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- [Feb 9 13:42:31] VERBOSE[1642] logger.c: [Feb 9 13:42:31] Asterisk Event Logger restarted
- [Feb 9 13:42:31] VERBOSE[1642] logger.c: [Feb 9 13:42:31] Asterisk Queue Logger restarted
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
- <--- SIP read from 192.168.10.252:5060 --->
- INVITE sip:10012@192.168.10.10 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- To: <sip:10012@192.168.10.10>
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 1 INVITE
- Max-Forwards: 70
- Contact: <sip:10013@192.168.10.252:5060>;reg-id=1
- X-Serialnumber: 0004134110AC
- P-Key-Flags: resolution="31x13", keys="4"
- User-Agent: snom870/8.3.6
- Accept: application/sdp
- Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
- 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: 374
- v=0
- o=root 1564119033 1564119033 IN IP4 192.168.10.252
- s=call
- c=IN IP4 192.168.10.252
- t=0 0
- m=audio 54030 RTP/AVP 0 8 9 99 3 18 4 101
- a=rtpmap:0 pcmu/8000
- a=rtpmap:8 pcma/8000
- a=rtpmap:9 g722/8000
- a=rtpmap:99 g726-32/8000
- a=rtpmap:3 gsm/8000
- a=rtpmap:18 g729/8000
- a=rtpmap:4 g723/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------->
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: INVITE sip:10012@192.168.10.10 SIP/2.0 (38)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;rport (70)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd (54)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.10> (29)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 1 INVITE (14)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:10013@192.168.10.252:5060>;reg-id=1 (49)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 8: X-Serialnumber: 0004134110AC (28)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 9: P-Key-Flags: resolution="31x13", keys="4" (41)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 10: User-Agent: snom870/8.3.6 (25)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 11: Accept: application/sdp (23)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE (96)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 13: Allow-Events: talk, hold, refer, call-info (42)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 14: Supported: timer, 100rel, replaces, from-change (47)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 15: Session-Expires: 3600;refresher=uas (35)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 16: Min-SE: 90 (10)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 17: Content-Type: application/sdp (29)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 18: Content-Length: 374 (19)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 19: (0)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: v=0 (3)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: o=root 1564119033 1564119033 IN IP4 192.168.10.252 (50)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: s=call (6)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: c=IN IP4 192.168.10.252 (23)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: t=0 0 (5)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: m=audio 54030 RTP/AVP 0 8 9 99 3 18 4 101 (41)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:0 pcmu/8000 (20)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:99 g726-32/8000 (24)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=fmtp:101 0-16 (15)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=ptime:20 (10)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=sendrecv (10)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (19 headers 17 lines) ---
- [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 192.168.10.0
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Allocating new SIP dialog for f1882b3c0d38-xtxggv3ikaiv - INVITE (With RTP)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change"
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found SIP option: -timer-
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Matched SIP option: timer
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found SIP option: -100rel-
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Matched SIP option: 100rel
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found SIP option: -replaces-
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Matched SIP option: replaces
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found SIP option: -from-change-
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Found no match for SIP option: from-change (Please file bug report!)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Sending to 192.168.10.252 : 5060 (NAT)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Using INVITE request as basis request - f1882b3c0d38-xtxggv3ikaiv
- [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 0.0.0.0
- [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 0.0.0.0
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
- <--- Reliably Transmitting (NAT) to 192.168.10.252:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;received=192.168.10.252;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- To: <sip:10012@192.168.10.10>;tag=as2d7f3984
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 1 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="629537b8"
- Content-Length: 0
- <------------>
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Scheduling destruction of SIP dialog 'f1882b3c0d38-xtxggv3ikaiv' in 32000 ms (Method: INVITE)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found user '10013'
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
- <--- SIP read from 192.168.10.252:5060 --->
- ACK sip:10012@192.168.10.10 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- To: <sip:10012@192.168.10.10>;tag=as2d7f3984
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 1 ACK
- Max-Forwards: 70
- Contact: <sip:10013@192.168.10.252:5060>;reg-id=1
- Content-Length: 0
- <------------->
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: ACK sip:10012@192.168.10.10 SIP/2.0 (35)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-wimizwgzqxdn;rport (70)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd (54)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.10>;tag=as2d7f3984 (44)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 1 ACK (11)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:10013@192.168.10.252:5060>;reg-id=1 (49)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 8: Content-Length: 0 (17)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 9: (0)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (9 headers 0 lines) ---
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #230
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Stopping retransmission on 'f1882b3c0d38-xtxggv3ikaiv' of Response 1: Match Found
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
- <--- SIP read from 192.168.10.252:5060 --->
- INVITE sip:10012@192.168.10.10 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- To: <sip:10012@192.168.10.10>
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 2 INVITE
- Max-Forwards: 70
- Contact: <sip:10013@192.168.10.252:5060>;reg-id=1
- X-Serialnumber: 0004134110AC
- P-Key-Flags: resolution="31x13", keys="4"
- User-Agent: snom870/8.3.6
- Accept: application/sdp
- Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
- 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="10013",realm="asterisk",nonce="629537b8",uri="sip:10012@192.168.10.10",response="0674eb6676f2bb09c78276ad58a42c88",algorithm=MD5
- Content-Type: application/sdp
- Content-Length: 374
- v=0
- o=root 1564119033 1564119033 IN IP4 192.168.10.252
- s=call
- c=IN IP4 192.168.10.252
- t=0 0
- m=audio 54030 RTP/AVP 0 8 9 99 3 18 4 101
- a=rtpmap:0 pcmu/8000
- a=rtpmap:8 pcma/8000
- a=rtpmap:9 g722/8000
- a=rtpmap:99 g726-32/8000
- a=rtpmap:3 gsm/8000
- a=rtpmap:18 g729/8000
- a=rtpmap:4 g723/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------->
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: INVITE sip:10012@192.168.10.10 SIP/2.0 (38)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;rport (70)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd (54)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.10> (29)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 2 INVITE (14)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:10013@192.168.10.252:5060>;reg-id=1 (49)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 8: X-Serialnumber: 0004134110AC (28)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 9: P-Key-Flags: resolution="31x13", keys="4" (41)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 10: User-Agent: snom870/8.3.6 (25)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 11: Accept: application/sdp (23)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE (96)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 13: Allow-Events: talk, hold, refer, call-info (42)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 14: Supported: timer, 100rel, replaces, from-change (47)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 15: Session-Expires: 3600;refresher=uas (35)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 16: Min-SE: 90 (10)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 17: Proxy-Authorization: Digest username="10013",realm="asterisk",nonce="629537b8",uri="sip:10012@192.168.10.10",response="0674eb6676f2bb09c78276ad58a42c88",algorithm=MD5 (166)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 18: Content-Type: application/sdp (29)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 19: Content-Length: 374 (19)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 20: (0)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: v=0 (3)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: o=root 1564119033 1564119033 IN IP4 192.168.10.252 (50)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: s=call (6)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: c=IN IP4 192.168.10.252 (23)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: t=0 0 (5)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: m=audio 54030 RTP/AVP 0 8 9 99 3 18 4 101 (41)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:0 pcmu/8000 (20)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:99 g726-32/8000 (24)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=fmtp:101 0-16 (15)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=ptime:20 (10)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Line: a=sendrecv (10)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (20 headers 17 lines) ---
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Sending to 192.168.10.252 : 5060 (NAT)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Using INVITE request as basis request - f1882b3c0d38-xtxggv3ikaiv
- [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 0.0.0.0
- [Feb 9 13:42:36] DEBUG[1571] acl.c: ##### Testing 192.168.10.252 with 0.0.0.0
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found user '10013'
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 0
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 8
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 9
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 99
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 3
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 18
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 4
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found RTP audio format 101
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Peer audio RTP is at port 192.168.10.252:54030
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format pcmu for ID 0
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format pcma for ID 8
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format g722 for ID 9
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format g726-32 for ID 99
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format gsm for ID 3
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format g729 for ID 18
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format g723 for ID 4
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Found audio description format telephone-event for ID 101
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Got unsupported a:fmtp in SDP offer
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: T38 state changed to 0 on channel <none>
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Capabilities: us - 0x8 (alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x8 (alaw)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Peer audio RTP is at port 192.168.10.252:54030
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: We're settling with these formats: 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Checking SIP call limits for device 10013
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Updating call counter for incoming call
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Call from peer '10013' is 1 out of 50
- [Feb 9 13:42:36] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] Looking for 10012 in from-internal (domain 192.168.10.10)
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 2 (In use)
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:36] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '2' (In use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** Our native formats are 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** Joint capabilities are 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** Our capabilities are 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: This channel will not be able to handle video.
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c2d44996af4-i4ocu4zvtwre
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: build_route: Contact hop: <sip:10013@192.168.10.252:5060>;reg-id=1
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: Parsing <sip:10012@192.168.10.253:5060> for address/port to send to
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] list_route: hop: <sip:10013@192.168.10.252:5060>
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: SIP/10013-0993e6e0: New call is still down.... Trying...
