[Jun 9 14:11:16] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.237:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.237;branch=z9hG4bKac1502339809 Max-Forwards: 70 From: ;tag=1c1502243045 To: Call-ID: 4743299462942008195958@192.168.161.237 CSeq: 78237 REGISTER Authorization: Digest username="104",realm="siprealm",nonce="488bb21b",uri="sip:pbx.example.net",algorithm=MD5,response="892a3dedb79019f00269e47cf3e9b9b7" Contact: ;expires=180 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 180 User-Agent: acMP114-01/v.5.20A.032.001 Content-Length: 0 <-------------> [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.237;branch=z9hG4bKac1502339809 (59) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 3: From: ;tag=1c1502243045 (48) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 4: To: (29) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 4743299462942008195958@192.168.161.237 (47) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 6: CSeq: 78237 REGISTER (20) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 7: Authorization: Digest username="104",realm="siprealm",nonce="488bb21b",uri="sip:pbx.example.net",algorithm=MD5,response="892a3dedb79019f00269e47cf3e9b9b7" (154) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 8: Contact: ;expires=180 (46) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 9: Supported: em,timer,replaces,path,resource-priority (51) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 10: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 11: Expires: 180 (12) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 12: User-Agent: acMP114-01/v.5.20A.032.001 (38) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:16] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: = Found Their Call ID: 4743299462942008195958@192.168.161.237 Their Tag 1c1502243045 Our tag: as0f2a9650 [Jun 9 14:11:16] DEBUG[23613] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jun 9 14:11:16] VERBOSE[23613] logger.c: Using latest REGISTER request as basis request [Jun 9 14:11:16] VERBOSE[23613] logger.c: Sending to 192.168.161.237 : 5060 (no NAT) [Jun 9 14:11:16] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.237:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.237;branch=z9hG4bKac1502339809;received=192.168.161.237 From: ;tag=1c1502243045 To: Call-ID: 4743299462942008195958@192.168.161.237 CSeq: 78237 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:16] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.237:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.237;branch=z9hG4bKac1502339809;received=192.168.161.237 From: ;tag=1c1502243045 To: ;tag=as0f2a9650 Call-ID: 4743299462942008195958@192.168.161.237 CSeq: 78237 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 180 Contact: ;expires=180 Date: Mon, 09 Jun 2008 12:11:16 GMT Content-Length: 0 <------------> [Jun 9 14:11:16] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/104 [Jun 9 14:11:16] VERBOSE[23613] logger.c: Scheduling destruction of SIP dialog '4743299462942008195958@192.168.161.237' in 32000 ms (Method: REGISTER) [Jun 9 14:11:16] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 104 [Jun 9 14:11:16] DEBUG[23576] chan_sip.c: Checking device state for peer 104 [Jun 9 14:11:16] DEBUG[23576] devicestate.c: Changing state for SIP/104 - state 1 (Not in use) [Jun 9 14:11:16] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 104 [Jun 9 14:11:16] DEBUG[23576] chan_sip.c: Checking device state for peer 104 [Jun 9 14:11:16] DEBUG[23615] app_queue.c: Device 'SIP/104' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 9 14:11:16] VERBOSE[25781] logger.c: == Manager 'lmt' logged off from 127.0.0.1 [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.216:5060 ---> INVITE sip:401@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-k9hf78sjb991;rport From: "Toestel 400" ;tag=ow35sk9e64 To: Call-ID: 3c617304b5ff-hqq001xrvvkf CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/7.1.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 224 v=0 o=root 17093307 17093307 IN IP4 192.168.161.216 s=call c=IN IP4 192.168.161.216 t=0 0 m=audio 18780 RTP/AVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: INVITE sip:401@pbx.example.net;user=phone SIP/2.0 (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-k9hf78sjb991;rport (71) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 400" ;tag=ow35sk9e64 (60) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: (40) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617304b5ff-hqq001xrvvkf (34) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: CSeq: 1 INVITE (14) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 8: P-Key-Flags: keys="3" (21) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 9: User-Agent: snom320/7.1.33 (26) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 10: Accept: application/sdp (23) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 12: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 13: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 14: Session-Expires: 3600;refresher=uas (35) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 15: Min-SE: 90 (10) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 16: Content-Type: application/sdp (29) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 17: Content-Length: 224 (19) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 18: (0) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: o=root 17093307 17093307 IN IP4 192.168.161.216 (47) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: s=call (6) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.216 (24) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: m=audio 18780 RTP/AVP 18 101 (28) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:20] VERBOSE[23613] logger.c: --- (18 headers 11 lines) --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743299462942008195958@192.168.161.237 Their Tag 1c1502243045 Our tag: as0f2a9650 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743318622942008195958@192.168.161.237 Their Tag 1c1502214188 Our tag: as2363565d [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743312262942008195958@192.168.161.237 Their Tag 1c1502198413 Our tag: as14e39483 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743305952942008195958@192.168.161.237 Their Tag 1c1502133902 Our tag: as3584dd5b [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ada48-mkwj7m38q3nu Their Tag wbtjjopkwe Our tag: as28583bdb [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701a4fb9-rd37qdoo1uyb Their Tag cyb8d7epkd Our tag: as2313fe09 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad395-ajo6vj1qc0vm Their Tag 97qngh5alh Our tag: as39395c26 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad4c6-phe3ytd26k13 Their Tag hf67o5uhr4 Our tag: as73260cbd [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad52a-gwy8p5tlpim3 Their Tag sjongh64mn Our tag: as56f28897 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad45e-7fxbtgnl1u8t Their Tag tstmut2dxd Our tag: as602e2d77 [Jun 9 14:11:20] DEBUG[23613] acl.c: ##### Testing 192.168.161.216 with 192.168.161.0 [Jun 9 14:11:20] DEBUG[23613] acl.c: ##### Testing 192.168.161.216 with 192.168.200.0 [Jun 9 14:11:20] DEBUG[23613] acl.c: ##### Testing 192.168.161.216 with 192.168.1.0 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Setting NAT on RTP to Off [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Setting NAT on VRTP to Off [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Setting NAT on UDPTL to Off [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for 3c617304b5ff-hqq001xrvvkf - INVITE (With RTP) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Found SIP option: -timer- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Matched SIP option: timer [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Found SIP option: -100rel- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Matched SIP option: 100rel [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Found SIP option: -replaces- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Matched SIP option: replaces [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Found SIP option: -from-change- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Found no match for SIP option: from-change (Please file bug report!) [Jun 9 14:11:20] VERBOSE[23613] logger.c: Sending to 192.168.161.216 : 5060 (NAT) [Jun 9 14:11:20] VERBOSE[23613] logger.c: Using INVITE request as basis request - 3c617304b5ff-hqq001xrvvkf [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Setting NAT on RTP to Off [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Setting NAT on VRTP to Off [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Setting NAT on UDPTL to Off [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.161.216:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-k9hf78sjb991;received=192.168.161.216;rport=5060 From: "Toestel 400" ;tag=ow35sk9e64 To: ;tag=as01710d2a Call-ID: 3c617304b5ff-hqq001xrvvkf CSeq: 1 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="538900ee" Content-Length: 0 <------------> [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:20] VERBOSE[23613] logger.c: Scheduling destruction of SIP dialog '3c617304b5ff-hqq001xrvvkf' in 32000 ms (Method: INVITE) [Jun 9 14:11:20] VERBOSE[23613] logger.c: Found user '400' [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.216:5060 ---> ACK sip:401@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-k9hf78sjb991;rport From: "Toestel 400" ;tag=ow35sk9e64 To: ;tag=as01710d2a Call-ID: 3c617304b5ff-hqq001xrvvkf CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: ACK sip:401@pbx.example.net;user=phone SIP/2.0 (46) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-k9hf78sjb991;rport (71) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 400" ;tag=ow35sk9e64 (60) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=as01710d2a (55) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617304b5ff-hqq001xrvvkf (34) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: CSeq: 1 ACK (11) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 8: Content-Length: 0 (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 9: (0) [Jun 9 14:11:20] VERBOSE[23613] logger.c: --- (9 headers 0 lines) --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c617304b5ff-hqq001xrvvkf Their Tag ow35sk9e64 Our tag: as01710d2a [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1327 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Stopping retransmission on '3c617304b5ff-hqq001xrvvkf' of Response 1: Match Found [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.216:5060 ---> INVITE sip:401@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-9nq0i54h11hq;rport From: "Toestel 400" ;tag=ow35sk9e64 To: Call-ID: 3c617304b5ff-hqq001xrvvkf CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/7.1.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="400",realm="siprealm",nonce="538900ee",uri="sip:401@pbx.example.net;user=phone",response="01509d04bc8d66fd0f72189f8e1c92a4",algorithm=MD5 Content-Type: application/sdp Content-Length: 224 v=0 o=root 17093307 17093307 IN IP4 192.168.161.216 s=call c=IN IP4 192.168.161.216 t=0 0 m=audio 18780 RTP/AVP 18 101 a=rtpmap:18 g729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: INVITE sip:401@pbx.example.net;user=phone SIP/2.0 (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-9nq0i54h11hq;rport (71) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 400" ;tag=ow35sk9e64 (60) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: (40) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617304b5ff-hqq001xrvvkf (34) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: CSeq: 2 INVITE (14) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 8: P-Key-Flags: keys="3" (21) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 9: User-Agent: snom320/7.1.33 (26) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 10: Accept: application/sdp (23) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 12: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 13: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 14: Session-Expires: 3600;refresher=uas (35) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 15: Min-SE: 90 (10) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 16: Proxy-Authorization: Digest username="400",realm="siprealm",nonce="538900ee",uri="sip:401@pbx.example.net;user=phone",response="01509d04bc8d66fd0f72189f8e1c92a4",algorithm=MD5 (175) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 17: Content-Type: application/sdp (29) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 18: Content-Length: 224 (19) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 19: (0) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: o=root 17093307 17093307 IN IP4 192.168.161.216 (47) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: s=call (6) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.216 (24) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: m=audio 18780 RTP/AVP 18 101 (28) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:20] VERBOSE[23613] logger.c: --- (19 headers 11 lines) --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c617304b5ff-hqq001xrvvkf Their Tag ow35sk9e64 Our tag: as01710d2a [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 9 14:11:20] VERBOSE[23613] logger.c: Sending to 192.168.161.216 : 5060 (NAT) [Jun 9 14:11:20] VERBOSE[23613] logger.c: Using INVITE request as basis request - 3c617304b5ff-hqq001xrvvkf [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Setting NAT on RTP to Off [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Setting NAT on VRTP to Off [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Setting NAT on UDPTL to Off [Jun 9 14:11:20] VERBOSE[23613] logger.c: Found user '400' [Jun 9 14:11:20] VERBOSE[23613] logger.c: Found RTP audio format 18 [Jun 9 14:11:20] VERBOSE[23613] logger.c: Found RTP audio format 101 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Peer doesn't provide T.38 UDPTL [Jun 9 14:11:20] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.216:18780 [Jun 9 14:11:20] VERBOSE[23613] logger.c: Found audio description format g729 for ID 18 [Jun 9 14:11:20] VERBOSE[23613] logger.c: Found audio description format telephone-event for ID 101 [Jun 9 14:11:20] VERBOSE[23613] logger.c: Got unsupported a:fmtp in SDP offer [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: T38 state changed to 0 on channel [Jun 9 14:11:20] VERBOSE[23613] logger.c: Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) [Jun 9 14:11:20] VERBOSE[23613] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 9 14:11:20] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.216:18780 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: We're settling with these formats: 0x100 (g729) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Checking SIP call limits for device 400 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Updating call counter for incoming call [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Call from peer '400' is 1 out of 30 [Jun 9 14:11:20] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/400 [Jun 9 14:11:20] VERBOSE[23613] logger.c: Looking for 401 in user-01 (domain pbx.example.net) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: *** Our native formats are 0x100 (g729) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: *** Joint capabilities are 0x100 (g729) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: *** Our capabilities are 0x100 (g729) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x100 (g729) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: This channel will not be able to handle video. [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: build_route: Contact hop: ;flow-id=1 [Jun 9 14:11:20] VERBOSE[23613] logger.c: list_route: hop: [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: SIP/400-08908960: New call is still down.... Trying... [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.216:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-9nq0i54h11hq;received=192.168.161.216;rport=5060 From: "Toestel 400" ;tag=ow35sk9e64 To: Call-ID: 3c617304b5ff-hqq001xrvvkf CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:20] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/400-08908960 [Jun 9 14:11:20] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/400 [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 400 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 400 [Jun 9 14:11:20] DEBUG[23576] devicestate.c: Changing state for SIP/400 - state 2 (In use) [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 400 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 400 [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 400 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 400 [Jun 9 14:11:20] VERBOSE[23576] logger.c: Reliably Transmitting (no NAT) to 192.168.161.250:5060: NOTIFY sip:401@192.168.161.250:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4845d44f;rport From: ;tag=as39395c26 To: ;tag=97qngh5alh Contact: Call-ID: 3c26701ad395-ajo6vj1qc0vm CSeq: 121 NOTIFY User-Agent: atCOM PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 206 confirmed --- [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:20] VERBOSE[23576] logger.c: Extension Changed 400[subscriptions] new state InUse for Notify User 401 [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 400-08908960 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 400-08908960 [Jun 9 14:11:20] DEBUG[25839] pbx.c: Launching 'Dial' [Jun 9 14:11:20] VERBOSE[25839] logger.c: -- Executing [401@user-01:1] Dial("SIP/400-08908960", "SIP/401") in new stack [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Asked to create a SIP channel with formats: 0x100 (g729) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Setting NAT on RTP to Off [Jun 9 14:11:20] DEBUG[25839] acl.c: ##### Testing 192.168.161.250 with 192.168.161.0 [Jun 9 14:11:20] DEBUG[25839] acl.c: ##### Testing 192.168.161.250 with 192.168.200.0 [Jun 9 14:11:20] DEBUG[25839] acl.c: ##### Testing 192.168.161.250 with 192.168.1.0 [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: *** Our capabilities are 0x8 (alaw) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: *** Our preferred formats from the incoming channel are 0x100 (g729) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: This channel will not be able to handle video. [Jun 9 14:11:20] DEBUG[25839] channel.c: Not copying variable SIPCALLID. [Jun 9 14:11:20] DEBUG[25839] channel.c: Not copying variable SIPUSERAGENT. [Jun 9 14:11:20] DEBUG[25839] channel.c: Not copying variable SIPDOMAIN. [Jun 9 14:11:20] DEBUG[25839] channel.c: Not copying variable SIPURI. [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Outgoing Call for 401 [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Updating call counter for outgoing call [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Call to peer '401' is 1 out of 30 [Jun 9 14:11:20] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Our T38 capability (3856), joint T38 capability (3856) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: False [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: ** Our prefcodec: 0x100 (g729) [Jun 9 14:11:20] VERBOSE[25839] logger.c: Audio is at 192.168.161.100 port 19240 [Jun 9 14:11:20] VERBOSE[25839] logger.c: Adding codec 0x8 (alaw) to SDP [Jun 9 14:11:20] VERBOSE[25839] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: -- Done with adding codecs to SDP [Jun 9 14:11:20] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=30) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 0: INVITE sip:401@192.168.161.250:5060 SIP/2.0 (43) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK544d9e60;rport (66) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 2: From: "Toestel 400" ;tag=as7bac7baf (60) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 4: Contact: (34) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 5: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net (57) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 9: Remote-Party-ID: "Toestel 400" ;privacy=off;screen=no (78) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 10: Date: Mon, 09 Jun 2008 12:11:20 GMT (35) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 12: Supported: replaces (19) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 14: Content-Length: 246 (19) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Header 15: (0) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: o=root 23546 23546 IN IP4 192.168.161.100 (41) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: s=session (9) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: c=IN IP4 192.168.161.100 (24) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: m=audio 19240 RTP/AVP 8 101 (27) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:20] VERBOSE[25839] logger.c: Reliably Transmitting (no NAT) to 192.168.161.250:5060: INVITE sip:401@192.168.161.250:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK544d9e60;rport From: "Toestel 400" ;tag=as7bac7baf To: Contact: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 102 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Toestel 