foxtrot*CLI> sip debug SIP Debugging enabled -- Accepting call from '07803034440' to '444608' on channel 0/1, span 1 [Aug 7 18:03:02] DEBUG[8132]: chan_zap.c:1384 zt_enable_ec: Enabled echo cancellation on channel 1 -- Executing [444608@isdn10:1] Goto("Zap/1-1", "common|7708|1") in new stack -- Goto (common,7708,1) -- Executing [7708@common:1] Answer("Zap/1-1", "") in new stack [Aug 7 18:03:02] DEBUG[8124]: channel.c:885 channel_find_locked: Avoiding initial deadlock for channel '0x90d69c0' -- Executing [7708@common:2] Dial("Zap/1-1", "SIP/7708|120|g") in new stack Audio is at 192.168.6.6 port 18782 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.6.252:5060: INVITE sip:7708@192.168.6.252:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK42514e3e;rport From: "07803034440" ;tag=as1eb408cc To: Contact: Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 Aug 2006 17:03:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 271 v=0 o=root 8073 8073 IN IP4 192.168.6.6 s=session c=IN IP4 192.168.6.6 t=0 0 m=audio 18782 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Called 7708 [Aug 7 18:03:02] DEBUG[8124]: channel.c:885 channel_find_locked: Avoiding initial deadlock for channel '0x90d69c0' foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK42514e3e;rport From: "07803034440" ;tag=as1eb408cc To: Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 Date: Mon, 07 Aug 2006 17:03:02 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Length: 0 --- (10 headers 0 lines)--- foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK42514e3e;rport From: "07803034440" ;tag=as1eb408cc To: ;tag=000c8533cc9900641bee4edb-65bcbf53 Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 Date: Mon, 07 Aug 2006 17:03:02 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/7708-09180220 is ringing [Aug 7 18:03:02] DEBUG[8645]: chan_zap.c:4918 zt_indicate: Requested indication 3 on channel Zap/1-1 foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK42514e3e;rport From: "07803034440" ;tag=as1eb408cc To: ;tag=000c8533cc9900641bee4edb-65bcbf53 Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 Date: Mon, 07 Aug 2006 17:03:05 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 198 v=0 o=Cisco-SIPUA 6166 25689 IN IP4 192.168.6.252 s=SIP Call c=IN IP4 192.168.6.252 t=0 0 m=audio 18820 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (11 headers 9 lines)--- [Aug 7 18:03:05] DEBUG[8135]: chan_sip.c:1996 __sip_ack: Acked pending invite 102 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.6.252:18820 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.6.252:18820 list_route: hop: [Aug 7 18:03:05] DEBUG[8135]: chan_sip.c:5343 reqprep: Strict routing enforced for session 127f2910376754bd1ff7c00578ea2198@192.168.6.6 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.6.252, port 5060 Transmitting (no NAT) to 192.168.6.252:5060: ACK sip:7708@192.168.6.252:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK0f258de1;rport From: "07803034440" ;tag=as1eb408cc To: ;tag=000c8533cc9900641bee4edb-65bcbf53 Contact: Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/7708-09180220 answered Zap/1-1 [Aug 7 18:03:05] DEBUG[8645]: chan_zap.c:4918 zt_indicate: Requested indication -1 on channel Zap/1-1 foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: INVITE sip:07803034440@192.168.6.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK2180df96 From: ;tag=000c8533cc9900641bee4edb-65bcbf53 To: "07803034440" ;tag=as1eb408cc Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 Date: Mon, 07 Aug 2006 17:03:10 GMT CSeq: 101 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 193 Remote-Party-ID: "7708" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 20276 12844 IN IP4 192.168.6.252 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 18820 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (12 headers 9 lines)--- Sending to 192.168.6.252 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:18820 