Started music on hold, class 'default', on SIP/80634-b7e371e0 pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK7ab04589;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 102 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 231 v=0 o=AudiocodesGW 1139153857 1139153773 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (12 headers 11 lines) --- pbx-gr*CLI> Found RTP audio format 8 pbx-gr*CLI> Found RTP audio format 101 pbx-gr*CLI> [Jul 21 13:07:29] DEBUG[28720]: chan_sip.c:5232 process_sdp: Peer doesn't provide T.38 UDPTL pbx-gr*CLI> Peer audio RTP is at port 132.64.4.44:6000 pbx-gr*CLI> Found audio description format pcma for ID 8 pbx-gr*CLI> Found audio description format telephone-event for ID 101 pbx-gr*CLI> Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) pbx-gr*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbx-gr*CLI> Peer audio RTP is at port 132.64.4.44:6000 pbx-gr*CLI> [Jul 21 13:07:29] DEBUG[28720]: chan_sip.c:5971 reqprep: Strict routing enforced for session 113915894411200062657@132.64.4.44 pbx-gr*CLI> set_destination: Parsing for address/port to send to pbx-gr*CLI> set_destination: set destination to 132.64.4.44, port 5060 pbx-gr*CLI> Transmitting (no NAT) to 132.64.4.44:5060: ACK sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK05ed114b;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> ACK sip:80634@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.249;branch=z9hG4bKac18e179A3BB3614 From: ;tag=352ED078-8CE3133D To: "System group" ;tag=as46d4a29e CSeq: 1 ACK Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Max-Forwards: 70 Content-Length: 0 <-------------> pbx-gr*CLI> --- (11 headers 0 lines) --- pbx-gr*CLI> Really destroying SIP dialog '3e16ebcc3cbc68e85fc23d4c06c5a488@132.64.9.163' Method: NOTIFY pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> REFER sip:80634@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.249;branch=z9hG4bK82c52067EB3B6132 From: ;tag=352ED078-8CE3133D To: "System group" ;tag=as46d4a29e CSeq: 2 REFER Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Refer-To: sip:89444@132.64.9.163 Referred-By: Max-Forwards: 70 Content-Length: 0 <-------------> pbx-gr*CLI> --- (12 headers 0 lines) --- pbx-gr*CLI> Call 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 got a SIP call transfer from caller: (REFER)! pbx-gr*CLI> SIP transfer to extension 89444@huji-remote-ag by 80625@132.64.9.163 pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.249:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 132.64.4.249;branch=z9hG4bK82c52067EB3B6132;received=132.64.4.249 From: ;tag=352ED078-8CE3133D To: "System group" ;tag=as46d4a29e Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[2976]: chan_sip.c:5971 reqprep: Strict routing enforced for session 113915894411200062657@132.64.4.44 pbx-gr*CLI> set_destination: Parsing for address/port to send to pbx-gr*CLI> set_destination: set destination to 132.64.4.44, port 5060 pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28720]: chan_sip.c:5971 reqprep: Audio is at 132.64.9.163 port 8154 Strict routing enforced for session 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 set_destination: Parsing for address/port to send to Adding codec 0x8 (alaw) to SDP pbx-gr*CLI> set_destination: set destination to 132.64.4.249, port 5060 pbx-gr*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.249:5060: NOTIFY sip:80625@132.64.4.249 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2ed0a4dc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D Contact: Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 21 SIP/2.0 183 Ringing --- pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE User-Agent: 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: 239 v=0 o=root 28709 28711 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28720]: chan_sip.c:5971 reqprep: -- Stopped music on hold on SIP/80634-b7e371e0 Strict routing enforced for session 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 pbx-gr*CLI> set_destination: Parsing for address/port to send to pbx-gr*CLI> set_destination: set destination to 132.64.4.249, port 5060 pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.249:5060: NOTIFY sip:80625@132.64.4.249 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK347aabcc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D Contact: Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 Ok --- pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[2976]: chan_sip.c:3270 update_call_counter: Call to peer '80625' removed from call limit 5 Scheduling destruction of SIP dialog '214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163' in 32000 ms (Method: REFER) pbx-gr*CLI> == Spawn extension (huji-remote-ag, 89444, 0) exited non-zero on 'SIP/80634-b7e371e0' pbx-gr*CLI> -- Executing [89444@huji-remote-ag:1] NoOp("SIP/80634-b7e371e0", "") in