400" ;privacy=off;screen=no Date: Mon, 09 Jun 2008 12:11:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 246 v=0 o=root 23546 23546 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 19240 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 --- [Jun 9 14:11:20] DEBUG[25839] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:20] VERBOSE[25839] logger.c: -- Called 401 [Jun 9 14:11:20] DEBUG[25839] channel.c: Set channel SIP/401-089415f0 to read format slin [Jun 9 14:11:20] DEBUG[25839] channel.c: Set channel SIP/400-08908960 to write format slin [Jun 9 14:11:20] DEBUG[25839] channel.c: Set channel SIP/400-08908960 to read format slin [Jun 9 14:11:20] DEBUG[25839] channel.c: Set channel SIP/401-089415f0 to write format slin [Jun 9 14:11:20] DEBUG[23615] app_queue.c: Device 'SIP/400' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:20] DEBUG[23576] devicestate.c: Changing state for SIP/400-08908960 - state 4 (Invalid) [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 400 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 400 [Jun 9 14:11:20] DEBUG[23576] devicestate.c: Changing state for SIP/400 - state 2 (In use) [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 400 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 400 [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:20] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 6 (Ringing) [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:20] DEBUG[23615] app_queue.c: Device 'SIP/400-08908960' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:20] DEBUG[23615] app_queue.c: Device 'SIP/400' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:20] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4845d44f;rport=5060 From: ;tag=as39395c26 To: ;tag=97qngh5alh Call-ID: 3c26701ad395-ajo6vj1qc0vm CSeq: 121 NOTIFY Content-Length: 0 <-------------> [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 Ok (14) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4845d44f;rport=5060 (71) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: ;tag=as39395c26 (57) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=97qngh5alh (44) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c26701ad395-ajo6vj1qc0vm (34) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: CSeq: 121 NOTIFY (16) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: Content-Length: 0 (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: (0) [Jun 9 14:11:20] VERBOSE[23613] logger.c: --- (7 headers 0 lines) --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag Our tag: as7bac7baf [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c617304b5ff-hqq001xrvvkf Their Tag ow35sk9e64 Our tag: as03ae9445 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743299462942008195958@192.168.161.237 Their Tag 1c1502243045 Our tag: as0f2a9650 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743318622942008195958@192.168.161.237 Their Tag 1c1502214188 Our tag: as2363565d [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743312262942008195958@192.168.161.237 Their Tag 1c1502198413 Our tag: as14e39483 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743305952942008195958@192.168.161.237 Their Tag 1c1502133902 Our tag: as3584dd5b [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ada48-mkwj7m38q3nu Their Tag wbtjjopkwe Our tag: as28583bdb [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701a4fb9-rd37qdoo1uyb Their Tag cyb8d7epkd Our tag: as2313fe09 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c26701ad395-ajo6vj1qc0vm Their Tag 97qngh5alh Our tag: as39395c26 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Acked pending invite 121 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1329 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Stopping retransmission on '3c26701ad395-ajo6vj1qc0vm' of Request 121: Match Found [Jun 9 14:11:20] VERBOSE[23613] logger.c: SIP Response message for INCOMING dialog NOTIFY arrived [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK544d9e60;rport=5060 From: "Toestel 400" ;tag=as7bac7baf To: ;tag=j41casm0gq Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK544d9e60;rport=5060 (71) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 400" ;tag=as7bac7baf (60) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=j41casm0gq (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net (57) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 8: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 9: Content-Length: 0 (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 10: (0) [Jun 9 14:11:20] VERBOSE[23613] logger.c: --- (10 headers 0 lines) --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = Found Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag Our tag: as7bac7baf [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1330 - INVITE (got response) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '28aa6e4d058373f806130cd33cee1f63@pbx.example.net' Request 102: Found [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: SIP response 180 to standard invite [Jun 9 14:11:20] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/401-089415f0 [Jun 9 14:11:20] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:20] VERBOSE[25839] logger.c: -- SIP/401-089415f0 is ringing [Jun 9 14:11:20] VERBOSE[25839] logger.c: <--- Transmitting (no NAT) to 192.168.161.216:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-9nq0i54h11hq;received=192.168.161.216;rport=5060 From: "Toestel 400" ;tag=ow35sk9e64 To: ;tag=as03ae9445 Call-ID: 3c617304b5ff-hqq001xrvvkf CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401-089415f0 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 401-089415f0 [Jun 9 14:11:20] DEBUG[23576] devicestate.c: Changing state for SIP/401-089415f0 - state 4 (Invalid) [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:20] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 6 (Ringing) [Jun 9 14:11:20] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:20] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:20] DEBUG[23615] app_queue.c: Device 'SIP/401-089415f0' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:20] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:20] DEBUG[23613] acl.c: ##### Testing 192.168.161.242 with 192.168.161.0 [Jun 9 14:11:20] DEBUG[23613] acl.c: ##### Testing 192.168.161.242 with 192.168.200.0 [Jun 9 14:11:20] DEBUG[23613] acl.c: ##### Testing 192.168.161.242 with 192.168.1.0 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:422@192.168.161.242:5060 SIP/2.0 (44) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1f42c092;rport (66) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as39849072 (62) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 1b0770c6539552550b978bbd0f1400c7@pbx.example.net (57) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:20 GMT (35) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:20] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.242:5060: OPTIONS sip:422@192.168.161.242:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1f42c092;rport From: "asterisk" ;tag=as39849072 To: Contact: Call-ID: 1b0770c6539552550b978bbd0f1400c7@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK544d9e60;rport=5060 From: "Toestel 400" ;tag=as7bac7baf To: ;tag=j41casm0gq Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK544d9e60;rport=5060 (71) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 400" ;tag=as7bac7baf (60) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=j41casm0gq (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net (57) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 8: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 9: Content-Length: 0 (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 10: (0) [Jun 9 14:11:20] VERBOSE[23613] logger.c: --- (10 headers 0 lines) --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = No match Their Call ID: 1b0770c6539552550b978bbd0f1400c7@pbx.example.net Their Tag Our tag: as39849072 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = Found Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '28aa6e4d058373f806130cd33cee1f63@pbx.example.net' Request 102: Found [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: SIP response 180 to standard invite [Jun 9 14:11:20] VERBOSE[25839] logger.c: -- SIP/401-089415f0 is ringing [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.242:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1f42c092;rport=5060 From: "asterisk" ;tag=as39849072 To: Call-ID: 1b0770c6539552550b978bbd0f1400c7@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom370/7.1.33 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1f42c092;rport=5060 (71) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as39849072 (62) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 1b0770c6539552550b978bbd0f1400c7@pbx.example.net (57) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom370/7.1.33 (26) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 8: Accept-Language: en (19) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 9: Accept: application/sdp (23) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 11: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 12: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:20] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = Found Their Call ID: 1b0770c6539552550b978bbd0f1400c7@pbx.example.net Their Tag Our tag: as39849072 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1332 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Stopping retransmission on '1b0770c6539552550b978bbd0f1400c7@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:20] VERBOSE[23613] logger.c: Really destroying SIP dialog '1b0770c6539552550b978bbd0f1400c7@pbx.example.net' Method: OPTIONS [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:20] DEBUG[23613] acl.c: ##### Testing 192.168.161.233 with 192.168.161.0 [Jun 9 14:11:20] DEBUG[23613] acl.c: ##### Testing 192.168.161.233 with 192.168.200.0 [Jun 9 14:11:20] DEBUG[23613] acl.c: ##### Testing 192.168.161.233 with 192.168.1.0 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:421@192.168.161.233:5060 SIP/2.0 (44) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK036fdb3d;rport (66) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as2f7555ec (62) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 61f72916177c067125ec2c130faa8843@pbx.example.net (57) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:20 GMT (35) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:20] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.233:5060: OPTIONS sip:421@192.168.161.233:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK036fdb3d;rport From: "asterisk" ;tag=as2f7555ec To: Contact: Call-ID: 61f72916177c067125ec2c130faa8843@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:20] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.233:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK036fdb3d;rport=5060 From: "asterisk" ;tag=as2f7555ec To: Call-ID: 61f72916177c067125ec2c130faa8843@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom360/7.1.33 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK036fdb3d;rport=5060 (71) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as2f7555ec (62) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 61f72916177c067125ec2c130faa8843@pbx.example.net (57) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 8: Accept-Language: en (19) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 9: Accept: application/sdp (23) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 11: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 12: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:20] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: = Found Their Call ID: 61f72916177c067125ec2c130faa8843@pbx.example.net Their Tag Our tag: as2f7555ec [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1335 [Jun 9 14:11:20] DEBUG[23613] chan_sip.c: Stopping retransmission on '61f72916177c067125ec2c130faa8843@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:20] VERBOSE[23613] logger.c: Really destroying SIP dialog '61f72916177c067125ec2c130faa8843@pbx.example.net' Method: OPTIONS [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.215 with 192.168.161.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.215 with 192.168.200.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.215 with 192.168.1.0 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:408@192.168.161.215:5060 SIP/2.0 (44) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK72275b6c;rport (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as411cde17 (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 40b5c148026c9ecd03ddabf73917fad5@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:21 GMT (35) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:21] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.215:5060: OPTIONS sip:408@192.168.161.215:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK72275b6c;rport From: "asterisk" ;tag=as411cde17 To: Contact: Call-ID: 40b5c148026c9ecd03ddabf73917fad5@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.215:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK72275b6c;rport=5060 From: "asterisk" ;tag=as411cde17 To: Call-ID: 40b5c148026c9ecd03ddabf73917fad5@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom300/7.1.33 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK72275b6c;rport=5060 (71) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as411cde17 (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 40b5c148026c9ecd03ddabf73917fad5@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom300/7.1.33 (26) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Accept-Language: en (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Accept: application/sdp (23) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:21] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = Found Their Call ID: 40b5c148026c9ecd03ddabf73917fad5@pbx.example.net Their Tag Our tag: as411cde17 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1338 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Stopping retransmission on '40b5c148026c9ecd03ddabf73917fad5@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:21] VERBOSE[23613] logger.c: Really destroying SIP dialog '40b5c148026c9ecd03ddabf73917fad5@pbx.example.net' Method: OPTIONS [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.243 with 192.168.161.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.243 with 192.168.200.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.243 with 192.168.1.0 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:406@192.168.161.243:5060 SIP/2.0 (44) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3076d5bd;rport (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as17d3ad56 (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 66b155a0163078096c42e9ff70f1dcc4@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:21 GMT (35) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:21] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.243:5060: OPTIONS sip:406@192.168.161.243:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3076d5bd;rport From: "asterisk" ;tag=as17d3ad56 To: Contact: Call-ID: 66b155a0163078096c42e9ff70f1dcc4@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.243:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3076d5bd;rport=5060 From: "asterisk" ;tag=as17d3ad56 To: Call-ID: 66b155a0163078096c42e9ff70f1dcc4@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/7.1.33 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK3076d5bd;rport=5060 (71) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as17d3ad56 (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 66b155a0163078096c42e9ff70f1dcc4@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom320/7.1.33 (26) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Accept-Language: en (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Accept: application/sdp (23) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:21] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = Found Their Call ID: 66b155a0163078096c42e9ff70f1dcc4@pbx.example.net Their Tag Our tag: as17d3ad56 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1341 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Stopping retransmission on '66b155a0163078096c42e9ff70f1dcc4@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:21] VERBOSE[23613] logger.c: Really destroying SIP dialog '66b155a0163078096c42e9ff70f1dcc4@pbx.example.net' Method: OPTIONS [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.247 with 192.168.161.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.247 with 192.168.200.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.247 with 192.168.1.0 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:404@192.168.161.247:5060 SIP/2.0 (44) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6bc87d9b;rport (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as336666e7 (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 2b05b7cf27a9078b164e2d87309e1af6@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:21 GMT (35) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:21] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.247:5060: OPTIONS sip:404@192.168.161.247:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6bc87d9b;rport From: "asterisk" ;tag=as336666e7 To: Contact: Call-ID: 2b05b7cf27a9078b164e2d87309e1af6@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6bc87d9b;rport=5060 From: "asterisk" ;tag=as336666e7 To: Call-ID: 2b05b7cf27a9078b164e2d87309e1af6@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom360/7.1.33 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6bc87d9b;rport=5060 (71) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as336666e7 (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 2b05b7cf27a9078b164e2d87309e1af6@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Accept-Language: en (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Accept: application/sdp (23) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:21] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = Found Their Call ID: 2b05b7cf27a9078b164e2d87309e1af6@pbx.example.net Their Tag Our tag: as336666e7 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1344 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Stopping retransmission on '2b05b7cf27a9078b164e2d87309e1af6@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:21] VERBOSE[23613] logger.c: Really destroying SIP dialog '2b05b7cf27a9078b164e2d87309e1af6@pbx.example.net' Method: OPTIONS [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.248 with 192.168.161.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.248 with 192.168.200.