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 0.0.0.0:18820 Audio is at 192.168.6.6 port 18782 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.6.252:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK2180df96;received=192.168.6.252 From: ;tag=000c8533cc9900641bee4edb-65bcbf53 To: "07803034440" ;tag=as1eb408cc Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 224 v=0 o=root 8073 8074 IN IP4 192.168.6.6 s=session c=IN IP4 192.168.6.6 t=0 0 m=audio 18782 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- [Aug 7 18:03:10] DEBUG[8645]: chan_zap.c:4918 zt_indicate: Requested indication 16 on channel Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 [Aug 7 18:03:10] DEBUG[8645]: channel.c:1837 ast_settimeout: Scheduling timer at 160 sample intervals [Aug 7 18:03:10] DEBUG[8645]: channel.c:1837 ast_settimeout: Scheduling timer at 0 sample intervals foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: ACK sip:07803034440@192.168.6.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK513a9530 From: ;tag=000c8533cc9900641bee4edb-65bcbf53 To: "07803034440" ;tag=as1eb408cc Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 Date: Mon, 07 Aug 2006 17:03:10 GMT CSeq: 101 ACK User-Agent: CSCO/7 Content-Length: 0 --- (9 headers 0 lines)--- foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: INVITE sip:7709@192.168.6.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK59432eb3 From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 Date: Mon, 07 Aug 2006 17:03:13 GMT CSeq: 101 INVITE User-Agent: CSCO/7 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 248 Accept: application/sdp Remote-Party-ID: "7708" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 23703 4934 IN IP4 192.168.6.252 s=SIP Call c=IN IP4 192.168.6.252 t=0 0 m=audio 18822 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (14 headers 11 lines)--- Sending to 192.168.6.252 : 5060 (no NAT) Using INVITE request as basis request - 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 Reliably Transmitting (no NAT) to 192.168.6.252:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK59432eb3;received=192.168.6.252 From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: ;tag=as052cbef2 Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a9ff2a5" Content-Length: 0 --- Scheduling destruction of SIP dialog '000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252' in 32000 ms (Method: INVITE) Found user '7708' foxtrot*CLI> <-- SIP read from 192.168.6.252:50765: ACK sip:7709@192.168.6.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK59432eb3 From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: ;tag=as052cbef2 Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 Date: Mon, 07 Aug 2006 17:03:13 GMT CSeq: 101 ACK Content-Length: 0 --- (8 headers 0 lines)--- foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: INVITE sip:7709@192.168.6.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK510bef14 From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 Date: Mon, 07 Aug 2006 17:03:13 GMT CSeq: 102 INVITE User-Agent: CSCO/7 Contact: Proxy-Authorization: Digest username="7708",realm="asterisk",uri="sip:192.168.6.6",response="8993822a77f5eee1f0688bf7b9dde735",nonce="7a9ff2a5",algorithm=MD5 Expires: 180 Content-Type: application/sdp Content-Length: 248 Remote-Party-ID: "7708" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 23703 4934 IN IP4 192.168.6.252 s=SIP Call c=IN IP4 192.168.6.252 t=0 0 m=audio 18822 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (14 headers 11 lines)--- Sending to 192.168.6.252 : 5060 (no NAT) Using INVITE request as basis request - 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 Found user '7708' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.6.252:18822 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), 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) Peer audio RTP is at port 192.168.6.252:18822 Looking for 7709 in from-sip (domain 