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: set destination to 192.168.10.253, port 5060
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
- <--- Transmitting (NAT) to 192.168.10.252:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;received=192.168.10.252;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- To: <sip:10012@192.168.10.10>
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 2 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:10012@192.168.10.10>
- Content-Length: 0
- <------------>
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Reliably Transmitting (NAT) to 192.168.10.253:5060:
- NOTIFY sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2fb5c7af;rport
- From: <sip:10013@192.168.10.10>;tag=as6d58a6fc
- To: <sip:10012@192.168.10.10>;tag=5e5cbuyyd1
- Contact: <sip:10013@192.168.10.10>
- Call-ID: 3c2d44996af4-i4ocu4zvtwre
- CSeq: 113 NOTIFY
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Event: dialog
- Content-Type: application/dialog-info+xml
- Subscription-State: active
- Content-Length: 208
- <?xml version="1.0"?>
- <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:10013@192.168.10.10">
- <dialog id="10013">
- <state>confirmed</state>
- </dialog>
- </dialog-info>
- ---
- [Feb 9 13:42:36] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Extension Changed 10013[ext-local] new state InUse for Notify User 10012
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c27bbfb7594-mfl0ssslkb6l
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: Parsing <sip:10014@192.168.10.251:5060> for address/port to send to
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: set destination to 192.168.10.251, port 5060
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Reliably Transmitting (NAT) to 192.168.10.251:5060:
- NOTIFY sip:10014@192.168.10.251:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2021a1b8;rport
- From: <sip:10013@192.168.10.10>;tag=as647499f8
- To: <sip:10014@192.168.10.10>;tag=wesbz3i941
- Contact: <sip:10013@192.168.10.10>
- Call-ID: 3c27bbfb7594-mfl0ssslkb6l
- CSeq: 113 NOTIFY
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Event: dialog
- Content-Type: application/dialog-info+xml
- Subscription-State: active
- Content-Length: 208
- <?xml version="1.0"?>
- <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:10013@192.168.10.10">
- <dialog id="10013">
- <state>confirmed</state>
- </dialog>
- </dialog-info>
- ---
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Macro'
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [10012@from-internal:1] Macro("SIP/10013-0993e6e0", "exten-vm|novm|10012") in new stack
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Extension Changed 10013[ext-local] new state InUse for Notify User 10014
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Macro'
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 2 (In use)
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:1] Macro("SIP/10013-0993e6e0", "user-callerid") in new stack
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:36] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '2' (In use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:1] Set("SIP/10013-0993e6e0", "AMPUSER=10013") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:2] GotoIf("SIP/10013-0993e6e0", "0?report") in new stack
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Not taking any branch
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'ExecIf'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:3] ExecIf("SIP/10013-0993e6e0", "1|Set|REALCALLERIDNUM=10013") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: ExecIf
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:4] Set("SIP/10013-0993e6e0", "AMPUSER=10013") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is 'Test3'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:5] Set("SIP/10013-0993e6e0", "AMPUSERCIDNAME=Test3") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:6] GotoIf("SIP/10013-0993e6e0", "0?report") in new stack
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Not taking any branch
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '1'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '10013'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:7] Set("SIP/10013-0993e6e0", "AMPUSERCID=10013") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:8] Set("SIP/10013-0993e6e0", "CALLERID(all)="Test3" <10013>") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:9] GotoIf("SIP/10013-0993e6e0", "0?continue") in new stack
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Not taking any branch
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '-1'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '64'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:10] Set("SIP/10013-0993e6e0", "__TTL=64") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:11] GotoIf("SIP/10013-0993e6e0", "1?continue") in new stack
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Goto (macro-user-callerid,s,18)
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '"Test3" <10013>'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'NoOp'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-user-callerid:18] NoOp("SIP/10013-0993e6e0", "Using CallerID "Test3" <10013>") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Noop
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Macro
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:2] Set("SIP/10013-0993e6e0", "RingGroupMethod=none") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:3] Set("SIP/10013-0993e6e0", "VMBOX=novm") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:4] Set("SIP/10013-0993e6e0", "EXTTOCALL=10012") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CFU'
- [Feb 9 13:42:36] DEBUG[1643] func_db.c: DB: CFU/10012 not found in database.
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is ''
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:5] Set("SIP/10013-0993e6e0", "CFUEXT=") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CFB'
- [Feb 9 13:42:36] DEBUG[1643] func_db.c: DB: CFB/10012 not found in database.
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is ''
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:6] Set("SIP/10013-0993e6e0", "CFBEXT=") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '0'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '""'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Set'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:7] Set("SIP/10013-0993e6e0", "RT=""") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Set
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Macro'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:8] Macro("SIP/10013-0993e6e0", "record-enable|10012|IN") in new stack
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-record-enable:1] GotoIf("SIP/10013-0993e6e0", "1?check") in new stack
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Goto (macro-record-enable,s,4)
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Function result is '20100209-134236'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'AGI'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-record-enable:4] AGI("SIP/10013-0993e6e0", "recordingcheck|20100209-134236|1265719356.18") in new stack
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
- <--- SIP read from 192.168.10.251:5060 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2021a1b8;rport=5060
- From: <sip:10013@192.168.10.10>;tag=as647499f8
- To: <sip:10014@192.168.10.10>;tag=wesbz3i941
- Call-ID: 3c27bbfb7594-mfl0ssslkb6l
- CSeq: 113 NOTIFY
- Content-Length: 0
- <------------->
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2021a1b8;rport=5060 (69)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: <sip:10013@192.168.10.10>;tag=as647499f8 (46)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10014@192.168.10.10>;tag=wesbz3i941 (44)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bbfb7594-mfl0ssslkb6l (34)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 113 NOTIFY (16)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: (0)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (7 headers 0 lines) ---
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Acked pending invite 113
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #233
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bbfb7594-mfl0ssslkb6l' of Request 113: Match Found
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] SIP Response message for INCOMING dialog NOTIFY arrived
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
- <--- SIP read from 192.168.10.253:5060 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2fb5c7af;rport=5060
- From: <sip:10013@192.168.10.10>;tag=as6d58a6fc
- To: <sip:10012@192.168.10.10>;tag=5e5cbuyyd1
- Call-ID: 3c2d44996af4-i4ocu4zvtwre
- CSeq: 113 NOTIFY
- Content-Length: 0
- <------------->
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK2fb5c7af;rport=5060 (69)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: <sip:10013@192.168.10.10>;tag=as6d58a6fc (46)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.10>;tag=5e5cbuyyd1 (44)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c2d44996af4-i4ocu4zvtwre (34)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 113 NOTIFY (16)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: (0)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (7 headers 0 lines) ---
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Acked pending invite 113
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #232
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c2d44996af4-i4ocu4zvtwre' of Request 113: Match Found
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] SIP Response message for INCOMING dialog NOTIFY arrived
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] recordingcheck|20100209-134236|1265719356.18: Inbound recording not enabled
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- AGI Script recordingcheck completed, returning 0
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: AGI
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'MacroExit'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-record-enable:5] MacroExit("SIP/10013-0993e6e0", "") in new stack
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: Macro
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Macro'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-exten-vm:9] Macro("SIP/10013-0993e6e0", "dial||tr|10012") in new stack
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'GotoIf'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-dial:1] GotoIf("SIP/10013-0993e6e0", "1?dial") in new stack
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Goto (macro-dial,s,3)
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: GotoIf
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'AGI'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-dial:3] AGI("SIP/10013-0993e6e0", "dialparties.agi") in new stack
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: Starting New Dialparties.agi
- [Feb 9 13:42:36] DEBUG[1646] manager.c: Manager received command 'login'
- [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Parsing '/etc/asterisk/manager.conf': [Feb 9 13:42:36] DEBUG[1646] config.c: Parsing /etc/asterisk/manager.conf
- [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] Found
- [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Parsing '/etc/asterisk/manager_additional.conf': [Feb 9 13:42:36] DEBUG[1646] config.c: Parsing /etc/asterisk/manager_additional.conf
- [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] Found
- [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Parsing '/etc/asterisk/manager_custom.conf': [Feb 9 13:42:36] DEBUG[1646] config.c: Parsing /etc/asterisk/manager_custom.conf
- [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] Found
- [Feb 9 13:42:36] DEBUG[1646] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer
- [Feb 9 13:42:36] DEBUG[1646] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer
- [Feb 9 13:42:36] DEBUG[1646] acl.c: ##### Testing 127.0.0.1 with 0.0.0.0
- [Feb 9 13:42:36] DEBUG[1646] acl.c: ##### Testing 127.0.0.1 with 127.0.0.0
- [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Manager 'admin' logged on from 127.0.0.1
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: Caller ID name is 'Test3' number is '10013'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: USE_CONFIRMATION: 'FALSE'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: RINGGROUP_INDEX: ''
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: Methodology of ring is 'none'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: Added extension 10012 to extension map
- [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CF'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: Extension 10012 cf is disabled
- [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'DND'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: Extension 10012 do not disturb is disabled
- [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CFB'
- [Feb 9 13:42:36] DEBUG[1643] db.c: Unable to find key '10012' in family 'CFU'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] > dialparties.agi: extnum 10012 has: cw: 1; hascfb: 0 [] hascfu: 0 []
- [Feb 9 13:42:36] DEBUG[1646] manager.c: Manager received command 'ExtensionState'
- [Feb 9 13:42:36] DEBUG[1646] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:36] DEBUG[1646] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] dialparties.agi: ExtensionState: 0
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: dbset CALLTRACE/10012 to 10013
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- dialparties.agi: Filtered ARG3: 10012
- [Feb 9 13:42:36] DEBUG[1646] manager.c: Manager received command 'Logoff'
- [Feb 9 13:42:36] VERBOSE[1646] logger.c: [Feb 9 13:42:36] == Manager 'admin' logged off from 127.0.0.1
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- AGI Script dialparties.agi completed, returning 0
- [Feb 9 13:42:36] DEBUG[1643] app_macro.c: Executed application: AGI
- [Feb 9 13:42:36] DEBUG[1643] pbx.c: Launching 'Dial'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Executing [s@macro-dial:7] Dial("SIP/10013-0993e6e0", "SIP/10012||tr") in new stack
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Setting NAT on RTP to On
- [Feb 9 13:42:36] DEBUG[1643] acl.c: ##### Testing 192.168.10.253 with 192.168.10.0
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** Our native formats are 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** Joint capabilities are 0x0 (nothing)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** Our capabilities are 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: This channel will not be able to handle video.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALEDTIME.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ANSWEREDTIME.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALEDPEERNAME.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALEDPEERNUMBER.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALSTATUS.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable MACRO_DEPTH.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable AGISTATUS.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ds.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable FILTERED_DIAL.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DIALSTATUS_CW.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Copying hard-transferable variable KEEPCID.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ARG3.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ARG2.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable ARG1.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable MACRO_PRIORITY.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable MACRO_CONTEXT.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable MACRO_EXTEN.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable RT.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable CFBEXT.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable CFUEXT.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable EXTTOCALL.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable VMBOX.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable RingGroupMethod.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Copying hard-transferable variable TTL.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable AMPUSERCID.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable DB_RESULT.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable AMPUSERCIDNAME.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable AMPUSER.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable REALCALLERIDNUM.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable SIPCALLID.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable SIPUSERAGENT.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable SIPDOMAIN.