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.248 with 192.168.1.0 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:403@192.168.161.248:5060 SIP/2.0 (44) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK779ee854;rport (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as14efc70b (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 7f64ffc72c13885b355e23b26ee2b425@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:21 GMT (35) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:21] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.248:5060: OPTIONS sip:403@192.168.161.248:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK779ee854;rport From: "asterisk" ;tag=as14efc70b To: Contact: Call-ID: 7f64ffc72c13885b355e23b26ee2b425@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.248:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK779ee854;rport=5060 From: "asterisk" ;tag=as14efc70b To: Call-ID: 7f64ffc72c13885b355e23b26ee2b425@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/7.1.33 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK779ee854;rport=5060 (71) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as14efc70b (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 7f64ffc72c13885b355e23b26ee2b425@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom320/7.1.33 (26) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Accept-Language: en (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Accept: application/sdp (23) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:21] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = Found Their Call ID: 7f64ffc72c13885b355e23b26ee2b425@pbx.example.net Their Tag Our tag: as14efc70b [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1347 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Stopping retransmission on '7f64ffc72c13885b355e23b26ee2b425@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:21] VERBOSE[23613] logger.c: Really destroying SIP dialog '7f64ffc72c13885b355e23b26ee2b425@pbx.example.net' Method: OPTIONS [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK544d9e60;rport=5060 From: "Toestel 400" ;tag=as7bac7baf To: ;tag=j41casm0gq Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom360/7.1.33 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 226 v=0 o=root 1824544477 1824544478 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 11842 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 Ok (14) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK544d9e60;rport=5060 (71) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 400" ;tag=as7bac7baf (60) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=j41casm0gq (49) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Content-Type: application/sdp (29) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 226 (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 13: (0) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: o=root 1824544477 1824544478 IN IP4 192.168.161.250 (51) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: s=call (6) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.250 (24) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: m=audio 11842 RTP/AVP 8 101 (27) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:21] VERBOSE[23613] logger.c: --- (13 headers 11 lines) --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = Found Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Acked pending invite 102 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Stopping retransmission on '28aa6e4d058373f806130cd33cee1f63@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: SIP response 200 to standard invite [Jun 9 14:11:21] VERBOSE[23613] logger.c: Found RTP audio format 8 [Jun 9 14:11:21] VERBOSE[23613] logger.c: Found RTP audio format 101 [Jun 9 14:11:21] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.250:11842 [Jun 9 14:11:21] VERBOSE[23613] logger.c: Found audio description format pcma for ID 8 [Jun 9 14:11:21] VERBOSE[23613] logger.c: Found audio description format telephone-event for ID 101 [Jun 9 14:11:21] VERBOSE[23613] logger.c: Got unsupported a:fmtp in SDP offer [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: T38 state changed to 0 on channel SIP/401-089415f0 [Jun 9 14:11:21] VERBOSE[23613] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jun 9 14:11:21] VERBOSE[23613] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 9 14:11:21] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.250:11842 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: We have an owner, now see if we need to change this call [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Updating call counter for outgoing call [Jun 9 14:11:21] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: build_route: Contact hop: ;flow-id=1 [Jun 9 14:11:21] VERBOSE[23613] logger.c: list_route: hop: [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Strict routing enforced for session 28aa6e4d058373f806130cd33cee1f63@pbx.example.net [Jun 9 14:11:21] VERBOSE[23613] logger.c: set_destination: Parsing for address/port to send to [Jun 9 14:11:21] VERBOSE[23613] logger.c: set_destination: set destination to 192.168.161.250, port 5060 [Jun 9 14:11:21] VERBOSE[23613] logger.c: Transmitting (no NAT) to 192.168.161.250:5060: ACK sip:401@192.168.161.250:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK4579e819;rport From: "Toestel 400" ;tag=as7bac7baf To: ;tag=j41casm0gq Contact: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 102 ACK User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Toestel 400" ;privacy=off;screen=no Content-Length: 0 --- [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:21] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 2 (In use) [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:21] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/401-089415f0 [Jun 9 14:11:21] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:21] VERBOSE[25839] logger.c: -- SIP/401-089415f0 answered SIP/400-08908960 [Jun 9 14:11:21] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/400-08908960 [Jun 9 14:11:21] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/400 [Jun 9 14:11:21] DEBUG[25839] chan_sip.c: SIP answering channel: SIP/400-08908960 [Jun 9 14:11:21] DEBUG[25839] chan_sip.c: Setting framing from config on incoming call [Jun 9 14:11:21] DEBUG[25839] chan_sip.c: ** Our capability: 0x100 (g729) Video flag: True [Jun 9 14:11:21] DEBUG[25839] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 9 14:11:21] VERBOSE[25839] logger.c: Audio is at 192.168.161.100 port 11752 [Jun 9 14:11:21] VERBOSE[25839] logger.c: Adding codec 0x100 (g729) to SDP [Jun 9 14:11:21] VERBOSE[25839] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 9 14:11:21] DEBUG[25839] chan_sip.c: -- Done with adding codecs to SDP [Jun 9 14:11:21] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:21] DEBUG[25839] chan_sip.c: Done building SDP. Settling with this capability: 0x100 (g729) [Jun 9 14:11:21] VERBOSE[25839] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.161.216:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-9nq0i54h11hq;received=192.168.161.216;rport=5060 From: "Toestel 400" ;tag=ow35sk9e64 To: ;tag=as03ae9445 Call-ID: 3c617304b5ff-hqq001xrvvkf CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 269 v=0 o=root 23546 23546 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 11752 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jun 9 14:11:21] DEBUG[25839] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:21] DEBUG[25839] rtp.c: Channel codec0 = 256 is not codec1 = 8, cannot native bridge in RTP. [Jun 9 14:11:21] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401-089415f0 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer 401-089415f0 [Jun 9 14:11:21] DEBUG[23576] devicestate.c: Changing state for SIP/401-089415f0 - state 4 (Invalid) [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:21] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 2 (In use) [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 400-08908960 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer 400-08908960 [Jun 9 14:11:21] DEBUG[23615] app_queue.c: Device 'SIP/401-089415f0' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:21] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:21] DEBUG[23576] devicestate.c: Changing state for SIP/400-08908960 - state 4 (Invalid) [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 400 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer 400 [Jun 9 14:11:21] DEBUG[23576] devicestate.c: Changing state for SIP/400 - state 2 (In use) [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 400 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer 400 [Jun 9 14:11:21] DEBUG[23615] app_queue.c: Device 'SIP/400-08908960' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:21] DEBUG[23615] app_queue.c: Device 'SIP/400' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.216:5060 ---> ACK sip:401@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-t1r8k3stl70d;rport From: "Toestel 400" ;tag=ow35sk9e64 To: ;tag=as03ae9445 Call-ID: 3c617304b5ff-hqq001xrvvkf CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: ACK sip:401@192.168.161.100 SIP/2.0 (35) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.216:5060;branch=z9hG4bK-t1r8k3stl70d;rport (71) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 400" ;tag=ow35sk9e64 (60) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=as03ae9445 (55) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617304b5ff-hqq001xrvvkf (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: CSeq: 2 ACK (11) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Content-Length: 0 (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: (0) [Jun 9 14:11:21] VERBOSE[23613] logger.c: --- (9 headers 0 lines) --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c617304b5ff-hqq001xrvvkf Their Tag ow35sk9e64 Our tag: as03ae9445 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1350 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Stopping retransmission on '3c617304b5ff-hqq001xrvvkf' of Response 2: Match Found [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.249 with 192.168.161.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.249 with 192.168.200.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.249 with 192.168.1.0 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:402@192.168.161.249:5060 SIP/2.0 (44) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2679ce37;rport (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as2be002fd (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 47b07a6306f26f8641429c792339c590@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:21 GMT (35) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:21] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.249:5060: OPTIONS sip:402@192.168.161.249:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2679ce37;rport From: "asterisk" ;tag=as2be002fd To: Contact: Call-ID: 47b07a6306f26f8641429c792339c590@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2679ce37;rport=5060 From: "asterisk" ;tag=as2be002fd To: Call-ID: 47b07a6306f26f8641429c792339c590@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom360/7.1.33 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2679ce37;rport=5060 (71) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as2be002fd (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 47b07a6306f26f8641429c792339c590@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Accept-Language: en (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Accept: application/sdp (23) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:21] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = Found Their Call ID: 47b07a6306f26f8641429c792339c590@pbx.example.net Their Tag Our tag: as2be002fd [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1351 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Stopping retransmission on '47b07a6306f26f8641429c792339c590@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:21] VERBOSE[23613] logger.c: Really destroying SIP dialog '47b07a6306f26f8641429c792339c590@pbx.example.net' Method: OPTIONS [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.4:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.4;branch=z9hG4bKac923713048 Max-Forwards: 70 From: ;tag=1c923705775 To: Call-ID: 868521015562008142045@192.168.161.3 CSeq: 7677 REGISTER Contact: ;expires=180 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 180 User-Agent: acM1000-01/v.5.20A.032.001 Content-Length: 0 <-------------> [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.4;branch=z9hG4bKac923713048 (56) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: From: ;tag=1c923705775 (54) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: To: (36) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 868521015562008142045@192.168.161.3 (44) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: CSeq: 7677 REGISTER (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: Contact: ;expires=180 (51) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Supported: em,timer,replaces,path,resource-priority (51) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Expires: 180 (12) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: User-Agent: acM1000-01/v.5.20A.032.001 (38) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 13: (0) [Jun 9 14:11:21] VERBOSE[23613] logger.c: --- (13 headers 0 lines) --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c617304b5ff-hqq001xrvvkf Their Tag ow35sk9e64 Our tag: as03ae9445 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743299462942008195958@192.168.161.237 Their Tag 1c1502243045 Our tag: as0f2a9650 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743318622942008195958@192.168.161.237 Their Tag 1c1502214188 Our tag: as2363565d [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743312262942008195958@192.168.161.237 Their Tag 1c1502198413 Our tag: as14e39483 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743305952942008195958@192.168.161.237 Their Tag 1c1502133902 Our tag: as3584dd5b [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ada48-mkwj7m38q3nu Their Tag wbtjjopkwe Our tag: as28583bdb [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701a4fb9-rd37qdoo1uyb Their Tag cyb8d7epkd Our tag: as2313fe09 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad395-ajo6vj1qc0vm Their Tag 97qngh5alh Our tag: as39395c26 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad4c6-phe3ytd26k13 Their Tag hf67o5uhr4 Our tag: as73260cbd [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad52a-gwy8p5tlpim3 Their Tag sjongh64mn Our tag: as56f28897 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad45e-7fxbtgnl1u8t Their Tag tstmut2dxd Our tag: as602e2d77 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.4 with 192.168.161.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.4 with 192.168.200.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.4 with 192.168.1.0 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for 868521015562008142045@192.168.161.3 - REGISTER (No RTP) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jun 9 14:11:21] VERBOSE[23613] logger.c: Using latest REGISTER request as basis request [Jun 9 14:11:21] VERBOSE[23613] logger.c: Sending to 192.168.161.4 : 5060 (no NAT) [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.4:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.4;branch=z9hG4bKac923713048;received=192.168.161.4 From: ;tag=1c923705775 To: Call-ID: 868521015562008142045@192.168.161.3 CSeq: 7677 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.4:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.4;branch=z9hG4bKac923713048;received=192.168.161.4 From: ;tag=1c923705775 To: ;tag=as704c8058 Call-ID: 868521015562008142045@192.168.161.3 CSeq: 7677 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="14da9d1e" Content-Length: 0 <------------> [Jun 9 14:11:21] VERBOSE[23613] logger.c: Scheduling destruction of SIP dialog '868521015562008142045@192.168.161.3' in 32000 ms (Method: REGISTER) [Jun 9 14:11:21] DEBUG[25839] rtp.c: Ooh, format changed from unknown to g729 [Jun 9 14:11:21] DEBUG[25839] rtp.c: Created smoother: format: 256 ms: 20 len: 20 [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.4:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.4;branch=z9hG4bKac924144045 Max-Forwards: 70 From: ;tag=1c923705775 To: Call-ID: 868521015562008142045@192.168.161.3 CSeq: 7678 REGISTER Authorization: Digest username="acM1000-01",realm="siprealm",nonce="14da9d1e",uri="sip:pbx.example.net",algorithm=MD5,response="6ab4e9ae5b95a3054171d8bc577c2217" Contact: ;expires=180 Supported: em,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 180 User-Agent: acM1000-01/v.5.20A.032.001 Content-Length: 0 <-------------> [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.4;branch=z9hG4bKac924144045 (56) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: From: ;tag=1c923705775 (54) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: To: (36) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 868521015562008142045@192.168.161.3 (44) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: CSeq: 7678 REGISTER (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: Authorization: Digest username="acM1000-01",realm="siprealm",nonce="14da9d1e",uri="sip:pbx.example.net",algorithm=MD5,response="6ab4e9ae5b95a3054171d8bc577c2217" (161) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Contact: ;expires=180 (51) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Supported: em,timer,replaces,path,resource-priority (51) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Expires: 180 (12) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: User-Agent: acM1000-01/v.5.20A.032.001 (38) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:21] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: = Found Their Call ID: 868521015562008142045@192.168.161.3 Their Tag 1c923705775 Our tag: as704c8058 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jun 9 14:11:21] VERBOSE[23613] logger.c: Using latest REGISTER request as basis request [Jun 9 14:11:21] VERBOSE[23613] logger.c: Sending to 192.168.161.4 : 5060 (no NAT) [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.4:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.4;branch=z9hG4bKac924144045;received=192.168.161.4 From: ;tag=1c923705775 To: Call-ID: 868521015562008142045@192.168.161.3 CSeq: 7678 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:21] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.4;branch=z9hG4bKac924144045;received=192.168.161.4 From: ;tag=1c923705775 To: ;tag=as704c8058 Call-ID: 868521015562008142045@192.168.161.3 CSeq: 7678 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 180 Contact: ;expires=180 Date: Mon, 09 Jun 2008 12:11:21 GMT Content-Length: 0 <------------> [Jun 9 14:11:21] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/acM1000-01 [Jun 9 14:11:21] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/acM1000 [Jun 9 14:11:21] VERBOSE[23613] logger.c: Scheduling destruction of SIP dialog '868521015562008142045@192.168.161.3' in 32000 ms (Method: REGISTER) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.250 with 192.168.161.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.250 with 192.168.200.0 [Jun 9 14:11:21] DEBUG[23613] acl.c: ##### Testing 192.168.161.250 with 192.168.1.0 [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:401@192.168.161.250:5060 SIP/2.0 (44) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1135deaf;rport (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as3cc23cac (62) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 1640a08518ad3f521c8aeb4a55a2e61f@pbx.example.net (57) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:21 GMT (35) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:21] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.250:5060: OPTIONS sip:401@192.168.161.250:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1135deaf;rport From: "asterisk" ;tag=as3cc23cac To: Contact: Call-ID: 1640a08518ad3f521c8aeb4a55a2e61f@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:21] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - acM1000-01 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer acM1000-01 [Jun 9 14:11:21] DEBUG[23576] devicestate.c: Changing state for SIP/acM1000-01 - state 1 (Not in use) [Jun 9 14:11:21] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - acM1000 [Jun 9 14:11:21] DEBUG[23576] chan_sip.c: Checking device state for peer acM1000 [Jun 9 14:11:21] DEBUG[23615] app_queue.c: Device 'SIP/acM1000-01' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 9 14:11:21] DEBUG[23576] devicestate.c: Changing state for SIP/acM1000 - state 4 (Invalid) [Jun 9 14:11:21] DEBUG[23615] app_queue.c: Device 'SIP/acM1000' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:22] DEBUG[25839] rtp.c: Ooh, format changed from unknown to alaw [Jun 9 14:11:22] DEBUG[25839] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.216 with 192.168.161.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.216 with 192.168.200.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.216 with 192.168.1.0 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:400@192.168.161.216:5060 SIP/2.0 (44) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6e5ec745;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as473ad244 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 593433d46f64abdd5ba795dd4c9972a1@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:22 GMT (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:22] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.216:5060: OPTIONS sip:400@192.168.161.216:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6e5ec745;rport From: "asterisk" ;tag=as473ad244 To: Contact: Call-ID: 593433d46f64abdd5ba795dd4c9972a1@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:22] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.216:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6e5ec745;rport=5060 From: "asterisk" ;tag=as473ad244 To: Call-ID: 593433d46f64abdd5ba795dd4c9972a1@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/7.1.33 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK6e5ec745;rport=5060 (71) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as473ad244 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 593433d46f64abdd5ba795dd4c9972a1@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom320/7.1.33 (26) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Accept-Language: en (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Accept: application/sdp (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:22] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: = Found Their Call ID: 593433d46f64abdd5ba795dd4c9972a1@pbx.example.net Their Tag Our tag: as473ad244 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1361 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Stopping retransmission on '593433d46f64abdd5ba795dd4c9972a1@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:22] VERBOSE[23613] logger.c: Really destroying SIP dialog '593433d46f64abdd5ba795dd4c9972a1@pbx.example.net' Method: OPTIONS [Jun 9 14:11:22] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1135deaf;rport=5060 From: "asterisk" ;tag=as3cc23cac To: Call-ID: 1640a08518ad3f521c8aeb4a55a2e61f@pbx.example.net CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom360/7.1.33 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 <-------------> [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK1135deaf;rport=5060 (71) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as3cc23cac (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 1640a08518ad3f521c8aeb4a55a2e61f@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Accept-Language: en (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Accept: application/sdp (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:22] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: = Found Their Call ID: 1640a08518ad3f521c8aeb4a55a2e61f@pbx.example.net Their Tag Our tag: as3cc23cac [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1359 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Stopping retransmission on '1640a08518ad3f521c8aeb4a55a2e61f@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:22] VERBOSE[23613] logger.c: Really destroying SIP dialog '1640a08518ad3f521c8aeb4a55a2e61f@pbx.example.net' Method: OPTIONS [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.161.