<------------> [Mar 29 14:59:30] DEBUG[2735]: chan_sip.c:14817 sipsock_read: Invalid SIP message - rejected , no callid, len 328 anarki*CLI> <--- SIP read from 10.10.10.23:5061 ---> INVITE sip:0123456789@172.16.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-a8bb1448 From: fax ;tag=31668792b665b3b8o1 To: Remote-Party-ID: fax ;screen=yes;party=calling Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 101 INVITE Max-Forwards: 70 Contact: fax Expires: 240 User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 446 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 1094563 1094563 IN IP4 10.10.10.23 s=- c=IN IP4 10.10.10.23 t=0 0 m=audio 16472 RTP/AVP 8 0 2 4 18 96 97 98 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: INVITE sip:0123456789@172.16.0.25 SIP/2.0 (40) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-a8bb1448 (59) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: fax ;tag=31668792b665b3b8o1 (59) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: (31) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Remote-Party-ID: fax ;screen=yes;party=calling (72) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 (40) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: CSeq: 101 INVITE (16) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Contact: fax (47) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: Expires: 240 (12) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: User-Agent: Linksys/SPA2102-3.3.6 (33) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 11: Content-Length: 446 (19) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 12: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 13: Supported: x-sipura (19) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 14: Content-Type: application/sdp (29) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 15: (0) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: v=0 (3) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: o=- 1094563 1094563 IN IP4 10.10.10.23 (40) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: s=- (3) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: c=IN IP4 10.10.10.23 (22) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: t=0 0 (5) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: m=audio 16472 RTP/AVP 8 0 2 4 18 96 97 98 100 101 (49) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:18 G729a/8000 (22) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:96 G726-40/8000 (24) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:97 G726-24/8000 (24) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:98 G726-16/8000 (24) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:100 NSE/8000 (21) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:100 192-193 (18) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=ptime:30 (10) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=sendrecv (10) --- (15 headers 20 lines) --- [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2575 do_setnat: Setting NAT on RTP to Off [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2580 do_setnat: Setting NAT on VRTP to Off [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2585 do_setnat: Setting NAT on UDPTL to Off [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4310 sip_alloc: Allocating new SIP dialog for 7cc2f01a-2b9d24d0@10.10.10.23 - INVITE (With RTP) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:14633 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:1680 parse_sip_options: Begin: parsing SIP "Supported: x-sipura" [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:1688 parse_sip_options: Found SIP option: -x-sipura- [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:1700 parse_sip_options: Found private SIP option, not supported: x-sipura Sending to 10.10.10.23 : 5061 (no NAT) Using INVITE request as basis request - 7cc2f01a-2b9d24d0@10.10.10.23 [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2575 do_setnat: Setting NAT on RTP to On [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2580 do_setnat: Setting NAT on VRTP to On [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2585 do_setnat: Setting NAT on UDPTL to On <--- Reliably Transmitting (NAT) to 10.10.10.23:5061 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-a8bb1448;received=10.10.10.23 From: fax ;tag=31668792b665b3b8o1 To: ;tag=as25be85d2 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 101 INVITE User-Agent: blabla.ch 1.4PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="blabla.ch", nonce="03e45a7d" Content-Length: 0 <------------> [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:1975 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #1340 Scheduling destruction of SIP dialog '7cc2f01a-2b9d24d0@10.10.10.23' in 32000 ms (Method: INVITE) Found user 'blabla' anarki*CLI> <--- SIP read from 10.10.10.23:5061 ---> ACK sip:0123456789@172.16.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-a8bb1448 From: fax ;tag=31668792b665b3b8o1 To: ;tag=as25be85d2 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 101 ACK Max-Forwards: 70 Contact: fax User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 <-------------> [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: ACK sip:0123456789@172.16.0.25 SIP/2.0 (37) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-a8bb1448 (59) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: fax ;tag=31668792b665b3b8o1 (59) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=as25be85d2 (46) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 (40) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 101 ACK (13) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Max-Forwards: 70 (16) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Contact: fax (47) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: User-Agent: Linksys/SPA2102-3.3.6 (33) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: Content-Length: 0 (17) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 7cc2f01a-2b9d24d0@10.10.10.23 Their Tag 31668792b665b3b8o1 Our tag: as25be85d2 [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:14633 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2079 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1340 [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2089 __sip_ack: Stopping retransmission on '7cc2f01a-2b9d24d0@10.10.10.23' of Response 101: Match Not Found anarki*CLI> <--- SIP read from 10.10.10.23:5061 ---> INVITE sip:0123456789@172.16.