new stack pbx-gr*CLI> -- Executing [89444@huji-remote-ag:2] MYSQL("SIP/80634-b7e371e0", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack [Jul 21 13:07:31] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80625-08bf9468' AND host = 'dynamic' pbx-gr*CLI> -- Executing [89444@huji-remote-ag:3] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT name from sip_users where name='89444'") in new stack pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: -- Executing [89444@huji-remote-ag:4] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 name") in new stack pbx-gr*CLI> MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80625-08bf9468' -- Executing [89444@huji-remote-ag:5] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack -- Executing [89444@huji-remote-ag:6] NoOp("SIP/80634-b7e371e0", "89444") in new stack -- Executing [89444@huji-remote-ag:7] GotoIf("SIP/80634-b7e371e0", "1?8:11") in new stack -- Goto (huji-remote-ag,89444,8) -- Executing [89444@huji-remote-ag:8] MYSQL("SIP/80634-b7e371e0", "Disconnect 19") in new stack -- Executing [89444@huji-remote-ag:9] Set("SIP/80634-b7e371e0", "_To=89444") in new stack -- Executing [89444@huji-remote-ag:10] Goto("SIP/80634-b7e371e0", "huji-local|_806XX|StartLocal") in new stack -- Goto (huji-local,_806XX,3) -- Executing [_806XX@huji-local:3] GotoIf("SIP/80634-b7e371e0", "0?4:7") in new stack -- Goto (huji-local,_806XX,7) -- Executing [_806XX@huji-local:7] Set("SIP/80634-b7e371e0", "_From=80634") in new stack -- Executing [_806XX@huji-local:8] Set("SIP/80634-b7e371e0", "_FromName=System group") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:9] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-8") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:10] GotoIf("SIP/80634-b7e371e0", "0?11:14") in new stack pbx-gr*CLI> -- Goto (huji-local,_806XX,14) pbx-gr*CLI> -- Executing [_806XX@huji-local:14] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-9") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:15] GotoIf("SIP/80634-b7e371e0", "1?16:17") in new stack pbx-gr*CLI> -- Goto (huji-local,_806XX,16) pbx-gr*CLI> -- Executing [_806XX@huji-local:16] Set("SIP/80634-b7e371e0", "DB(89444/LastCaller)=80634") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:17] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-10") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:18] GotoIf("SIP/80634-b7e371e0", "1?19:20") in new stack pbx-gr*CLI> -- Goto (huji-local,_806XX,19) pbx-gr*CLI> -- Executing [_806XX@huji-local:19] Set("SIP/80634-b7e371e0", "DB(80634/LastCalled)=89444") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:20] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-11") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:21] Set("SIP/80634-b7e371e0", "DB(89444/CallingID)="System group" <80634>") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:22] MYSQL("SIP/80634-b7e371e0", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:23] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT additional_numbers from shared_lines where orig_number='89444'") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:24] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 aEXTEN") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:25] NoOp("SIP/80634-b7e371e0", "") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:26] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:27] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT waiting_call_enabled from sip_users where name='89444'") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:28] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 WaitingCallEnabled") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:29] NoOp("SIP/80634-b7e371e0", "NO") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:30] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:31] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT callerid from sip_users where name='89444'") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:32] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 CalledName") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:33] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:34] GotoIf("SIP/80634-b7e371e0", "0?35:41") in new stack pbx-gr*CLI> -- Goto (huji-local,_806XX,41) pbx-gr*CLI> -- Executing [_806XX@huji-local:41] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-12") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:42] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT regserver from sip_users where name='89444'") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:43] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 RegServer") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:44] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:45] MYSQL("SIP/80634-b7e371e0", "Query resID 19 SELECT