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.200.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.1.0 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:104@192.168.161.237 SIP/2.0 (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK45bcb2db;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as2f239a86 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: (29) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 7e20fecc25ca06583c1833df6cb64ed5@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:22 GMT (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:22] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.237:5060: OPTIONS sip:104@192.168.161.237 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK45bcb2db;rport From: "asterisk" ;tag=as2f239a86 To: Contact: Call-ID: 7e20fecc25ca06583c1833df6cb64ed5@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:22] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.237:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK45bcb2db;rport From: "asterisk" ;tag=as2f239a86 To: ;tag=1c1517141190 Call-ID: 7e20fecc25ca06583c1833df6cb64ed5@pbx.example.net CSeq: 102 OPTIONS Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: acMP114-01/v.5.20A.032.001 X-Resources: telchs=4/0;mediachs=0/0 Accept: application/sdp, application/simple-message-summary, message/sipfrag Content-Type: application/sdp Content-Length: 557 v=0 o=acMP114-01 1517144572 1517144444 IN IP4 192.168.161.237 s=Phone-Call c=IN IP4 192.168.161.237 t=0 0 m=audio 10000 RTP/AVP 18 8 118 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:118 PCMA/8000 a=gpmd:118 vbd=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:10001 IN IP4 192.168.161.237 m=image 10002 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:122 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK45bcb2db;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as2f239a86 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=1c1517141190 (46) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 7e20fecc25ca06583c1833df6cb64ed5@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: Contact: (30) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: Supported: em,100rel,timer,replaces,path,resource-priority (58) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Server: acMP114-01/v.5.20A.032.001 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: X-Resources: telchs=4/0;mediachs=0/0 (36) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Accept: application/sdp, application/simple-message-summary, message/sipfrag (76) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 557 (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: o=acMP114-01 1517144572 1517144444 IN IP4 192.168.161.237 (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: s=Phone-Call (12) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.237 (24) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=audio 10000 RTP/AVP 18 8 118 101 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:118 PCMA/8000 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=gpmd:118 vbd=yes (18) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtcp:10001 IN IP4 192.168.161.237 (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=image 10002 udptl t38 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxVersion:0 (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38MaxBitRate:14400 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxBuffer:1024 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxDatagram:122 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxRateManagement:transferredTCF (37) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) [Jun 9 14:11:22] VERBOSE[23613] logger.c: --- (14 headers 23 lines) --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: = Found Their Call ID: 7e20fecc25ca06583c1833df6cb64ed5@pbx.example.net Their Tag Our tag: as2f239a86 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1365 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Stopping retransmission on '7e20fecc25ca06583c1833df6cb64ed5@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:22] VERBOSE[23613] logger.c: Really destroying SIP dialog '7e20fecc25ca06583c1833df6cb64ed5@pbx.example.net' Method: OPTIONS [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.161.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.200.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.1.0 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:103@192.168.161.237 SIP/2.0 (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK17503b65;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as0e1cbfb8 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: (29) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 5172c4587d864cb332e4706621c7248b@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:22 GMT (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:22] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.237:5060: OPTIONS sip:103@192.168.161.237 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK17503b65;rport From: "asterisk" ;tag=as0e1cbfb8 To: Contact: Call-ID: 5172c4587d864cb332e4706621c7248b@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:22] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.237:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK17503b65;rport From: "asterisk" ;tag=as0e1cbfb8 To: ;tag=1c1517390928 Call-ID: 5172c4587d864cb332e4706621c7248b@pbx.example.net CSeq: 102 OPTIONS Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: acMP114-01/v.5.20A.032.001 X-Resources: telchs=4/0;mediachs=0/0 Accept: application/sdp, application/simple-message-summary, message/sipfrag Content-Type: application/sdp Content-Length: 557 v=0 o=acMP114-01 1517394240 1517394111 IN IP4 192.168.161.237 s=Phone-Call c=IN IP4 192.168.161.237 t=0 0 m=audio 10000 RTP/AVP 18 8 118 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:118 PCMA/8000 a=gpmd:118 vbd=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:10001 IN IP4 192.168.161.237 m=image 10002 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:122 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK17503b65;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as0e1cbfb8 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=1c1517390928 (46) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 5172c4587d864cb332e4706621c7248b@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: Contact: (30) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: Supported: em,100rel,timer,replaces,path,resource-priority (58) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Server: acMP114-01/v.5.20A.032.001 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: X-Resources: telchs=4/0;mediachs=0/0 (36) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Accept: application/sdp, application/simple-message-summary, message/sipfrag (76) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 557 (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: o=acMP114-01 1517394240 1517394111 IN IP4 192.168.161.237 (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: s=Phone-Call (12) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.237 (24) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=audio 10000 RTP/AVP 18 8 118 101 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:118 PCMA/8000 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=gpmd:118 vbd=yes (18) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtcp:10001 IN IP4 192.168.161.237 (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=image 10002 udptl t38 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxVersion:0 (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38MaxBitRate:14400 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxBuffer:1024 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxDatagram:122 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxRateManagement:transferredTCF (37) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) [Jun 9 14:11:22] VERBOSE[23613] logger.c: --- (14 headers 23 lines) --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: = Found Their Call ID: 5172c4587d864cb332e4706621c7248b@pbx.example.net Their Tag Our tag: as0e1cbfb8 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1368 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Stopping retransmission on '5172c4587d864cb332e4706621c7248b@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:22] VERBOSE[23613] logger.c: Really destroying SIP dialog '5172c4587d864cb332e4706621c7248b@pbx.example.net' Method: OPTIONS [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.161.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.200.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.1.0 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:102@192.168.161.237 SIP/2.0 (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK71753b63;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as28ea8cd7 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: (29) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 6301e00424ea3ec70eb4658527f3e22f@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:22 GMT (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:22] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.237:5060: OPTIONS sip:102@192.168.161.237 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK71753b63;rport From: "asterisk" ;tag=as28ea8cd7 To: Contact: Call-ID: 6301e00424ea3ec70eb4658527f3e22f@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:22] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.237:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK71753b63;rport From: "asterisk" ;tag=as28ea8cd7 To: ;tag=1c1517639851 Call-ID: 6301e00424ea3ec70eb4658527f3e22f@pbx.example.net CSeq: 102 OPTIONS Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: acMP114-01/v.5.20A.032.001 X-Resources: telchs=4/0;mediachs=0/0 Accept: application/sdp, application/simple-message-summary, message/sipfrag Content-Type: application/sdp Content-Length: 557 v=0 o=acMP114-01 1517643165 1517643036 IN IP4 192.168.161.237 s=Phone-Call c=IN IP4 192.168.161.237 t=0 0 m=audio 10000 RTP/AVP 18 8 118 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:118 PCMA/8000 a=gpmd:118 vbd=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:10001 IN IP4 192.168.161.237 m=image 10002 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:122 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK71753b63;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as28ea8cd7 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=1c1517639851 (46) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 6301e00424ea3ec70eb4658527f3e22f@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: Contact: (30) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: Supported: em,100rel,timer,replaces,path,resource-priority (58) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Server: acMP114-01/v.5.20A.032.001 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: X-Resources: telchs=4/0;mediachs=0/0 (36) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Accept: application/sdp, application/simple-message-summary, message/sipfrag (76) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 557 (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: o=acMP114-01 1517643165 1517643036 IN IP4 192.168.161.237 (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: s=Phone-Call (12) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.237 (24) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=audio 10000 RTP/AVP 18 8 118 101 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:118 PCMA/8000 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=gpmd:118 vbd=yes (18) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtcp:10001 IN IP4 192.168.161.237 (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=image 10002 udptl t38 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxVersion:0 (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38MaxBitRate:14400 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxBuffer:1024 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxDatagram:122 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxRateManagement:transferredTCF (37) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) [Jun 9 14:11:22] VERBOSE[23613] logger.c: --- (14 headers 23 lines) --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: = Found Their Call ID: 6301e00424ea3ec70eb4658527f3e22f@pbx.example.net Their Tag Our tag: as28ea8cd7 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1371 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Stopping retransmission on '6301e00424ea3ec70eb4658527f3e22f@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:22] VERBOSE[23613] logger.c: Really destroying SIP dialog '6301e00424ea3ec70eb4658527f3e22f@pbx.example.net' Method: OPTIONS [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.161.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.200.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.237 with 192.168.1.0 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:101@192.168.161.237 SIP/2.0 (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2669c080;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as275d8d35 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: (29) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 177e33ad2c58d0cb27fe2f6064c06c0d@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:22 GMT (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:22] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.237:5060: OPTIONS sip:101@192.168.161.237 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2669c080;rport From: "asterisk" ;tag=as275d8d35 To: Contact: Call-ID: 177e33ad2c58d0cb27fe2f6064c06c0d@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:22] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.237:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2669c080;rport From: "asterisk" ;tag=as275d8d35 To: ;tag=1c1517889957 Call-ID: 177e33ad2c58d0cb27fe2f6064c06c0d@pbx.example.net CSeq: 102 OPTIONS Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: acMP114-01/v.5.20A.032.001 X-Resources: telchs=4/0;mediachs=0/0 Accept: application/sdp, application/simple-message-summary, message/sipfrag Content-Type: application/sdp Content-Length: 557 v=0 o=acMP114-01 1517893248 1517893120 IN IP4 192.168.161.237 s=Phone-Call c=IN IP4 192.168.161.237 t=0 0 m=audio 10000 RTP/AVP 18 8 118 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:118 PCMA/8000 a=gpmd:118 vbd=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:10001 IN IP4 192.168.161.237 m=image 10002 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:122 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK2669c080;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as275d8d35 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=1c1517889957 (46) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 177e33ad2c58d0cb27fe2f6064c06c0d@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: Contact: (30) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: Supported: em,100rel,timer,replaces,path,resource-priority (58) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Server: acMP114-01/v.5.20A.032.001 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: X-Resources: telchs=4/0;mediachs=0/0 (36) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Accept: application/sdp, application/simple-message-summary, message/sipfrag (76) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 557 (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: o=acMP114-01 1517893248 1517893120 IN IP4 192.168.161.237 (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: s=Phone-Call (12) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.237 (24) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=audio 10000 RTP/AVP 18 8 118 101 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:118 PCMA/8000 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=gpmd:118 vbd=yes (18) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtcp:10001 IN IP4 192.168.161.237 (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=image 10002 udptl t38 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxVersion:0 (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38MaxBitRate:14400 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxBuffer:1024 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxDatagram:122 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxRateManagement:transferredTCF (37) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) [Jun 9 14:11:22] VERBOSE[23613] logger.c: --- (14 headers 23 lines) --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: = Found Their Call ID: 177e33ad2c58d0cb27fe2f6064c06c0d@pbx.example.net Their Tag Our tag: as275d8d35 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1374 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Stopping retransmission on '177e33ad2c58d0cb27fe2f6064c06c0d@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:22] VERBOSE[23613] logger.c: Really destroying SIP dialog '177e33ad2c58d0cb27fe2f6064c06c0d@pbx.example.net' Method: OPTIONS [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.4 with 192.168.161.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.4 with 192.168.200.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.4 with 192.168.1.0 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:acM1000-01@192.168.161.4 SIP/2.0 (44) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK60dcc1bb;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as57c50472 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 4e6b94f038fc152a68633a182f9e0c5c@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:22 GMT (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:22] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.4:5060: OPTIONS sip:acM1000-01@192.168.161.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK60dcc1bb;rport From: "asterisk" ;tag=as57c50472 To: Contact: Call-ID: 4e6b94f038fc152a68633a182f9e0c5c@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:22] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK60dcc1bb;rport From: "asterisk" ;tag=as57c50472 To: ;tag=1c939389623 Call-ID: 4e6b94f038fc152a68633a182f9e0c5c@pbx.example.net CSeq: 102 OPTIONS Contact: Supported: em,100rel,timer,replaces,path,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: acM1000-01/v.5.20A.032.001 X-Resources: telchs=29/0;mediachs=0/0 Accept: application/sdp, application/simple-message-summary, message/sipfrag Content-Type: application/sdp Content-Length: 549 v=0 o=acM1000-01 939396842 939396501 IN IP4 192.168.161.4 s=Phone-Call c=IN IP4 192.168.161.4 t=0 0 m=audio 10000 RTP/AVP 18 8 118 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:118 PCMA/8000 a=gpmd:118 vbd=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:10001 IN IP4 192.168.161.4 m=image 10002 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:122 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK60dcc1bb;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as57c50472 (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=1c939389623 (50) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 4e6b94f038fc152a68633a182f9e0c5c@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: Contact: (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: Supported: em,100rel,timer,replaces,path,resource-priority (58) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE (86) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Server: acM1000-01/v.5.20A.032.001 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: X-Resources: telchs=29/0;mediachs=0/0 (37) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Accept: application/sdp, application/simple-message-summary, message/sipfrag (76) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 549 (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: o=acM1000-01 939396842 939396501 IN IP4 192.168.161.4 (53) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: s=Phone-Call (12) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.4 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=audio 10000 RTP/AVP 18 8 118 101 (34) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:18 annexb=no (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:118 PCMA/8000 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=gpmd:118 vbd=yes (18) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-15 (15) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=rtcp:10001 IN IP4 192.168.161.4 (33) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: m=image 10002 udptl t38 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxVersion:0 (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38MaxBitRate:14400 (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxBuffer:1024 (22) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxMaxDatagram:122 (23) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxRateManagement:transferredTCF (37) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) [Jun 9 14:11:22] VERBOSE[23613] logger.c: --- (14 headers 23 lines) --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: = Found Their Call ID: 4e6b94f038fc152a68633a182f9e0c5c@pbx.example.net Their Tag Our tag: as57c50472 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1377 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Stopping retransmission on '4e6b94f038fc152a68633a182f9e0c5c@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:22] VERBOSE[23613] logger.c: Really destroying SIP dialog '4e6b94f038fc152a68633a182f9e0c5c@pbx.example.net' Method: OPTIONS [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.3 with 192.168.161.