Asterisk 1.4.13, Copyright (C) 1999 - 2007 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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Binding sippeers to mysql/asterisk/sip_users == Binding sipusers to mysql/asterisk/sip_users == Binding voicemail to mysql/asterisk/voicemail_users Connected to Asterisk 1.4.13 currently running on pbx-dev (pid = 2620) pbx-dev*CLI> Verbosity is at least 6 pbx-dev*CLI> <--- SIP read from 132.64.4.148:5060 ---> INVITE sip:85684@132.64.9.164:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bK6d1c9ca62C8553D From: "80619" ;tag=D7256334-A63B9859 To: CSeq: 1 INVITE Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1192614252 1192614252 IN IP4 132.64.4.148 s=Polycom IP Phone c=IN IP4 132.64.4.148 t=0 0 m=audio 2230 RTP/AVP 8 0 18 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> pbx-dev*CLI> --- (14 headers 11 lines) --- pbx-dev*CLI> Sending to 132.64.4.148 : 5060 (no NAT) pbx-dev*CLI> Using INVITE request as basis request - 9b360d78-421b5146-8585953b@132.64.4.148 pbx-dev*CLI> <--- Reliably Transmitting (no NAT) to 132.64.4.148:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bK6d1c9ca62C8553D;received=132.64.4.148 From: "80619" ;tag=D7256334-A63B9859 To: ;tag=as16062e1b Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 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="11792329" Content-Length: 0 <------------> pbx-dev*CLI> Scheduling destruction of SIP dialog '9b360d78-421b5146-8585953b@132.64.4.148' in 32000 ms (Method: INVITE) pbx-dev*CLI> Found user '80619' pbx-dev*CLI> <--- SIP read from 132.64.4.148:5060 ---> ACK sip:85684@132.64.9.164:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bK6d1c9ca62C8553D From: "80619" ;tag=D7256334-A63B9859 To: ;tag=as16062e1b CSeq: 1 ACK Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 <-------------> pbx-dev*CLI> --- (11 headers 0 lines) --- pbx-dev*CLI> <--- SIP read from 132.64.4.148:5060 ---> INVITE sip:85684@132.64.9.164:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bKca4d7c9764C63142 From: "80619" ;tag=D7256334-A63B9859 To: CSeq: 2 INVITE Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="80619", realm="cc.huji.ac.il", nonce="11792329", uri="sip:85684@132.64.9.164:5060;user=phone", response="68878bb534bdbf8cecaf0c8b11caf16f", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1192614252 1192614252 IN IP4 132.64.4.148 s=Polycom IP Phone c=IN IP4 132.64.4.148 t=0 0 m=audio 2230 RTP/AVP 8 0 18 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> pbx-dev*CLI> --- (15 headers 11 lines) --- pbx-dev*CLI> Sending to 132.64.4.148 : 5060 (no NAT) Using INVITE request as basis request - 9b360d78-421b5146-8585953b@132.64.4.148 pbx-dev*CLI> Found user '80619' pbx-dev*CLI> Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 pbx-dev*CLI> [Oct 17 11:47:23] DEBUG[2627]: chan_sip.c:5079 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.148:2230 pbx-dev*CLI> Found description format PCMA for ID 8 Found description format PCMU for ID 0 pbx-dev*CLI> Found description format G729 for ID 18 Found description format telephone-event for ID 101 pbx-dev*CLI> Capabilities: us - 0xc040e (gsm|ulaw|alaw|ilbc|h261|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbx-dev*CLI> Peer audio RTP is at port 132.64.4.148:2230 Peer video RTP is at port 132.64.4.148:6400 pbx-dev*CLI> [Oct 17 11:47:23] DEBUG[2627]: chan_sip.c:3211 update_call_counter: Call from peer '80619' is 1 out of 5 Looking for 85684 in huji-remote-gr (domain 132.64.9.164) list_route: hop: pbx-dev*CLI> <--- Transmitting (no NAT) to 132.64.4.148:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bKca4d7c9764C63142;received=132.64.4.148 From: "80619" ;tag=D7256334-A63B9859 To: Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-dev*CLI> -- Executing [85684@huji-remote-gr:1] NoOp("SIP/80619-09c9e610", "") in new stack pbx-dev*CLI> -- Executing [85684@huji-remote-gr:2] MYSQL("SIP/80619-09c9e610", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack pbx-dev*CLI> -- Executing [85684@huji-remote-gr:3] MYSQL("SIP/80619-09c9e610", "Query resID 1 SELECT name from sip_users where name='85684'") in new stack pbx-dev*CLI> -- Executing [85684@huji-remote-gr:4] MYSQL("SIP/80619-09c9e610", "Fetch FetchId 2 name") in new stack pbx-dev*CLI> -- Executing [85684@huji-remote-gr:5] MYSQL("SIP/80619-09c9e610", "Clear 2") in new stack pbx-dev*CLI> -- Executing [85684@huji-remote-gr:6] NoOp("SIP/80619-09c9e610", "85684") in new stack pbx-dev*CLI> -- Executing [85684@huji-remote-gr:7] GotoIf("SIP/80619-09c9e610", "1?8:11") in new stack pbx-dev*CLI> -- Goto (huji-remote-gr,85684,8) pbx-dev*CLI> -- Executing [85684@huji-remote-gr:8] MYSQL("SIP/80619-09c9e610", "Disconnect 1") in new stack pbx-dev*CLI> -- Executing [85684@huji-remote-gr:9] Set("SIP/80619-09c9e610", "_To=85684") in new stack -- Executing [85684@huji-remote-gr:10] Goto("SIP/80619-09c9e610", "huji-local|_806XX|StartLocal") in new stack pbx-dev*CLI> -- Goto (huji-local,_806XX,3) -- Executing [_806XX@huji-local:3] Set("SIP/80619-09c9e610", "_From=80619") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:4] Set("SIP/80619-09c9e610", "DB(85684/LastCaller)=80619") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:5] Set("SIP/80619-09c9e610", "DB(80619/LastCalled)=85684") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:6] MYSQL("SIP/80619-09c9e610", "Connect connid localhost asterisk NotMe43e080 asterisk") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:7] MYSQL("SIP/80619-09c9e610", "Query resID 1 SELECT additional_numbers from shared_lines where orig_number='85684'") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:8] MYSQL("SIP/80619-09c9e610", "Fetch FetchId 2 aEXTEN") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:9] NoOp("SIP/80619-09c9e610", "&SIP/80611") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:10] MYSQL("SIP/80619-09c9e610", "Clear 2") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:11] MYSQL("SIP/80619-09c9e610", "Query resID 1 SELECT regserver from sip_users where name='85684'") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:12] MYSQL("SIP/80619-09c9e610", "Fetch FetchId 2 REG_SERVER") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:13] NoOp("SIP/80619-09c9e610", "132.64.9.164") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:14] MYSQL("SIP/80619-09c9e610", "Clear 2") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:15] MYSQL("SIP/80619-09c9e610", "Query resID 1 SELECT callerid from sip_users where name='85684'") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:16] MYSQL("SIP/80619-09c9e610", "Fetch FetchId 2 CalledName") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:17] MYSQL("SIP/80619-09c9e610", "Clear 2") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:18] GotoIf("SIP/80619-09c9e610", "0?19:25") in new stack pbx-dev*CLI> -- Goto (huji-local,_806XX,25) pbx-dev*CLI> -- Executing [_806XX@huji-local:25] NoOp("SIP/80619-09c9e610", "Finish if-huji-local-277") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:26] MYSQL("SIP/80619-09c9e610", "Disconnect 1") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:27] Set("SIP/80619-09c9e610", "CALLEDID(all)="Yehavi Test <85684>" <85684>") in new stack pbx-dev*CLI> <--- Transmitting (no NAT) to 132.64.4.148:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bKca4d7c9764C63142;received=132.64.4.148 From: "80619" ;tag=D7256334-A63B9859 To: ;tag=as60761c8d Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Remote-Party-ID: "Yehavi Test" ;party=called;privacy=off;screen=no <------------> pbx-dev*CLI> -- Executing [_806XX@huji-local:28] NoOp("SIP/80619-09c9e610", ""132.64.9.164"") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:29] NoOp("SIP/80619-09c9e610", ""132.64.9.164"") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:30] NoOp("SIP/80619-09c9e610", ""To: 85684"") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:31] NoOp("SIP/80619-09c9e610", ""aExt: &SIP/80611"") in new stack pbx-dev*CLI> -- Executing [_806XX@huji-local:32] GotoIf("SIP/80619-09c9e610", "0?33:35") in new stack pbx-dev*CLI> -- Goto (huji-local,_806XX,35) pbx-dev*CLI> -- Executing [_806XX@huji-local:35] Dial("SIP/80619-09c9e610", "SIP/85684&SIP/80611|20|u") in new stack pbx-dev*CLI> [Oct 17 11:47:23] DEBUG[3268]: chan_sip.c:3211 update_call_counter: Call to peer '85684' is 1 out of 5 pbx-dev*CLI> Video is at 132.64.9.164 port 58328 pbx-dev*CLI> Audio is at 132.64.9.164 port 44064 Adding codec 0x8 (alaw) to SDP pbx-dev*CLI> Adding codec 0x2 (gsm) to SDP pbx-dev*CLI> Adding codec 0x400 (ilbc) to SDP Adding codec 0x4 (ulaw) to SDP pbx-dev*CLI> Adding codec 0x40000 (h261) to SDP pbx-dev*CLI> Adding codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP pbx-dev*CLI> Reliably Transmitting (no NAT) to 132.64.4.137:2091: INVITE sip:85684@132.64.4.137:2091;line=atg2eq38 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK5b7c1775;rport From: "Yehavi Test" ;tag=as6761dc89 To: Contact: Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 17 Oct 2007 09:47:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "Yehavi Test" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 429 v=0 o=root 2620 2620 IN IP4 132.64.9.164 s=session c=IN IP4 132.64.9.164 b=CT:384 t=0 0 m=audio 44064 RTP/AVP 8 3 97 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 58328 RTP/AVP 31 34 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=sendrecv --- pbx-dev*CLI> -- Called 85684 pbx-dev*CLI> [Oct 17 11:47:23] DEBUG[3268]: chan_sip.c:3211 update_call_counter: Call to peer '80611' is 1 out of 5 pbx-dev*CLI> Video is at 132.64.9.164 port 28632 Audio is at 132.64.9.164 port 32060 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x40000 (h261) to SDP Adding codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 132.64.4.148:5060: NOTIFY sip:80619@132.64.4.148 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK1e7caa52;rport From: ;tag=as1246390c To: "80619" ;tag=78EEDEF9-2CB460E6 Contact: Call-ID: 1efe2fdd-79d9d5db-e5a32718@132.64.4.148 CSeq: 159 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 352
--- pbx-dev*CLI> Extension Changed 80611 new state Ringing for Notify User 80619 pbx-dev*CLI> Reliably Transmitting (no NAT) to 132.64.4.120:5060: INVITE sip:80611@132.64.4.120:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK46ad9a3c;rport From: "Yehavi Test" ;tag=as2d6fd875 To: Contact: Call-ID: 3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 17 Oct 2007 09:47:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "Yehavi Test" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 429 v=0 o=root 2620 2620 IN IP4 132.64.9.164 s=session c=IN IP4 132.64.9.164 b=CT:384 t=0 0 m=audio 32060 RTP/AVP 8 3 97 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 28632 RTP/AVP 31 34 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=sendrecv --- pbx-dev*CLI> -- Called 80611 pbx-dev*CLI> <--- SIP read from 132.64.4.120:5060 ---> REGISTER sip:pbx-dev.cc.huji.ac.il SIP/2.0 Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK74b571b1705d19b From: Line1 ATAtest ;tag=1732664774 To: Line1 ATAtest Call-ID: 2629037634@132.64.4.120 CSeq: 46 REGISTER Contact: Line1 ATAtest ;expires=900 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) Authorization: Digest username="80681",realm="cc.huji.ac.il",nonce="51090755",uri="sip:pbx-dev.cc.huji.ac.il",response="9a3b01b9f36e65980d58bc4f7cc6381f" Content-Length: 0 <-------------> pbx-dev*CLI> --- (10 headers 0 lines) --- pbx-dev*CLI> Using latest REGISTER request as basis request Sending to 132.64.4.120 : 5060 (no NAT) pbx-dev*CLI> <--- Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK74b571b1705d19b;received=132.64.4.120 From: Line1 ATAtest ;tag=1732664774 To: Line1 ATAtest Call-ID: 2629037634@132.64.4.120 CSeq: 46 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: C pbx-dev*CLI> ontent-Length: 0 <------------> pbx-dev*CLI> <--- Transmitting (no NAT) to 132.64.4.120:5060 ---> SIP/2.0 403 Forbidden (Bad auth) Via: SIP/2.0/UDP 132.64.4.120:5060;branch=z9hG4bK74b571b1705d19b;received=132.64.4.120 From: Line1 ATAtest ;tag=1732664774 To: Line1 ATAtest ;tag=as0e569142 Call-ID: 2629037634@132.64.4.120 CSeq: 46 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> pbx-dev*CLI> [Oct 17 11:47:23] NOTICE[2627]: chan_sip.c:14913 handle_request_register: Registration from 'Line1 ATAtest ' failed for '132.64.4.120' - Wrong password Scheduling destruction of SIP dialog '2629037634@132.64.4.120' in 32000 ms (Method: REGISTER) pbx-dev*CLI> <--- SIP read from 132.64.4.120:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK46ad9a3c;rport From: "Yehavi Test" ;tag=as2d6fd875 To: ;tag=1776719707 Call-ID: 3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164 CSeq: 102 INVITE Server: Cisco ATA 186 v3.2.0 atasip (041111A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> pbx-dev*CLI> --- (9 headers 0 lines) --- pbx-dev*CLI> <--- SIP read from 132.64.4.120:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK46ad9a3c;rport From: "Yehavi Test" ;tag=as2d6fd875 To: ;tag=1776719707 Call-ID: 3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164 CSeq: 102 INVITE Contact: line2 ATAtest Server: Cisco ATA 186 v3.2.0 atasip (041111A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> pbx-dev*CLI> --- (10 headers 0 lines) --- pbx-dev*CLI> -- SIP/80611-09c9cbf0 is ringing pbx-dev*CLI> <--- Transmitting (no NAT) to 132.64.4.148:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bKca4d7c9764C63142;received=132.64.4.148 From: "80619" ;tag=D7256334-A63B9859 To: ;tag=as60761c8d Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-dev*CLI> <--- SIP read from 132.64.4.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK1e7caa52;rport From: ;tag=as1246390c To: "80619" ;tag=78EEDEF9-2CB460E6 CSeq: 159 NOTIFY Call-ID: 1efe2fdd-79d9d5db-e5a32718@132.64.4.148 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Content-Length: 0 <-------------> pbx-dev*CLI> --- (10 headers 0 lines) --- pbx-dev*CLI> SIP Response message for INCOMING dialog NOTIFY arrived pbx-dev*CLI> <--- SIP read from 132.64.4.137:2091 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK5b7c1775;rport=5060 From: "Yehavi Test" ;tag=as6761dc89 To: ;tag=iujivyik7z Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> pbx-dev*CLI> --- (10 headers 0 lines) --- pbx-dev*CLI> -- SIP/85684-09ca6b80 is ringing pbx-dev*CLI> <--- SIP read from 132.64.4.137:2091 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK5b7c1775;rport=5060 From: "Yehavi Test" ;tag=as6761dc89 To: ;tag=iujivyik7z Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> pbx-dev*CLI> --- (10 headers 0 lines) --- pbx-dev*CLI> -- SIP/85684-09ca6b80 is ringing pbx-dev*CLI> <--- SIP read from 132.64.4.137:2091 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK5b7c1775;rport=5060 From: "Yehavi Test" ;tag=as6761dc89 To: ;tag=iujivyik7z Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.5.12 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO A pbx-dev*CLI> llow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 370 v=0 o=root 39801348 39801349 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 65418 RTP/AVP 8 3 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8vKVEY5bed6p+8chV6zWbHbYLfE0uVGGlxxqyh3d a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> pbx-dev*CLI> --- (13 headers 15 lines) --- pbx-dev*CLI> Found RTP audio format 8 pbx-dev*CLI> Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 101 pbx-dev*CLI> [Oct 17 11:47:24] DEBUG[2627]: chan_sip.c:5079 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.137:65418 pbx-dev*CLI> Found description format pcma for ID 8 Found description format gsm for ID 3 Found description format pcmu for ID 0 Found description format telephone-event for ID 101 pbx-dev*CLI> Capabilities: us - 0xc040e (gsm|ulaw|alaw|ilbc|h261|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbx-dev*CLI> Peer audio RTP is at port 132.64.4.137:65418 Peer video RTP is at port 132.64.4.137:6400 list_route: hop: [Oct 17 11:47:24] DEBUG[2627]: chan_sip.c:5799 reqprep: Strict routing enforced for session 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2091 pbx-dev*CLI> Transmitting (no NAT) to 132.64.4.137:2091: ACK sip:85684@132.64.4.137:2091;line=atg2eq38 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK4a7b1b2a;rport From: "Yehavi Test" ;tag=as6761dc89 To: ;tag=iujivyik7z Contact: Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-dev*CLI> -- SIP/85684-09ca6b80 answered SIP/80619-09c9e610 pbx-dev*CLI> [Oct 17 11:47:24] DEBUG[3268]: chan_sip.c:3185 update_call_counter: Call to peer '80611' removed from call limit 5 pbx-dev*CLI> Reliably Transmitting (no NAT) to 132.64.4.148:5060: NOTIFY sip:80619@132.64.4.148 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK293650d1;rport From: ;tag=as1246390c To: "80619" ;tag=78EEDEF9-2CB460E6 Contact: Call-ID: 1efe2fdd-79d9d5db-e5a32718@132.64.4.148 CSeq: 160 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 347
--- pbx-dev*CLI> Scheduling destruction of SIP dialog '3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164' in 32000 ms (Method: INVITE) Extension Changed 80611 new state Idle for Notify User 80619 pbx-dev*CLI> Reliably Transmitting (no NAT) to 132.64.4.120:5060: CANCEL sip:80611@132.64.4.120:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK46ad9a3c;rport From: "Yehavi Test" ;tag=as2d6fd875 To: Call-ID: 3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164' in 32000 ms (Method: INVITE) pbx-dev*CLI> [Oct 17 11:47:24] NOTICE[3268]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/80611-09c9cbf0' not posted pbx-dev*CLI> Audio is at 132.64.9.164 port 16622 pbx-dev*CLI> Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP pbx-dev*CLI> <--- Reliably Transmitting (no NAT) to 132.64.4.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bKca4d7c9764C63142;received=132.64.4.148 From: "80619" ;tag=D7256334-A63B9859 To: ;tag=as60761c8d Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 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: 262 v=0 o=root 2620 2620 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 65418 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-dev*CLI> -- Native bridging SIP/80619-09c9e610 and SIP/85684-09ca6b80 pbx-dev*CLI> [Oct 17 11:47:24] DEBUG[3268]: chan_sip.c:5799 reqprep: Strict routing enforced for session 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2091 pbx-dev*CLI> Audio is at 132.64.9.164 port 44064 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP pbx-dev*CLI> Reliably Transmitting (no NAT) to 132.64.4.137:2091: INVITE sip:85684@132.64.4.137:2091;line=atg2eq38 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK6f32b84f;rport From: "Yehavi Test" ;tag=as6761dc89 To: ;tag=iujivyik7z Contact: Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 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: 261 v=0 o=root 2620 2621 IN IP4 132.64.4.148 s=session c=IN IP4 132.64.4.148 t=0 0 m=audio 2230 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-dev*CLI> <--- SIP read from 132.64.4.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK46ad9a3c;rport From: "Yehavi Test" ;tag=as2d6fd875 To: ;tag=1776719707 Call-ID: 3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164 CSeq: 102 CANCEL Server: Cisco ATA 186 v3.2.0 atasip (041111A) Supported: replaces Content-Length: 0 <-------------> pbx-dev*CLI> --- (9 headers 0 lines) --- pbx-dev*CLI> <--- SIP read from 132.64.4.120:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK46ad9a3c;rport From: "Yehavi Test" ;tag=as2d6fd875 To: ;tag=1776719707 Call-ID: 3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164 CSeq: 102 INVITE Server: Cisco ATA 186 v3.2.0 atasip (041111A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> pbx-dev*CLI> --- (9 headers 0 lines) --- pbx-dev*CLI> Transmitting (no NAT) to 132.64.4.120:5060: ACK sip:80611@132.64.4.120:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK46ad9a3c;rport From: "Yehavi Test" ;tag=as2d6fd875 To: ;tag=1776719707 Contact: Call-ID: 3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-dev*CLI> [Oct 17 11:47:24] DEBUG[2627]: chan_sip.c:3185 update_call_counter: Call to peer '80611' removed from call limit 5 pbx-dev*CLI> Really destroying SIP dialog '3194b69b60b04cf12cc0fdac2b312d79@132.64.9.164' Method: INVITE pbx-dev*CLI> <--- SIP read from 132.64.4.