fullcontact from sip_users where name='89444'") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:46] MYSQL("SIP/80634-b7e371e0", "Fetch FetchId 20 FullContact") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:47] MYSQL("SIP/80634-b7e371e0", "Clear 20") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:48] MYSQL("SIP/80634-b7e371e0", "Disconnect 19") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:49] Set("SIP/80634-b7e371e0", "CONNECTEDLINE(all)="yehavi@vms.huji.ac.il <89444>" <89444>") in new stack pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[2976]: chan_sip.c:5971 reqprep: Strict routing enforced for session 113915894411200062657@132.64.4.44 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.44, port 5060 Audio is at 132.64.9.163 port 8154 pbx-gr*CLI> Adding codec 0x8 (alaw) to SDP pbx-gr*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "yehavi@vms.huji.ac.il" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 239 v=0 o=root 28709 28712 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> -- Executing [_806XX@huji-local:50] GotoIf("SIP/80634-b7e371e0", "0?51:56") in new stack pbx-gr*CLI> -- Goto (huji-local,_806XX,56) pbx-gr*CLI> -- Executing [_806XX@huji-local:56] GotoIf("SIP/80634-b7e371e0", "0?57:61") in new stack pbx-gr*CLI> -- Goto (huji-local,_806XX,61) pbx-gr*CLI> -- Executing [_806XX@huji-local:61] NoOp("SIP/80634-b7e371e0", "Finish if-if-huji-local-14-15") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:62] Set("SIP/80634-b7e371e0", "_PICKUPMARK=89444") in new stack pbx-gr*CLI> -- Executing [_806XX@huji-local:63] GotoIf("SIP/80634-b7e371e0", "0?64:67") in new stack -- Goto (huji-local,_806XX,67) -- Executing [_806XX@huji-local:67] Dial("SIP/80634-b7e371e0", "SIP/89444|20|") in new stack pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[2976]: chan_sip.c:3296 update_call_counter: Call to peer '89444' is 1 out of 2 pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.133:5060: NOTIFY sip:84138@132.64.4.133 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK5c3997be;rport From: ;tag=as6b5be321 To: "84138" ;tag=40362A31-153A73BA Contact: Call-ID: e9a2195-93d08823-f9ef8b6c@132.64.4.133 CSeq: 224 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 356
--- pbx-gr*CLI> Audio is at 132.64.9.163 port 9972 Extension Changed *789444[huji-local] new state Ringing for Notify User 84138 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.137:2048: INVITE sip:89444@132.64.4.137:2048 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3185f609;rport From: "System group" ;tag=as4fb72119 To: Contact: Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 Jul 2008 10:07:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "System group" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 237 v=0 o=root 28709 28709 IN IP4 132.64.4.44 s=session c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> -- Called 89444 pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[2976]: chan_sip.c:5971 reqprep: Strict routing enforced for session 113915894411200062657@132.64.4.44 pbx-gr*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.44, port 5060 Audio is at 132.64.9.163 port 8154 pbx-gr*CLI> Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK459aa686;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 105 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "yehavi@vms.huji.ac.il" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 239 v=0 o=root 28709 28713 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 231 v=0 o=AudiocodesGW 1139153857 1139153774 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (12 headers 11 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2ed0a4dc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D CSeq: 103 NOTIFY Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (11 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.133:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK5c3997be;rport From: ;tag=as6b5be321 To: "84138" ;tag=40362A31-153A73BA CSeq: 224 NOTIFY Call-ID: e9a2195-93d08823-f9ef8b6c@132.64.4.133 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> SIP Response message for INCOMING dialog NOTIFY arrived pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> BYE sip:80634@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.249;branch=z9hG4bK4125373581CF0F30 From: ;tag=352ED078-8CE3133D To: "System group" ;tag=as46d4a29e CSeq: 3 BYE Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Max-Forwards: 70 Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> Sending to 132.64.4.249 : 5060 (no NAT) pbx-gr*CLI> Scheduling destruction of SIP dialog '214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163' in 32000 ms (Method: BYE) pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.249;branch=z9hG4bK4125373581CF0F30;received=132.64.4.249 From: ;tag=352ED078-8CE3133D To: "System group" ;tag=as46d4a29e Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK459aa686;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 105 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (11 headers 0 lines) --- pbx-gr*CLI> -- Got SIP response 500 "Server Internal Error" back from 132.64.4.44 pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28720]: chan_sip.c:5971 reqprep: Strict routing enforced for session 113915894411200062657@132.64.4.44 set_destination: Parsing for address/port to send to pbx-gr*CLI> set_destination: set destination to 132.64.4.44, port 5060 pbx-gr*CLI> Transmitting (no NAT) to 132.64.4.44:5060: ACK sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK459aa686;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK347aabcc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D CSeq: 104 NOTIFY Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> SIP Response message for INCOMING dialog NOTIFY arrived pbx-gr*CLI> -- Incoming call: Got SIP response 500 "Internal Server Error" back from 132.64.4.249 pbx-gr*CLI> <--- SIP read from 132.64.4.137:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3185f609;rport=5060 From: "System group" ;tag=as4fb72119 To: ;tag=aj6u2fy1bf Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '89444-08b7e680' AND host = 'dynamic' pbx-gr*CLI> -- SIP/89444-08b7e680 is ringing pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:31] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '89444-08b7e680' pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 231 v=0 o=AudiocodesGW 1139153857 1139153774 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (12 headers 11 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (11 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.137:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3185f609;rport=5060 From: "System group" ;tag=as4fb72119 To: ;tag=aj6u2fy1bf Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> -- SIP/89444-08b7e680 is ringing pbx-gr*CLI> Retransmitting #1 (no NAT) to 132.64.4.249:5060: NOTIFY sip:80625@132.64.4.249 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2ed0a4dc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D Contact: Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 21 SIP/2.0 183 Ringing --- pbx-gr*CLI> Retransmitting #1 (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE User-Agent: 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: 239 v=0 o=root 28709 28711 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> Retransmitting #1 (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "yehavi@vms.huji.ac.il" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 239 v=0 o=root 28709 28712 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 231 v=0 o=AudiocodesGW 1139153857 1139153774 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (12 headers 11 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2ed0a4dc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D CSeq: 103 NOTIFY Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (11 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 231 v=0 o=AudiocodesGW 1139153857 1139153774 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (12 headers 11 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (11 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.137:2048 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3185f609;rport=5060 From: "System group" ;tag=as4fb72119 To: ;tag=aj6u2fy1bf Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> -- SIP/89444-08b7e680 is ringing pbx-gr*CLI> Retransmitting #2 (no NAT) to 132.64.4.249:5060: NOTIFY sip:80625@132.64.4.249 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2ed0a4dc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D Contact: Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 21 SIP/2.0 183 Ringing --- pbx-gr*CLI> Retransmitting #2 (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE User-Agent: 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: 239 v=0 o=root 28709 28711 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> Retransmitting #2 (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "yehavi@vms.huji.ac.il" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 239 v=0 o=root 28709 28712 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 231 v=0 o=AudiocodesGW 1139153857 1139153774 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (12 headers 11 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2ed0a4dc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D CSeq: 103 NOTIFY Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (11 headers 0 lines) --- pbx-gr*CLI> Really destroying SIP dialog '100d1830-cd0f8b80-13c4-3d7b3bce-8ddb2b-3d7b3bce' Method: REGISTER pbx-gr*CLI> <--- SIP read from 132.64.4.137:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK3185f609;rport=5060 