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.3 with 192.168.200.0 [Jun 9 14:11:22] DEBUG[23613] acl.c: ##### Testing 192.168.161.3 with 192.168.1.0 [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 0: OPTIONS sip:acMP404-01@192.168.161.3:5060 SIP/2.0 (49) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK08ee34e4;rport (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 2: From: "asterisk" ;tag=as599bbe8c (62) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 3: To: (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 4: Contact: (39) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 5: Call-ID: 3cf884f17e32689c79ef9c3b291336c1@pbx.example.net (57) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 9: Date: Mon, 09 Jun 2008 12:11:22 GMT (35) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 11: Supported: replaces (19) [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 0 (17) [Jun 9 14:11:22] VERBOSE[23613] logger.c: Reliably Transmitting (no NAT) to 192.168.161.3:5060: OPTIONS sip:acMP404-01@192.168.161.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK08ee34e4;rport From: "asterisk" ;tag=as599bbe8c To: Contact: Call-ID: 3cf884f17e32689c79ef9c3b291336c1@pbx.example.net CSeq: 102 OPTIONS User-Agent: atCOM PBX Max-Forwards: 70 Date: Mon, 09 Jun 2008 12:11:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 9 14:11:22] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:23] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.3:5060 ---> SIP/2.0 501 Not Implemented From: "asterisk";tag=as599bbe8c To: ;tag=139fb30-0-13c4-8069b-7b5ae2c2-8069b Call-ID: 3cf884f17e32689c79ef9c3b291336c1@pbx.example.net CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.161.100:5060;rport=5060;branch=z9hG4bK08ee34e4 Supported: replaces,timer,100rel Content-Length: 0 <-------------> [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 501 Not Implemented (27) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 1: From: "asterisk";tag=as599bbe8c (61) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 2: To: ;tag=139fb30-0-13c4-8069b-7b5ae2c2-8069b (79) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 3: Call-ID: 3cf884f17e32689c79ef9c3b291336c1@pbx.example.net (57) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.161.100:5060;rport=5060;branch=z9hG4bK08ee34e4 (71) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 6: Supported: replaces,timer,100rel (32) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 7: Content-Length: 0 (17) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 8: (0) [Jun 9 14:11:23] VERBOSE[23613] logger.c: --- (8 headers 0 lines) --- [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3cf884f17e32689c79ef9c3b291336c1@pbx.example.net Their Tag Our tag: as599bbe8c [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1380 [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Stopping retransmission on '3cf884f17e32689c79ef9c3b291336c1@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:23] VERBOSE[23613] logger.c: Really destroying SIP dialog '3cf884f17e32689c79ef9c3b291336c1@pbx.example.net' Method: OPTIONS [Jun 9 14:11:23] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> INVITE sip:400@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-tadmx9qs8mlk;rport From: ;tag=j41casm0gq To: "Toestel 400" ;tag=as7bac7baf Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.1.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 226 v=0 o=root 1824544477 1824544479 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 11842 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 0: INVITE sip:400@192.168.161.100 SIP/2.0 (38) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-tadmx9qs8mlk;rport (71) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 2: From: ;tag=j41casm0gq (51) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 3: To: "Toestel 400" ;tag=as7bac7baf (58) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net (57) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 5: CSeq: 1 INVITE (14) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 8: P-Key-Flags: resolution="31x13", keys="4" (41) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 9: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 10: Accept: application/sdp (23) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 12: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 13: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 14: Session-Expires: 3600;refresher=uas (35) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 15: Min-SE: 90 (10) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 16: Content-Type: application/sdp (29) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 17: Content-Length: 226 (19) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 18: (0) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: o=root 1824544477 1824544479 IN IP4 192.168.161.250 (51) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: s=call (6) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.250 (24) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: m=audio 11842 RTP/AVP 8 101 (27) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Line: a=sendonly (10) [Jun 9 14:11:23] VERBOSE[23613] logger.c: --- (18 headers 11 lines) --- [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: = No match Their Call ID: 868521015562008142045@192.168.161.3 Their Tag 1c923705775 Our tag: as704c8058 [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: = Found Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Found SIP option: -timer- [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Matched SIP option: timer [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Found SIP option: -100rel- [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Matched SIP option: 100rel [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Found SIP option: -replaces- [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Matched SIP option: replaces [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Found SIP option: -from-change- [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Found no match for SIP option: from-change (Please file bug report!) [Jun 9 14:11:23] VERBOSE[23613] logger.c: Sending to 192.168.161.250 : 5060 (NAT) [Jun 9 14:11:23] VERBOSE[23613] logger.c: Found RTP audio format 8 [Jun 9 14:11:23] VERBOSE[23613] logger.c: Found RTP audio format 101 [Jun 9 14:11:23] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.250:11842 [Jun 9 14:11:23] VERBOSE[23613] logger.c: Found audio description format pcma for ID 8 [Jun 9 14:11:23] VERBOSE[23613] logger.c: Found audio description format telephone-event for ID 101 [Jun 9 14:11:23] VERBOSE[23613] logger.c: Got unsupported a:fmtp in SDP offer [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: T38 state changed to 0 on channel SIP/401-089415f0 [Jun 9 14:11:23] VERBOSE[23613] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jun 9 14:11:23] VERBOSE[23613] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 9 14:11:23] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.250:11842 [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: We have an owner, now see if we need to change this call [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Got a SIP re-invite for call 28aa6e4d058373f806130cd33cee1f63@pbx.example.net [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: SIP/401-089415f0: This call is UP.... [Jun 9 14:11:23] VERBOSE[23613] logger.c: <--- Transmitting (NAT) to 192.168.161.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-tadmx9qs8mlk;received=192.168.161.250;rport=5060 From: ;tag=j41casm0gq To: "Toestel 400" ;tag=as7bac7baf Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 1 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Setting framing from config on incoming call [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: ** Our prefcodec: 0x100 (g729) [Jun 9 14:11:23] VERBOSE[23613] logger.c: Audio is at 192.168.161.100 port 19240 [Jun 9 14:11:23] VERBOSE[23613] logger.c: Adding codec 0x8 (alaw) to SDP [Jun 9 14:11:23] VERBOSE[23613] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: -- Done with adding codecs to SDP [Jun 9 14:11:23] DEBUG[23613] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=30) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 9 14:11:23] VERBOSE[23613] logger.c: <--- Reliably Transmitting (NAT) to 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-tadmx9qs8mlk;received=192.168.161.250;rport=5060 From: ;tag=j41casm0gq To: "Toestel 400" ;tag=as7bac7baf Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 1 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 23546 23547 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 19240 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:23] VERBOSE[25839] logger.c: -- Started music on hold, class 'default', on SIP/400-08908960 [Jun 9 14:11:23] DEBUG[25839] channel.c: Scheduling timer at 160 sample intervals [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Generator got voice, switching to phase locked mode [Jun 9 14:11:23] DEBUG[25839] channel.c: Scheduling timer at 0 sample intervals [Jun 9 14:11:23] DEBUG[25839] channel.c: Set channel SIP/400-08908960 to write format g729 [Jun 9 14:11:23] DEBUG[25839] res_musiconhold.c: SIP/400-08908960 Opened file 1 '/var/lib/asterisk/moh/default/calm-river' [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> ACK sip:400@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-b3fgsl55agxb;rport From: ;tag=j41casm0gq To: "Toestel 400" ;tag=as7bac7baf Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 0: ACK sip:400@192.168.161.100 SIP/2.0 (35) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-b3fgsl55agxb;rport (71) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 2: From: ;tag=j41casm0gq (51) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 3: To: "Toestel 400" ;tag=as7bac7baf (58) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net (57) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 5: CSeq: 1 ACK (11) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 8: Content-Length: 0 (17) [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Header 9: (0) [Jun 9 14:11:23] VERBOSE[23613] logger.c: --- (9 headers 0 lines) --- [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: = No match Their Call ID: 868521015562008142045@192.168.161.3 Their Tag 1c923705775 Our tag: as704c8058 [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: = Found Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1383 [Jun 9 14:11:23] DEBUG[23613] chan_sip.c: Stopping retransmission on '28aa6e4d058373f806130cd33cee1f63@pbx.example.net' of Response 1: Match Found [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:23] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> INVITE sip:402@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-mwzbc7i4bs6p;rport From: "Toestel 401" ;tag=5gx8xpwpe4 To: Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.1.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 226 v=0 o=root 1521396104 1521396104 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 17384 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 0: INVITE sip:402@pbx.example.net;user=phone SIP/2.0 (49) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-mwzbc7i4bs6p;rport (71) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 401" ;tag=5gx8xpwpe4 (60) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 3: To: (40) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617340bfe0-92e68vgyoo4t (34) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 5: CSeq: 1 INVITE (14) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 8: P-Key-Flags: resolution="31x13", keys="4" (41) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 9: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 10: Accept: application/sdp (23) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 12: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 13: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 14: Session-Expires: 3600;refresher=uas (35) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 15: Min-SE: 90 (10) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 16: Content-Type: application/sdp (29) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 17: Content-Length: 226 (19) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 18: (0) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: o=root 1521396104 1521396104 IN IP4 192.168.161.250 (51) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: s=call (6) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.250 (24) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: m=audio 17384 RTP/AVP 8 101 (27) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:24] VERBOSE[23613] logger.c: --- (18 headers 11 lines) --- [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 868521015562008142045@192.168.161.3 Their Tag 1c923705775 Our tag: as704c8058 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c617304b5ff-hqq001xrvvkf Their Tag ow35sk9e64 Our tag: as03ae9445 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743299462942008195958@192.168.161.237 Their Tag 1c1502243045 Our tag: as0f2a9650 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743318622942008195958@192.168.161.237 Their Tag 1c1502214188 Our tag: as2363565d [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743312262942008195958@192.168.161.237 Their Tag 1c1502198413 Our tag: as14e39483 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743305952942008195958@192.168.161.237 Their Tag 1c1502133902 Our tag: as3584dd5b [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ada48-mkwj7m38q3nu Their Tag wbtjjopkwe Our tag: as28583bdb [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701a4fb9-rd37qdoo1uyb Their Tag cyb8d7epkd Our tag: as2313fe09 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad395-ajo6vj1qc0vm Their Tag 97qngh5alh Our tag: as39395c26 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad4c6-phe3ytd26k13 Their Tag hf67o5uhr4 Our tag: as73260cbd [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad52a-gwy8p5tlpim3 Their Tag sjongh64mn Our tag: as56f28897 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad45e-7fxbtgnl1u8t Their Tag tstmut2dxd Our tag: as602e2d77 [Jun 9 14:11:24] DEBUG[23613] acl.c: ##### Testing 192.168.161.250 with 192.168.161.0 [Jun 9 14:11:24] DEBUG[23613] acl.c: ##### Testing 192.168.161.250 with 192.168.200.0 [Jun 9 14:11:24] DEBUG[23613] acl.c: ##### Testing 192.168.161.250 with 192.168.1.0 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Setting NAT on RTP to Off [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Setting NAT on VRTP to Off [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Setting NAT on UDPTL to Off [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for 3c617340bfe0-92e68vgyoo4t - INVITE (With RTP) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change" [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Found SIP option: -timer- [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Matched SIP option: timer [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Found SIP option: -100rel- [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Matched SIP option: 100rel [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Found SIP option: -replaces- [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Matched SIP option: replaces [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Found SIP option: -from-change- [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Found no match for SIP option: from-change (Please file bug report!) [Jun 9 14:11:24] VERBOSE[23613] logger.c: Sending to 192.168.161.250 : 5060 (NAT) [Jun 9 14:11:24] VERBOSE[23613] logger.c: Using INVITE request as basis request - 3c617340bfe0-92e68vgyoo4t [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Setting NAT on RTP to Off [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Setting NAT on VRTP to Off [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Setting NAT on UDPTL to Off [Jun 9 14:11:24] VERBOSE[23613] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-mwzbc7i4bs6p;received=192.168.161.250;rport=5060 From: "Toestel 401" ;tag=5gx8xpwpe4 To: ;tag=as41edb3fd Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 1 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="29d61c5f" Content-Length: 0 <------------> [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:24] VERBOSE[23613] logger.c: Scheduling destruction of SIP dialog '3c617340bfe0-92e68vgyoo4t' in 32000 ms (Method: INVITE) [Jun 9 14:11:24] VERBOSE[23613] logger.c: Found user '401' [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> ACK sip:402@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-mwzbc7i4bs6p;rport From: "Toestel 401" ;tag=5gx8xpwpe4 To: ;tag=as41edb3fd Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 0: ACK sip:402@pbx.example.net;user=phone SIP/2.0 (46) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-mwzbc7i4bs6p;rport (71) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 401" ;tag=5gx8xpwpe4 (60) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=as41edb3fd (55) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617340bfe0-92e68vgyoo4t (34) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 5: CSeq: 1 ACK (11) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 8: Content-Length: 0 (17) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 9: (0) [Jun 9 14:11:24] VERBOSE[23613] logger.c: --- (9 headers 0 lines) --- [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c617340bfe0-92e68vgyoo4t Their Tag 5gx8xpwpe4 Our tag: as41edb3fd [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1384 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Stopping retransmission on '3c617340bfe0-92e68vgyoo4t' of Response 1: Match Found [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> INVITE sip:402@pbx.example.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-elydtnhmgh3o;rport From: "Toestel 401" ;tag=5gx8xpwpe4 To: Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.1.