Hi.
Well, i check the debug option an i can view this information:
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 43]: INVITE sip:XXXXXXXXX@192.168.100.21 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK1815663290;rport
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 34]: To: sip:XXXXXXXXX@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 16]: CSeq: 620 INVITE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 48]: Contact: “Sergio” sip:1001@192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 8 [ 40]: User-Agent: Grandstream GXP1625 1.0.1.12
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 9 [ 13]: Privacy: none
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 10 [ 56]: P-Preferred-Identity: “Sergio” sip:1001@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 11 [ 32]: Supported: replaces, path, timer
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 12 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 14 [ 47]: Accept: application/sdp, application/dtmf-relay
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 15 [ 21]: Content-Length: 337
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 16 [ 0]:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 0 [ 3]: v=0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 1 [ 38]: o=1001 8000 8000 IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 5 [ 35]: m=audio 5004 RTP/AVP 0 8 18 9 2 101
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 6 [ 10]: a=sendrecv
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 8 [ 10]: a=ptime:20
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 12 [ 20]: a=rtpmap:9 G722/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 13 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-15
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (16 headers 16 lines) —
[Nov 4 16:13:16] DEBUG[14741] acl.c: For destination ‘192.168.100.32’, our source address is ‘192.168.100.21’.
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741] netsock2.c: Splitting ‘192.168.100.32:5060’ into…
[Nov 4 16:13:16] DEBUG[14741] netsock2.c: …host ‘192.168.100.32’ and port ‘5060’.
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: Sending to 192.168.100.32:5060 (no NAT)
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Allocating new SIP dialog for 1695553247-5060-64@BJC.BGI.BAA.DC - INVITE (No RTP)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Begin: parsing SIP “Supported: replaces, path, timer”
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Found SIP option: -replaces-
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Matched SIP option: replaces
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Found SIP option: -path-
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Matched SIP option: path
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Found SIP option: -timer-
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] sip/reqresp_parser.c: Matched SIP option: timer
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.32:5060’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.32’ and port ‘5060’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Sending to 192.168.100.32:5060 (no NAT)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Initializing initreq for method INVITE - callid 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Using INVITE request as basis request - 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found peer ‘1001’ for ‘1001’ from 192.168.100.32:5060
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 0 [ 24]: SIP/2.0 401 Unauthorized
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 1 [ 96]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK1815663290;received=192.168.100.32;rport=5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 3 [ 49]: To: sip:XXXXXXXXX@192.168.100.21;tag=as45df46a2
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 5 [ 16]: CSeq: 620 INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 6 [ 20]: Server: Asterisk PBX
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 9 [ 74]: WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3453120a”
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 10 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 11 [ 0]:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1601
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Trying to put ‘SIP/2.0 401’ onto UDP socket destined for 192.168.100.32:5060
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Scheduling destruction of SIP dialog '1695553247-5060-64@BJC.BGI.BAA.DC’ in 32000 ms (Method: INVITE)
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 40]: ACK sip:XXXXXXXXX@192.168.100.21 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK1815663290;rport
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 49]: To: sip:XXXXXXXXX@192.168.100.21;tag=as45df46a2
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 13]: CSeq: 620 ACK
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [ 0]:
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (7 headers 0 lines) —
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1601
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Stopping retransmission on '1695553247-5060-64@BJC.BGI.BAA.DC’ of Response 620: Match Found
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 43]: INVITE sip:XXXXXXXXX@192.168.100.21 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK573606872;rport
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 34]: To: sip:XXXXXXXXX@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 16]: CSeq: 621 INVITE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 48]: Contact: “Sergio” sip:1001@192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [169]: Authorization: Digest username=“1001”, realm=“asterisk”, nonce=“3453120a”, uri="sip:XXXXXXXXX@192.168.100.21", response=“53d2dac71b43c9a28a3ca6dd46435ef6”, algorithm=MD5
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 9 [ 40]: User-Agent: Grandstream GXP1625 1.0.1.12
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 10 [ 13]: Privacy: none
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 11 [ 56]: P-Preferred-Identity: “Sergio” sip:1001@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 12 [ 32]: Supported: replaces, path, timer
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 13 [ 89]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 15 [ 47]: Accept: application/sdp, application/dtmf-relay
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 16 [ 21]: Content-Length: 337
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 17 [ 0]:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 0 [ 3]: v=0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 1 [ 38]: o=1001 8000 8000 IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.100.32
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 5 [ 35]: m=audio 5004 RTP/AVP 0 8 18 9 2 101
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 6 [ 10]: a=sendrecv