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK293650d1;rport From: ;tag=as1246390c To: "80619" ;tag=78EEDEF9-2CB460E6 CSeq: 160 NOTIFY Call-ID: 1efe2fdd-79d9d5db-e5a32718@132.64.4.148 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Content-Length: 0 <-------------> pbx-dev*CLI> --- (10 headers 0 lines) --- pbx-dev*CLI> SIP Response message for INCOMING dialog NOTIFY arrived pbx-dev*CLI> <--- SIP read from 132.64.4.148:5060 ---> ACK sip:85684@132.64.9.164 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bK254e59be814E7C11 From: "80619" ;tag=D7256334-A63B9859 To: ;tag=as60761c8d CSeq: 2 ACK Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 pbx-dev*CLI> Proxy-Authorization: Digest username="80619", realm="cc.huji.ac.il", nonce="11792329", uri="sip:85684@132.64.9.164:5060;user=phone", response="68878bb534bdbf8cecaf0c8b11caf16f", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> pbx-dev*CLI> --- (12 headers 0 lines) --- pbx-dev*CLI> <--- SIP read from 132.64.4.137:2091 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK6f32b84f;rport=5060 From: "Yehavi Test" ;tag=as6761dc89 To: ;tag=iujivyik7z Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 CSeq: 103 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.5.12 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO A pbx-dev*CLI> llow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 347 v=0 o=root 39801348 39801350 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 65418 RTP/AVP 8 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8vKVEY5bed6p+8chV6zWbHbYLfE0uVGGlxxqyh3d a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv <-------------> pbx-dev*CLI> --- (13 headers 14 lines) --- pbx-dev*CLI> Found RTP audio format 8 pbx-dev*CLI> Found RTP audio format 0 Found RTP audio format 101 pbx-dev*CLI> [Oct 17 11:47:25] DEBUG[2627]: chan_sip.c:5079 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.137:65418 pbx-dev*CLI> Found description format pcma for ID 8 Found description format pcmu for ID 0 Found description format telephone-event for ID 101 pbx-dev*CLI> Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:65418 pbx-dev*CLI> Peer video RTP is at port 132.64.4.137:6400 pbx-dev*CLI> [Oct 17 11:47:25] DEBUG[2627]: chan_sip.c:5799 reqprep: Strict routing enforced for session 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.137, port 2091 Transmitting (no NAT) to 132.64.4.137:2091: ACK sip:85684@132.64.4.137:2091;line=atg2eq38 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK14ccc3e2;rport From: "Yehavi Test" ;tag=as6761dc89 To: ;tag=iujivyik7z Contact: Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-dev*CLI> [Oct 17 11:47:25] DEBUG[3268]: chan_sip.c:5799 reqprep: Strict routing enforced for session 9b360d78-421b5146-8585953b@132.64.4.148 pbx-dev*CLI> set_destination: Parsing for address/port to send to pbx-dev*CLI> set_destination: set destination to 132.64.4.148, port 5060 Audio is at 132.64.9.164 port 16622 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP pbx-dev*CLI> Reliably Transmitting (no NAT) to 132.64.4.148:5060: INVITE sip:80619@132.64.4.148 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK3fba246d;rport From: ;tag=as60761c8d To: "80619" ;tag=D7256334-A63B9859 Contact: Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces pbx-dev*CLI> X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 262 v=0 o=root 2620 2621 IN IP4 132.64.4.137 s=session c=IN IP4 132.64.4.137 t=0 0 m=audio 65418 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-dev*CLI> <--- SIP read from 132.64.4.