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-d14f11cc From: fax ;tag=31668792b665b3b8o1 To: Remote-Party-ID: fax ;screen=yes;party=calling Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="blabla",realm="blabla.ch",nonce="03e45a7d",uri="sip:0123456789@172.16.0.25",algorithm=MD5,response="dd87757ea155cc744333a0a47ca42f8a" Contact: fax Expires: 240 User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 446 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 1094563 1094563 IN IP4 10.10.10.23 s=- c=IN IP4 10.10.10.23 t=0 0 m=audio 16472 RTP/AVP 8 0 2 4 18 96 97 98 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: INVITE sip:0123456789@172.16.0.25 SIP/2.0 (40) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-d14f11cc (59) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: fax ;tag=31668792b665b3b8o1 (59) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: (31) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Remote-Party-ID: fax ;screen=yes;party=calling (72) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 (40) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Proxy-Authorization: Digest username="blabla",realm="blabla.ch",nonce="03e45a7d",uri="sip:0123456789@172.16.0.25",algorithm=MD5,response="dd87757ea155cc744333a0a47ca42f8a" (171) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: Contact: fax (47) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: Expires: 240 (12) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 11: User-Agent: Linksys/SPA2102-3.3.6 (33) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 12: Content-Length: 446 (19) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 13: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 14: Supported: x-sipura (19) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 15: Content-Type: application/sdp (29) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 16: (0) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: v=0 (3) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: o=- 1094563 1094563 IN IP4 10.10.10.23 (40) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: s=- (3) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: c=IN IP4 10.10.10.23 (22) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: t=0 0 (5) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: m=audio 16472 RTP/AVP 8 0 2 4 18 96 97 98 100 101 (49) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:18 G729a/8000 (22) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:96 G726-40/8000 (24) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:97 G726-24/8000 (24) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:98 G726-16/8000 (24) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:100 NSE/8000 (21) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:100 192-193 (18) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=ptime:30 (10) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=sendrecv (10) --- (16 headers 20 lines) --- [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 7cc2f01a-2b9d24d0@10.10.10.23 Their Tag 31668792b665b3b8o1 Our tag: as25be85d2 [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:14633 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:1680 parse_sip_options: Begin: parsing SIP "Supported: x-sipura" [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:1688 parse_sip_options: Found SIP option: -x-sipura- [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:1700 parse_sip_options: Found private SIP option, not supported: x-sipura Sending to 10.10.10.23 : 5061 (NAT) Using INVITE request as basis request - 7cc2f01a-2b9d24d0@10.10.10.23 [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2575 do_setnat: Setting NAT on RTP to On [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2580 do_setnat: Setting NAT on VRTP to On [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2585 do_setnat: Setting NAT on UDPTL to On Found user 'blabla' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4899 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 10.10.10.23:16472 Found description format PCMA for ID 8 Found description format PCMU for ID 0 Found description format G726-32 for ID 2 Found description format G723 for ID 4 Found description format G729a for ID 18 Found description format G726-40 for ID 96 Found description format G726-24 for ID 97 Found description format G726-16 for ID 98 Found description format NSE for ID 100 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:5129 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0xa (gsm|alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.10.23:16472 [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:5209 process_sdp: We're settling with these formats: 0x8 (alaw) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:13401 handle_request_invite: Checking SIP call limits for device blabla [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:3003 update_call_counter: Updating call counter for incoming call Looking for 0123456789 in privileged-prod (domain 172.16.0.25) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:3805 sip_new: *** Our native formats are 0x8 (alaw) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:3806 sip_new: *** Joint capabilities are 0x8 (alaw) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:3807 sip_new: *** Our capabilities are 0xa (gsm|alaw) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:3808 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:3831 sip_new: This channel will not be able to handle video. [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:7980 build_route: build_route: Contact hop: fax list_route: hop: [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:13476 handle_request_invite: SIP/blabla-b7830da8: New call is still down.... Trying... <--- Transmitting (NAT) to 10.10.10.23:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-d14f11cc;received=10.10.10.23 From: fax ;tag=31668792b665b3b8o1 To: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 102 INVITE User-Agent: blabla.ch 1.4PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 29 14:59:35] DEBUG[2735]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/blabla-b7830da8 [Mar 29 14:59:35] DEBUG[2711]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - blabla [Mar 29 14:59:35] DEBUG[2711]: chan_sip.c:15244 sip_devicestate: Checking device state for peer blabla [Mar 29 14:59:35] DEBUG[2711]: devicestate.c:287 