192.168.6.6) list_route: hop: Transmitting (no NAT) to 192.168.6.252:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK510bef14;received=192.168.6.252 From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- -- Executing [7709@from-sip:1] Answer("SIP/7708-09195d30", "") in new stack Audio is at 192.168.6.6 port 18852 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.6.252:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK510bef14;received=192.168.6.252 From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: ;tag=as05dffdf2 Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 248 v=0 o=root 8073 8073 IN IP4 192.168.6.6 s=session c=IN IP4 192.168.6.6 t=0 0 m=audio 18852 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Executing [7709@from-sip:2] Dial("SIP/7708-09195d30", "Local/7707@from-sip/n") in new stack -- Called 7707@from-sip/n -- Executing [7707@from-sip:1] Answer("Local/7707@from-sip-bb9e,2", "") in new stack [Aug 7 18:03:13] DEBUG[8124]: channel.c:885 channel_find_locked: Avoiding initial deadlock for channel '0x919b540' -- Local/7707@from-sip-bb9e,1 answered SIP/7708-09195d30 -- Executing [7707@from-sip:2] Dial("Local/7707@from-sip-bb9e,2", "SIP/7707|120|g") in new stack Audio is at 192.168.6.6 port 15318 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.6.100:5060: INVITE sip:7707@192.168.6.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK3c37abe6;rport From: "Extension 7708" ;tag=as75a9859a To: Contact: Call-ID: 6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 Aug 2006 17:03:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 271 v=0 o=root 8073 8073 IN IP4 192.168.6.6 s=session c=IN IP4 192.168.6.6 t=0 0 m=audio 15318 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Called 7707 foxtrot*CLI> <-- SIP read from 192.168.6.100:5060: SIP/2.0 100 Trying To: From: "Extension 7708" ;tag=as75a9859a Call-ID: 6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK3c37abe6 Server: Sipura/SPA941-4.1.8 Content-Length: 0 --- (8 headers 0 lines)--- foxtrot*CLI> <-- SIP read from 192.168.6.100:5060: SIP/2.0 180 Ringing To: ;tag=aa0599d43d6b09di0 From: "Extension 7708" ;tag=as75a9859a Call-ID: 6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK3c37abe6 Server: Sipura/SPA941-4.1.8 Content-Length: 0 --- (8 headers 0 lines)--- -- SIP/7707-091a01c0 is ringing [Aug 7 18:03:13] DEBUG[8650]: channel.c:3521 ast_generic_bridge: Got a FRAME_CONTROL (3) frame on channel Local/7707@from-sip-bb9e,1 [Aug 7 18:03:13] DEBUG[8650]: channel.c:3793 ast_channel_bridge: Bridge stops bridging channels SIP/7708-09195d30 and Local/7707@from-sip-bb9e,1 [Aug 7 18:03:13] DEBUG[8650]: channel.c:2216 ast_indicate_data: Driver for channel 'SIP/7708-09195d30' does not support indication 3, emulating it [Aug 7 18:03:13] DEBUG[8650]: channel.c:1837 ast_settimeout: Scheduling timer at 160 sample intervals [Aug 7 18:03:14] DEBUG[8650]: channel.c:1837 ast_settimeout: Scheduling timer at 0 sample intervals foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: ACK sip:7709@192.168.6.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK6dbadb50 From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: ;tag=as05dffdf2 Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 Date: Mon, 07 Aug 2006 17:03:14 GMT CSeq: 102 ACK User-Agent: CSCO/7 Proxy-Authorization: Digest username="7708",realm="asterisk",uri="sip:192.168.6.6",response="8993822a77f5eee1f0688bf7b9dde735",nonce="7a9ff2a5",algorithm=MD5 Content-Length: 0 --- (10 headers 0 lines)--- [Aug 7 18:03:15] ERROR[8135]: rtp.c:2093 ast_rtcp_write_rr: RTCP RR transmission error to, rtcp halted Interrupted system call [Aug 7 18:03:15] NOTICE[8135]: sched.c:283 ast_sched_del: Attempted to delete nonexistent schedule entry 76! foxtrot*CLI> <-- SIP read from 192.168.6.100:5060: SIP/2.0 200 OK To: ;tag=aa0599d43d6b09di0 From: "Extension 7708" ;tag=as75a9859a Call-ID: 6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK3c37abe6 Contact: "7707 Test" Server: Sipura/SPA941-4.1.8 Content-Length: 212 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 44592610 44592610 IN IP4 192.168.6.100 s=- c=IN IP4 192.168.6.100 t=0 0 m=audio 16430 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (11 headers 11 lines)--- [Aug 7 18:03:21] DEBUG[8135]: chan_sip.c:1996 __sip_ack: Acked pending invite 102 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.6.100:16430 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.6.100:16430 list_route: hop: [Aug 7 18:03:21] DEBUG[8135]: chan_sip.c:5343 reqprep: Strict routing enforced for session 6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.6.100, port 5060 Transmitting (no NAT) to 192.168.6.100:5060: ACK sip:7707@192.168.6.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK1713cc36;rport From: "Extension 7708" ;tag=as75a9859a To: ;tag=aa0599d43d6b09di0 Contact: Call-ID: 6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/7707-091a01c0 answered Local/7707@from-sip-bb9e,2 [Aug 7 18:03:21] DEBUG[8650]: channel.c:3521 ast_generic_bridge: Got a FRAME_CONTROL (-1) frame on channel Local/7707@from-sip-bb9e,1 [Aug 7 18:03:21] DEBUG[8650]: channel.c:3793 ast_channel_bridge: Bridge stops bridging channels SIP/7708-09195d30 and Local/7707@from-sip-bb9e,1 [Aug 7 18:03:21] DEBUG[8650]: channel.c:1837 ast_settimeout: Scheduling timer at 0 sample intervals foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: INVITE sip:7709@192.168.6.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK6f40aa9f From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: ;tag=as05dffdf2 Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 Date: Mon, 07 Aug 2006 17:03:29 GMT CSeq: 103 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 243 Remote-Party-ID: "7708" ;party=calling;id-type=subscriber;privacy=off;screen=no Proxy-Authorization: Digest username="7708",realm="asterisk",uri="sip:192.168.6.6",response="8993822a77f5eee1f0688bf7b9dde735",nonce="7a9ff2a5",algorithm=MD5 v=0 o=Cisco-SIPUA 17404 10790 IN IP4 192.168.6.252 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 18822 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (13 headers 11 lines)--- Sending to 192.168.6.252 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:18822 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), 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) Peer audio RTP is at port 0.0.0.0:18822 Audio is at 192.168.6.6 port 18852 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.6.252:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK6f40aa9f;received=192.168.6.252 From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: ;tag=as05dffdf2 Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 248 v=0 o=root 8073 8074 IN IP4 192.168.6.6 s=session c=IN IP4 192.168.6.6 t=0 0 m=audio 18852 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Started music on hold, class 'default', on SIP/7707-091a01c0 [Aug 7 18:03:29] DEBUG[8654]: channel.c:1837 ast_settimeout: Scheduling timer at 160 sample intervals [Aug 7 18:03:29] DEBUG[8654]: channel.c:1837 ast_settimeout: Scheduling timer at 0 sample intervals foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: ACK sip:7709@192.168.6.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK136fcaf9 From: "7708" ;tag=000c8533cc99006547568d31-7f28d25c To: ;tag=as05dffdf2 Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 Date: Mon, 07 Aug 2006 17:03:29 GMT CSeq: 103 ACK User-Agent: CSCO/7 Proxy-Authorization: Digest username="7708",realm="asterisk",uri="sip:192.168.6.6",response="8993822a77f5eee1f0688bf7b9dde735",nonce="7a9ff2a5",algorithm=MD5 Content-Length: 0 --- (10 headers 0 lines)--- foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: REFER sip:07803034440@192.168.6.