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK3fba246d;rport From: ;tag=as60761c8d To: "80619" ;tag=D7256334-A63B9859 CSeq: 102 INVITE Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Content-Type: application/sdp Content-Length: 199 v=0 o=- 1192614252 1192614253 IN IP4 132.64.4.148 s=Polycom IP Phone c=IN IP4 132.64.4.148 t=0 0 m=audio 2230 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <-------------> pbx-dev*CLI> --- (11 headers 9 lines) --- pbx-dev*CLI> Found RTP audio format 8 pbx-dev*CLI> Found RTP audio format 101 pbx-dev*CLI> [Oct 17 11:47:25] DEBUG[2627]: chan_sip.c:5079 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.148:2230 pbx-dev*CLI> Found description format PCMA for ID 8 Found description format telephone-event for ID 101 pbx-dev*CLI> Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.148:2230 Peer video RTP is at port 132.64.4.148:6400 pbx-dev*CLI> list_route: hop: [Oct 17 11:47:25] DEBUG[2627]: chan_sip.c:5799 reqprep: Strict routing enforced for session 9b360d78-421b5146-8585953b@132.64.4.148 set_destination: Parsing for address/port to send to set_destination: set destination to 132.64.4.148, port 5060 pbx-dev*CLI> Transmitting (no NAT) to 132.64.4.148:5060: ACK sip:80619@132.64.4.148 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK7e0ec675;rport From: ;tag=as60761c8d To: "80619" ;tag=D7256334-A63B9859 Contact: Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-dev*CLI> <--- SIP read from 132.64.4.137:2091 ---> BYE sip:80619@132.64.9.164 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2091;branch=z9hG4bK-s4rfv8tegl4d;rport From: ;tag=iujivyik7z To: "Yehavi Test" ;tag=as6761dc89 Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 CSeq: 1 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom320/6.5.12 RTP-RxStat: Total_Rx_Pkts=175,Rx_Pkts=175,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=3 RTP-TxStat: Total_Tx_Pkts=179,Tx_Pkts=179,Remote_Tx_Pkts=101 Content-Length: 0 <-------------> pbx-dev*CLI> --- (12 headers 0 lines) --- pbx-dev*CLI> Sending to 132.64.4.137 : 2091 (NAT) pbx-dev*CLI> <--- Transmitting (NAT) to 132.64.4.137:2091 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.4.137:2091;branch=z9hG4bK-s4rfv8tegl4d;received=132.64.4.137;rport=2091 From: ;tag=iujivyik7z To: "Yehavi Test" ;tag=as6761dc89 Call-ID: 7ecdad2d09136fda38e79d24144139d8@132.64.9.164 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx-dev*CLI> [Oct 17 11:47:29] DEBUG[3268]: chan_sip.c:5799 reqprep: Strict routing enforced for session 9b360d78-421b5146-8585953b@132.64.4.148 pbx-dev*CLI> set_destination: Parsing for address/port to send to pbx-dev*CLI> set_destination: set destination to 132.64.4.148, port 5060 pbx-dev*CLI> Audio is at 132.64.9.164 port 16622 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP pbx-dev*CLI> Reliably Transmitting (no NAT) to 132.64.4.148:5060: INVITE sip:80619@132.64.4.148 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK2d20c337;rport From: ;tag=as60761c8d To: "80619" ;tag=D7256334-A63B9859 Contact: Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces pbx-dev*CLI> X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 238 v=0 o=root 2620 2622 IN IP4 132.64.9.164 s=session c=IN IP4 132.64.9.164 t=0 0 m=audio 16622 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-dev*CLI> [Oct 17 11:47:29] DEBUG[3268]: chan_sip.c:3185 update_call_counter: Call to peer '85684' removed from call limit 5 == Spawn extension (huji-local, _806XX, 35) exited non-zero on 'SIP/80619-09c9e610' -- Executing [h@huji-local:1] ResetCDR("SIP/80619-09c9e610", "w") in new stack pbx-dev*CLI> -- Executing [h@huji-local:2] NoOp("SIP/80619-09c9e610", "80619") in new stack pbx-dev*CLI> Really destroying SIP dialog '7ecdad2d09136fda38e79d24144139d8@132.64.9.164' Method: BYE [Oct 17 