- [Feb 9 13:42:36] DEBUG[1643] channel.c: Not copying variable SIPURI.
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Outgoing Call for 10012
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Updating call counter for outgoing call
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Call to peer '10012' is 1 out of 50
- [Feb 9 13:42:36] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Our T38 capability (0), joint T38 capability (0)
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: False
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 6 (Ringing)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: ** Our prefcodec: 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] Audio is at 192.168.10.10 port 18198
- [Feb 9 13:42:36] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] Adding codec 0x8 (alaw) to SDP
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] Adding non-codec 0x1 (telephone-event) to SDP
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: -- Done with adding codecs to SDP
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 0: INVITE sip:10012@192.168.10.253:5060 SIP/2.0 (44)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport (64)
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae (54)
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 3: To: <sip:10012@192.168.10.253:5060> (35)
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c27bbfb6fc4-t8z10gt0v8fu
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 4: Contact: <sip:10013@192.168.10.10> (34)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 5: Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10 (55)
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: Parsing <sip:10014@192.168.10.251:5060> for address/port to send to
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 6: CSeq: 102 INVITE (16)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24)
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] set_destination: set destination to 192.168.10.251, port 5060
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 8: Max-Forwards: 70 (16)
- [Feb 9 13:42:36] DEBUG[1565] channel.c: Avoiding initial deadlock for channel '0x9942958'
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 9: Date: Tue, 09 Feb 2010 12:42:36 GMT (35)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 11: Supported: replaces (19)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 12: Content-Type: application/sdp (29)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 13: Content-Length: 240 (19)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Header 14: (0)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: v=0 (3)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: o=root 1562 1562 IN IP4 192.168.10.10 (37)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: s=session (9)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: c=IN IP4 192.168.10.10 (22)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: t=0 0 (5)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: m=audio 18198 RTP/AVP 8 101 (27)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=fmtp:101 0-16 (15)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=silenceSupp:off - - - - (25)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=ptime:20 (10)
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: Line: a=sendrecv (10)
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] Reliably Transmitting (NAT) to 192.168.10.253:5060:
- INVITE sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>
- Contact: <sip:10013@192.168.10.10>
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Date: Tue, 09 Feb 2010 12:42:36 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Type: application/sdp
- Content-Length: 240
- v=0
- o=root 1562 1562 IN IP4 192.168.10.10
- s=session
- c=IN IP4 192.168.10.10
- t=0 0
- m=audio 18198 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
- ---
- [Feb 9 13:42:36] DEBUG[1643] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:36] DEBUG[1565] channel.c: Avoiding initial deadlock for channel '0x9942958'
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- Called 10012
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36]
- <--- Transmitting (NAT) to 192.168.10.252:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;received=192.168.10.252;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- To: <sip:10012@192.168.10.10>;tag=as17f86ff1
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 2 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:10012@192.168.10.10>
- Content-Length: 0
- <------------>
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Sent call-pickup info to peer 10014
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Reliably Transmitting (NAT) to 192.168.10.251:5060:
- NOTIFY sip:10014@192.168.10.251:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK31065cb0;rport
- From: <sip:10012@192.168.10.10>;tag=as31f661fa
- To: <sip:10014@192.168.10.10>;tag=ttbbodh6oe
- Contact: <sip:10012@192.168.10.10>
- Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu
- CSeq: 113 NOTIFY
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Event: dialog
- Content-Type: application/dialog-info+xml
- Subscription-State: active
- Content-Length: 497
- <?xml version="1.0"?>
- <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:10012@192.168.10.10">
- <dialog id="10012" call-id="285302da47842cc114d1a20766a4c073@192.168.10.10" direction="recipient">
- <local><identity display="10012">10012</identity><target uri="sip:10014@192.168.10.10"/></local>
- <remote><identity display="Test3">sip:10013@192.168.10.10</identity><target uri="sip:10012@192.168.10.10"/></remote>
- <state>early</state>
- </dialog>
- </dialog-info>
- ---
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:36] VERBOSE[1565] logger.c: [Feb 9 13:42:36] Extension Changed 10012[ext-local] new state Ringing for Notify User 10014
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36]
- <--- SIP read from 192.168.10.253:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 INVITE
- Contact: <sip:10012@192.168.10.253:5060>;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
- <------------->
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae (54)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5 (50)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10 (55)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:10012@192.168.10.253:5060>;flow-id=1 (50)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 8: Allow-Events: talk, hold, refer, call-info (42)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 9: Content-Length: 0 (17)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: Header 10: (0)
- [Feb 9 13:42:36] VERBOSE[1571] logger.c: [Feb 9 13:42:36] --- (10 headers 0 lines) ---
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: *** SIP TIMER: Cancelling retransmission #234 - INVITE (got response)
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '285302da47842cc114d1a20766a4c073@192.168.10.10' Request 102: Found
- [Feb 9 13:42:36] DEBUG[1571] chan_sip.c: SIP response 180 to standard invite
- [Feb 9 13:42:36] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 6 (Ringing)
- [Feb 9 13:42:36] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:36] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:36] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:36] VERBOSE[1643] logger.c: [Feb 9 13:42:36] -- SIP/10012-09943518 is ringing
- [Feb 9 13:42:37] VERBOSE[1571] logger.c: [Feb 9 13:42:37]
- <--- SIP read from 192.168.10.251:5060 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK31065cb0;rport=5060
- From: <sip:10012@192.168.10.10>;tag=as31f661fa
- To: <sip:10014@192.168.10.10>;tag=ttbbodh6oe
- Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu
- CSeq: 113 NOTIFY
- Content-Length: 0
- <------------->
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK31065cb0;rport=5060 (69)
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 2: From: <sip:10012@192.168.10.10>;tag=as31f661fa (46)
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10014@192.168.10.10>;tag=ttbbodh6oe (44)
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu (34)
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 5: CSeq: 113 NOTIFY (16)
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Header 7: (0)
- [Feb 9 13:42:37] VERBOSE[1571] logger.c: [Feb 9 13:42:37] --- (7 headers 0 lines) ---
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Acked pending invite 113
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #236
- [Feb 9 13:42:37] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bbfb6fc4-t8z10gt0v8fu' of Request 113: Match Found
- [Feb 9 13:42:37] VERBOSE[1571] logger.c: [Feb 9 13:42:37] SIP Response message for INCOMING dialog NOTIFY arrived
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- SIP read from 192.168.10.251:5060 --->
- INVITE sip:10012@192.168.10.10 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;rport
- From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6
- To: "Test3" <sip:10013@192.168.10.10>
- Call-ID: 3c27bc5ad168-wohs43kd06vu
- CSeq: 1 INVITE
- Max-Forwards: 70
- Contact: <sip:10014@192.168.10.251:5060>;reg-id=1
- Replaces: 285302da47842cc114d1a20766a4c073@192.168.10.10
- P-Key-Flags: resolution="31x13", keys="4"
- User-Agent: snom360/7.3.30
- 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: 374
- v=0
- o=root 1866843021 1866843021 IN IP4 192.168.10.251
- s=call
- c=IN IP4 192.168.10.251
- t=0 0
- m=audio 56984 RTP/AVP 0 8 9 99 3 18 4 101
- a=rtpmap:0 pcmu/8000
- a=rtpmap:8 pcma/8000
- a=rtpmap:9 g722/8000
- a=rtpmap:99 g726-32/8000
- a=rtpmap:3 gsm/8000
- a=rtpmap:18 g729/8000
- a=rtpmap:4 g723/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------->
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: INVITE sip:10012@192.168.10.10 SIP/2.0 (38)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;rport (70)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6 (54)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:10013@192.168.10.10> (37)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 1 INVITE (14)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:10014@192.168.10.251:5060>;reg-id=1 (49)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Replaces: 285302da47842cc114d1a20766a4c073@192.168.10.10 (56)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: P-Key-Flags: resolution="31x13", keys="4" (41)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 10: User-Agent: snom360/7.3.30 (26)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 11: Accept: application/sdp (23)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 13: Allow-Events: talk, hold, refer, call-info (42)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 14: Supported: timer, 100rel, replaces, from-change (47)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 15: Session-Expires: 3600;refresher=uas (35)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 16: Min-SE: 90 (10)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 17: Content-Type: application/sdp (29)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 18: Content-Length: 374 (19)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 19: (0)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: v=0 (3)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: o=root 1866843021 1866843021 IN IP4 192.168.10.251 (50)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: s=call (6)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: c=IN IP4 192.168.10.251 (23)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: t=0 0 (5)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: m=audio 56984 RTP/AVP 0 8 9 99 3 18 4 101 (41)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:0 pcmu/8000 (20)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:99 g726-32/8000 (24)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=fmtp:101 0-16 (15)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=ptime:20 (10)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=sendrecv (10)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (19 headers 17 lines) ---