do_state_change: Changing state for SIP/blabla - state 1 (Not in use) [Mar 29 14:59:35] DEBUG[15103]: app_queue.c:546 changethread: Device 'SIP/blabla' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 29 14:59:35] DEBUG[2711]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - blabla [Mar 29 14:59:35] DEBUG[2711]: chan_sip.c:15244 sip_devicestate: Checking device state for peer blabla [Mar 29 14:59:35] DEBUG[15102]: pbx.c:1795 pbx_extension_helper: Launching 'Dial' -- Executing [0123456789@privileged-prod:1] Dial("SIP/blabla-b7830da8", "SIP/0123456789@172.16.0.20") in new stack [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:15310 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4310 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:3805 sip_new: *** Our native formats are 0x8 (alaw) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:3806 sip_new: *** Joint capabilities are 0x0 (nothing) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:3807 sip_new: *** Our capabilities are 0x40a (gsm|alaw|ilbc) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:3808 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:3810 sip_new: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:3831 sip_new: This channel will not be able to handle video. [Mar 29 14:59:35] DEBUG[15102]: rtp.c:1598 ast_rtp_make_compatible: Seeded SDP of 'SIP/172.16.0.20-081d1620' with that of 'SIP/blabla-b7830da8' [Mar 29 14:59:35] DEBUG[15102]: channel.c:3301 ast_channel_inherit_variables: Not copying variable STACK-privileged-prod-0123456789-1. [Mar 29 14:59:35] DEBUG[15102]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Mar 29 14:59:35] DEBUG[15102]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Mar 29 14:59:35] DEBUG[15102]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Mar 29 14:59:35] DEBUG[15102]: channel.c:3301 ast_channel_inherit_variables: Not copying variable SIPURI. [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:2830 sip_call: Outgoing Call for 0123456789 [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:3003 update_call_counter: Updating call counter for outgoing call [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:2845 sip_call: Our T38 capability (3856), joint T38 capability (3856) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:6188 add_sdp: ** Our capability: 0x40a (gsm|alaw|ilbc) Video flag: False [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:6189 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 172.16.0.25 port 10998 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:6320 add_sdp: -- Done with adding codecs to SDP [Mar 29 14:59:35] DEBUG[15102]: channel.c:2381 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=36) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:6365 add_sdp: Done building SDP. Settling with this capability: 0x40a (gsm|alaw|ilbc) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 0: INVITE sip:0123456789@172.16.0.20 SIP/2.0 (40) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK54024be6;rport (61) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 2: From: "anon" ;tag=as01a570eb (55) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 3: To: (31) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 4: Contact: (36) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 5: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 (52) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 7: User-Agent: blabla.ch 1.4PBX (26) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 9: Date: Thu, 29 Mar 2007 12:59:35 GMT (35) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 11: Supported: replaces (19) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 13: Content-Length: 304 (19) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 14: (0) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: v=0 (3) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: o=root 15102 15102 IN IP4 172.16.0.25 (36) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: s=session (9) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: c=IN IP4 172.16.0.25 (19) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: t=0 0 (5) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: m=audio 10998 RTP/AVP 8 3 97 101 (32) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=rtpmap:97 iLBC/8000 (21) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=fmtp:97 mode=30 (17) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=ptime:20 (10) [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 172.16.0.20:5060: INVITE sip:0123456789@172.16.0.20 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK54024be6;rport From: "anon" ;tag=as01a570eb To: Contact: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 102 INVITE User-Agent: blabla.ch 1.4PBX Max-Forwards: 70 Date: Thu, 29 Mar 2007 12:59:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 304 v=0 o=root 15102 15102 IN IP4 172.16.0.25 s=session c=IN IP4 172.16.0.25 t=0 0 m=audio 10998 RTP/AVP 8 3 97 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Mar 29 14:59:35] DEBUG[15102]: chan_sip.c:1975 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #1342 -- Called 0123456789@172.16.0.20 <--- SIP read from 172.16.0.20:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK54024be6;rport=5060 From: "anon" ;tag=as01a570eb To: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 102 INVITE <-------------> [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: SIP/2.0 100 Trying (18) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK54024be6;rport=5060 (86) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: "anon" ;tag=as01a570eb (55) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: (31) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 (52) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 102 INVITE (16) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: (0) --- (6 headers 0 lines) --- [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag Our tag: as01a570eb [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2122 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #1342 - INVITE (got response) [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:2131 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1500a200028c8c5509988c94046c25f2@172.16.0.25' Request 102: Found [Mar 29 14:59:35] DEBUG[2735]: chan_sip.c:11641 handle_response_invite: SIP response 100 to standard invite [Mar 29 14:59:36] DEBUG[15102]: chan_sip.c:6420 