11:47:29] DEBUG[3268]: db.c:197 ast_db_get: Unable to find key 'CallBack' in family '80619' pbx-dev*CLI> [Oct 17 11:47:29] DEBUG[3268]: func_db.c:70 function_db_read: DB: 80619/CallBack not found in database. -- Executing [h@huji-local:3] Set("SIP/80619-09c9e610", "tmp=") in new stack pbx-dev*CLI> -- Executing [h@huji-local:4] GotoIf("SIP/80619-09c9e610", "0?5:11") in new stack pbx-dev*CLI> -- Goto (huji-local,h,11) pbx-dev*CLI> -- Executing [h@huji-local:11] NoOp("SIP/80619-09c9e610", "Finish if-huji-local-283") in new stack pbx-dev*CLI> [Oct 17 11:47:29] DEBUG[3268]: db.c:197 ast_db_get: Unable to find key 'CallBack' in family '85684' pbx-dev*CLI> [Oct 17 11:47:29] DEBUG[3268]: func_db.c:70 function_db_read: DB: 85684/CallBack not found in database. -- Executing [h@huji-local:12] Set("SIP/80619-09c9e610", "tmp=") in new stack pbx-dev*CLI> -- Executing [h@huji-local:13] GotoIf("SIP/80619-09c9e610", "0?14:20") in new stack pbx-dev*CLI> -- Goto (huji-local,h,20) pbx-dev*CLI> -- Executing [h@huji-local:20] NoOp("SIP/80619-09c9e610", "Finish if-huji-local-284") in new stack pbx-dev*CLI> [Oct 17 11:47:29] DEBUG[3268]: chan_sip.c:3185 update_call_counter: Call from peer '80619' removed from call limit 5 pbx-dev*CLI> Scheduling destruction of SIP dialog '9b360d78-421b5146-8585953b@132.64.4.148' in 32000 ms (Method: ACK) pbx-dev*CLI> <--- SIP read from 132.64.4.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK2d20c337;rport From: ;tag=as60761c8d To: "80619" ;tag=D7256334-A63B9859 CSeq: 103 INVITE Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Content-Type: application/sdp Content-Length: 199 v=0 o=- 1192614252 1192614254 IN IP4 132.64.4.148 s=Polycom IP Phone c=IN IP4 132.64.4.148 t=0 0 m=audio 2230 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <-------------> pbx-dev*CLI> --- (11 headers 9 lines) --- pbx-dev*CLI> Found RTP audio format 8 pbx-dev*CLI> Found RTP audio format 101 pbx-dev*CLI> [Oct 17 11:47:29] DEBUG[2627]: chan_sip.c:5079 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.148:2230 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) pbx-dev*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) pbx-dev*CLI> Peer audio RTP is at port 132.64.4.148:2230 Peer video RTP is at port 132.64.4.148:6400 [Oct 17 11:47:29] DEBUG[2627]: chan_sip.c:5799 reqprep: Strict routing enforced for session 9b360d78-421b5146-8585953b@132.64.4.148 set_destination: Parsing for address/port to send to pbx-dev*CLI> set_destination: set destination to 132.64.4.148, port 5060 pbx-dev*CLI> Transmitting (no NAT) to 132.64.4.148:5060: ACK sip:80619@132.64.4.148 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK4db54f47;rport From: ;tag=as60761c8d To: "80619" ;tag=D7256334-A63B9859 Contact: Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-dev*CLI> [Oct 17 11:47:29] DEBUG[2627]: chan_sip.c:5799 reqprep: Strict routing enforced for session 9b360d78-421b5146-8585953b@132.64.4.148 set_destination: Parsing for address/port to send to pbx-dev*CLI> set_destination: set destination to 132.64.4.148, port 5060 pbx-dev*CLI> Reliably Transmitting (no NAT) to 132.64.4.148:5060: BYE sip:80619@132.64.4.148 SIP/2.0 Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK4fbef141;rport From: ;tag=as60761c8d To: "80619" ;tag=D7256334-A63B9859 Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-dev*CLI> Scheduling destruction of SIP dialog '9b360d78-421b5146-8585953b@132.64.4.148' in 32000 ms (Method: ACK) pbx-dev*CLI> <--- SIP read from 132.64.4.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK4fbef141;rport From: ;tag=as60761c8d To: "80619" ;tag=D7256334-A63B9859 CSeq: 104 BYE Call-ID: 9b360d78-421b5146-8585953b@132.64.4.148 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Content-Length: 0 <-------------> pbx-dev*CLI> --- (9 headers 0 lines) --- pbx-dev*CLI> Really destroying SIP dialog '9b360d78-421b5146-8585953b@132.64.4.148' Method: ACK pbx-dev*CLI> quit Executing last minute cleanups