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="401",realm="siprealm",nonce="29d61c5f",uri="sip:402@pbx.example.net;user=phone",response="5283b2cc26ed4d92451f7230dbf9708e",algorithm=MD5 Content-Type: application/sdp Content-Length: 226 v=0 o=root 1521396104 1521396104 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 17384 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 0: INVITE sip:402@pbx.example.net;user=phone SIP/2.0 (49) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-elydtnhmgh3o;rport (71) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 401" ;tag=5gx8xpwpe4 (60) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 3: To: (40) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617340bfe0-92e68vgyoo4t (34) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 5: CSeq: 2 INVITE (14) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 8: P-Key-Flags: resolution="31x13", keys="4" (41) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 9: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 10: Accept: application/sdp (23) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 12: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 13: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 14: Session-Expires: 3600;refresher=uas (35) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 15: Min-SE: 90 (10) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 16: Proxy-Authorization: Digest username="401",realm="siprealm",nonce="29d61c5f",uri="sip:402@pbx.example.net;user=phone",response="5283b2cc26ed4d92451f7230dbf9708e",algorithm=MD5 (175) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 17: Content-Type: application/sdp (29) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 18: Content-Length: 226 (19) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Header 19: (0) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: o=root 1521396104 1521396104 IN IP4 192.168.161.250 (51) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: s=call (6) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.250 (24) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: m=audio 17384 RTP/AVP 8 101 (27) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:24] VERBOSE[23613] logger.c: --- (19 headers 11 lines) --- [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c617340bfe0-92e68vgyoo4t Their Tag 5gx8xpwpe4 Our tag: as41edb3fd [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 9 14:11:24] VERBOSE[23613] logger.c: Sending to 192.168.161.250 : 5060 (NAT) [Jun 9 14:11:24] VERBOSE[23613] logger.c: Using INVITE request as basis request - 3c617340bfe0-92e68vgyoo4t [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Setting NAT on RTP to Off [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Setting NAT on VRTP to Off [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Setting NAT on UDPTL to Off [Jun 9 14:11:24] VERBOSE[23613] logger.c: Found user '401' [Jun 9 14:11:24] VERBOSE[23613] logger.c: Found RTP audio format 8 [Jun 9 14:11:24] VERBOSE[23613] logger.c: Found RTP audio format 101 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Peer doesn't provide T.38 UDPTL [Jun 9 14:11:24] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.250:17384 [Jun 9 14:11:24] VERBOSE[23613] logger.c: Found audio description format pcma for ID 8 [Jun 9 14:11:24] VERBOSE[23613] logger.c: Found audio description format telephone-event for ID 101 [Jun 9 14:11:24] VERBOSE[23613] logger.c: Got unsupported a:fmtp in SDP offer [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: T38 state changed to 0 on channel [Jun 9 14:11:24] VERBOSE[23613] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jun 9 14:11:24] VERBOSE[23613] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 9 14:11:24] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.250:17384 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Checking SIP call limits for device 401 [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Updating call counter for incoming call [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: Call from peer '401' is 2 out of 30 [Jun 9 14:11:24] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:24] VERBOSE[23613] logger.c: Looking for 402 in flex-14 (domain pbx.example.net) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: *** Our capabilities are 0x8 (alaw) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: This channel will not be able to handle video. [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: build_route: Contact hop: ;flow-id=1 [Jun 9 14:11:24] VERBOSE[23613] logger.c: list_route: hop: [Jun 9 14:11:24] DEBUG[23613] chan_sip.c: SIP/401-088e0b60: New call is still down.... Trying... [Jun 9 14:11:24] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-elydtnhmgh3o;received=192.168.161.250;rport=5060 From: "Toestel 401" ;tag=5gx8xpwpe4 To: Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:24] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/401-088e0b60 [Jun 9 14:11:24] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:24] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:24] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:24] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 2 (In use) [Jun 9 14:11:24] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:24] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:24] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401-088e0b60 [Jun 9 14:11:24] DEBUG[23576] chan_sip.c: Checking device state for peer 401-088e0b60 [Jun 9 14:11:24] DEBUG[25840] pbx.c: Launching 'Set' [Jun 9 14:11:24] VERBOSE[25840] logger.c: -- Executing [402@flex-14:1] Set("SIP/401-088e0b60", "ARRAY(_EXTTYPE|_STARTTIME)=flex,1213013484.756433") in new stack [Jun 9 14:11:24] DEBUG[25840] func_strings.c: array (_EXTTYPE|_STARTTIME=flex,1213013484.756433) [Jun 9 14:11:24] DEBUG[25840] func_strings.c: array set value (_EXTTYPE=flex) [Jun 9 14:11:24] DEBUG[25840] func_strings.c: array set value (_STARTTIME=1213013484.756433) [Jun 9 14:11:24] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:24] DEBUG[23576] devicestate.c: Changing state for SIP/401-088e0b60 - state 4 (Invalid) [Jun 9 14:11:24] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:24] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:24] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 2 (In use) [Jun 9 14:11:24] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:24] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:24] DEBUG[23615] app_queue.c: Device 'SIP/401-088e0b60' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:24] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25840] pbx.c: Function result is '0' [Jun 9 14:11:24] DEBUG[25840] pbx.c: Expression result is '0' [Jun 9 14:11:24] DEBUG[25840] pbx.c: Launching 'GotoIf' [Jun 9 14:11:24] VERBOSE[25840] logger.c: -- Executing [402@flex-14:2] GotoIf("SIP/401-088e0b60", "0?INBOUND:OUTBOUND") in new stack [Jun 9 14:11:24] VERBOSE[25840] logger.c: -- Goto (flex-14,402,8) [Jun 9 14:11:24] DEBUG[25840] pbx.c: Function result is '401' [Jun 9 14:11:24] DEBUG[25840] pbx.c: Launching 'Macro' [Jun 9 14:11:24] VERBOSE[25840] logger.c: -- Executing [402@flex-14:8] Macro("SIP/401-088e0b60", "flexext|SETEXT|401") in new stack [Jun 9 14:11:24] DEBUG[25840] pbx.c: Launching 'Goto' [Jun 9 14:11:24] VERBOSE[25840] logger.c: -- Executing [s@macro-flexext:1] Goto("SIP/401-088e0b60", "SETEXT") in new stack [Jun 9 14:11:24] VERBOSE[25840] logger.c: -- Goto (macro-flexext,s,60) [Jun 9 14:11:24] DEBUG[25840] app_macro.c: Executed application: Goto [Jun 9 14:11:24] DEBUG[25840] pbx.c: Launching 'NoOp' [Jun 9 14:11:24] VERBOSE[25840] logger.c: -- Executing [s@macro-flexext:60] NoOp("SIP/401-088e0b60", ".... Request to set the CallerID & Context for call from flex working enabled extension: 401 ....") in new stack [Jun 9 14:11:24] DEBUG[25840] app_macro.c: Executed application: NoOp [Jun 9 14:11:24] DEBUG[25840] pbx.c: Function result is '0' [Jun 9 14:11:24] DEBUG[25840] pbx.c: Expression result is '1' [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25840] pbx.c: Function result is '2' [Jun 9 14:11:24] DEBUG[25840] pbx.c: Function result is '1.000000' [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25840] pbx.c: Function result is 'SIP/401' [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25840] pbx.c: Function result is '0' [Jun 9 14:11:24] DEBUG[25840] pbx.c: Function result is '-1.000000' [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:24] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25840] pbx.c: Function result is '' [Jun 9 14:11:25] DEBUG[25840] pbx.c: Function result is 'SIP/401' [Jun 9 14:11:25] DEBUG[25840] pbx.c: Launching 'Set' [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- Executing [s@macro-flexext:61] Set("SIP/401-088e0b60", "FLEXDEVICE=SIP/401") in new stack [Jun 9 14:11:25] DEBUG[25840] app_macro.c: Executed application: Set [Jun 9 14:11:25] VERBOSE[25840] logger.c: > Found no rows [SELECT nr,name,usercontext FROM viewextensiondata WHERE devicename = 'SIP/401' AND type = 'flex'] [Jun 9 14:11:25] DEBUG[25840] pbx.c: Function result is '' [Jun 9 14:11:25] DEBUG[25840] pbx.c: Launching 'Set' [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- Executing [s@macro-flexext:62] Set("SIP/401-088e0b60", "ARRAY(FLEXEXT|FLEXNAME|FLEXCONTEXT)=") in new stack [Jun 9 14:11:25] DEBUG[25840] func_strings.c: array (FLEXEXT|FLEXNAME|FLEXCONTEXT=) [Jun 9 14:11:25] DEBUG[25840] func_strings.c: array set value (FLEXEXT=(null)) [Jun 9 14:11:25] DEBUG[25840] func_strings.c: array set value (FLEXNAME=(null)) [Jun 9 14:11:25] DEBUG[25840] func_strings.c: array set value (FLEXCONTEXT=(null)) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25840] app_macro.c: Executed application: Set [Jun 9 14:11:25] DEBUG[25840] pbx.c: Expression result is '0' [Jun 9 14:11:25] DEBUG[25840] pbx.c: Launching 'GotoIf' [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- Executing [s@macro-flexext:63] GotoIf("SIP/401-088e0b60", "0?LOGGEDIN") in new stack [Jun 9 14:11:25] DEBUG[25840] pbx.c: Not taking any branch [Jun 9 14:11:25] DEBUG[25840] app_macro.c: Executed application: GotoIf [Jun 9 14:11:25] DEBUG[25840] pbx.c: Launching 'Set' [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- Executing [s@macro-flexext:64] Set("SIP/401-088e0b60", "FLEXCONTEXT=user-14") in new stack [Jun 9 14:11:25] DEBUG[25840] app_macro.c: Executed application: Set [Jun 9 14:11:25] DEBUG[25840] pbx.c: Function result is '0' [Jun 9 14:11:25] DEBUG[25840] pbx.c: Expression result is '1' [Jun 9 14:11:25] DEBUG[25840] pbx.c: Launching 'ExecIf' [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- Executing [s@macro-flexext:65] ExecIf("SIP/401-088e0b60", "1|MacroExit|") in new stack [Jun 9 14:11:25] DEBUG[25840] pbx.c: Launching 'Goto' [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- Executing [402@flex-14:9] Goto("SIP/401-088e0b60", "user-14|402|1") in new stack [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- Goto (user-14,402,1) [Jun 9 14:11:25] DEBUG[25840] pbx.c: Launching 'Dial' [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- Executing [402@user-14:1] Dial("SIP/401-088e0b60", "SIP/402") in new stack [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Setting NAT on RTP to Off [Jun 9 14:11:25] DEBUG[25840] acl.c: ##### Testing 192.168.161.249 with 192.168.161.0 [Jun 9 14:11:25] DEBUG[25840] acl.c: ##### Testing 192.168.161.249 with 192.168.200.0 [Jun 9 14:11:25] DEBUG[25840] acl.c: ##### Testing 192.168.161.249 with 192.168.1.0 [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: *** Our capabilities are 0x8 (alaw) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: This channel will not be able to handle video. [Jun 9 14:11:25] DEBUG[25840] channel.c: Not copying variable MACRO_DEPTH. [Jun 9 14:11:25] DEBUG[25840] channel.c: Not copying variable FLEXCONTEXT. [Jun 9 14:11:25] DEBUG[25840] channel.c: Not copying variable FLEXNAME. [Jun 9 14:11:25] DEBUG[25840] channel.c: Not copying variable FLEXEXT. [Jun 9 14:11:25] DEBUG[25840] channel.c: Not copying variable FLEXDEVICE. [Jun 9 14:11:25] DEBUG[25840] channel.c: Copying soft-transferable variable STARTTIME. [Jun 9 14:11:25] DEBUG[25840] channel.c: Copying soft-transferable variable EXTTYPE. [Jun 9 14:11:25] DEBUG[25840] channel.c: Not copying variable SIPCALLID. [Jun 9 14:11:25] DEBUG[25840] channel.c: Not copying variable SIPUSERAGENT. [Jun 9 14:11:25] DEBUG[25840] channel.c: Not copying variable SIPDOMAIN. [Jun 9 14:11:25] DEBUG[25840] channel.c: Not copying variable SIPURI. [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Outgoing Call for 402 [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Updating call counter for outgoing call [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Call to peer '402' is 1 out of 30 [Jun 9 14:11:25] DEBUG[25840] devicestate.c: Notification of state change to be queued on device/channel SIP/402 [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Our T38 capability (3856), joint T38 capability (3856) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: False [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 9 14:11:25] VERBOSE[25840] logger.c: Audio is at 192.168.161.100 port 19822 [Jun 9 14:11:25] VERBOSE[25840] logger.c: Adding codec 0x8 (alaw) to SDP [Jun 9 14:11:25] VERBOSE[25840] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: -- Done with adding codecs to SDP [Jun 9 14:11:25] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 0: INVITE sip:402@192.168.161.249:5060 SIP/2.0 (43) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK55ba651d;rport (66) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 2: From: "Toestel 401" ;tag=as08c126ff (60) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 3: To: (34) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 4: Contact: (34) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 5: Call-ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net (57) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 7: User-Agent: atCOM PBX (21) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 9: Remote-Party-ID: "Toestel 401" ;privacy=off;screen=no (78) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 10: Date: Mon, 09 Jun 2008 12:11:25 GMT (35) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 12: Supported: replaces (19) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 14: Content-Length: 246 (19) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Header 15: (0) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: o=root 23546 23546 IN IP4 192.168.161.100 (41) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: s=session (9) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: c=IN IP4 192.168.161.100 (24) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: m=audio 19822 RTP/AVP 8 101 (27) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:25] VERBOSE[25840] logger.c: Reliably Transmitting (no NAT) to 192.168.161.249:5060: INVITE sip:402@192.168.161.249:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK55ba651d;rport From: "Toestel 401" ;tag=as08c126ff To: Contact: Call-ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net CSeq: 102 INVITE User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Toestel 401" ;privacy=off;screen=no Date: Mon, 09 Jun 2008 12:11:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 246 v=0 o=root 23546 23546 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 19822 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 --- [Jun 9 14:11:25] DEBUG[25840] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- Called 402 [Jun 9 14:11:25] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402 [Jun 9 14:11:25] DEBUG[23576] chan_sip.c: Checking device state for peer 402 [Jun 9 14:11:25] DEBUG[23576] devicestate.c: Changing state for SIP/402 - state 6 (Ringing) [Jun 9 14:11:25] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402 [Jun 9 14:11:25] DEBUG[23576] chan_sip.c: Checking device state for peer 402 [Jun 9 14:11:25] DEBUG[23615] app_queue.c: Device 'SIP/402' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.249:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK55ba651d;rport=5060 From: "Toestel 401" ;tag=as08c126ff To: ;tag=18rpfhgert Call-ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK55ba651d;rport=5060 (71) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 401" ;tag=as08c126ff (60) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=18rpfhgert (49) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net (57) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 8: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 9: Content-Length: 0 (17) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 10: (0) [Jun 9 14:11:25] VERBOSE[23613] logger.c: --- (10 headers 0 lines) --- [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: = Found Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag Our tag: as08c126ff [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: *** SIP TIMER: Cancelling retransmission #1386 - INVITE (got response) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net' Request 102: Found [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: SIP response 180 to standard invite [Jun 9 14:11:25] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/402-08943748 [Jun 9 14:11:25] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/402 [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- SIP/402-08943748 is ringing [Jun 9 14:11:25] VERBOSE[25840] logger.c: <--- Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-elydtnhmgh3o;received=192.168.161.250;rport=5060 From: "Toestel 401" ;tag=5gx8xpwpe4 To: ;tag=as145459f4 Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:25] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402-08943748 [Jun 9 14:11:25] DEBUG[23576] chan_sip.c: Checking device state for peer 402-08943748 [Jun 9 14:11:25] DEBUG[23576] devicestate.c: Changing state for SIP/402-08943748 - state 4 (Invalid) [Jun 9 14:11:25] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402 [Jun 9 14:11:25] DEBUG[23576] chan_sip.c: Checking device state for peer 402 [Jun 9 14:11:25] DEBUG[23576] devicestate.c: Changing state for SIP/402 - state 6 (Ringing) [Jun 9 14:11:25] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402 [Jun 9 14:11:25] DEBUG[23576] chan_sip.c: Checking device state for peer 402 [Jun 9 14:11:25] DEBUG[23615] app_queue.c: Device 'SIP/402-08943748' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:25] DEBUG[23615] app_queue.c: Device 'SIP/402' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.249:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK55ba651d;rport=5060 From: "Toestel 401" ;tag=as08c126ff To: ;tag=18rpfhgert Call-ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 <-------------> [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK55ba651d;rport=5060 (71) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 401" ;tag=as08c126ff (60) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=18rpfhgert (49) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net (57) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 8: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 9: Content-Length: 0 (17) [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: Header 10: (0) [Jun 9 14:11:25] VERBOSE[23613] logger.c: --- (10 headers 0 lines) --- [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: = Found Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net' Request 102: Found [Jun 9 14:11:25] DEBUG[23613] chan_sip.c: SIP response 180 to standard invite [Jun 9 14:11:25] VERBOSE[25840] logger.c: -- SIP/402-08943748 is ringing [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:25] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.249:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK55ba651d;rport=5060 From: "Toestel 401" ;tag=as08c126ff To: ;tag=18rpfhgert Call-ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom360/7.1.33 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Type: application/sdp Content-Length: 226 v=0 o=root 1773935955 1773935956 IN IP4 192.168.161.249 s=call c=IN IP4 192.168.161.249 t=0 0 m=audio 16142 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 Ok (14) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK55ba651d;rport=5060 (71) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 401" ;tag=as08c126ff (60) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=18rpfhgert (49) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net (57) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 8: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 9: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 10: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 11: Content-Type: application/sdp (29) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 12: Content-Length: 226 (19) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 13: (0) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: o=root 1773935955 1773935956 IN IP4 192.168.161.249 (51) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: s=call (6) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.249 (24) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: m=audio 16142 RTP/AVP 8 101 (27) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Line: a=sendrecv (10) [Jun 9 14:11:26] VERBOSE[23613] logger.c: --- (13 headers 11 lines) --- [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: = Found Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Acked pending invite 102 [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Stopping retransmission on '36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net' of Request 102: Match Found [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: SIP response 200 to standard invite [Jun 9 14:11:26] VERBOSE[23613] logger.c: Found RTP audio format 8 [Jun 9 14:11:26] VERBOSE[23613] logger.c: Found RTP audio format 101 [Jun 9 14:11:26] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.249:16142 [Jun 9 14:11:26] VERBOSE[23613] logger.c: Found audio description format pcma for ID 8 [Jun 9 14:11:26] VERBOSE[23613] logger.c: Found audio description format telephone-event for ID 101 [Jun 9 14:11:26] VERBOSE[23613] logger.c: Got unsupported a:fmtp in SDP offer [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: T38 state changed to 0 on channel SIP/402-08943748 [Jun 9 14:11:26] VERBOSE[23613] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jun 9 14:11:26] VERBOSE[23613] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 9 14:11:26] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.249:16142 [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: We have an owner, now see if we need to change this call [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Updating call counter for outgoing call [Jun 9 14:11:26] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/402 [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: build_route: Contact hop: ;flow-id=1 [Jun 9 