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 29 14:59:36] DEBUG[15102]: chan_sip.c:6188 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Mar 29 14:59:36] DEBUG[15102]: chan_sip.c:6189 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 172.16.0.25 port 11176 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Mar 29 14:59:36] DEBUG[15102]: chan_sip.c:6320 add_sdp: -- Done with adding codecs to SDP [Mar 29 14:59:36] DEBUG[15102]: channel.c:2381 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) [Mar 29 14:59:36] DEBUG[15102]: chan_sip.c:6365 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) anarki*CLI> <--- Transmitting (NAT) to 10.10.10.23:5061 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-d14f11cc;received=10.10.10.23 From: fax ;tag=31668792b665b3b8o1 To: ;tag=as6fa456c1 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 102 INVITE User-Agent: blabla.ch 1.4PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 236 v=0 o=root 15102 15102 IN IP4 172.16.0.25 s=session c=IN IP4 172.16.0.25 t=0 0 m=audio 11176 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 <------------> [Mar 29 14:59:36] DEBUG[15102]: rtp.c:2689 ast_rtp_write: Ooh, format changed from unknown to alaw [Mar 29 14:59:36] DEBUG[15102]: rtp.c:2706 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 anarki*CLI> <--- SIP read from 172.16.0.20:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK54024be6;rport=5060 From: "anon" ;tag=as01a570eb To: ;tag=510742046 Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 343 v=0 o=- 47867 1 IN IP4 172.16.0.20 s=Cisco SDP 0 c=IN IP4 172.16.0.20 t=0 0 m=audio 21346 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 200-202 a=X-cap: 2 image udptl t38 <-------------> [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: SIP/2.0 183 Session Progress (28) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK54024be6;rport=5060 (86) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: "anon" ;tag=as01a570eb (55) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=510742046 (45) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 (52) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 102 INVITE (16) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Contact: (55) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Content-Type: application/sdp (29) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Content-Length: 343 (19) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: (0) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: v=0 (3) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: o=- 47867 1 IN IP4 172.16.0.20 (29) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: s=Cisco SDP 0 (13) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: c=IN IP4 172.16.0.20 (19) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: t=0 0 (5) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: m=audio 21346 RTP/AVP 8 101 100 (31) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:100 X-NSE/8000 (23) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:100 200-202 (18) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-sqn:0 (9) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cap: 1 audio RTP/AVP 100 (28) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cpar: a=rtpmap:100 X-NSE/8000 (33) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cpar: a=fmtp:100 200-202 (28) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cap: 2 image udptl t38 (26) --- (9 headers 15 lines) --- [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag Our tag: as01a570eb [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:2131 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1500a200028c8c5509988c94046c25f2@172.16.0.25' Request 102: Found [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:11641 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 8 Found RTP audio format 101 Found RTP audio format 100 [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:4899 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 172.16.0.20:21346 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Found description format X-NSE for ID 100 Got unsupported a:fmtp in SDP offer [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:5129 process_sdp: T38 state changed to 0 on channel SIP/172.16.0.20-081d1620 Capabilities: us - 0x40a (gsm|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.0.20:21346 Peer video RTP is at port 172.16.0.20:63927 [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:5209 process_sdp: We're settling with these formats: 0x8 (alaw) [Mar 29 14:59:37] DEBUG[2735]: chan_sip.c:5216 process_sdp: We have an owner, now see if we need to change this call -- Call on SIP/172.16.0.20-081d1620 left from hold -- SIP/172.16.0.20-081d1620 is making progress passing it to SIP/blabla-b7830da8 [Mar 29 14:59:37] DEBUG[15102]: rtp.c:1527 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/blabla-b7830da8' with that of 'SIP/172.16.0.20-081d1620' [Mar 29 14:59:37] DEBUG[15102]: rtp.c:2689 ast_rtp_write: Ooh, format changed from unknown to alaw [Mar 29 14:59:37] DEBUG[15102]: rtp.c:2706 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 anarki*CLI> <--- SIP read from 172.16.0.20:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK54024be6;rport=5060 From: "anon" ;tag=as01a570eb To: ;tag=510742046 Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 102 INVITE Contact: Content-Length: 0 <-------------> [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK54024be6;rport=5060 (86) [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: "anon" ;tag=as01a570eb (55) [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=510742046 (45) [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 (52) [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 102 INVITE (16) [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Contact: (55) [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Content-Length: 0 (17) [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag 510742046 Our tag: as01a570eb [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:2131 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1500a200028c8c5509988c94046c25f2@172.16.0.25' Request 102: Found [Mar 29 14:59:38] DEBUG[2735]: chan_sip.c:11641 handle_response_invite: SIP response 180 to standard invite [Mar 29 14:59:38] DEBUG[2735]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/172.16.0.20-081d1620 [Mar 29 