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 8 [ 10]: a=ptime:20
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 9 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 12 [ 20]: a=rtpmap:9 G722/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 13 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Body 15 [ 15]: a=fmtp:101 0-15
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (17 headers 16 lines) —
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.32:5060’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.32’ and port ‘5060’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Sending to 192.168.100.32:5060 (no NAT)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Initializing initreq for method INVITE - callid 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Using INVITE request as basis request - 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found peer ‘1001’ for ‘1001’ from 192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Allocated port 10066 for RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: RTP instance ‘0xf3b3b294’ is setup and ready to go
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] netsock2.c: Using SIP RTP CoS mark 5
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Setting NAT on RTP to Off
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP o=1001 8000 8000 IN IP4 192.168.100.32… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP s=SIP Call… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.32’ into…
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.32’ and port ‘’.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.100.32… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing session-level SDP t=0 0… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 0 (0xf3d27eac) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 8
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 8 (0xf3d2799c) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 18
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 18 (0xf3b3e83c) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 9
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 9 (0xf3d0a864) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 2
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 2 (0xf3d14be4) based on m type on 0xf489085c
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found RTP audio format 101
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Setting payload 101 (0xf3d48144) based on m type on 0xf489085c
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=sendrecv… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format PCMU for ID 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=ptime:20… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format G729 for ID 18
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format G722 for ID 9
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format G726-32 for ID 2
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000… OK.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15… UNSUPPORTED OR FAILED.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|g726|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Peer audio RTP is at port 192.168.100.32:5004
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 0 (0xf3d89d7c) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 2 (0xf3d0a864) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 8 (0xf3d27eac) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 9 (0xf3b3e83c) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 18 (0xf3d2799c) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] rtp_engine.c: Copying payload 101 (0xf3d14be4) from 0xf489085c to 0xf3b3b3bc
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance ‘0xf3b3b294’
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: We’re settling with these formats: (ulaw)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Checking SIP call limits for device 1001
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Updating call counter for incoming call
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Call from peer ‘1001’ is 1 out of 2147483647
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: No provider found, checking channel drivers for SIP - 1001
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] DEBUG[14477] chan_sip.c: Checking device state for peer 1001
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: Splitting ‘192.168.100.21’ into…
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: Changing state for SIP/1001 - state 2 (In use)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] netsock2.c: …host ‘192.168.100.21’ and port ‘’.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Looking for XXXXXXXXX in DLPN_DialPlan1 (domain 192.168.100.21)
[Nov 4 16:13:16] DEBUG[14814] app_queue.c: Device ‘SIP/1001’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Nov 4 16:13:16] DEBUG[14674] app_queue.c: Extension ‘1001@default’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Incoming INVITE with ‘timer’ option supported
[Nov 4 16:13:16] DEBUG[14394] threadpool.c: Increasing threadpool stasis-core’s size by 1
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** Our native formats are (ulaw)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** Joint capabilities are (ulaw)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** Our capabilities are (ulaw|gsm)
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: This channel will not be able to handle video.
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] sip/route.c: sip_route_dump: route/path hop: sip:1001@192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: SIP/1001-0000005f: New call is still down… Trying…
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 1 [ 95]: Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK573606872;received=192.168.100.32;rport=5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 2 [ 54]: From: “Sergio” sip:1001@192.168.100.21;tag=259365463
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 3 [ 34]: To: sip:XXXXXXXXX@192.168.100.21
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 4 [ 42]: Call-ID: 1695553247-5060-64@BJC.BGI.BAA.DC
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 5 [ 16]: CSeq: 621 INVITE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 6 [ 20]: Server: Asterisk PBX
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 7 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 9 [ 35]: Session-Expires: 1800;refresher=uas
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 10 [ 44]: Contact: sip:XXXXXXXXX@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 11 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 12 [ 0]:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Trying to put ‘SIP/2.0 100’ onto UDP socket destined for 192.168.100.32:5060
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: No provider found, checking channel drivers for SIP - 1001