- [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 192.168.10.0
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Allocating new SIP dialog for 3c27bc5ad168-wohs43kd06vu - INVITE (With RTP)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, from-change"
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found SIP option: -timer-
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Matched SIP option: timer
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found SIP option: -100rel-
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Matched SIP option: 100rel
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found SIP option: -replaces-
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Matched SIP option: replaces
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found SIP option: -from-change-
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found no match for SIP option: from-change (Please file bug report!)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: INVITE part of call transfer. Replaces [285302da47842cc114d1a20766a4c073@192.168.10.10]
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : 285302da47842cc114d1a20766a4c073@192.168.10.10 Fromtag: <no from tag> Totag: <no to tag>
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found call with callid 285302da47842cc114d1a20766a4c073@192.168.10.10 (ourtag=as7741abae, theirtag=p88g5axhr5)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Sending to 192.168.10.251 : 5060 (NAT)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Using INVITE request as basis request - 3c27bc5ad168-wohs43kd06vu
- [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 0.0.0.0
- [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 0.0.0.0
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- Reliably Transmitting (NAT) to 192.168.10.251:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;received=192.168.10.251;rport=5060
- From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6
- To: "Test3" <sip:10013@192.168.10.10>;tag=as5a4bc795
- Call-ID: 3c27bc5ad168-wohs43kd06vu
- CSeq: 1 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="517a6259"
- Content-Length: 0
- <------------>
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Scheduling destruction of SIP dialog '3c27bc5ad168-wohs43kd06vu' in 32000 ms (Method: INVITE)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found user '10014'
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- SIP read from 192.168.10.251:5060 --->
- ACK sip:10012@192.168.10.10 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;rport
- From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6
- To: "Test3" <sip:10013@192.168.10.10>;tag=as5a4bc795
- Call-ID: 3c27bc5ad168-wohs43kd06vu
- CSeq: 1 ACK
- Max-Forwards: 70
- Contact: <sip:10014@192.168.10.251:5060>;reg-id=1
- Content-Length: 0
- <------------->
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: ACK sip:10012@192.168.10.10 SIP/2.0 (35)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-ajx4n5bea72r;rport (70)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6 (54)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:10013@192.168.10.10>;tag=as5a4bc795 (52)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 1 ACK (11)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:10014@192.168.10.251:5060>;reg-id=1 (49)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Content-Length: 0 (17)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: (0)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (9 headers 0 lines) ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #237
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bc5ad168-wohs43kd06vu' of Response 1: Match Found
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- SIP read from 192.168.10.251:5060 --->
- INVITE sip:10012@192.168.10.10 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-qeooevjtigx9;rport
- From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6
- To: "Test3" <sip:10013@192.168.10.10>
- Call-ID: 3c27bc5ad168-wohs43kd06vu
- CSeq: 2 INVITE
- Max-Forwards: 70
- Contact: <sip:10014@192.168.10.251:5060>;reg-id=1
- Replaces: 285302da47842cc114d1a20766a4c073@192.168.10.10
- P-Key-Flags: resolution="31x13", keys="4"
- User-Agent: snom360/7.3.30
- 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="10014",realm="asterisk",nonce="517a6259",uri="sip:10012@192.168.10.10",response="7c7b135179e96dde5516c178fc21f003",algorithm=MD5
- Content-Type: application/sdp
- Content-Length: 374
- v=0
- o=root 1866843021 1866843021 IN IP4 192.168.10.251
- s=call
- c=IN IP4 192.168.10.251
- t=0 0
- m=audio 56984 RTP/AVP 0 8 9 99 3 18 4 101
- a=rtpmap:0 pcmu/8000
- a=rtpmap:8 pcma/8000
- a=rtpmap:9 g722/8000
- a=rtpmap:99 g726-32/8000
- a=rtpmap:3 gsm/8000
- a=rtpmap:18 g729/8000
- a=rtpmap:4 g723/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------->
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: INVITE sip:10012@192.168.10.10 SIP/2.0 (38)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-qeooevjtigx9;rport (70)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6 (54)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:10013@192.168.10.10> (37)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 2 INVITE (14)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:10014@192.168.10.251:5060>;reg-id=1 (49)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Replaces: 285302da47842cc114d1a20766a4c073@192.168.10.10 (56)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: P-Key-Flags: resolution="31x13", keys="4" (41)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 10: User-Agent: snom360/7.3.30 (26)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 11: Accept: application/sdp (23)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 12: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 13: Allow-Events: talk, hold, refer, call-info (42)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 14: Supported: timer, 100rel, replaces, from-change (47)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 15: Session-Expires: 3600;refresher=uas (35)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 16: Min-SE: 90 (10)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 17: Proxy-Authorization: Digest username="10014",realm="asterisk",nonce="517a6259",uri="sip:10012@192.168.10.10",response="7c7b135179e96dde5516c178fc21f003",algorithm=MD5 (166)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 18: Content-Type: application/sdp (29)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 19: Content-Length: 374 (19)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 20: (0)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: v=0 (3)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: o=root 1866843021 1866843021 IN IP4 192.168.10.251 (50)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: s=call (6)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: c=IN IP4 192.168.10.251 (23)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: t=0 0 (5)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: m=audio 56984 RTP/AVP 0 8 9 99 3 18 4 101 (41)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:0 pcmu/8000 (20)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:99 g726-32/8000 (24)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=fmtp:101 0-16 (15)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=ptime:20 (10)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Line: a=sendrecv (10)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (20 headers 17 lines) ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: INVITE part of call transfer. Replaces [285302da47842cc114d1a20766a4c073@192.168.10.10]
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : 285302da47842cc114d1a20766a4c073@192.168.10.10 Fromtag: <no from tag> Totag: <no to tag>
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Found call with callid 285302da47842cc114d1a20766a4c073@192.168.10.10 (ourtag=as7741abae, theirtag=p88g5axhr5)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Sending to 192.168.10.251 : 5060 (NAT)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Using INVITE request as basis request - 3c27bc5ad168-wohs43kd06vu
- [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 0.0.0.0
- [Feb 9 13:42:38] DEBUG[1571] acl.c: ##### Testing 192.168.10.251 with 0.0.0.0
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting NAT on RTP to On
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found user '10014'
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 0
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 8
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 9
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 99
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 3
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 18
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 4
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found RTP audio format 101
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Peer audio RTP is at port 192.168.10.251:56984
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format pcmu for ID 0
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format pcma for ID 8
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format g722 for ID 9
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format g726-32 for ID 99
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format gsm for ID 3
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format g729 for ID 18
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format g723 for ID 4
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Found audio description format telephone-event for ID 101
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Got unsupported a:fmtp in SDP offer
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: T38 state changed to 0 on channel <none>
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Capabilities: us - 0x8 (alaw), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x8 (alaw)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Peer audio RTP is at port 192.168.10.251:56984
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: We're settling with these formats: 0x8 (alaw)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Checking SIP call limits for device 10014
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Updating call counter for incoming call
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Call from peer '10014' is 1 out of 50
- [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Looking for 10012 in from-internal (domain 192.168.10.10)
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 2 (In use)
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '2' (In use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** Our native formats are 0x8 (alaw)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** Joint capabilities are 0x8 (alaw)
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** Our capabilities are 0x8 (alaw)
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c2d44997030-61xfuv9xjhtd
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: This channel will not be able to handle video.