14:59:38] DEBUG[2711]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 172.16.0.20 [Mar 29 14:59:38] DEBUG[2711]: chan_sip.c:15244 sip_devicestate: -- SIP/172.16.0.20-081d1620 is ringing Checking device state for peer 172.16.0.20 [Mar 29 14:59:38] DEBUG[15102]: rtp.c:1527 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/blabla-b7830da8' with that of 'SIP/172.16.0.20-081d1620' anarki*CLI> <--- Transmitting (NAT) to 10.10.10.23:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-d14f11cc;received=10.10.10.23 From: fax ;tag=31668792b665b3b8o1 To: ;tag=as6fa456c1 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 102 INVITE User-Agent: blabla.ch 1.4PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 29 14:59:38] DEBUG[2711]: devicestate.c:287 do_state_change: Changing state for SIP/172.16.0.20 - state 6 (Ringing) [Mar 29 14:59:38] DEBUG[15104]: app_queue.c:546 changethread: Device 'SIP/172.16.0.20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Mar 29 14:59:39] DEBUG[15102]: rtp.c:873 ast_rtcp_read: Got RTCP report of 92 bytes [Mar 29 14:59:39] DEBUG[15102]: rtp.c:873 ast_rtcp_read: Got RTCP report of 88 bytes [Mar 29 14:59:41] DEBUG[15102]: rtp.c:873 ast_rtcp_read: Got RTCP report of 88 bytes [Mar 29 14:59:42] DEBUG[15102]: rtp.c:873 ast_rtcp_read: Got RTCP report of 92 bytes anarki*CLI> <--- SIP read from 172.16.0.20:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK54024be6;rport=5060 From: "anon" ;tag=as01a570eb To: ;tag=510742046 Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 102 INVITE Contact: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Supported: timer Content-Type: application/sdp Content-Length: 343 v=0 o=- 47867 1 IN IP4 172.16.0.20 s=Cisco SDP 0 c=IN IP4 172.16.0.20 t=0 0 m=audio 21346 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 200-202 a=X-cap: 2 image udptl t38 <-------------> [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: SIP/2.0 200 Ok (14) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK54024be6;rport=5060 (86) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: "anon" ;tag=as01a570eb (55) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=510742046 (45) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 (52) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 102 INVITE (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Contact: (55) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE (69) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Supported: timer (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: Content-Type: application/sdp (29) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: Content-Length: 343 (19) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 11: (0) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: v=0 (3) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: o=- 47867 1 IN IP4 172.16.0.20 (29) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: s=Cisco SDP 0 (13) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: c=IN IP4 172.16.0.20 (19) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: t=0 0 (5) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: m=audio 21346 RTP/AVP 8 101 100 (31) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:100 X-NSE/8000 (23) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:100 200-202 (18) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-sqn:0 (9) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cap: 1 audio RTP/AVP 100 (28) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cpar: a=rtpmap:100 X-NSE/8000 (33) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cpar: a=fmtp:100 200-202 (28) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cap: 2 image udptl t38 (26) --- (11 headers 15 lines) --- [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag 510742046 Our tag: as01a570eb [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2071 __sip_ack: Acked pending invite 102 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2089 __sip_ack: Stopping retransmission on '1500a200028c8c5509988c94046c25f2@172.16.0.25' of Request 102: Match Not Found [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:11641 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 8 Found RTP audio format 101 Found RTP audio format 100 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4899 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 172.16.0.20:21346 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Found description format X-NSE for ID 100 Got unsupported a:fmtp in SDP offer [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5129 process_sdp: T38 state changed to 0 on channel SIP/172.16.0.20-081d1620 Capabilities: us - 0x40a (gsm|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.0.20:21346 Peer video RTP is at port 172.16.0.20:7688 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5209 process_sdp: We're settling with these formats: 0x8 (alaw) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5216 process_sdp: We have an owner, now see if we need to change this call [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:3003 update_call_counter: Updating call counter for outgoing call [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:7980 build_route: build_route: Contact hop: list_route: hop: [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5643 reqprep: Strict routing enforced for session 1500a200028c8c5509988c94046c25f2@172.16.0.25 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.0.20, port 5060 Transmitting (no NAT) to 172.16.0.20:5060: ACK sip:0123456789@172.16.0.20:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK5926cff1;rport From: "anon" ;tag=as01a570eb To: ;tag=510742046 Contact: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 102 ACK User-Agent: blabla.ch 1.4PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/172.16.0.20-081d1620 left from hold [Mar 29 14:59:44] DEBUG[15102]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/172.16.0.20-081d1620 -- SIP/172.16.0.20-081d1620 answered SIP/blabla-b7830da8 [Mar 29 14:59:44] DEBUG[15102]: rtp.c:1527 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/blabla-b7830da8' with that of 'SIP/172.16.0.20-081d1620' [Mar 29 14:59:44] DEBUG[15102]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/blabla-b7830da8 [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:3463 sip_answer: SIP answering channel: SIP/blabla-b7830da8 [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:6420 