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK496a67c6 From: ;tag=000c8533cc9900641bee4edb-65bcbf53 To: "07803034440" ;tag=as1eb408cc Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 Date: Mon, 07 Aug 2006 17:03:29 GMT CSeq: 102 REFER User-Agent: CSCO/7 Contact: Content-Length: 0 Remote-Party-ID: "7708" ;party=calling;id-type=subscriber;privacy=off;screen=no Refer-To: Referred-By: --- (13 headers 0 lines)--- Call 127f2910376754bd1ff7c00578ea2198@192.168.6.6 got a SIP call transfer from callee: (REFER)! SIP transfer to extension 7709@from-sip by 7708@192.168.6.252:5060 Transmitting (no NAT) to 192.168.6.252:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK496a67c6;received=192.168.6.252 From: ;tag=000c8533cc9900641bee4edb-65bcbf53 To: "07803034440" ;tag=as1eb408cc Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 CSeq: 102 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- -- Stopped music on hold on Zap/1-1 [Aug 7 18:03:29] DEBUG[8135]: channel.c:1837 ast_settimeout: Scheduling timer at 0 sample intervals [Aug 7 18:03:29] DEBUG[8135]: channel.c:2963 ast_channel_masquerade: Planning to masquerade channel Zap/1-1 into the structure of SIP/7708-09195d30 [Aug 7 18:03:29] DEBUG[8135]: channel.c:2976 ast_channel_masquerade: Done planning to masquerade channel Zap/1-1 into the structure of SIP/7708-09195d30 [Aug 7 18:03:29] DEBUG[8135]: chan_sip.c:12213 attempt_transfer: SIP transfer: Succeeded to masquerade channels. [Aug 7 18:03:29] DEBUG[8135]: chan_sip.c:5343 reqprep: Strict routing enforced for session 127f2910376754bd1ff7c00578ea2198@192.168.6.6 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.6.252, port 5060 Reliably Transmitting (no NAT) to 192.168.6.252:5060: NOTIFY sip:7708@192.168.6.252:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK1a9fa568;rport From: "07803034440" ;tag=as1eb408cc To: ;tag=000c8533cc9900641bee4edb-65bcbf53 Contact: Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=102 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 --- [Aug 7 18:03:29] DEBUG[8135]: chan_sip.c:13138 local_attended_transfer: SIP attended transfer: Unlocking channel SIP/7708-09195d30 [Aug 7 18:03:29] DEBUG[8645]: chan_zap.c:4918 zt_indicate: Requested indication 17 on channel Zap/1-1 [Aug 7 18:03:29] DEBUG[8650]: chan_sip.c:5343 reqprep: Strict routing enforced for session 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.6.252, port 5060 Reliably Transmitting (no NAT) to 192.168.6.252:5060: BYE sip:7708@192.168.6.252:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK4018221e;rport From: ;tag=as05dffdf2 To: "7708" ;tag=000c8533cc99006547568d31-7f28d25c Contact: Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Aug 7 18:03:29] DEBUG[8650]: chan_zap.c:3354 zt_fixup: New owner for channel 1 is Zap/1-1 [Aug 7 18:03:29] DEBUG[8650]: chan_zap.c:1353 update_conf: Updated conferencing on 1, with 0 conference users [Aug 7 18:03:29] DEBUG[8650]: chan_zap.c:1353 update_conf: Updated conferencing on 1, with 0 conference users [Aug 7 18:03:29] DEBUG[8650]: channel.c:3313 ast_do_masquerade: Released clone lock on 'SIP/7708-09195d30' [Aug 7 18:03:29] DEBUG[8645]: channel.c:3499 ast_generic_bridge: Didn't get a frame from channel: SIP/7708-09195d30 [Aug 7 18:03:29] DEBUG[8645]: channel.c:3793 ast_channel_bridge: Bridge stops bridging channels SIP/7708-09195d30 and SIP/7708-09180220 Scheduling destruction of SIP dialog '127f2910376754bd1ff7c00578ea2198@192.168.6.6' in 32000 ms (Method: REFER) == Spawn extension (common, 7708, 2) exited non-zero on 'SIP/7708-09195d30' foxtrot*CLI> <-- SIP read from 192.168.6.252:50766: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK1a9fa568;rport From: "07803034440" ;tag=as1eb408cc To: ;tag=000c8533cc9900641bee4edb-65bcbf53 Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 Date: Mon, 07 Aug 2006 17:03:29 GMT CSeq: 103 NOTIFY Content-Length: 0 --- (8 headers 0 lines)--- Really destroying SIP dialog '127f2910376754bd1ff7c00578ea2198@192.168.6.6' Method: REFER foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK4018221e;rport From: ;tag=as05dffdf2 To: "7708" ;tag=000c8533cc99006547568d31-7f28d25c Call-ID: 000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252 Date: Mon, 07 Aug 2006 17:03:29 GMT CSeq: 102 BYE Server: CSCO/7 Content-Length: 0 RTP-RxStat: Dur=15,Pkt=756,Oct=120960,LatePkt=0,LostPkt=0,AvgJit=10 RTP-TxStat: Dur=15,Pkt=756,Oct=120960 --- (11 headers 0 lines)--- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '000c8533-cc990042-17dbcf35-1e0041ee@192.168.6.252' Method: ACK foxtrot*CLI> <-- SIP read from 192.168.6.252:50616: BYE sip:07803034440@192.168.6.