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: build_route: Contact hop: <sip:10014@192.168.10.251:5060>;reg-id=1
- [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] set_destination: Parsing <sip:10012@192.168.10.253:5060> for address/port to send to
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] list_route: hop: <sip:10014@192.168.10.251:5060>
- [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] set_destination: set destination to 192.168.10.253, port 5060
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Sending this call to the invite/replcaes handler 3c27bc5ad168-wohs43kd06vu
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Attended transfer attempted to replace call with no bridge (maybe ringing). Channel SIP/10012-09943518!
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP transfer: Invite Replace incoming channel should replace and hang up channel SIP/10012-09943518 (one call leg)
- [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] Reliably Transmitting (NAT) to 192.168.10.253:5060:
- NOTIFY sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK24e5733b;rport
- From: <sip:10014@192.168.10.10>;tag=as68be196f
- To: <sip:10012@192.168.10.10>;tag=nw4c6bdjzk
- Contact: <sip:10014@192.168.10.10>
- Call-ID: 3c2d44997030-61xfuv9xjhtd
- CSeq: 113 NOTIFY
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Event: dialog
- Content-Type: application/dialog-info+xml
- Subscription-State: active
- Content-Length: 208
- <?xml version="1.0"?>
- <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:10014@192.168.10.10">
- <dialog id="10014">
- <state>confirmed</state>
- </dialog>
- </dialog-info>
- ---
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- Transmitting (NAT) to 192.168.10.251:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-qeooevjtigx9;received=192.168.10.251;rport=5060
- From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6
- To: "Test3" <sip:10013@192.168.10.10>
- Call-ID: 3c27bc5ad168-wohs43kd06vu
- CSeq: 2 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:10012@192.168.10.10>
- Content-Length: 0
- <------------>
- [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] Extension Changed 10014[ext-local] new state InUse for Notify User 10012
- [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting framing from config on incoming call
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 2 (In use)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Audio is at 192.168.10.10 port 19568
- [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '2' (In use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Adding codec 0x8 (alaw) to SDP
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Adding non-codec 0x1 (telephone-event) to SDP
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- Done with adding codecs to SDP
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- Reliably Transmitting (NAT) to 192.168.10.251:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-qeooevjtigx9;received=192.168.10.251;rport=5060
- From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6
- To: "Test3" <sip:10013@192.168.10.10>;tag=as673f2fe0
- Call-ID: 3c27bc5ad168-wohs43kd06vu
- CSeq: 2 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:10012@192.168.10.10>
- Content-Type: application/sdp
- Content-Length: 240
- v=0
- o=root 1562 1562 IN IP4 192.168.10.10
- s=session
- c=IN IP4 192.168.10.10
- t=0 0
- m=audio 19568 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
- <------------>
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/Replaces: preparing to masquerade SIP/10014-0994c8c8 into SIP/10012-09943518
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:38] DEBUG[1571] channel.c: Planning to masquerade channel SIP/10014-0994c8c8 into the structure of SIP/10012-09943518
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:38] DEBUG[1571] channel.c: Done planning to masquerade channel SIP/10014-0994c8c8 into the structure of SIP/10012-09943518
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 2 (In use)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/Replaces: Going to masquerade SIP/10014-0994c8c8 into SIP/10012-09943518
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '2' (In use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:38] DEBUG[1571] channel.c: Actually Masquerading SIP/10014-0994c8c8(6) into the structure of SIP/10012-09943518(5)
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:38] DEBUG[1571] channel.c: Got clone lock for masquerade on 'SIP/10014-0994c8c8' at 0x97e1868
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP Fixup: New owner for dialogue 285302da47842cc114d1a20766a4c073@192.168.10.10: SIP/10014-0994c8c8<MASQ> (Old parent: SIP/10014-0994c8c8)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Hangup call SIP/10014-0994c8c8<MASQ>, SIP callid 285302da47842cc114d1a20766a4c073@192.168.10.10)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: update_call_counter(10012) - decrement call limit counter on hangup
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Updating call counter for outgoing call
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Call to peer '10012' removed from call limit 50
- [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Hanging up channel in state Ringing (not UP)
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 1 (Not in use)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Scheduling destruction of SIP dialog '285302da47842cc114d1a20766a4c073@192.168.10.10' in 6400 ms (Method: INVITE)
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '285302da47842cc114d1a20766a4c073@192.168.10.10' Request 102: Found
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Reliably Transmitting (NAT) to 192.168.10.253:5060:
- CANCEL sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 CANCEL
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Scheduling destruction of SIP dialog '285302da47842cc114d1a20766a4c073@192.168.10.10' in 6400 ms (Method: INVITE)
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c27bbfb6fc4-t8z10gt0v8fu
- [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] set_destination: Parsing <sip:10014@192.168.10.251:5060> for address/port to send to
- [Feb 9 13:42:38] DEBUG[1571] channel.c: Putting channel SIP/10014-0994c8c8 in 8/8 formats
- [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] set_destination: set destination to 192.168.10.251, port 5060
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP Fixup: New owner for dialogue 3c27bc5ad168-wohs43kd06vu: SIP/10014-0994c8c8 (Old parent: SIP/10012-09943518<ZOMBIE>)
- [Feb 9 13:42:38] DEBUG[1571] channel.c: Released clone lock on 'SIP/10012-09943518<ZOMBIE>'
- [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] Reliably Transmitting (NAT) to 192.168.10.251:5060:
- NOTIFY sip:10014@192.168.10.251:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK05ef9d45;rport
- From: <sip:10012@192.168.10.10>;tag=as31f661fa
- To: <sip:10014@192.168.10.10>;tag=ttbbodh6oe
- Contact: <sip:10012@192.168.10.10>
- Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu
- CSeq: 114 NOTIFY
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Event: dialog
- Content-Type: application/dialog-info+xml
- Subscription-State: active
- Content-Length: 209
- <?xml version="1.0"?>
- <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="12" state="full" entity="sip:10012@192.168.10.10">
- <dialog id="10012">
- <state>terminated</state>
- </dialog>
- </dialog-info>
- ---
- [Feb 9 13:42:38] DEBUG[1571] channel.c: Done Masquerading SIP/10014-0994c8c8 (6)
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Invite/Replace: Could successfully read frame from RING channel!