14:11:26] VERBOSE[23613] logger.c: list_route: hop: [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Strict routing enforced for session 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net [Jun 9 14:11:26] VERBOSE[23613] logger.c: set_destination: Parsing for address/port to send to [Jun 9 14:11:26] VERBOSE[23613] logger.c: set_destination: set destination to 192.168.161.249, port 5060 [Jun 9 14:11:26] VERBOSE[23613] logger.c: Transmitting (no NAT) to 192.168.161.249:5060: ACK sip:402@192.168.161.249:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK18d5505b;rport From: "Toestel 401" ;tag=as08c126ff To: ;tag=18rpfhgert Contact: Call-ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net CSeq: 102 ACK User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Toestel 401" ;privacy=off;screen=no Content-Length: 0 --- [Jun 9 14:11:26] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402 [Jun 9 14:11:26] DEBUG[23576] chan_sip.c: Checking device state for peer 402 [Jun 9 14:11:26] DEBUG[23576] devicestate.c: Changing state for SIP/402 - state 2 (In use) [Jun 9 14:11:26] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402 [Jun 9 14:11:26] DEBUG[23576] chan_sip.c: Checking device state for peer 402 [Jun 9 14:11:26] DEBUG[25840] devicestate.c: Notification of state change to be queued on device/channel SIP/402-08943748 [Jun 9 14:11:26] DEBUG[25840] devicestate.c: Notification of state change to be queued on device/channel SIP/402 [Jun 9 14:11:26] VERBOSE[25840] logger.c: -- SIP/402-08943748 answered SIP/401-088e0b60 [Jun 9 14:11:26] DEBUG[25840] devicestate.c: Notification of state change to be queued on device/channel SIP/401-088e0b60 [Jun 9 14:11:26] DEBUG[25840] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:26] DEBUG[25840] chan_sip.c: SIP answering channel: SIP/401-088e0b60 [Jun 9 14:11:26] DEBUG[25840] chan_sip.c: Setting framing from config on incoming call [Jun 9 14:11:26] DEBUG[25840] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Jun 9 14:11:26] DEBUG[25840] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 9 14:11:26] VERBOSE[25840] logger.c: Audio is at 192.168.161.100 port 11368 [Jun 9 14:11:26] VERBOSE[25840] logger.c: Adding codec 0x8 (alaw) to SDP [Jun 9 14:11:26] VERBOSE[25840] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 9 14:11:26] DEBUG[25840] chan_sip.c: -- Done with adding codecs to SDP [Jun 9 14:11:26] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=33) [Jun 9 14:11:26] DEBUG[25840] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 9 14:11:26] VERBOSE[25840] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-elydtnhmgh3o;received=192.168.161.250;rport=5060 From: "Toestel 401" ;tag=5gx8xpwpe4 To: ;tag=as145459f4 Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 2 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 23546 23546 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 11368 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 <------------> [Jun 9 14:11:26] DEBUG[25840] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:26] VERBOSE[25840] logger.c: -- Packet2Packet bridging SIP/401-088e0b60 and SIP/402-08943748 [Jun 9 14:11:26] DEBUG[23615] app_queue.c: Device 'SIP/402' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:26] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402-08943748 [Jun 9 14:11:26] DEBUG[23576] chan_sip.c: Checking device state for peer 402-08943748 [Jun 9 14:11:26] DEBUG[23576] devicestate.c: Changing state for SIP/402-08943748 - state 4 (Invalid) [Jun 9 14:11:26] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402 [Jun 9 14:11:26] DEBUG[23576] chan_sip.c: Checking device state for peer 402 [Jun 9 14:11:26] DEBUG[23576] devicestate.c: Changing state for SIP/402 - state 2 (In use) [Jun 9 14:11:26] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 402 [Jun 9 14:11:26] DEBUG[23576] chan_sip.c: Checking device state for peer 402 [Jun 9 14:11:26] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401-088e0b60 [Jun 9 14:11:26] DEBUG[23576] chan_sip.c: Checking device state for peer 401-088e0b60 [Jun 9 14:11:26] DEBUG[23615] app_queue.c: Device 'SIP/402-08943748' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:26] DEBUG[23615] app_queue.c: Device 'SIP/402' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:26] DEBUG[23576] devicestate.c: Changing state for SIP/401-088e0b60 - state 4 (Invalid) [Jun 9 14:11:26] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:26] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:26] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 2 (In use) [Jun 9 14:11:26] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:26] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:26] DEBUG[23615] app_queue.c: Device 'SIP/401-088e0b60' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:26] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> ACK sip:402@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-xc13app69k20;rport From: "Toestel 401" ;tag=5gx8xpwpe4 To: ;tag=as145459f4 Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 0: ACK sip:402@192.168.161.100 SIP/2.0 (35) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-xc13app69k20;rport (71) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 401" ;tag=5gx8xpwpe4 (60) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=as145459f4 (55) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617340bfe0-92e68vgyoo4t (34) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 5: CSeq: 2 ACK (11) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 8: Content-Length: 0 (17) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Header 9: (0) [Jun 9 14:11:26] VERBOSE[23613] logger.c: --- (9 headers 0 lines) --- [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: = No match Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c617340bfe0-92e68vgyoo4t Their Tag 5gx8xpwpe4 Our tag: as145459f4 [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1388 [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Stopping retransmission on '3c617340bfe0-92e68vgyoo4t' of Response 2: Match Found [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Auto destroying SIP dialog '3c26701a4fb9-rd37qdoo1uyb' [Jun 9 14:11:26] DEBUG[23613] chan_sip.c: Destroying SIP dialog 3c26701a4fb9-rd37qdoo1uyb [Jun 9 14:11:26] VERBOSE[23613] logger.c: Really destroying SIP dialog '3c26701a4fb9-rd37qdoo1uyb' Method: REGISTER [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:26] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> INVITE sip:402@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-27budkdc5bpi;rport From: "Toestel 401" ;tag=5gx8xpwpe4 To: ;tag=as145459f4 Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 3 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.1.33 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 226 v=0 o=root 1521396104 1521396105 IN IP4 192.168.161.250 s=call c=IN IP4 192.168.161.250 t=0 0 m=audio 17384 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendonly <-------------> [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 0: INVITE sip:402@192.168.161.100 SIP/2.0 (38) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-27budkdc5bpi;rport (71) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 401" ;tag=5gx8xpwpe4 (60) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=as145459f4 (55) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617340bfe0-92e68vgyoo4t (34) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 5: CSeq: 3 INVITE (14) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 8: P-Key-Flags: resolution="31x13", keys="4" (41) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 9: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 10: Accept: application/sdp (23) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 12: Allow-Events: talk, hold, refer, call-info (42) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 13: Supported: timer, 100rel, replaces, from-change (47) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 14: Session-Expires: 3600;refresher=uas (35) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 15: Min-SE: 90 (10) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 16: Content-Type: application/sdp (29) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 17: Content-Length: 226 (19) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 18: (0) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: v=0 (3) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: o=root 1521396104 1521396105 IN IP4 192.168.161.250 (51) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: s=call (6) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: c=IN IP4 192.168.161.250 (24) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: t=0 0 (5) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: m=audio 17384 RTP/AVP 8 101 (27) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: a=ptime:20 (10) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Line: a=sendonly (10) [Jun 9 14:11:27] VERBOSE[23613] logger.c: --- (18 headers 11 lines) --- [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: = No match Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c617340bfe0-92e68vgyoo4t Their Tag 5gx8xpwpe4 Our tag: as145459f4 [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jun 9 14:11:27] VERBOSE[23613] logger.c: Sending to 192.168.161.250 : 5060 (NAT) [Jun 9 14:11:27] VERBOSE[23613] logger.c: Found RTP audio format 8 [Jun 9 14:11:27] VERBOSE[23613] logger.c: Found RTP audio format 101 [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Peer doesn't provide T.38 UDPTL [Jun 9 14:11:27] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.250:17384 [Jun 9 14:11:27] VERBOSE[23613] logger.c: Found audio description format pcma for ID 8 [Jun 9 14:11:27] VERBOSE[23613] logger.c: Found audio description format telephone-event for ID 101 [Jun 9 14:11:27] VERBOSE[23613] logger.c: Got unsupported a:fmtp in SDP offer [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: T38 state changed to 0 on channel SIP/401-088e0b60 [Jun 9 14:11:27] VERBOSE[23613] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jun 9 14:11:27] VERBOSE[23613] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 9 14:11:27] VERBOSE[23613] logger.c: Peer audio RTP is at port 192.168.161.250:17384 [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: We have an owner, now see if we need to change this call [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Got a SIP re-invite for call 3c617340bfe0-92e68vgyoo4t [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: SIP/401-088e0b60: This call is UP.... [Jun 9 14:11:27] VERBOSE[23613] logger.c: <--- Transmitting (NAT) to 192.168.161.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-27budkdc5bpi;received=192.168.161.250;rport=5060 From: "Toestel 401" ;tag=5gx8xpwpe4 To: ;tag=as145459f4 Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 3 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Setting framing from config on incoming call [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jun 9 14:11:27] VERBOSE[23613] logger.c: Audio is at 192.168.161.100 port 11368 [Jun 9 14:11:27] VERBOSE[23613] logger.c: Adding codec 0x8 (alaw) to SDP [Jun 9 14:11:27] VERBOSE[23613] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: -- Done with adding codecs to SDP [Jun 9 14:11:27] DEBUG[23613] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=33) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jun 9 14:11:27] VERBOSE[23613] logger.c: <--- Reliably Transmitting (NAT) to 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-27budkdc5bpi;received=192.168.161.250;rport=5060 From: "Toestel 401" ;tag=5gx8xpwpe4 To: ;tag=as145459f4 Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 3 INVITE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 246 v=0 o=root 23546 23547 IN IP4 192.168.161.100 s=session c=IN IP4 192.168.161.100 t=0 0 m=audio 11368 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:27] VERBOSE[25840] logger.c: -- Started music on hold, class 'default', on SIP/402-08943748 [Jun 9 14:11:27] DEBUG[25840] channel.c: Scheduling timer at 160 sample intervals [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:27] DEBUG[25840] channel.c: Generator got voice, switching to phase locked mode [Jun 9 14:11:27] DEBUG[25840] channel.c: Scheduling timer at 0 sample intervals [Jun 9 14:11:27] DEBUG[25840] res_musiconhold.c: SIP/402-08943748 Opened file 2 '/var/lib/asterisk/moh/default/worldmix' [Jun 9 14:11:27] DEBUG[25840] rtp.c: Ooh, format changed from unknown to alaw [Jun 9 14:11:27] DEBUG[25840] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> ACK sip:402@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-yq5yss6p48se;rport From: "Toestel 401" ;tag=5gx8xpwpe4 To: ;tag=as145459f4 Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 3 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 0: ACK sip:402@192.168.161.100 SIP/2.0 (35) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-yq5yss6p48se;rport (71) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 401" ;tag=5gx8xpwpe4 (60) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=as145459f4 (55) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617340bfe0-92e68vgyoo4t (34) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 5: CSeq: 3 ACK (11) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 8: Content-Length: 0 (17) [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Header 9: (0) [Jun 9 14:11:27] VERBOSE[23613] logger.c: --- (9 headers 0 lines) --- [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: = No match Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c617340bfe0-92e68vgyoo4t Their Tag 5gx8xpwpe4 Our tag: as145459f4 [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1389 [Jun 9 14:11:27] DEBUG[23613] chan_sip.c: Stopping retransmission on '3c617340bfe0-92e68vgyoo4t' of Response 3: Match Found [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:27] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:27] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] DEBUG[25840] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) [Jun 9 14:11:28] DEBUG[25839] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25) [Jun 9 14:11:28] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> REFER sip:400@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-h15al7a8el2c;rport From: ;tag=j41casm0gq To: "Toestel 400" ;tag=as7bac7baf Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 2 REFER Max-Forwards: 70 Contact: ;flow-id=1 Refer-To: sip:402@192.168.161.100?Replaces=3c617340bfe0-92e68vgyoo4t%3Bto-tag%3Das145459f4%3Bfrom-tag%3D5gx8xpwpe4 Referred-By: sip:401@pbx.example.net User-Agent: snom360/7.1.33 Content-Length: 0 <-------------> [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 0: REFER sip:400@192.168.161.100 SIP/2.0 (37) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-h15al7a8el2c;rport (71) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 2: From: ;tag=j41casm0gq (51) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 3: To: "Toestel 400" ;tag=as7bac7baf (58) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net (57) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 5: CSeq: 2 REFER (13) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 8: Refer-To: sip:402@192.168.161.100?Replaces=3c617340bfe0-92e68vgyoo4t%3Bto-tag%3Das145459f4%3Bfrom-tag%3D5gx8xpwpe4 (114) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 9: Referred-By: sip:401@pbx.example.net (36) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 10: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 11: Content-Length: 0 (17) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 12: (0) [Jun 9 14:11:28] VERBOSE[23613] logger.c: --- (12 headers 0 lines) --- [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c617340bfe0-92e68vgyoo4t Their Tag 5gx8xpwpe4 Our tag: as145459f4 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 868521015562008142045@192.168.161.3 Their Tag 1c923705775 Our tag: as704c8058 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = Found Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Jun 9 14:11:28] VERBOSE[23613] logger.c: Call 28aa6e4d058373f806130cd33cee1f63@pbx.example.net got a SIP call transfer from caller: (REFER)! [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 3c617340bfe0-92e68vgyoo4t (No check of from/to tags) [Jun 9 14:11:28] VERBOSE[23613] logger.c: SIP transfer to extension 402@flex-14 by 401@pbx.example.net [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: SIP attended transfer: Transferer channel SIP/401-089415f0, transferee channel SIP/400-08908960 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/400-08908960' [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Looking for callid 3c617340bfe0-92e68vgyoo4t (fromtag 5gx8xpwpe4 totag as145459f4) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Matched INCOMING call - their tag is 5gx8xpwpe4 Our tag is as145459f4 [Jun 9 14:11:28] VERBOSE[23613] logger.c: <--- Transmitting (NAT) to 192.168.161.250:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-h15al7a8el2c;received=192.168.161.250;rport=5060 From: ;tag=j41casm0gq To: "Toestel 400" ;tag=as7bac7baf Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 2 REFER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: SIP attended transfer: trying to bridge SIP/401-088e0b60 and SIP/400-08908960 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Sip transfer:-------------------- [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: -- Transferer to PBX channel: SIP/401-089415f0 State Up [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: -- Transferer to PBX second channel (target): SIP/401-088e0b60 State Up [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: -- Bridged call to transferee: SIP/400-08908960 State Up [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: -- Bridged call to transfer target: SIP/402-08943748 State Up [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: -- END Sip transfer:-------------------- [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: SIP transfer: Four channels to handle [Jun 9 14:11:28] VERBOSE[23613] logger.c: -- Stopped music on hold on SIP/400-08908960 [Jun 9 14:11:28] DEBUG[23613] channel.c: Set channel SIP/400-08908960 to write format slin [Jun 9 14:11:28] DEBUG[23613] channel.c: Scheduling timer at 0 sample intervals [Jun 9 14:11:28] VERBOSE[23613] logger.c: -- Stopped music on hold on SIP/402-08943748 [Jun 9 14:11:28] DEBUG[23613] channel.c: Scheduling timer at 0 sample intervals [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: SIP transfer: trying to masquerade SIP/400-08908960 into SIP/401-088e0b60 [Jun 9 14:11:28] DEBUG[23613] channel.c: Planning to masquerade channel SIP/400-08908960 into the structure of SIP/401-088e0b60 [Jun 9 14:11:28] DEBUG[23613] channel.c: Done planning to masquerade channel SIP/400-08908960 into the structure of SIP/401-088e0b60 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Strict routing enforced for session 28aa6e4d058373f806130cd33cee1f63@pbx.example.net [Jun 9 14:11:28] VERBOSE[23613] logger.c: set_destination: Parsing for address/port to send to [Jun 9 14:11:28] VERBOSE[23613] logger.c: set_destination: set destination to 192.168.161.250, port 5060 [Jun 9 14:11:28] VERBOSE[23613] logger.c: Reliably Transmitting (NAT) to 192.168.161.250:5060: NOTIFY sip:401@192.168.161.250:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK5fc45185;rport From: "Toestel 400" ;tag=as7bac7baf To: ;tag=j41casm0gq Contact: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 103 NOTIFY User-Agent: atCOM PBX Max-Forwards: 70 Remote-Party-ID: "Toestel 400" ;privacy=off;screen=no Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: SIP attended transfer: Unlocking channel SIP/401-088e0b60 [Jun 9 14:11:28] DEBUG[25840] channel.c: Actually Masquerading SIP/400-08908960(6) into the structure of SIP/401-088e0b60(6) [Jun 9 14:11:28] DEBUG[25840] channel.c: Got clone lock for masquerade on 'SIP/400-08908960' at 0x87ef720 [Jun 9 14:11:28] DEBUG[25840] chan_sip.c: SIP Fixup: New owner for dialogue 3c617340bfe0-92e68vgyoo4t: SIP/400-08908960 (Old parent: SIP/400-08908960) [Jun 9 14:11:28] DEBUG[25840] chan_sip.c: Hangup call SIP/400-08908960, SIP callid 3c617340bfe0-92e68vgyoo4t) [Jun 9 14:11:28] DEBUG[25840] chan_sip.c: update_call_counter(401) - decrement call limit counter on hangup [Jun 9 14:11:28] DEBUG[25840] chan_sip.c: Updating call counter for incoming call [Jun 9 14:11:28] DEBUG[25840] chan_sip.c: Call from peer '401' removed from call limit 30 [Jun 9 14:11:28] DEBUG[25840] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:28] VERBOSE[25840] logger.c: Scheduling destruction of SIP dialog '3c617340bfe0-92e68vgyoo4t' in 32000 ms (Method: ACK) [Jun 9 14:11:28] DEBUG[25840] chan_sip.c: Strict routing enforced for session 3c617340bfe0-92e68vgyoo4t [Jun 9 14:11:28] VERBOSE[25840] logger.c: set_destination: Parsing for address/port to send to [Jun 9 14:11:28] VERBOSE[25840] logger.c: set_destination: set destination to 192.168.161.250, port 5060 [Jun 9 14:11:28] VERBOSE[25840] logger.c: Reliably Transmitting (NAT) to 192.168.161.250:5060: BYE sip:401@192.168.161.250:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK5617969c;rport