From: "System group" ;tag=as4fb72119 To: ;tag=aj6u2fy1bf Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 269 v=0 o=root 366336362 366336363 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 61332 RTP/AVP 8 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ObRKr+lTpwyJjbtpHuLFxILVz48//Q4Bxz298SJA a=rtpmap:8 pcma/8000 a=ptime:20 a=encryption:optional a=sendrecv <-------------> pbx-gr*CLI> --- (13 headers 11 lines) --- pbx-gr*CLI> Found RTP audio format 8 pbx-gr*CLI> [Jul 21 13:07:33] DEBUG[28720]: chan_sip.c:5232 process_sdp: Peer doesn't provide T.38 UDPTL pbx-gr*CLI> Peer audio RTP is at port 132.64.4.137:61332 pbx-gr*CLI> Found audio description format pcma for ID 8 pbx-gr*CLI> Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) pbx-gr*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) pbx-gr*CLI> Peer audio RTP is at port 132.64.4.137:61332 pbx-gr*CLI> list_route: hop: pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.133:5060: NOTIFY sip:84138@132.64.4.133 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK384e0166;rport From: ;tag=as6b5be321 To: "84138" ;tag=40362A31-153A73BA Contact: Call-ID: e9a2195-93d08823-f9ef8b6c@132.64.4.133 CSeq: 225 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 356
--- [Jul 21 13:07:33] DEBUG[28720]: chan_sip.c:5971 reqprep: Strict routing enforced for session 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 pbx-gr*CLI> set_destination: Parsing for address/port to send to pbx-gr*CLI> Extension Changed *789444[huji-local] new state InUse for Notify User 84138 pbx-gr*CLI> set_destination: set destination to 132.64.4.137, port 2048 pbx-gr*CLI> Transmitting (no NAT) to 132.64.4.137:2048: ACK sip:89444@132.64.4.137:2048 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK07224731;rport From: "System group" ;tag=as4fb72119 To: ;tag=aj6u2fy1bf Contact: Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-gr*CLI> -- SIP/89444-08b7e680 answered SIP/80634-b7e371e0 pbx-gr*CLI> -- Native bridging SIP/80634-b7e371e0 and SIP/89444-08b7e680 [Jul 21 13:07:33] DEBUG[2976]: chan_sip.c:5971 reqprep: Strict routing enforced for session 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2048 [Jul 21 13:07:33] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: Audio is at 132.64.9.163 port 9972 MySQL RealTime: Everything is fine. Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 132.64.4.137:2048: INVITE sip:89444@132.64.4.137:2048 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK13d7bd14;rport From: "System group" ;tag=as4fb72119 To: ;tag=aj6u2fy1bf Contact: Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 103 INVITE User-Agent: 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: 183 v=0 o=root 28709 28710 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 9972 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> [Jul 21 13:07:33] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '89444-08b7e680' AND host = 'dynamic' pbx-gr*CLI> [Jul 21 13:07:33] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:33] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '89444-08b7e680' pbx-gr*CLI> <--- SIP read from 132.64.4.133:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK384e0166;rport From: ;tag=as6b5be321 To: "84138" ;tag=40362A31-153A73BA CSeq: 225 NOTIFY Call-ID: e9a2195-93d08823-f9ef8b6c@132.64.4.133 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> SIP Response message for INCOMING dialog NOTIFY arrived pbx-gr*CLI> <--- SIP read from 132.64.4.137:2048 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK13d7bd14;rport=5060 From: "System group" ;tag=as4fb72119 To: ;tag=aj6u2fy1bf Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 103 INVITE Contact: ;flow-id=1 User-Agent: snom320/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 269 v=0 o=root 366336362 366336364 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 61332 RTP/AVP 8 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ObRKr+lTpwyJjbtpHuLFxILVz48//Q4Bxz298SJA a=rtpmap:8 pcma/8000 a=ptime:20 a=encryption:optional a=sendrecv <-------------> pbx-gr*CLI> --- (13 headers 11 lines) --- pbx-gr*CLI> Found RTP audio format 8 pbx-gr*CLI> [Jul 21 13:07:33] DEBUG[28720]: chan_sip.c:5232 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.137:61332 pbx-gr*CLI> Found audio description format pcma for ID 8 pbx-gr*CLI> Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) pbx-gr*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) pbx-gr*CLI> Peer audio RTP is at port 132.64.4.137:61332 pbx-gr*CLI> [Jul 21 13:07:33] DEBUG[28720]: chan_sip.c:5971 reqprep: Strict routing enforced for session 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 pbx-gr*CLI> set_destination: Parsing for address/port to send to pbx-gr*CLI> set_destination: set destination to 132.64.4.137, port 2048 