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:6188 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:6189 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 172.16.0.25 port 11176 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 172.16.0.20 [Mar 29 14:59:44] DEBUG[2711]: chan_sip.c:15244 sip_devicestate: Checking device state for peer 172.16.0.20 [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:287 do_state_change: Changing state for SIP/172.16.0.20 - state 2 (In use) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:6320 add_sdp: -- Done with adding codecs to SDP [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - blabla [Mar 29 14:59:44] DEBUG[15105]: app_queue.c:546 changethread: Device 'SIP/172.16.0.20' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Mar 29 14:59:44] DEBUG[2711]: chan_sip.c:15244 sip_devicestate: Checking device state for peer blabla [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:287 do_state_change: Changing state for SIP/blabla - state 1 (Not in use) [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - blabla [Mar 29 14:59:44] DEBUG[2711]: chan_sip.c:15244 sip_devicestate: Checking device state for peer blabla [Mar 29 14:59:44] DEBUG[15106]: app_queue.c:546 changethread: Device 'SIP/blabla' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 29 14:59:44] DEBUG[15102]: channel.c:2381 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:6365 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) anarki*CLI> <--- Reliably Transmitting (NAT) to 10.10.10.23:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-d14f11cc;received=10.10.10.23 From: fax ;tag=31668792b665b3b8o1 To: ;tag=as6fa456c1 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 102 INVITE User-Agent: blabla.ch 1.4PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 236 v=0 o=root 15102 15103 IN IP4 172.16.0.25 s=session c=IN IP4 172.16.0.25 t=0 0 m=audio 11176 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 <------------> [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:1975 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #1346 -- Native bridging SIP/blabla-b7830da8 and SIP/172.16.0.20-081d1620 [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:16912 sip_set_rtp_peer: Deferring reinvite on SIP '7cc2f01a-2b9d24d0@10.10.10.23' - It's audio will be redirected to IP 172.16.0.20 [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:16907 sip_set_rtp_peer: Sending reinvite on SIP '1500a200028c8c5509988c94046c25f2@172.16.0.25' - It's audio soon redirected to IP 10.10.10.23 [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:5643 reqprep: Strict routing enforced for session 1500a200028c8c5509988c94046c25f2@172.16.0.25 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.0.20, port 5060 [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:6188 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:6189 add_sdp: ** Our prefcodec: 0x8 (alaw) Audio is at 172.16.0.25 port 10998 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:6320 add_sdp: -- Done with adding codecs to SDP [Mar 29 14:59:44] DEBUG[15102]: channel.c:2381 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=36) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:6365 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:1621 initialize_initreq: Initializing already initialized SIP dialog 1500a200028c8c5509988c94046c25f2@172.16.0.25 (presumably reinvite) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 0: INVITE sip:0123456789@172.16.0.20:5060;transport=udp SIP/2.0 (59) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK36b6c128;rport (61) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 2: From: "anon" ;tag=as01a570eb (55) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=510742046 (45) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 4: Contact: (36) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 5: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 (52) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 6: CSeq: 103 INVITE (16) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 7: User-Agent: blabla.ch 1.4PBX (26) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 10: Supported: replaces (19) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 13: Content-Length: 242 (19) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4573 parse_request: Header 14: (0) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: v=0 (3) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: o=root 15102 15103 IN IP4 10.10.10.23 (39) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: s=session (9) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: c=IN IP4 10.10.10.23 (22) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: t=0 0 (5) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: m=audio 16472 RTP/AVP 8 101 (27) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=ptime:20 (10) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:4605 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 172.16.0.20:5060: INVITE sip:0123456789@172.16.0.20:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK36b6c128;rport From: "anon" ;tag=as01a570eb To: ;tag=510742046 Contact: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 103 INVITE User-Agent: blabla.ch 1.4PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 242 v=0 o=root 15102 15103 IN IP4 10.10.10.23 s=session c=IN IP4 10.10.10.23 t=0 0 m=audio 16472 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 --- [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:1975 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #1347 anarki*CLI> <--- SIP read from 10.10.10.23:5061 ---> ACK sip:0123456789@172.16.