- [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
- [Feb 9 13:42:38] VERBOSE[1565] logger.c: [Feb 9 13:42:38] Extension Changed 10012[ext-local] new state Idle for Notify User 10014
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: After transfer:----------------------------
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- C: SIP/10012-09943518<ZOMBIE> State Down
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- replacecall: SIP/10014-0994c8c8 State Up
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 1 (Not in use)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- P->owner: SIP/10014-0994c8c8 State Up
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: -- No call bridged to C->owner
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: End After transfer:----------------------------
- [Feb 9 13:42:38] DEBUG[1571] channel.c: Hanging up zombie 'SIP/10012-09943518<ZOMBIE>'
- [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 1 (Not in use)
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38] -- SIP/10014-0994c8c8 answered SIP/10013-0993e6e0
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:38] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: SIP answering channel: SIP/10013-0993e6e0
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 2 (In use)
- [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: Setting framing from config on incoming call
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True
- [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '2' (In use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38] Audio is at 192.168.10.10 port 10000
- [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38] Adding codec 0x8 (alaw) to SDP
- [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38] Adding non-codec 0x1 (telephone-event) to SDP
- [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: -- Done with adding codecs to SDP
- [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw)
- [Feb 9 13:42:38] VERBOSE[1643] logger.c: [Feb 9 13:42:38]
- <--- Reliably Transmitting (NAT) to 192.168.10.252:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-t21vmvd192u5;received=192.168.10.252;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- To: <sip:10012@192.168.10.10>;tag=as17f86ff1
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 2 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:10012@192.168.10.10>
- Content-Type: application/sdp
- Content-Length: 240
- v=0
- o=root 1562 1562 IN IP4 192.168.10.10
- s=session
- c=IN IP4 192.168.10.10
- t=0 0
- m=audio 10000 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
- <------------>
- [Feb 9 13:42:38] DEBUG[1643] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- SIP read from 192.168.10.252:5060 --->
- ACK sip:10012@192.168.10.10 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-lrtc2e94lbnh;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- To: <sip:10012@192.168.10.10>;tag=as17f86ff1
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 2 ACK
- Max-Forwards: 70
- Contact: <sip:10013@192.168.10.252:5060>;reg-id=1
- Content-Length: 0
- <------------->
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: ACK sip:10012@192.168.10.10 SIP/2.0 (35)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.252:5060;branch=z9hG4bK-lrtc2e94lbnh;rport (70)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd (54)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.10>;tag=as17f86ff1 (44)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 2 ACK (11)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:10013@192.168.10.252:5060>;reg-id=1 (49)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Content-Length: 0 (17)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: (0)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (9 headers 0 lines) ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #245
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on 'f1882b3c0d38-xtxggv3ikaiv' of Response 2: Match Found
- [Feb 9 13:42:38] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
- [Feb 9 13:42:38] DEBUG[1643] rtp.c: Ooh, format changed from unknown to alaw
- [Feb 9 13:42:38] DEBUG[1643] rtp.c: Created smoother: format: 8 ms: 20 len: 160
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- SIP read from 192.168.10.253:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 INVITE
- Contact: <sip:10012@192.168.10.253:5060>;flow-id=1
- Content-Length: 0
- <------------->
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae (54)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5 (50)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10 (55)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:10012@192.168.10.253:5060>;flow-id=1 (50)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: (0)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (8 headers 0 lines) ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Acked pending invite 102
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '285302da47842cc114d1a20766a4c073@192.168.10.10' of Request 102: Match Found
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP response 487 to standard invite
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Transmitting (NAT) to 192.168.10.253:5060:
- ACK sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5
- Contact: <sip:10013@192.168.10.10>
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 ACK
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Updating call counter for outgoing call
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Call to peer '10012' removed from call limit 50
- [Feb 9 13:42:38] DEBUG[1571] devicestate.c: Notification of state change to be queued on device/channel SIP/10012
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Setting SIP_ALREADYGONE on dialog 285302da47842cc114d1a20766a4c073@192.168.10.10
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: Changing state for SIP/10012 - state 1 (Not in use)
- [Feb 9 13:42:38] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10012
- [Feb 9 13:42:38] DEBUG[1587] app_queue.c: Device 'SIP/10012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:38] DEBUG[1565] chan_sip.c: Checking device state for peer 10012
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- SIP read from 192.168.10.251:5060 --->
- ACK sip:10012@192.168.10.10 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-5cikq0utxig2;rport
- From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6
- To: "Test3" <sip:10013@192.168.10.10>;tag=as673f2fe0
- Call-ID: 3c27bc5ad168-wohs43kd06vu
- CSeq: 2 ACK
- Max-Forwards: 70
- Contact: <sip:10014@192.168.10.251:5060>;reg-id=1
- Content-Length: 0
- <------------->
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: ACK sip:10012@192.168.10.10 SIP/2.0 (35)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.251:5060;branch=z9hG4bK-5cikq0utxig2;rport (70)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6 (54)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:10013@192.168.10.10>;tag=as673f2fe0 (52)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 2 ACK (11)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Max-Forwards: 70 (16)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Contact: <sip:10014@192.168.10.251:5060>;reg-id=1 (49)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: Content-Length: 0 (17)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 9: (0)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (9 headers 0 lines) ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #240
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bc5ad168-wohs43kd06vu' of Response 2: Match Found
- [Feb 9 13:42:38] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #239 (1) NOTIFY - 4
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #239))
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Retransmitting #1 (NAT) to 192.168.10.253:5060:
- NOTIFY sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK24e5733b;rport
- From: <sip:10014@192.168.10.10>;tag=as68be196f
- To: <sip:10012@192.168.10.10>;tag=nw4c6bdjzk
- Contact: <sip:10014@192.168.10.10>
- Call-ID: 3c2d44997030-61xfuv9xjhtd
- CSeq: 113 NOTIFY
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Event: dialog
- Content-Type: application/dialog-info+xml
- Subscription-State: active
- Content-Length: 208
- <?xml version="1.0"?>
- <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:10014@192.168.10.10">
- <dialog id="10014">
- <state>confirmed</state>
- </dialog>
- </dialog-info>
- ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (1) CANCEL - 14
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 200 ms (t1 100 ms (Retrans id #242))
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Retransmitting #1 (NAT) to 192.168.10.253:5060:
- CANCEL sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 CANCEL
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Feb 9 13:42:38] DEBUG[1643] rtp.c: Ooh, format changed from unknown to alaw
- [Feb 9 13:42:38] DEBUG[1643] rtp.c: Created smoother: format: 8 ms: 20 len: 160
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- SIP read from 192.168.10.251:5060 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK05ef9d45;rport=5060
- From: <sip:10012@192.168.10.10>;tag=as31f661fa
- To: <sip:10014@192.168.10.10>;tag=ttbbodh6oe
- Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu
- CSeq: 114 NOTIFY
- Content-Length: 0
- <------------->
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK05ef9d45;rport=5060 (69)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: <sip:10012@192.168.10.10>;tag=as31f661fa (46)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10014@192.168.10.10>;tag=ttbbodh6oe (44)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bbfb6fc4-t8z10gt0v8fu (34)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 114 NOTIFY (16)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: (0)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (7 headers 0 lines) ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Acked pending invite 114
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #244
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bbfb6fc4-t8z10gt0v8fu' of Request 114: Match Found
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] SIP Response message for INCOMING dialog NOTIFY arrived
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- SIP read from 192.168.10.253:5060 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK24e5733b;rport=5060
- From: <sip:10014@192.168.10.10>;tag=as68be196f
- To: <sip:10012@192.168.10.10>;tag=nw4c6bdjzk
- Call-ID: 3c2d44997030-61xfuv9xjhtd
- CSeq: 113 NOTIFY
- Content-Length: 0
- <------------->
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK24e5733b;rport=5060 (69)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: <sip:10014@192.168.10.10>;tag=as68be196f (46)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.10>;tag=nw4c6bdjzk (44)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c2d44997030-61xfuv9xjhtd (34)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 113 NOTIFY (16)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: (0)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (7 headers 0 lines) ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Acked pending invite 113
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #239
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c2d44997030-61xfuv9xjhtd' of Request 113: Match Found
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] SIP Response message for INCOMING dialog NOTIFY arrived
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38]
- <--- SIP read from 192.168.10.253:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 INVITE
- Contact: <sip:10012@192.168.10.253:5060>;flow-id=1
- Content-Length: 0
- <------------->
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae (54)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5 (50)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10 (55)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:10012@192.168.10.253:5060>;flow-id=1 (50)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Header 8: (0)
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] --- (8 headers 0 lines) ---