[Nov 4 16:13:16] DEBUG[14477] chan_sip.c: Checking device state for peer 1001
[Nov 4 16:13:16] DEBUG[14477] devicestate.c: Changing state for SIP/1001 - state 2 (In use)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ROUTER’ is ‘SIP/ROUTER’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘EXTEN’ is ‘XXXXXXXXX’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Macro’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [XXXXXXXXX@DLPN_DialPlan1:1] Macro(“SIP/1001-0000005f”, “trunkdial-failover-0.3,SIP/ROUTER/XXXXXXXXX,ROUTER,”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘FMCIDNUM’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘GotoIf’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:1] GotoIf(“SIP/1001-0000005f”, “0?1-fmsetcid,1”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Not taking any branch
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: GotoIf
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘GLOBAL_OUTBOUNDCIDNAME’ is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘GotoIf’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:2] GotoIf(“SIP/1001-0000005f”, “0?1-setgbobname,1”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Not taking any branch
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: GotoIf
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function CALLERID(num) result is ‘1001’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘CID_1001’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function CALLERID(num) result is ‘1001’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘CID_1001’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function IF(0? result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Set’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:3] Set(“SIP/1001-0000005f”, “CALLERID(num)=”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Set
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG5’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG5’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function IF(0? result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Set’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:4] Set(“SIP/1001-0000005f”, “CALLERID(all)=”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Set
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function CALLERID(num) result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘GotoIf’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:5] GotoIf(“SIP/1001-0000005f”, “0?1-dial,1”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Not taking any branch
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: GotoIf
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG3’ is ‘ROUTER’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘CID_ROUTER’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG3’ is ‘ROUTER’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘CID_ROUTER’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘GLOBAL_OUTBOUNDCID’ is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function IF(0? result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Set’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:6] Set(“SIP/1001-0000005f”, “CALLERID(all)=”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Set
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG5’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function LEN() result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Expression result is ‘0’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG5’ is NULL
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Function IF(0? result is ‘’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Set’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:7] Set(“SIP/1001-0000005f”, “CALLERID(all)=”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Set
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Goto’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [s@macro-trunkdial-failover-0.3:8] Goto(“SIP/1001-0000005f”, “1-dial,1”) in new stack
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Goto (macro-trunkdial-failover-0.3,1-dial,1)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] app_macro.c: Executed application: Goto
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Result of ‘ARG1’ is ‘SIP/ROUTER/XXXXXXXXX’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] pbx.c: Launching ‘Dial’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] pbx.c: Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial(“SIP/1001-0000005f”, “SIP/ROUTER/XXXXXXXXX”) in new stack
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Asked to create a SIP channel with formats: (ulaw)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Allocating new SIP dialog for 179f4c2e083d5b6842f220787003208a@192.168.100.22:5060 - INVITE (No RTP)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0xf46466f4’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] res_rtp_asterisk.c: Allocated port 19714 for RTP instance ‘0xf46466f4’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: RTP instance ‘0xf46466f4’ is setup and ready to go
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0xf46466f4’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] netsock2.c: Using SIP RTP CoS mark 5
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Setting NAT on RTP to Off
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] acl.c: For destination ‘192.168.1.1’, our source address is ‘192.168.100.21’.
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Setting NAT on RTP to Off
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: SIP call-id changed from ‘179f4c2e083d5b6842f220787003208a@192.168.100.22:5060’ to ‘17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** Our native formats are (ulaw)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** Joint capabilities are (ulaw)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** Our capabilities are (ulaw|alaw|gsm|g726|g729)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: This channel will not be able to handle video.
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] channel_internal_api.c: Channel Call ID changing from [C-000001bf] to [C-000001bf]
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 0 (0xf3d89d7c) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 2 (0xf3d0a864) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 8 (0xf3d27eac) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 9 (0xf3b3e83c) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 18 (0xf3d2799c) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Copying payload 101 (0xf3d14be4) from 0xf3b3b3bc to 0xf464681c
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] rtp_engine.c: Seeded SDP of ‘SIP/ROUTER-00000060’ with that of ‘SIP/1001-0000005f’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Outgoing Call for XXXXXXXXX
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Updating call counter for outgoing call