From: ;tag=as145459f4 To: "Toestel 401" ;tag=5gx8xpwpe4 Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 102 BYE User-Agent: atCOM PBX Max-Forwards: 70 Content-Length: 0 --- [Jun 9 14:11:28] DEBUG[25840] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Jun 9 14:11:28] DEBUG[25840] channel.c: Set channel SIP/400-08908960 to write format alaw [Jun 9 14:11:28] DEBUG[25840] channel.c: Set channel SIP/400-08908960 to read format alaw [Jun 9 14:11:28] DEBUG[25840] channel.c: Putting channel SIP/400-08908960 in 8/8 formats [Jun 9 14:11:28] DEBUG[25840] chan_sip.c: SIP Fixup: New owner for dialogue 3c617304b5ff-hqq001xrvvkf: SIP/400-08908960 (Old parent: SIP/401-088e0b60) [Jun 9 14:11:28] DEBUG[25840] channel.c: Released clone lock on 'SIP/401-088e0b60' [Jun 9 14:11:28] DEBUG[25840] channel.c: Done Masquerading SIP/400-08908960 (6) [Jun 9 14:11:28] DEBUG[25840] rtp.c: Oooh, formats changed, backing out [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 2 (In use) [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[25839] channel.c: Didn't get a frame from channel: SIP/401-088e0b60 [Jun 9 14:11:28] DEBUG[25839] channel.c: Bridge stops bridging channels SIP/401-088e0b60 and SIP/401-089415f0 [Jun 9 14:11:28] DEBUG[25839] channel.c: Hanging up channel 'SIP/401-089415f0' [Jun 9 14:11:28] DEBUG[25839] chan_sip.c: update_call_counter(401) - decrement call limit counter on hangup [Jun 9 14:11:28] DEBUG[25839] chan_sip.c: Updating call counter for outgoing call [Jun 9 14:11:28] DEBUG[25839] chan_sip.c: Call to peer '401' removed from call limit 30 [Jun 9 14:11:28] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:28] DEBUG[25839] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 28aa6e4d058373f806130cd33cee1f63@pbx.example.net. [Jun 9 14:11:28] VERBOSE[25839] logger.c: Scheduling destruction of SIP dialog '28aa6e4d058373f806130cd33cee1f63@pbx.example.net' in 32000 ms (Method: REFER) [Jun 9 14:11:28] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/401-089415f0 [Jun 9 14:11:28] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:28] DEBUG[25839] rtp.c: Channel '' has no RTP, not doing anything [Jun 9 14:11:28] DEBUG[25839] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jun 9 14:11:28] DEBUG[25839] pbx.c: Spawn extension (user-01,401,1) exited non-zero on 'SIP/401-088e0b60' [Jun 9 14:11:28] VERBOSE[25839] logger.c: == Spawn extension (user-01, 401, 1) exited non-zero on 'SIP/401-088e0b60' [Jun 9 14:11:28] DEBUG[25839] channel.c: Soft-Hanging up channel 'SIP/401-088e0b60' [Jun 9 14:11:28] DEBUG[25839] pbx.c: Launching 'Hangup' [Jun 9 14:11:28] VERBOSE[25839] logger.c: -- Executing [h@user-01:1] Hangup("SIP/401-088e0b60", "") in new stack [Jun 9 14:11:28] DEBUG[25839] pbx.c: Spawn extension (user-01,h,1) exited non-zero on 'SIP/401-088e0b60' [Jun 9 14:11:28] VERBOSE[25839] logger.c: == Spawn extension (user-01, h, 1) exited non-zero on 'SIP/401-088e0b60' [Jun 9 14:11:28] DEBUG[25839] channel.c: Hanging up zombie 'SIP/401-088e0b60' [Jun 9 14:11:28] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/401-088e0b60 [Jun 9 14:11:28] DEBUG[25839] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:28] DEBUG[25839] cdr.c: Dropping CDR ! [Jun 9 14:11:28] NOTICE[25839] cdr.c: CDR on channel 'SIP/401-089415f0' not posted [Jun 9 14:11:28] NOTICE[25839] cdr.c: CDR on channel 'SIP/400-08908960' not posted [Jun 9 14:11:28] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 1 (Not in use) [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401-089415f0 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401-089415f0 [Jun 9 14:11:28] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 9 14:11:28] DEBUG[23576] devicestate.c: Changing state for SIP/401-089415f0 - state 4 (Invalid) [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 1 (Not in use) [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401-088e0b60 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401-088e0b60 [Jun 9 14:11:28] DEBUG[23615] app_queue.c: Device 'SIP/401-089415f0' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:28] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 9 14:11:28] DEBUG[23576] devicestate.c: Changing state for SIP/401-088e0b60 - state 4 (Invalid) [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 1 (Not in use) [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[23615] app_queue.c: Device 'SIP/401-088e0b60' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jun 9 14:11:28] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 9 14:11:28] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK5fc45185;rport=5060 From: "Toestel 400" ;tag=as7bac7baf To: ;tag=j41casm0gq Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 103 NOTIFY Contact: ;flow-id=1 Content-Length: 0 <-------------> [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 Ok (14) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK5fc45185;rport=5060 (71) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 2: From: "Toestel 400" ;tag=as7bac7baf (60) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 3: To: ;tag=j41casm0gq (49) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net (57) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 5: CSeq: 103 NOTIFY (16) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 7: Content-Length: 0 (17) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 8: (0) [Jun 9 14:11:28] VERBOSE[23613] logger.c: --- (8 headers 0 lines) --- [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c617340bfe0-92e68vgyoo4t Their Tag 5gx8xpwpe4 Our tag: as145459f4 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 868521015562008142045@192.168.161.3 Their Tag 1c923705775 Our tag: as704c8058 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = Found Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1391 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Stopping retransmission on '28aa6e4d058373f806130cd33cee1f63@pbx.example.net' of Request 103: Match Found [Jun 9 14:11:28] VERBOSE[23613] logger.c: SIP Response message for INCOMING dialog NOTIFY arrived [Jun 9 14:11:28] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> BYE sip:400@192.168.161.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-6fovbxytnnsf;rport From: ;tag=j41casm0gq To: "Toestel 400" ;tag=as7bac7baf Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.33 RTP-RxStat: Total_Rx_Pkts=85,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=88,Tx_Pkts=0,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 0: BYE sip:400@192.168.161.100 SIP/2.0 (35) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-6fovbxytnnsf;rport (71) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 2: From: ;tag=j41casm0gq (51) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 3: To: "Toestel 400" ;tag=as7bac7baf (58) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net (57) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 5: CSeq: 3 BYE (11) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1 (49) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 8: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 9: RTP-RxStat: Total_Rx_Pkts=85,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (75) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 10: RTP-TxStat: Total_Tx_Pkts=88,Tx_Pkts=0,Remote_Tx_Pkts=0 (55) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 11: Content-Length: 0 (17) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 12: (0) [Jun 9 14:11:28] VERBOSE[23613] logger.c: --- (12 headers 0 lines) --- [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c617340bfe0-92e68vgyoo4t Their Tag 5gx8xpwpe4 Our tag: as145459f4 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 868521015562008142045@192.168.161.3 Their Tag 1c923705775 Our tag: as704c8058 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = Found Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jun 9 14:11:28] VERBOSE[23613] logger.c: Sending to 192.168.161.250 : 5060 (NAT) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Setting SIP_ALREADYGONE on dialog 28aa6e4d058373f806130cd33cee1f63@pbx.example.net [Jun 9 14:11:28] VERBOSE[23613] logger.c: Scheduling destruction of SIP dialog '28aa6e4d058373f806130cd33cee1f63@pbx.example.net' in 32000 ms (Method: BYE) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Received bye, no owner, selfdestruct soon. [Jun 9 14:11:28] VERBOSE[23613] logger.c: <--- Transmitting (NAT) to 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.250:5060;branch=z9hG4bK-6fovbxytnnsf;received=192.168.161.250;rport=5060 From: ;tag=j41casm0gq To: "Toestel 400" ;tag=as7bac7baf Call-ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net CSeq: 3 BYE User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:28] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK5617969c;rport=5060 From: ;tag=as145459f4 To: "Toestel 401" ;tag=5gx8xpwpe4 Call-ID: 3c617340bfe0-92e68vgyoo4t CSeq: 102 BYE Contact: ;flow-id=1 User-Agent: snom360/7.1.33 RTP-RxStat: Total_Rx_Pkts=78,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=85,Tx_Pkts=0,Remote_Tx_Pkts=0 Content-Length: 0 <-------------> [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.100:5060;branch=z9hG4bK5617969c;rport=5060 (71) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 2: From: ;tag=as145459f4 (57) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 3: To: "Toestel 401" ;tag=5gx8xpwpe4 (58) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c617340bfe0-92e68vgyoo4t (34) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 5: CSeq: 102 BYE (13) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 6: Contact: ;flow-id=1 (49) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 7: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 8: RTP-RxStat: Total_Rx_Pkts=78,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (75) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 9: RTP-TxStat: Total_Tx_Pkts=85,Tx_Pkts=0,Remote_Tx_Pkts=0 (55) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 10: Content-Length: 0 (17) [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Header 11: (0) [Jun 9 14:11:28] VERBOSE[23613] logger.c: --- (11 headers 0 lines) --- [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = No match Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c617340bfe0-92e68vgyoo4t Their Tag 5gx8xpwpe4 Our tag: as145459f4 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1393 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Stopping retransmission on '3c617340bfe0-92e68vgyoo4t' of Request 102: Match Found [Jun 9 14:11:28] VERBOSE[23613] logger.c: SIP Response message for INCOMING dialog BYE arrived [Jun 9 14:11:28] VERBOSE[23613] logger.c: Really destroying SIP dialog '3c617340bfe0-92e68vgyoo4t' Method: ACK [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Updating call counter for incoming call [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: Call from peer '401' removed from call limit 30 [Jun 9 14:11:28] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/401 [Jun 9 14:11:28] DEBUG[23613] chan_sip.c: This call did not properly clean up call limits. Call ID 3c617340bfe0-92e68vgyoo4t [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[23576] devicestate.c: Changing state for SIP/401 - state 1 (Not in use) [Jun 9 14:11:28] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 401 [Jun 9 14:11:28] DEBUG[23576] chan_sip.c: Checking device state for peer 401 [Jun 9 14:11:28] DEBUG[23615] app_queue.c: Device 'SIP/401' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 9 14:11:30] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.247:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-t41e9zp9s7y2;rport From: "Sebastiaan" ;tag=i0s2kmgwwh To: "Sebastiaan" Call-ID: 3c26701a1d89-bca3s0s1uv13 CSeq: 20269 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="" User-Agent: snom360/7.1.33 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.161.247 Expires: 600 Content-Length: 0 <-------------> [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-t41e9zp9s7y2;rport (71) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 2: From: "Sebastiaan" ;tag=i0s2kmgwwh (59) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 3: To: "Sebastiaan" (42) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c26701a1d89-bca3s0s1uv13 (34) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 5: CSeq: 20269 REGISTER (20) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="" (119) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 8: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 9: Supported: gruu (15) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 10: Allow-Events: dialog (20) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 11: X-Real-IP: 192.168.161.247 (26) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 12: Expires: 600 (12) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 13: Content-Length: 0 (17) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 14: (0) [Jun 9 14:11:30] VERBOSE[23613] logger.c: --- (14 headers 0 lines) --- [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 36ec10c97d7bcc0924ad7f5e3c2bb30b@pbx.example.net Their Tag 18rpfhgert Our tag: as08c126ff [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 868521015562008142045@192.168.161.3 Their Tag 1c923705775 Our tag: as704c8058 [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 28aa6e4d058373f806130cd33cee1f63@pbx.example.net Their Tag j41casm0gq Our tag: as7bac7baf [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c617304b5ff-hqq001xrvvkf Their Tag ow35sk9e64 Our tag: as03ae9445 [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743299462942008195958@192.168.161.237 Their Tag 1c1502243045 Our tag: as0f2a9650 [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743318622942008195958@192.168.161.237 Their Tag 1c1502214188 Our tag: as2363565d [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743312262942008195958@192.168.161.237 Their Tag 1c1502198413 Our tag: as14e39483 [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 4743305952942008195958@192.168.161.237 Their Tag 1c1502133902 Our tag: as3584dd5b [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ada48-mkwj7m38q3nu Their Tag wbtjjopkwe Our tag: as28583bdb [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad395-ajo6vj1qc0vm Their Tag 97qngh5alh Our tag: as39395c26 [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad4c6-phe3ytd26k13 Their Tag hf67o5uhr4 Our tag: as73260cbd [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad52a-gwy8p5tlpim3 Their Tag sjongh64mn Our tag: as56f28897 [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = No match Their Call ID: 3c26701ad45e-7fxbtgnl1u8t Their Tag tstmut2dxd Our tag: as602e2d77 [Jun 9 14:11:30] DEBUG[23613] acl.c: ##### Testing 192.168.161.247 with 192.168.161.0 [Jun 9 14:11:30] DEBUG[23613] acl.c: ##### Testing 192.168.161.247 with 192.168.200.0 [Jun 9 14:11:30] DEBUG[23613] acl.c: ##### Testing 192.168.161.247 with 192.168.1.0 [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Allocating new SIP dialog for 3c26701a1d89-bca3s0s1uv13 - REGISTER (No RTP) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jun 9 14:11:30] VERBOSE[23613] logger.c: Using latest REGISTER request as basis request [Jun 9 14:11:30] VERBOSE[23613] logger.c: Sending to 192.168.161.247 : 5060 (NAT) [Jun 9 14:11:30] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.247:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-t41e9zp9s7y2;received=192.168.161.247;rport=5060 From: "Sebastiaan" ;tag=i0s2kmgwwh To: "Sebastiaan" Call-ID: 3c26701a1d89-bca3s0s1uv13 CSeq: 20269 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:30] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.247:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-t41e9zp9s7y2;received=192.168.161.247;rport=5060 From: "Sebastiaan" ;tag=i0s2kmgwwh To: "Sebastiaan" ;tag=as4d77f5e9 Call-ID: 3c26701a1d89-bca3s0s1uv13 CSeq: 20269 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="siprealm", nonce="5d1f7d23" Content-Length: 0 <------------> [Jun 9 14:11:30] VERBOSE[23613] logger.c: Scheduling destruction of SIP dialog '3c26701a1d89-bca3s0s1uv13' in 32000 ms (Method: REGISTER) [Jun 9 14:11:30] VERBOSE[23613] logger.c: <--- SIP read from 192.168.161.247:5060 ---> REGISTER sip:pbx.example.net SIP/2.0 Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-abm7131p5zy4;rport From: "Sebastiaan" ;tag=i0s2kmgwwh To: "Sebastiaan" Call-ID: 3c26701a1d89-bca3s0s1uv13 CSeq: 20270 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="" User-Agent: snom360/7.1.33 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.161.247 Authorization: Digest username="404",realm="siprealm",nonce="5d1f7d23",uri="sip:pbx.example.net",response="b46e547c3169da7c8ec9ac01f1fd4f0d",algorithm=MD5 Expires: 600 Content-Length: 0 <-------------> [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 0: REGISTER sip:pbx.example.net SIP/2.0 (36) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-abm7131p5zy4;rport (71) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 2: From: "Sebastiaan" ;tag=i0s2kmgwwh (59) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 3: To: "Sebastiaan" (42) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 4: Call-ID: 3c26701a1d89-bca3s0s1uv13 (34) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 5: CSeq: 20270 REGISTER (20) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="" (119) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 8: User-Agent: snom360/7.1.33 (26) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 9: Supported: gruu (15) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 10: Allow-Events: dialog (20) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 11: X-Real-IP: 192.168.161.247 (26) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 12: Authorization: Digest username="404",realm="siprealm",nonce="5d1f7d23",uri="sip:pbx.example.net",response="b46e547c3169da7c8ec9ac01f1fd4f0d",algorithm=MD5 (154) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 13: Expires: 600 (12) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 14: Content-Length: 0 (17) [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: Header 15: (0) [Jun 9 14:11:30] VERBOSE[23613] logger.c: --- (15 headers 0 lines) --- [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: = Found Their Call ID: 3c26701a1d89-bca3s0s1uv13 Their Tag i0s2kmgwwh Our tag: as4d77f5e9 [Jun 9 14:11:30] DEBUG[23613] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jun 9 14:11:30] VERBOSE[23613] logger.c: Using latest REGISTER request as basis request [Jun 9 14:11:30] VERBOSE[23613] logger.c: Sending to 192.168.161.247 : 5060 (NAT) [Jun 9 14:11:30] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.247:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-abm7131p5zy4;received=192.168.161.247;rport=5060 From: "Sebastiaan" ;tag=i0s2kmgwwh To: "Sebastiaan" Call-ID: 3c26701a1d89-bca3s0s1uv13 CSeq: 20270 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jun 9 14:11:30] VERBOSE[23613] logger.c: <--- Transmitting (no NAT) to 192.168.161.247:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.161.247:5060;branch=z9hG4bK-abm7131p5zy4;received=192.168.161.247;rport=5060 From: "Sebastiaan" ;tag=i0s2kmgwwh To: "Sebastiaan" ;tag=as4d77f5e9 Call-ID: 3c26701a1d89-bca3s0s1uv13 CSeq: 20270 REGISTER User-Agent: atCOM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 600 Contact: ;expires=600 Date: Mon, 09 Jun 2008 12:11:30 GMT Content-Length: 0 <------------> [Jun 9 14:11:30] DEBUG[23613] devicestate.c: Notification of state change to be queued on device/channel SIP/404 [Jun 9 14:11:30] VERBOSE[23613] logger.c: Scheduling destruction of SIP dialog '3c26701a1d89-bca3s0s1uv13' in 32000 ms (Method: REGISTER) [Jun 9 14:11:30] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 404 [Jun 9 14:11:30] DEBUG[23576] chan_sip.c: Checking device state for peer 404 [Jun 9 14:11:30] DEBUG[23576] devicestate.c: Changing state for SIP/404 - state 1 (Not in use) [Jun 9 14:11:30] DEBUG[23576] devicestate.c: No provider found, checking channel drivers for SIP - 404 [Jun 9 14:11:30] DEBUG[23576] chan_sip.c: Checking device state for peer 404 [Jun 9 14:11:30] DEBUG[23615] app_queue.c: Device 'SIP/404' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jun 9 14:11:34] DEBUG[25872] manager.c: Manager received command 'login' [Jun 9 14:11:34] VERBOSE[25872] logger.c: == Parsing '/etc/asterisk/manager.conf': [Jun 9 14:11:34] DEBUG[25872] config.c: Parsing /etc/asterisk/manager.conf [Jun 9 14:11:34] VERBOSE[25872] logger.c: Found [Jun 9 14:11:34] VERBOSE[25872] logger.c: == Parsing '/etc/asterisk/settings/manager.conf': [Jun 9 14:11:34] DEBUG[25872] config.c: Parsing /etc/asterisk/settings/manager.conf [Jun 9 14:11:34] VERBOSE[25872] logger.c: Found [Jun 9 14:11:34] DEBUG[25872] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer [Jun 9 14:11:34] DEBUG[25872] acl.c: 127.0.0.1/255.255.255.255/255.255.255.255 appended to acl for peer [Jun 9 14:11:34] DEBUG[25872] acl.c: ##### Testing 127.0.0.1 with 0.0.0.0 [Jun 9 14:11:34] DEBUG[25872] acl.c: ##### Testing 127.0.0.1 with 127.0.0.1 [Jun 9 14:11:34] VERBOSE[25872] logger.c: == Manager 'lmt' logged on from 127.0.0.1 [Jun 9 14:11:34] DEBUG[25872] manager.c: Manager received command 'Command'