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=z9hG4bKf7383f0eFFCB40E1 From: "80619" ;tag=E0C75B8-3AE877D To: CSeq: 1 INVITE Call-ID: 80b3c87c-9fab560a-8282d09f@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=- 1192614217 1192614217 IN IP4 132.64.4.148 s=Polycom IP Phone c=IN IP4 132.64.4.148 t=0 0 m=audio 2228 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 - 80b3c87c-9fab560a-8282d09f@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=z9hG4bKf7383f0eFFCB40E1;received=132.64.4.148 From: "80619" ;tag=E0C75B8-3AE877D To: ;tag=as07852af2 Call-ID: 80b3c87c-9fab560a-8282d09f@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="2f2d7e0f" Content-Length: 0 <------------> pbx-dev*CLI> Scheduling destruction of SIP dialog '80b3c87c-9fab560a-8282d09f@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=z9hG4bKf7383f0eFFCB40E1 From: "80619" ;tag=E0C75B8-3AE877D To: ;tag=as07852af2 CSeq: 1 ACK Call-ID: 80b3c87c-9fab560a-8282d09f@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=z9hG4bKea65437b3956D586 From: "80619" ;tag=E0C75B8-3AE877D To: CSeq: 2 INVITE Call-ID: 80b3c87c-9fab560a-8282d09f@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="2f2d7e0f", uri="sip:85684@132.64.9.164:5060;user=phone", response="8fb846baf673adf2be4b9db2d24fc761", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1192614217 1192614217 IN IP4 132.64.4.148 s=Polycom IP Phone c=IN IP4 132.64.4.148 t=0 0 m=audio 2228 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) pbx-dev*CLI> Using INVITE request as basis request - 80b3c87c-9fab560a-8282d09f@132.64.4.148 pbx-dev*CLI> Found user '80619' pbx-dev*CLI> Found RTP audio format 8 pbx-dev*CLI> Found RTP audio format 0 pbx-dev*CLI> Found RTP audio format 18 Found RTP audio format 101 pbx-dev*CLI> [Oct 17 11:46:48] DEBUG[2627]: chan_sip.c:5079 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.148:2228 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) pbx-dev*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.148:2228 pbx-dev*CLI> Peer video RTP is at port 132.64.4.148:6400 pbx-dev*CLI> [Oct 17 11:46:48] DEBUG[2627]: chan_sip.c:3211 update_call_counter: Call from peer '80619' is 1 out of 5 pbx-dev*CLI> 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=z9hG4bKea65437b3956D586;received=132.64.4.148 From: "80619" ;tag=E0C75B8-3AE877D To: Call-ID: 80b3c87c-9fab560a-8282d09f@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 -- 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-250") 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=z9hG4bKea65437b3956D586;received=132.64.4.148 From: "80619" ;tag=E0C75B8-3AE877D To: ;tag=as47bababe Call-ID: 80b3c87c-9fab560a-8282d09f@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|uL(3600000:60000:30000)") in new stack pbx-dev*CLI> -- Limit Data for this call: pbx-dev*CLI> > timelimit = 3600000 > play_warning = 60000 > play_to_caller = yes > play_to_callee = no > warning_freq = 30000 > start_sound = (null) > warning_sound = timeleft > end_sound = (null) pbx-dev*CLI> [Oct 17 11:46:48] DEBUG[3243]: 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 55634 pbx-dev*CLI> Audio is at 132.64.9.164 port 27684 pbx-dev*CLI> Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x400 (ilbc) to SDP pbx-dev*CLI> 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 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=z9hG4bK0e134359;rport From: "Yehavi Test" ;tag=as3a62088f To: Contact: Call-ID: 2dc454f334153b8a0b1ba71b41001509@132.64.9.164 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 17 Oct 2007 09:46:48 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 27684 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 55634 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 11:46:48] DEBUG[3243]: chan_sip.c:3211 update_call_counter: Call to peer '80611' is 1 out of 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=z9hG4bK03c0ed9f;rport From: ;tag=as1246390c To: "80619" ;tag=78EEDEF9-2CB460E6 Contact: Call-ID: 1efe2fdd-79d9d5db-e5a32718@132.64.4.148 CSeq: 157 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 352