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: Stopping retransmission on '285302da47842cc114d1a20766a4c073@192.168.10.10' of Request 102: Match Not Found
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (2) CANCEL - 14
- [Feb 9 13:42:38] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 400 ms (t1 100 ms (Retrans id #242))
- [Feb 9 13:42:38] VERBOSE[1571] logger.c: [Feb 9 13:42:38] Retransmitting #2 (NAT) to 192.168.10.253:5060:
- CANCEL sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 CANCEL
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Feb 9 13:42:39] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (3) CANCEL - 14
- [Feb 9 13:42:39] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 800 ms (t1 100 ms (Retrans id #242))
- [Feb 9 13:42:39] VERBOSE[1571] logger.c: [Feb 9 13:42:39] Retransmitting #3 (NAT) to 192.168.10.253:5060:
- CANCEL sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 CANCEL
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (4) CANCEL - 14
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 1600 ms (t1 100 ms (Retrans id #242))
- [Feb 9 13:42:40] VERBOSE[1571] logger.c: [Feb 9 13:42:40] Retransmitting #4 (NAT) to 192.168.10.253:5060:
- CANCEL sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 CANCEL
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Feb 9 13:42:40] VERBOSE[1571] logger.c: [Feb 9 13:42:40]
- <--- SIP read from 192.168.10.253:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 INVITE
- Contact: <sip:10012@192.168.10.253:5060>;flow-id=1
- Content-Length: 0
- <------------->
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae (54)
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5 (50)
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10 (55)
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:10012@192.168.10.253:5060>;flow-id=1 (50)
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Header 8: (0)
- [Feb 9 13:42:40] VERBOSE[1571] logger.c: [Feb 9 13:42:40] --- (8 headers 0 lines) ---
- [Feb 9 13:42:40] DEBUG[1571] chan_sip.c: Stopping retransmission on '285302da47842cc114d1a20766a4c073@192.168.10.10' of Request 102: Match Not Found
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (5) CANCEL - 14
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 3200 ms (t1 100 ms (Retrans id #242))
- [Feb 9 13:42:41] VERBOSE[1571] logger.c: [Feb 9 13:42:41] Retransmitting #5 (NAT) to 192.168.10.253:5060:
- CANCEL sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 CANCEL
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Feb 9 13:42:41] VERBOSE[1571] logger.c: [Feb 9 13:42:41]
- <--- SIP read from 192.168.10.253:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 INVITE
- Contact: <sip:10012@192.168.10.253:5060>;flow-id=1
- Content-Length: 0
- <------------->
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae (54)
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5 (50)
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10 (55)
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:10012@192.168.10.253:5060>;flow-id=1 (50)
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Header 8: (0)
- [Feb 9 13:42:41] VERBOSE[1571] logger.c: [Feb 9 13:42:41] --- (8 headers 0 lines) ---
- [Feb 9 13:42:41] DEBUG[1571] chan_sip.c: Stopping retransmission on '285302da47842cc114d1a20766a4c073@192.168.10.10' of Request 102: Match Not Found
- [Feb 9 13:42:43] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
- [Feb 9 13:42:43] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: SIP TIMER: Rescheduling retransmission #242 (6) CANCEL - 14
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 100 ms (Retrans id #242))
- [Feb 9 13:42:44] VERBOSE[1571] logger.c: [Feb 9 13:42:44] Retransmitting #6 (NAT) to 192.168.10.253:5060:
- CANCEL sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 CANCEL
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Content-Length: 0
- ---
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Auto destroying SIP dialog '285302da47842cc114d1a20766a4c073@192.168.10.10'
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Destroying SIP dialog 285302da47842cc114d1a20766a4c073@192.168.10.10
- [Feb 9 13:42:44] VERBOSE[1571] logger.c: [Feb 9 13:42:44] Really destroying SIP dialog '285302da47842cc114d1a20766a4c073@192.168.10.10' Method: INVITE
- [Feb 9 13:42:44] VERBOSE[1571] logger.c: [Feb 9 13:42:44]
- <--- SIP read from 192.168.10.253:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae
- To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5
- Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10
- CSeq: 102 INVITE
- Contact: <sip:10012@192.168.10.253:5060>;flow-id=1
- Content-Length: 0
- <------------->
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30)
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0b7be4e4;rport=5060 (69)
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=as7741abae (54)
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.253:5060>;tag=p88g5axhr5 (50)
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 285302da47842cc114d1a20766a4c073@192.168.10.10 (55)
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:10012@192.168.10.253:5060>;flow-id=1 (50)
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 7: Content-Length: 0 (17)
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Header 8: (0)
- [Feb 9 13:42:44] VERBOSE[1571] logger.c: [Feb 9 13:42:44] --- (8 headers 0 lines) ---
- [Feb 9 13:42:44] DEBUG[1571] chan_sip.c: Invalid SIP message - rejected , no callid, len 359
- [Feb 9 13:42:47] VERBOSE[1642] logger.c: [Feb 9 13:42:47] -- Remote UNIX connection disconnected
- [Feb 9 13:42:48] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
- [Feb 9 13:42:48] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
- [Feb 9 13:42:53] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
- [Feb 9 13:42:53] DEBUG[1643] rtp.c: Got RTCP report of 68 bytes
- [Feb 9 13:42:56] DEBUG[1562] channel.c: Soft-Hanging up channel 'SIP/10014-0994c8c8'
- [Feb 9 13:42:56] DEBUG[1562] channel.c: Soft-Hanging up channel 'SIP/10013-0993e6e0'
- [Feb 9 13:42:56] DEBUG[1643] channel.c: Didn't get a frame from channel: SIP/10014-0994c8c8
- [Feb 9 13:42:56] DEBUG[1643] channel.c: Bridge stops bridging channels SIP/10013-0993e6e0 and SIP/10014-0994c8c8
- [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'Macro'
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [h@macro-dial:1] Macro("SIP/10013-0993e6e0", "hangupcall") in new stack
- [Feb 9 13:42:56] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'GotoIf'
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [s@macro-hangupcall:1] GotoIf("SIP/10013-0993e6e0", "1?skiprg") in new stack
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Goto (macro-hangupcall,s,4)
- [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Executed application: GotoIf
- [Feb 9 13:42:56] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'GotoIf'
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [s@macro-hangupcall:4] GotoIf("SIP/10013-0993e6e0", "1?skipblkvm") in new stack
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Goto (macro-hangupcall,s,7)
- [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Executed application: GotoIf
- [Feb 9 13:42:56] DEBUG[1643] pbx.c: Expression result is '1'
- [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'GotoIf'
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [s@macro-hangupcall:7] GotoIf("SIP/10013-0993e6e0", "1?theend") in new stack
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Goto (macro-hangupcall,s,9)
- [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Executed application: GotoIf
- [Feb 9 13:42:56] DEBUG[1643] pbx.c: Launching 'Hangup'
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] -- Executing [s@macro-hangupcall:9] Hangup("SIP/10013-0993e6e0", "") in new stack
- [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Spawn extension (macro-hangupcall,s,9) exited non-zero on 'SIP/10013-0993e6e0' in macro 'hangupcall'
- [Feb 9 13:42:56] DEBUG[1643] res_features.c: Spawn h extension (macro-dial,h,1) exited non-zero on 'SIP/10013-0993e6e0'
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] == Spawn h extension (macro-dial, h, 1) exited non-zero on 'SIP/10013-0993e6e0'
- [Feb 9 13:42:56] DEBUG[1643] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
- [Feb 9 13:42:56] DEBUG[1643] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield) VALUES ('2010-02-09 13:42:36','\"Test3\" <10013>','10013','10012','from-internal', 'SIP/10013-0993e6e0','SIP/10014-0994c8c8','Dial','SIP/10012||tr',20,18,'ANSWERED',3,'','')
- [Feb 9 13:42:56] DEBUG[1643] channel.c: Hanging up channel 'SIP/10014-0994c8c8'
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Hangup call SIP/10014-0994c8c8, SIP callid 3c27bc5ad168-wohs43kd06vu)
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: update_call_counter(10014) - decrement call limit counter on hangup
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Updating call counter for incoming call
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Call from peer '10014' removed from call limit 50
- [Feb 9 13:42:56] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] Scheduling destruction of SIP dialog '3c27bc5ad168-wohs43kd06vu' in 32000 ms (Method: ACK)
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 1 (Not in use)
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Strict routing enforced for session 3c27bc5ad168-wohs43kd06vu
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:10014@192.168.10.251:5060> for address/port to send to
- [Feb 9 13:42:56] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.251, port 5060
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.251:5060:
- BYE sip:10014@192.168.10.251:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK6793e7c9;rport
- From: "Test3" <sip:10013@192.168.10.10>;tag=as673f2fe0
- To: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6
- Call-ID: 3c27bc5ad168-wohs43kd06vu
- CSeq: 102 BYE
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- ---
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:56] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10014
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:56] DEBUG[1643] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:56] DEBUG[1643] app_dial.c: Exiting with DIALSTATUS=ANSWER.
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c2d44997030-61xfuv9xjhtd
- [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Spawn extension (macro-dial,s,7) exited non-zero on 'SIP/10013-0993e6e0' in macro 'dial'
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:10012@192.168.10.253:5060> for address/port to send to
- [Feb 9 13:42:56] DEBUG[1643] app_macro.c: Spawn extension (macro-exten-vm,s,9) exited non-zero on 'SIP/10013-0993e6e0' in macro 'exten-vm'
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.253, port 5060
- [Feb 9 13:42:56] DEBUG[1643] pbx.c: Spawn extension (from-internal,10012,1) exited non-zero on 'SIP/10013-0993e6e0'
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.253:5060:
- NOTIFY sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK52b93ef3;rport
- From: <sip:10014@192.168.10.10>;tag=as68be196f
- To: <sip:10012@192.168.10.10>;tag=nw4c6bdjzk
- Contact: <sip:10014@192.168.10.10>
- Call-ID: 3c2d44997030-61xfuv9xjhtd
- CSeq: 114 NOTIFY
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Event: dialog
- Content-Type: application/dialog-info+xml
- Subscription-State: active
- Content-Length: 209
- <?xml version="1.0"?>
- <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="12" state="full" entity="sip:10014@192.168.10.10">
- <dialog id="10014">
- <state>terminated</state>
- </dialog>
- </dialog-info>
- ---
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] == Spawn extension (from-internal, 10012, 1) exited non-zero on 'SIP/10013-0993e6e0'
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:56] DEBUG[1643] channel.c: Soft-Hanging up channel 'SIP/10013-0993e6e0'
- [Feb 9 13:42:56] DEBUG[1643] channel.c: Hanging up channel 'SIP/10013-0993e6e0'
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Extension Changed 10014[ext-local] new state Idle for Notify User 10012
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Hangup call SIP/10013-0993e6e0, SIP callid f1882b3c0d38-xtxggv3ikaiv)