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: ** Our capability: (ulaw|alaw|gsm|g726|g729) Video flag: False Text flag: False
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: ** Our prefcodec: (ulaw)
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Audio is at 19714
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec ulaw to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec alaw to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec gsm to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec g726 to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding codec g729 to SDP
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: – Done with adding codecs to SDP
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|alaw|gsm|g726|g729)
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Initializing initreq for method INVITE - callid 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 0 [ 40]: INVITE sip:XXXXXXXXX@192.168.1.1 SIP/2.0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK1f56d062
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 3 [ 61]: From: “asterisk” sip:asterisk@192.168.100.21;tag=as410dcd29
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 4 [ 31]: To: sip:XXXXXXXXX@192.168.1.1
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 5 [ 43]: Contact: sip:asterisk@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 6 [ 61]: Call-ID: 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 9 [ 35]: Date: Wed, 04 Nov 2015 15:13:16 GMT
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.1:5060:
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 0 [ 40]: INVITE sip:XXXXXXXXX@192.168.1.1 SIP/2.0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK1f56d062
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 3 [ 61]: From: “asterisk” sip:asterisk@192.168.100.21;tag=as410dcd29
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 4 [ 31]: To: sip:XXXXXXXXX@192.168.1.1
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 5 [ 43]: Contact: sip:asterisk@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 6 [ 61]: Call-ID: 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 9 [ 35]: Date: Wed, 04 Nov 2015 15:13:16 GMT
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 13 [ 19]: Content-Length: 374
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Header 14 [ 0]:
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 0 [ 3]: v=0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 1 [ 48]: o=root 262529552 262529552 IN IP4 192.168.100.21
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 2 [ 21]: s=Asterisk PBX 13.1.0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.100.21
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 5 [ 36]: m=audio 19714 RTP/AVP 0 8 3 2 18 101
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 9 [ 23]: a=rtpmap:2 G726-32/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 10 [ 21]: a=rtpmap:18 G729/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 11 [ 19]: a=fmtp:18 annexb=no
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 12 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 13 [ 15]: a=fmtp:101 0-16
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 14 [ 10]: a=ptime:20
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 15 [ 14]: a=maxptime:150
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Body 16 [ 10]: a=sendrecv
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #1604
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Trying to put ‘INVITE sip:’ onto UDP socket destined for 192.168.1.1:5060
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] app_dial.c: Called SIP/ROUTER/XXXXXXXXX
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c:
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 0 [ 21]: SIP/2.0 404 Not Found
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 1 [ 80]: Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK1f56d062;received=192.168.1.2
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 2 [ 61]: From: “asterisk” sip:asterisk@192.168.100.21;tag=as410dcd29
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 3 [ 68]: To: sip:XXXXXXXXX@192.168.1.1;tag=490977b9c933825154b0b9671e5afc51
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 4 [ 61]: Call-ID: 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 6 [ 43]: User-Agent: DSL Router/DSL Router-00.96.315
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 7 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741] chan_sip.c: Header 8 [ 0]:
[Nov 4 16:13:16] VERBOSE[14741] chan_sip.c: — (8 headers 0 lines) —
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Acked pending invite 102
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1604
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Stopping retransmission on ‘17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060’ of Request 102: Match Found
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: SIP response 404 to standard invite
[Nov 4 16:13:16] VERBOSE[14741][C-000001bf] chan_sip.c: Transmitting (no NAT) to 192.168.1.1:5060:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 0 [ 37]: ACK sip:XXXXXXXXX@192.168.1.1 SIP/2.0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.100.21:5060;branch=z9hG4bK1f56d062
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 3 [ 61]: From: “asterisk” sip:asterisk@192.168.100.21;tag=as410dcd29
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 4 [ 68]: To: sip:XXXXXXXXX@192.168.1.1;tag=490977b9c933825154b0b9671e5afc51
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 5 [ 43]: Contact: sip:asterisk@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 6 [ 61]: Call-ID: 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 7 [ 13]: CSeq: 102 ACK
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 9 [ 17]: Content-Length: 0
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Header 10 [ 0]:
[Nov 4 16:13:16] DEBUG[14741][C-000001bf] chan_sip.c: Trying to put ‘ACK sip:644’ onto UDP socket destined for 192.168.1.1:5060
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] channel.c: Hanging up channel ‘SIP/ROUTER-00000060’
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] chan_sip.c: Hangup call SIP/ROUTER-00000060, SIP callid 17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060
[Nov 4 16:13:16] DEBUG[14664] cdr.c: Finalized CDR for SIP/1001-0000005f - start 1446649996.779049 answer 0.000000 end 1446649996.819008 dispo NO ANSWER
[Nov 4 16:13:16] DEBUG[24726][C-000001bf] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0xf46466f4’
[Nov 4 16:13:16] VERBOSE[24726][C-000001bf] chan_sip.c: Scheduling destruction of SIP dialog ‘17ed7ac67cb18fc83bc73ff972eeb94e@192.168.100.21:5060’ in 32000 ms (Method: INVITE)
[Nov 4 16:13:16] DEBUG[14664] cdr.c: Finalized CDR for SIP/ROUTER-00000060 - start 1446649996.789172 answer 0.000000 end 1446649996.819905 dispo NO ANSWER
[Nov 4 16:13:16] DEBUG[14664] cdr.c: CDR for SIP/ROUTER-00000060 is dialed and has no Party B; discarding