--- pbx-dev*CLI> Video is at 132.64.9.164 port 51602 pbx-dev*CLI> Extension Changed 80611 new state Ringing for Notify User 80619 pbx-dev*CLI> Audio is at 132.64.9.164 port 10892 pbx-dev*CLI> Adding codec 0x8 (alaw) to SDP pbx-dev*CLI> Adding codec 0x2 (gsm) to SDP pbx-dev*CLI> Adding codec 0x400 (ilbc) to SDP pbx-dev*CLI> 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 132.64.4.120:5060: 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=z9hG4bK05397891;rport From: "Yehavi Test" ;tag=as34900b16 To: Contact: Call-ID: 2fd42d4a5b599cb465c8ed0a392896fb@132.64.9.164 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 17 Oct 2007 09:46:48 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 10892 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 51602 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 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 132.64.9.164:5060;branch=z9hG4bK05397891;rport From: "Yehavi Test" ;tag=as34900b16 To: ;tag=1776719707 Call-ID: 2fd42d4a5b599cb465c8ed0a392896fb@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=z9hG4bK05397891;rport From: "Yehavi Test" ;tag=as34900b16 To: ;tag=1776719707 Call-ID: 2fd42d4a5b599cb465c8ed0a392896fb@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/80611-09c8d590 is ringing <--- Transmitting (no NAT) to 132.64.4.148:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bKea65437b3956D586;received=132.64.4.148 From: "80619" ;tag=E0C75B8-3AE877D To: ;tag=as47bababe Call-ID: 80b3c87c-9fab560a-8282d09f@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=z9hG4bK03c0ed9f;rport From: ;tag=as1246390c To: "80619" ;tag=78EEDEF9-2CB460E6 CSeq: 157 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=z9hG4bK0e134359;rport=5060 From: "Yehavi Test" ;tag=as3a62088f To: ;tag=v35ijmlali Call-ID: 2dc454f334153b8a0b1ba71b41001509@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/85684-09c9cbf0 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=z9hG4bK0e134359;rport=5060 From: "Yehavi Test" ;tag=as3a62088f To: ;tag=v35ijmlali Call-ID: 2dc454f334153b8a0b1ba71b41001509@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/85684-09c9cbf0 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=z9hG4bK0e134359;rport=5060 From: "Yehavi Test" ;tag=as3a62088f To: ;tag=v35ijmlali Call-ID: 2dc454f334153b8a0b1ba71b41001509@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: 372 v=0 o=root 432510983 432510984 IN IP4 132.64.4.137 s=call c=IN IP4 132.64.4.137 t=0 0 m=audio 59008 RTP/AVP 8 3 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZHi+HCLGI99rp0vh4aQ6qFiP6E8fUpqPUmHbgzJm 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 11:46:49] DEBUG[2627]: chan_sip.c:5079 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 132.64.4.137:59008 Found description format pcma for ID 8 Found description format gsm for ID 3 Found description format pcmu for ID 0 pbx-dev*CLI> 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) pbx-dev*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 132.64.4.137:59008 Peer video RTP is at port 132.64.4.137:6400 list_route: hop: pbx-dev*CLI> [Oct 17 11:46:49] DEBUG[2627]: chan_sip.c:5799 reqprep: Strict routing enforced for session 2dc454f334153b8a0b1ba71b41001509@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=z9hG4bK46fc18b2;rport From: "Yehavi Test" ;tag=as3a62088f To: ;tag=v35ijmlali Contact: Call-ID: 2dc454f334153b8a0b1ba71b41001509@132.64.9.164 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-dev*CLI> -- SIP/85684-09c9cbf0 answered SIP/80619-09c9e610 pbx-dev*CLI> [Oct 17 11:46:49] DEBUG[3243]: chan_sip.c:3185 update_call_counter: Call to peer '80611' removed from call limit 5 pbx-dev*CLI> Scheduling destruction of SIP dialog '2fd42d4a5b599cb465c8ed0a392896fb@132.64.9.164' in 32000 ms (Method: INVITE) 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=z9hG4bK2c0561b4;rport From: ;tag=as1246390c To: "80619" ;tag=78EEDEF9-2CB460E6 Contact: Call-ID: 1efe2fdd-79d9d5db-e5a32718@132.64.4.148 CSeq: 158 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 347