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: update_call_counter(10013) - decrement call limit counter on hangup
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Updating call counter for incoming call
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Call from peer '10013' removed from call limit 50
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: Changing state for SIP/10014 - state 1 (Not in use)
- [Feb 9 13:42:56] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10014
- [Feb 9 13:42:56] DEBUG[1587] app_queue.c: Device 'SIP/10014' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] Scheduling destruction of SIP dialog 'f1882b3c0d38-xtxggv3ikaiv' in 32000 ms (Method: ACK)
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: Strict routing enforced for session f1882b3c0d38-xtxggv3ikaiv
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:10013@192.168.10.252:5060> for address/port to send to
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10014
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.252, port 5060
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:56] VERBOSE[1643] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.252:5060:
- BYE sip:10013@192.168.10.252:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK68b5a119;rport
- From: <sip:10012@192.168.10.10>;tag=as17f86ff1
- To: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 102 BYE
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- ---
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 1 (Not in use)
- [Feb 9 13:42:56] DEBUG[1643] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:56] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:56] DEBUG[1643] devicestate.c: Notification of state change to be queued on device/channel SIP/10013
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c2d44996af4-i4ocu4zvtwre
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:10012@192.168.10.253:5060> for address/port to send to
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.253, port 5060
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.253:5060:
- NOTIFY sip:10012@192.168.10.253:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK0ed066d1;rport
- From: <sip:10013@192.168.10.10>;tag=as6d58a6fc
- To: <sip:10012@192.168.10.10>;tag=5e5cbuyyd1
- Contact: <sip:10013@192.168.10.10>
- Call-ID: 3c2d44996af4-i4ocu4zvtwre
- CSeq: 114 NOTIFY
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Event: dialog
- Content-Type: application/dialog-info+xml
- Subscription-State: active
- Content-Length: 209
- <?xml version="1.0"?>
- <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="12" state="full" entity="sip:10013@192.168.10.10">
- <dialog id="10013">
- <state>terminated</state>
- </dialog>
- </dialog-info>
- ---
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Extension Changed 10013[ext-local] new state Idle for Notify User 10012
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Strict routing enforced for session 3c27bbfb7594-mfl0ssslkb6l
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: Parsing <sip:10014@192.168.10.251:5060> for address/port to send to
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] set_destination: set destination to 192.168.10.251, port 5060
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Reliably Transmitting (NAT) to 192.168.10.251:5060:
- NOTIFY sip:10014@192.168.10.251:5060 SIP/2.0
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5e26394a;rport
- From: <sip:10013@192.168.10.10>;tag=as647499f8
- To: <sip:10014@192.168.10.10>;tag=wesbz3i941
- Contact: <sip:10013@192.168.10.10>
- Call-ID: 3c27bbfb7594-mfl0ssslkb6l
- CSeq: 114 NOTIFY
- User-Agent: Asterisk PBX
- Max-Forwards: 70
- Event: dialog
- Content-Type: application/dialog-info+xml
- Subscription-State: active
- Content-Length: 209
- <?xml version="1.0"?>
- <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="12" state="full" entity="sip:10013@192.168.10.10">
- <dialog id="10013">
- <state>terminated</state>
- </dialog>
- </dialog-info>
- ---
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
- [Feb 9 13:42:56] VERBOSE[1565] logger.c: [Feb 9 13:42:56] Extension Changed 10013[ext-local] new state Idle for Notify User 10014
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: Changing state for SIP/10013 - state 1 (Not in use)
- [Feb 9 13:42:56] DEBUG[1565] devicestate.c: No provider found, checking channel drivers for SIP - 10013
- [Feb 9 13:42:56] DEBUG[1565] chan_sip.c: Checking device state for peer 10013
- [Feb 9 13:42:56] DEBUG[1587] app_queue.c: Device 'SIP/10013' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56]
- <--- SIP read from 192.168.10.253:5060 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK52b93ef3;rport=5060
- From: <sip:10014@192.168.10.10>;tag=as68be196f
- To: <sip:10012@192.168.10.10>;tag=nw4c6bdjzk
- Call-ID: 3c2d44997030-61xfuv9xjhtd
- CSeq: 114 NOTIFY
- Content-Length: 0
- <------------->
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK52b93ef3;rport=5060 (69)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 2: From: <sip:10014@192.168.10.10>;tag=as68be196f (46)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10012@192.168.10.10>;tag=nw4c6bdjzk (44)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c2d44997030-61xfuv9xjhtd (34)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 5: CSeq: 114 NOTIFY (16)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 7: (0)
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] --- (7 headers 0 lines) ---
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Acked pending invite 114
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #250
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c2d44997030-61xfuv9xjhtd' of Request 114: Match Found
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] SIP Response message for INCOMING dialog NOTIFY arrived
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56]
- <--- SIP read from 192.168.10.252:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK68b5a119;rport=5060
- From: <sip:10012@192.168.10.10>;tag=as17f86ff1
- To: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd
- Call-ID: f1882b3c0d38-xtxggv3ikaiv
- CSeq: 102 BYE
- Contact: <sip:10013@192.168.10.252:5060>;reg-id=1
- User-Agent: snom870/8.3.6
- RTP-RxStat: Total_Rx_Pkts=899,Rx_Pkts=899,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
- RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=939,Remote_Tx_Pkts=740
- Content-Length: 0
- <------------->
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK68b5a119;rport=5060 (69)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 2: From: <sip:10012@192.168.10.10>;tag=as17f86ff1 (46)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 3: To: "Test3" <sip:10013@192.168.10.10>;tag=ac9v354jjd (52)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 4: Call-ID: f1882b3c0d38-xtxggv3ikaiv (34)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 BYE (13)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:10013@192.168.10.252:5060>;reg-id=1 (49)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 7: User-Agent: snom870/8.3.6 (25)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 8: RTP-RxStat: Total_Rx_Pkts=899,Rx_Pkts=899,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (78)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 9: RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=939,Remote_Tx_Pkts=740 (58)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 10: Content-Length: 0 (17)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 11: (0)
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] --- (11 headers 0 lines) ---
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #252
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Stopping retransmission on 'f1882b3c0d38-xtxggv3ikaiv' of Request 102: Match Found
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] SIP Response message for INCOMING dialog BYE arrived
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] Really destroying SIP dialog 'f1882b3c0d38-xtxggv3ikaiv' Method: ACK
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56]
- <--- SIP read from 192.168.10.251:5060 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5e26394a;rport=5060
- From: <sip:10013@192.168.10.10>;tag=as647499f8
- To: <sip:10014@192.168.10.10>;tag=wesbz3i941
- Call-ID: 3c27bbfb7594-mfl0ssslkb6l
- CSeq: 114 NOTIFY
- Content-Length: 0
- <------------->
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 Ok (14)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK5e26394a;rport=5060 (69)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 2: From: <sip:10013@192.168.10.10>;tag=as647499f8 (46)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 3: To: <sip:10014@192.168.10.10>;tag=wesbz3i941 (44)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bbfb7594-mfl0ssslkb6l (34)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 5: CSeq: 114 NOTIFY (16)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 6: Content-Length: 0 (17)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 7: (0)
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] --- (7 headers 0 lines) ---
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Acked pending invite 114
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #254
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bbfb7594-mfl0ssslkb6l' of Request 114: Match Found
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] SIP Response message for INCOMING dialog NOTIFY arrived
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56]
- <--- SIP read from 192.168.10.251:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK6793e7c9;rport=5060
- From: "Test3" <sip:10013@192.168.10.10>;tag=as673f2fe0
- To: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6
- Call-ID: 3c27bc5ad168-wohs43kd06vu
- CSeq: 102 BYE
- Contact: <sip:10014@192.168.10.251:5060>;reg-id=1
- User-Agent: snom360/7.3.30
- RTP-RxStat: Total_Rx_Pkts=910,Rx_Pkts=910,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
- RTP-TxStat: Total_Tx_Pkts=919,Tx_Pkts=919,Remote_Tx_Pkts=751
- Content-Length: 0
- <------------->
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.10:5060;branch=z9hG4bK6793e7c9;rport=5060 (69)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 2: From: "Test3" <sip:10013@192.168.10.10>;tag=as673f2fe0 (54)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 3: To: "Test4" <sip:10014@192.168.10.10>;tag=j0kehrl2e6 (52)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 4: Call-ID: 3c27bc5ad168-wohs43kd06vu (34)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 5: CSeq: 102 BYE (13)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 6: Contact: <sip:10014@192.168.10.251:5060>;reg-id=1 (49)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 7: User-Agent: snom360/7.3.30 (26)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 8: RTP-RxStat: Total_Rx_Pkts=910,Rx_Pkts=910,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (78)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 9: RTP-TxStat: Total_Tx_Pkts=919,Tx_Pkts=919,Remote_Tx_Pkts=751 (60)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 10: Content-Length: 0 (17)
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Header 11: (0)
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] --- (11 headers 0 lines) ---
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #249
- [Feb 9 13:42:56] DEBUG[1571] chan_sip.c: Stopping retransmission on '3c27bc5ad168-wohs43kd06vu' of Request 102: Match Found
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] SIP Response message for INCOMING dialog BYE arrived
- [Feb 9 13:42:56] VERBOSE[1571] logger.c: [Feb 9 13:42:56] Really destroying SIP dialog '3c27bc5ad168-wohs43kd06vu' Method: ACK
- [Feb 9 13:42:56] VERBOSE[1562] logger.c: [Feb 9 13:42:56] Executing last minute cleanups
- [Feb 9 13:42:56] VERBOSE[1562] logger.c: [Feb 9 13:42:56] == Destroying musiconhold processes
- [Feb 9 13:42:56] DEBUG[1562] res_musiconhold.c: Destroying MOH class 'default'
- [Feb 9 13:42:56] DEBUG[1562] res_musiconhold.c: Destroying MOH class 'none'
- [Feb 9 13:42:56] DEBUG[1562] asterisk.c: Asterisk ending (0).
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