pbx-gr*CLI> Transmitting (no NAT) to 132.64.4.137:2048: ACK sip:89444@132.64.4.137:2048 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK5098612e;rport From: "System group" ;tag=as4fb72119 To: ;tag=aj6u2fy1bf Contact: Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-gr*CLI> [Jul 21 13:07:34] NOTICE[2976]: rtp.c:788 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 132.64.4.137 pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 231 v=0 o=AudiocodesGW 1139153857 1139153774 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (12 headers 11 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (11 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.135:5060 ---> SUBSCRIBE sip:80609@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.135;branch=z9hG4bK51ecfef95A0CF4FC From: "84989" ;tag=6BAAD020-24BF17BF To: CSeq: 1 SUBSCRIBE Call-ID: 3668c40b-138ad456-aa3f3ecd@132.64.4.135 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.2.0084 Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> pbx-gr*CLI> --- (14 headers 0 lines) --- pbx-gr*CLI> Creating new subscription pbx-gr*CLI> Sending to 132.64.4.135 : 5060 (no NAT) pbx-gr*CLI> Found peer '84989' pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.135:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.135;branch=z9hG4bK51ecfef95A0CF4FC;received=132.64.4.135 From: "84989" ;tag=6BAAD020-24BF17BF To: ;tag=as062f7467 Call-ID: 3668c40b-138ad456-aa3f3ecd@132.64.4.135 CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="cc.huji.ac.il", nonce="6a733588" Content-Length: 0 <------------> pbx-gr*CLI> Scheduling destruction of SIP dialog '3668c40b-138ad456-aa3f3ecd@132.64.4.135' in 32000 ms (Method: SUBSCRIBE) pbx-gr*CLI> <--- SIP read from 132.64.4.135:5060 ---> SUBSCRIBE sip:80609@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.135;branch=z9hG4bKbd83b2da129DC361 From: "84989" ;tag=6BAAD020-24BF17BF To: CSeq: 2 SUBSCRIBE Call-ID: 3668c40b-138ad456-aa3f3ecd@132.64.4.135 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.2.0084 Accept: application/xpidf+xml,text/xml+msrtc.pidf Authorization: Digest username="84989", realm="cc.huji.ac.il", nonce="6a733588", uri="sip:80609@132.64.9.163", response="6e57ea0f315b66cf455970494735f738", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> pbx-gr*CLI> --- (15 headers 0 lines) --- pbx-gr*CLI> Creating new subscription pbx-gr*CLI> Sending to 132.64.4.135 : 5060 (no NAT) pbx-gr*CLI> Found peer '84989' pbx-gr*CLI> Looking for 80609 in huji-local (domain 132.64.9.163) pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.135:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 132.64.4.135;branch=z9hG4bKbd83b2da129DC361;received=132.64.4.135 From: "84989" ;tag=6BAAD020-24BF17BF To: ;tag=as062f7467 Call-ID: 3668c40b-138ad456-aa3f3ecd@132.64.4.135 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> pbx-gr*CLI> Really destroying SIP dialog '3668c40b-138ad456-aa3f3ecd@132.64.4.135' Method: SUBSCRIBE pbx-gr*CLI> Retransmitting #3 (no NAT) to 132.64.4.249:5060: NOTIFY sip:80625@132.64.4.249 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2ed0a4dc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D Contact: Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: active Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 21 SIP/2.0 183 Ringing --- pbx-gr*CLI> Retransmitting #3 (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE User-Agent: 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: 239 v=0 o=root 28709 28711 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> <--- SIP read from 132.64.4.137:2048 ---> BYE sip:80634@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2048;branch=z9hG4bK-1bxzraayn7nq;rport From: ;tag=aj6u2fy1bf To: "System group" ;tag=as4fb72119 Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 1 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom320/7.1.30 RTP-RxStat: Total_Rx_Pkts=64,Rx_Pkts=64,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=50,Tx_Pkts=50,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> pbx-gr*CLI> --- (12 headers 0 lines) --- pbx-gr*CLI> Sending to 132.64.4.137 : 2048 (NAT) pbx-gr*CLI> <--- Transmitting (NAT) to 132.64.4.137:2048 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2048;branch=z9hG4bK-1bxzraayn7nq;received=132.64.4.137;rport=2048 From: ;tag=aj6u2fy1bf To: "System group" ;tag=as4fb72119 Call-ID: 40f737bc6fcfd912264ae85332a22a22@132.64.9.163 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[2976]: chan_sip.c:3270 update_call_counter: Call to peer '89444' removed from call limit 2 pbx-gr*CLI> == Spawn extension (huji-local, _806XX, 67) exited non-zero on 'SIP/80634-b7e371e0' pbx-gr*CLI> Reliably Transmitting (no NAT) to 132.64.4.133:5060: NOTIFY sip:84138@132.64.4.133 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK51642e19;rport From: ;tag=as6b5be321 To: "84138" ;tag=40362A31-153A73BA Contact: Call-ID: e9a2195-93d08823-f9ef8b6c@132.64.4.133 CSeq: 226 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 351