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK581a38b0 From: ;tag=000c8533cc9900641bee4edb-65bcbf53 To: "07803034440" ;tag=as1eb408cc Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 Date: Mon, 07 Aug 2006 17:03:30 GMT CSeq: 103 BYE User-Agent: CSCO/7 Content-Length: 0 RTP-RxStat: Dur=25,Pkt=244,Oct=39040,LatePkt=0,LostPkt=0,AvgJit=10 RTP-TxStat: Dur=25,Pkt=241,Oct=38560 --- (11 headers 0 lines)--- Sending to 192.168.6.252 : 5060 (no NAT) Transmitting (no NAT) to 192.168.6.252:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.252:5060;branch=z9hG4bK581a38b0;received=192.168.6.252 From: ;tag=000c8533cc9900641bee4edb-65bcbf53 To: "07803034440" ;tag=as1eb408cc Call-ID: 127f2910376754bd1ff7c00578ea2198@192.168.6.6 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '127f2910376754bd1ff7c00578ea2198@192.168.6.6' Method: BYE -- Channel 0/1, span 1 got hangup request [Aug 7 18:03:34] DEBUG[8650]: channel.c:3499 ast_generic_bridge: Didn't get a frame from channel: Zap/1-1 [Aug 7 18:03:34] DEBUG[8650]: channel.c:3793 ast_channel_bridge: Bridge stops bridging channels Zap/1-1 and Local/7707@from-sip-bb9e,1 == Spawn extension (from-sip, 7709, 2) exited non-zero on 'Zap/1-1' -- Executing [h@from-sip:1] NoOp("Zap/1-1", "++++ HANGUP FROM-SIP ++++ [ANSWER/16]") in new stack [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1596 pbx_substitute_variables_helper_full: Expression result is '1' -- Executing [h@from-sip:2] GotoIf("Zap/1-1", "1?end") in new stack -- Goto (from-sip,h,6) -- Executing [h@from-sip:6] NoOp("Zap/1-1", "EndOfCall") in new stack [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '07803034440' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '07803034440' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'h' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'from-sip' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'SIP/7708-09195d30' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'Local/7707@from-sip-bb9e,1' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'NoOp' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'EndOfCall' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-07 18:03:13' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-07 18:03:13' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-07 18:03:34' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '21' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '21' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '1154970193.18' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' [Aug 7 18:03:34] DEBUG[8654]: channel.c:3499 ast_generic_bridge: Didn't get a frame from channel: Local/7707@from-sip-bb9e,2 [Aug 7 18:03:34] DEBUG[8654]: channel.c:3793 ast_channel_bridge: Bridge stops bridging channels Local/7707@from-sip-bb9e,2 and SIP/7707-091a01c0 -- Stopped music on hold on SIP/7707-091a01c0 [Aug 7 18:03:34] DEBUG[8654]: chan_sip.c:5343 reqprep: Strict routing enforced for session 6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.6.100, port 5060 Reliably Transmitting (no NAT) to 192.168.6.100:5060: BYE sip:7707@192.168.6.100:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK35206e31;rport From: "Extension 7708" ;tag=as75a9859a To: ;tag=aa0599d43d6b09di0 Contact: Call-ID: 6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (from-sip, 7707, 2) exited non-zero on 'Local/7707@from-sip-bb9e,2' -- Executing [h@from-sip:1] NoOp("Local/7707@from-sip-bb9e,2", "++++ HANGUP FROM-SIP ++++ [ANSWER/16]") in new stack [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1596 