--- 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=z9hG4bK05397891;rport From: "Yehavi Test" ;tag=as34900b16 To: Call-ID: 2fd42d4a5b599cb465c8ed0a392896fb@132.64.9.164 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-dev*CLI> Extension Changed 80611 new state Idle for Notify User 80619 pbx-dev*CLI> Scheduling destruction of SIP dialog '2fd42d4a5b599cb465c8ed0a392896fb@132.64.9.164' in 32000 ms (Method: INVITE) pbx-dev*CLI> [Oct 17 11:46:49] NOTICE[3243]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/80611-09c8d590' not posted pbx-dev*CLI> Audio is at 132.64.9.164 port 6058 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- 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=z9hG4bKea65437b3956D586;received=132.64.4.148 From: "80619" ;tag=E0C75B8-3AE877D To: ;tag=as47bababe Call-ID: 80b3c87c-9fab560a-8282d09f@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 59008 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=z9hG4bK05397891;rport From: "Yehavi Test" ;tag=as34900b16 To: ;tag=1776719707 Call-ID: 2fd42d4a5b599cb465c8ed0a392896fb@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=z9hG4bK05397891;rport From: "Yehavi Test" ;tag=as34900b16 To: ;tag=1776719707 Call-ID: 2fd42d4a5b599cb465c8ed0a392896fb@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=z9hG4bK05397891;rport From: "Yehavi Test" ;tag=as34900b16 To: ;tag=1776719707 Contact: Call-ID: 2fd42d4a5b599cb465c8ed0a392896fb@132.64.9.164 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- pbx-dev*CLI> [Oct 17 11:46:49] 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 '2fd42d4a5b599cb465c8ed0a392896fb@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=z9hG4bK2c0561b4;rport From: ;tag=as1246390c To: "80619" ;tag=78EEDEF9-2CB460E6 CSeq: 158 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=z9hG4bK7cb4ad827D112235 From: "80619" ;tag=E0C75B8-3AE877D To: ;tag=as47bababe CSeq: 2 ACK Call-ID: 80b3c87c-9fab560a-8282d09f@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 Proxy-Authorization: Digest username="80619", realm="cc.huji.ac.il", nonce="2f2d7e0f", uri="sip:85684@132.64.9.164:5060;user=phone", response="8fb846baf673adf2be4b9db2d24fc761", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> pbx-dev*CLI> --- (12 headers 0 lines) --- pbx-dev*CLI> [Oct 17 11:46:52] DEBUG[2623]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. pbx-dev*CLI> [Oct 17 11:46:52] DEBUG[2623]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name = '80681' pbx-dev*CLI> [Oct 17 11:46:52] DEBUG[2623]: db.c:197 ast_db_get: Unable to find key '80681' in family 'SIP/Registry' 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=z9hG4bKc9a041d5c12fbd7c From: Line1 ATAtest ;tag=1732664774 To: Line1 ATAtest Call-ID: 2629037634@132.64.4.120 CSeq: 45 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) --- Using latest REGISTER request as basis request pbx-dev*CLI> 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=z9hG4bKc9a041d5c12fbd7c;received=132.64.4.120 From: Line1 ATAtest ;tag=1732664774 To: Line1 ATAtest Call-ID: 2629037634@132.64.4.120 CSeq: 45 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: pbx-dev*CLI> Content-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=z9hG4bKc9a041d5c12fbd7c;received=132.64.4.120 From: Line1 ATAtest ;tag=1732664774 To: Line1 ATAtest ;tag=as0e569142 Call-ID: 2629037634@132.64.4.120 CSeq: 45 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> pbx-dev*CLI> [Oct 17 11:46:53] 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 132.64.4.137:2091 ---> BYE sip:80619@132.64.9.164 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.137:2091;branch=z9hG4bK-856qmusjglbb;rport From: ;tag=v35ijmlali To: "Yehavi Test" ;tag=as3a62088f Call-ID: 2dc454f334153b8a0b1ba71b41001509@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=159,Rx_Pkts=159,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=160,Tx_Pkts=160,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> pbx-dev*CLI> --- (12 headers 0 lines) --- 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-856qmusjglbb;received=132.64.4.137;rport=2091 From: ;tag=v35ijmlali To: "Yehavi Test" ;tag=as3a62088f Call-ID: 2dc454f334153b8a0b1ba71b41001509@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 11:46:53] DEBUG[3243]: chan_sip.c:3185 update_call_counter: Call to peer '85684' removed from call limit 5 pbx-dev*CLI> == Spawn extension (huji-local, _806XX, 35) exited non-zero on 'SIP/80619-09c9e610' pbx-dev*CLI> -- 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 Really destroying SIP dialog '2dc454f334153b8a0b1ba71b41001509@132.64.9.164' Method: BYE [Oct 17 11:46:53] DEBUG[3243]: db.c:197 ast_db_get: Unable to find key 'CallBack' in family '80619' [Oct 17 11:46:53] DEBUG[3243]: 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-256") in new stack pbx-dev*CLI> [Oct 17 11:46:53] DEBUG[3243]: db.c:197 ast_db_get: Unable to find key 'CallBack' in family '85684' pbx-dev*CLI> [Oct 17 11:46:53] DEBUG[3243]: 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-257") in new stack pbx-dev*CLI> [Oct 17 11:46:53] DEBUG[3243]: chan_sip.c:3185 update_call_counter: Call from peer '80619' removed from call limit 5 Scheduling destruction of SIP dialog '80b3c87c-9fab560a-8282d09f@132.64.4.148' in 32000 ms (Method: ACK) [Oct 17 11:46:53] DEBUG[3243]: chan_sip.c:5799 reqprep: Strict routing enforced for session 80b3c87c-9fab560a-8282d09f@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> 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=z9hG4bK64344e83;rport From: ;tag=as47bababe To: "80619" ;tag=E0C75B8-3AE877D Call-ID: 80b3c87c-9fab560a-8282d09f@132.64.4.148 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 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=z9hG4bK64344e83;rport From: ;tag=as47bababe To: "80619" ;tag=E0C75B8-3AE877D CSeq: 102 BYE Call-ID: 80b3c87c-9fab560a-8282d09f@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> SIP Response message for INCOMING dialog BYE arrived pbx-dev*CLI> Really destroying SIP dialog '80b3c87c-9fab560a-8282d09f@132.64.4.148' Method: ACK pbx-dev*CLI> quit <--- SIP read from 132.64.4.148:5060 ---> SUBSCRIBE sip:80606@132.64.9.164:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bKdd6021b0D9E11EB3 From: "80619" ;tag=B9926FA-921F670F To: CSeq: 1 SUBSCRIBE Call-ID: e420cdfe-7003006c-ddb35651@132.64.4.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> pbx-dev*CLI> quit --- (14 headers 0 lines) --- pbx-dev*CLI> quit Creating new subscription pbx-dev*CLI> quit Sending to 132.64.4.148 : 5060 (no NAT) pbx-dev*CLI> quit Found peer '80619' pbx-dev*CLI> quit <--- Transmitting (no NAT) to 132.64.4.148:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bKdd6021b0D9E11EB3;received=132.64.4.148 From: "80619" ;tag=B9926FA-921F670F To: ;tag=as2d22e08f Call-ID: e420cdfe-7003006c-ddb35651@132.64.4.148 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="34a80b82" Content-Length: 0 <------------> pbx-dev*CLI> quit Scheduling destruction of SIP dialog 'e420cdfe-7003006c-ddb35651@132.64.4.148' in 32000 ms (Method: SUBSCRIBE) pbx-dev*CLI> quit <--- SIP read from 132.64.4.148:5060 ---> SUBSCRIBE sip:80606@132.64.9.164:5060 SIP/2.0 Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bK14960eedDF69CFA8 From: "80619" ;tag=B9926FA-921F670F To: CSeq: 2 SUBSCRIBE Call-ID: e420cdfe-7003006c-ddb35651@132.64.4.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.3.0127 Accept: application/xpidf+xml,text/xml+msrtc.pidf Authorization: Digest username="80619", realm="cc.huji.ac.il", nonce="34a80b82", uri="sip:80606@132.64.9.164:5060", response="090dc04583bc3f43f2c192e20c7a8333", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> pbx-dev*CLI> quit --- (15 headers 0 lines) --- pbx-dev*CLI> quit Creating new subscription Sending to 132.64.4.148 : 5060 (no NAT) pbx-dev*CLI> quit Found peer '80619' pbx-dev*CLI> quit Looking for 80606 in huji-remote-gr (domain 132.64.9.164) pbx-dev*CLI> quit [Oct 17 11:46:54] DEBUG[2627]: chan_sip.c:14814 handle_request_subscribe: Adding subscription for extension 80606 context huji-local for peer 80619 Scheduling destruction of SIP dialog 'e420cdfe-7003006c-ddb35651@132.64.4.148' in 3610000 ms (Method: SUBSCRIBE) pbx-dev*CLI> quit [Oct 17 11:46:54] NOTICE[2627]: chan_sip.c:14833 handle_request_subscribe: Got SUBSCRIBE for extension 80606@huji-local from 132.64.4.148, but there is no hint for that extension. pbx-dev*CLI> quit <--- Transmitting (no NAT) to 132.64.4.148:5060 ---> SIP/2.0 404 Not found Via: SIP/2.0/UDP 132.64.4.148;branch=z9hG4bK14960eedDF69CFA8;received=132.64.4.148 From: "80619" ;tag=B9926FA-921F670F To: ;tag=as2d22e08f Call-ID: e420cdfe-7003006c-ddb35651@132.64.4.148 CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> pbx-dev*CLI> quit Really destroying SIP dialog 'e420cdfe-7003006c-ddb35651@132.64.4.148' Method: SUBSCRIBE pbx-dev*CLI> quit Executing last minute cleanups