--- pbx-gr*CLI> -- Executing [h@huji-local:1] ResetCDR("SIP/80634-b7e371e0", "w") in new stack pbx-gr*CLI> Extension Changed *789444[huji-local] new state Idle for Notify User 84138 pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: -- Executing [h@huji-local:2] NoOp("SIP/80634-b7e371e0", "80634") in new stack MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '89444-08b7e680' AND host = 'dynamic' pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[2976]: func_db.c:70 function_db_read: DB: 80634/CallBack not found in database. -- Executing [h@huji-local:3] Set("SIP/80634-b7e371e0", "tmp=") in new stack pbx-gr*CLI> -- Executing [h@huji-local:4] GotoIf("SIP/80634-b7e371e0", "0?5:11") in new stack pbx-gr*CLI> -- Goto (huji-local,h,11) -- Executing [h@huji-local:11] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-21") in new stack [Jul 21 13:07:35] DEBUG[2976]: func_db.c:70 function_db_read: DB: 89444/CallBack not found in database. -- Executing [h@huji-local:12] Set("SIP/80634-b7e371e0", "tmp=") in new stack pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> -- Executing [h@huji-local:13] GotoIf("SIP/80634-b7e371e0", "0?14:20") in new stack [Jul 21 13:07:35] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '89444-08b7e680' -- Goto (huji-local,h,20) pbx-gr*CLI> -- Executing [h@huji-local:20] NoOp("SIP/80634-b7e371e0", "Finish if-huji-local-22") in new stack pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[2976]: chan_sip.c:3270 update_call_counter: Call from peer '80634' removed from call limit 5 pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80634-b7e371e0' AND host = 'dynamic' pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[28717]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-gr*CLI> [Jul 21 13:07:35] DEBUG[28717]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80634-b7e371e0' pbx-gr*CLI> Retransmitting #3 (no NAT) to 132.64.4.44:5060: INVITE sip:80634@132.64.4.44 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Contact: Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "yehavi@vms.huji.ac.il" ;party=calling;privacy=off;screen=yes Content-Type: application/sdp Content-Length: 239 v=0 o=root 28709 28712 IN IP4 132.64.9.163 s=session c=IN IP4 132.64.9.163 t=0 0 m=audio 8154 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx-gr*CLI> Really destroying SIP dialog '40f737bc6fcfd912264ae85332a22a22@132.64.9.163' Method: BYE pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK798aaad3;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 103 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 231 v=0 o=AudiocodesGW 1139153857 1139153774 IN IP4 132.64.4.44 s=Phone-Call c=IN IP4 132.64.4.44 t=0 0 m=audio 6000 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> pbx-gr*CLI> --- (12 headers 11 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK2ed0a4dc;rport From: "System group" ;tag=as46d4a29e To: ;tag=352ED078-8CE3133D CSeq: 103 NOTIFY Call-ID: 214ab51e2194e30f6443c1217eb6f9f7@132.64.9.163 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> SIP/2.0 500 Server Internal Error Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK4f9462b5;rport From: ;tag=as4c014dea To: "Yehavi (Test)" ;tag=1c1139159289 Call-ID: 113915894411200062657@132.64.4.44 CSeq: 104 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (11 headers 0 lines) --- pbx-gr*CLI> <--- SIP read from 132.64.4.133:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.163:5060;branch=z9hG4bK51642e19;rport From: ;tag=as6b5be321 To: "84138" ;tag=40362A31-153A73BA CSeq: 226 NOTIFY Call-ID: e9a2195-93d08823-f9ef8b6c@132.64.4.133 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 <-------------> pbx-gr*CLI> --- (10 headers 0 lines) --- pbx-gr*CLI> SIP Response message for INCOMING dialog NOTIFY arrived pbx-gr*CLI> <--- SIP read from 132.64.4.44:5060 ---> BYE sip:80625@132.64.9.163 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1151938525 Max-Forwards: 70 From: "Yehavi (Test)" ;tag=1c1139159289 To: ;tag=as4c014dea Call-ID: 113915894411200062657@132.64.4.44 CSeq: 3 BYE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.60A.016.003 Content-Length: 0 <-------------> pbx-gr*CLI> --- (12 headers 0 lines) --- pbx-gr*CLI> Sending to 132.64.4.44 : 5060 (no NAT) pbx-gr*CLI> Scheduling destruction of SIP dialog '113915894411200062657@132.64.4.44' in 32000 ms (Method: BYE) pbx-gr*CLI> <--- Transmitting (no NAT) to 132.64.4.44:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.44;branch=z9hG4bKac1151938525;received=132.64.4.44 From: "Yehavi (Test)" ;tag=1c1139159289 To: ;tag=as4c014dea Call-ID: 113915894411200062657@132.64.4.44 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-gr*CLI> quit