pbx_substitute_variables_helper_full: Expression result is '1' -- Executing [h@from-sip:2] GotoIf("Local/7707@from-sip-bb9e,2", "1?end") in new stack -- Goto (from-sip,h,6) -- Executing [h@from-sip:6] NoOp("Local/7707@from-sip-bb9e,2", "EndOfCall") in new stack [Aug 7 18:03:34] DEBUG[8124]: channel.c:885 channel_find_locked: Avoiding initial deadlock for channel '0x919b540' > cdr_odbc: Query Successful! [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '07803034440' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '07803034440' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'h' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'from-sip' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'Zap/1-1' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'SIP/7708-09180220' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'NoOp' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'EndOfCall' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-07 18:03:02' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-07 18:03:02' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-07 18:03:34' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '32' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '32' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '1154970182.16' [Aug 7 18:03:34] DEBUG[8650]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' [Aug 7 18:03:34] DEBUG[8124]: channel.c:885 channel_find_locked: Avoiding initial deadlock for channel '0x919b540' > cdr_odbc: Query Successful! [Aug 7 18:03:34] DEBUG[8650]: chan_zap.c:2886 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/1-1 [Aug 7 18:03:34] DEBUG[8650]: chan_zap.c:2375 zt_hangup: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1 [Aug 7 18:03:34] DEBUG[8650]: chan_zap.c:2529 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call [Aug 7 18:03:34] DEBUG[8650]: chan_zap.c:1416 zt_disable_ec: disabled echo cancellation on channel 1 [Aug 7 18:03:34] DEBUG[8650]: chan_zap.c:2803 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1 [Aug 7 18:03:34] DEBUG[8650]: chan_zap.c:1353 update_conf: Updated conferencing on 1, with 0 conference users [Aug 7 18:03:34] DEBUG[8650]: chan_zap.c:2882 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 [Aug 7 18:03:34] DEBUG[8650]: chan_zap.c:1416 zt_disable_ec: disabled echo cancellation on channel 1 -- Hungup 'Zap/1-1' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '"Extension 7708" <7707>' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '7707' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'h' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'from-sip' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'Local/7707@from-sip-bb9e,2' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'SIP/7707-091a01c0' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'NoOp' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'EndOfCall' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-07 18:03:13' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-07 18:03:13' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-07 18:03:34' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '21' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '21' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '1154970193.20' [Aug 7 18:03:34] DEBUG[8654]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' [Aug 7 18:03:34] DEBUG[8124]: channel.c:885 channel_find_locked: Avoiding initial deadlock for channel '0x919b540' > cdr_odbc: Query Successful! foxtrot*CLI> <-- SIP read from 192.168.6.100:5060: SIP/2.0 200 OK To: ;tag=aa0599d43d6b09di0 From: "Extension 7708" ;tag=as75a9859a Call-ID: 6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6 CSeq: 103 BYE Via: SIP/2.0/UDP 192.168.6.6:5060;branch=z9hG4bK35206e31 Server: Sipura/SPA941-4.1.8 Content-Length: 0 --- (8 headers 0 lines)--- Really destroying SIP dialog '6ca7cae233f0d5cb2fa2db6a0eebe215@192.168.6.6' Method: INVITE foxtrot*CLI>