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-ac672214 From: fax ;tag=31668792b665b3b8o1 To: ;tag=as6fa456c1 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="blabla",realm="blabla.ch",nonce="03e45a7d",uri="sip:0123456789@172.16.0.25",algorithm=MD5,response="e6d0efd56e7c77914e4a809a229ace58" Contact: fax User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 <-------------> [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: ACK sip:0123456789@172.16.0.25 SIP/2.0 (37) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-ac672214 (59) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: fax ;tag=31668792b665b3b8o1 (59) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=as6fa456c1 (46) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 (40) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 102 ACK (13) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Max-Forwards: 70 (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Proxy-Authorization: Digest username="blabla",realm="blabla.ch",nonce="03e45a7d",uri="sip:0123456789@172.16.0.25",algorithm=MD5,response="e6d0efd56e7c77914e4a809a229ace58" (171) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Contact: fax (47) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: User-Agent: Linksys/SPA2102-3.3.6 (33) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: Content-Length: 0 (17) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = No match Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag 510742046 Our tag: as01a570eb [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 7cc2f01a-2b9d24d0@10.10.10.23 Their Tag 31668792b665b3b8o1 Our tag: as6fa456c1 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:14633 handle_request: **** Received ACK (6) - Command in SIP ACK [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2079 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1346 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2089 __sip_ack: Stopping retransmission on '7cc2f01a-2b9d24d0@10.10.10.23' of Response 102: Match Not Found [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:11622 check_pendings: Sending pending reinvite on '7cc2f01a-2b9d24d0@10.10.10.23' [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5643 reqprep: Strict routing enforced for session 7cc2f01a-2b9d24d0@10.10.10.23 set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.23, port 5061 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:6188 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:6189 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 172.16.0.25 port 11176 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:6320 add_sdp: -- Done with adding codecs to SDP [Mar 29 14:59:44] DEBUG[2735]: channel.c:2381 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:6365 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:1621 initialize_initreq: Initializing already initialized SIP dialog 7cc2f01a-2b9d24d0@10.10.10.23 (presumably reinvite) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: INVITE sip:blabla@10.10.10.23:5061 SIP/2.0 (47) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK71f92d17;rport (61) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: ;tag=as6fa456c1 (48) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: fax ;tag=31668792b665b3b8o1 (57) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Contact: (36) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 (40) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: CSeq: 102 INVITE (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: User-Agent: blabla.ch 1.4PBX (26) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Max-Forwards: 70 (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: Supported: replaces (19) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 12: Content-Type: application/sdp (29) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 13: Content-Length: 236 (19) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 14: (0) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: v=0 (3) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: o=root 15102 15104 IN IP4 172.16.0.20 (36) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: s=session (9) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: c=IN IP4 172.16.0.20 (19) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: t=0 0 (5) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: m=audio 21346 RTP/AVP 8 101 (27) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:101 0-16 (15) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=silenceSupp:off - - - - (25) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=ptime:20 (10) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=sendrecv (10) Reliably Transmitting (NAT) to 10.10.10.23:5061: INVITE sip:blabla@10.10.10.23:5061 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK71f92d17;rport From: ;tag=as6fa456c1 To: fax ;tag=31668792b665b3b8o1 Contact: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 102 INVITE User-Agent: blabla.ch 1.4PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 236 v=0 o=root 15102 15104 IN IP4 172.16.0.20 s=session c=IN IP4 172.16.0.20 t=0 0 m=audio 21346 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 --- [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:1975 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #1348 anarki*CLI> <--- SIP read from 10.10.10.23:5061 ---> INVITE sip:0123456789@172.16.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-8df10e6d From: fax ;tag=31668792b665b3b8o1 To: ;tag=as6fa456c1 Remote-Party-ID: fax ;screen=yes;party=calling Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="blabla",realm="blabla.ch",nonce="03e45a7d",uri="sip:0123456789@172.16.0.25",algorithm=MD5,response="dd87757ea155cc744333a0a47ca42f8a" Contact: fax Expires: 30 User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 269 Content-Type: application/sdp v=0 o=- 1095410 1095410 IN IP4 10.10.10.23 s=- c=IN IP4 10.10.10.23 t=0 0 m=image 16472 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: INVITE sip:0123456789@172.16.0.25 SIP/2.0 (40) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-8df10e6d (59) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: fax ;tag=31668792b665b3b8o1 (59) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=as6fa456c1 (46) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Remote-Party-ID: fax ;screen=yes;party=calling (72) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 (40) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: CSeq: 103 INVITE (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Max-Forwards: 70 (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Proxy-Authorization: Digest username="blabla",realm="blabla.ch",nonce="03e45a7d",uri="sip:0123456789@172.16.0.25",algorithm=MD5,response="dd87757ea155cc744333a0a47ca42f8a" (171) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: Contact: fax (47) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: Expires: 30 (11) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 11: User-Agent: Linksys/SPA2102-3.3.6 (33) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 12: Content-Length: 269 (19) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 13: Content-Type: application/sdp (29) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 14: (0) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: v=0 (3) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: o=- 1095410 1095410 IN IP4 10.10.10.23 (40) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: s=- (3) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: c=IN IP4 10.10.10.23 (22) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: t=0 0 (5) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: m=image 16472 udptl t38 (23) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=T38FaxVersion:0 (17) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=T38MaxBitRate:14400 (21) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=T38FaxRateManagement:transferredTCF (37) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=T38FaxMaxBuffer:200 (21) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=T38FaxMaxDatagram:200 (23) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=T38FaxUdpEC:t38UDPRedundancy (30) --- (14 headers 12 lines) --- [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = No match Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag 510742046 Our tag: as01a570eb [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 7cc2f01a-2b9d24d0@10.10.10.23 Their Tag 31668792b665b3b8o1 Our tag: as6fa456c1 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:14633 handle_request: **** Received INVITE (5) - Command in SIP INVITE <--- Transmitting (NAT) to 10.10.10.23:5061 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-8df10e6d;received=10.10.10.23 From: fax ;tag=31668792b665b3b8o1 To: ;tag=as6fa456c1 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 103 INVITE User-Agent: blabla.ch 1.4PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:13197 handle_request_invite: Got INVITE on call where we already have pending INVITE, deferring that - 7cc2f01a-2b9d24d0@10.10.10.23 anarki*CLI> <--- SIP read from 172.16.0.20:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK36b6c128;rport=5060 From: "anon" ;tag=as01a570eb To: ;tag=510742046 Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 103 INVITE Contact: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Supported: timer Content-Type: application/sdp Content-Length: 343 v=0 o=- 47867 2 IN IP4 172.16.0.20 s=Cisco SDP 0 c=IN IP4 172.16.0.20 t=0 0 m=audio 21346 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 200-202 a=X-cap: 2 image udptl t38 <-------------> [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: SIP/2.0 200 Ok (14) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK36b6c128;rport=5060 (86) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: "anon" ;tag=as01a570eb (55) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=510742046 (45) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 (52) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 103 INVITE (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Contact: (55) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE (69) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Supported: timer (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: Content-Type: application/sdp (29) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: Content-Length: 343 (19) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 11: (0) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: v=0 (3) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: o=- 47867 2 IN IP4 172.16.0.20 (29) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: s=Cisco SDP 0 (13) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: c=IN IP4 172.16.0.20 (19) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: t=0 0 (5) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: m=audio 21346 RTP/AVP 8 101 100 (31) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:101 0-15 (15) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=rtpmap:100 X-NSE/8000 (23) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=fmtp:100 200-202 (18) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-sqn:0 (9) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cap: 1 audio RTP/AVP 100 (28) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cpar: a=rtpmap:100 X-NSE/8000 (33) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cpar: a=fmtp:100 200-202 (28) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4605 parse_request: Line: a=X-cap: 2 image udptl t38 (26) --- (11 headers 15 lines) --- [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag 510742046 Our tag: as01a570eb [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2071 __sip_ack: Acked pending invite 103 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2079 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1347 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2089 __sip_ack: Stopping retransmission on '1500a200028c8c5509988c94046c25f2@172.16.0.25' of Request 103: Match Not Found [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:11639 handle_response_invite: SIP response 200 to RE-invite on outgoing call 1500a200028c8c5509988c94046c25f2@172.16.0.25 Found RTP audio format 8 Found RTP audio format 101 Found RTP audio format 100 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4899 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 172.16.0.20:21346 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Found description format X-NSE for ID 100 Got unsupported a:fmtp in SDP offer [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5129 process_sdp: T38 state changed to 0 on channel SIP/172.16.0.20-081d1620 Capabilities: us - 0x40a (gsm|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.16.0.20:21346 Peer video RTP is at port 172.16.0.20:58039 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5209 process_sdp: We're settling with these formats: 0x8 (alaw) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5216 process_sdp: We have an owner, now see if we need to change this call [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:3003 update_call_counter: Updating call counter for outgoing call [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:7919 build_route: build_route: Retaining previous route: [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5643 reqprep: Strict routing enforced for session 1500a200028c8c5509988c94046c25f2@172.16.0.25 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.0.20, port 5060 Transmitting (no NAT) to 172.16.0.20:5060: ACK sip:0123456789@172.16.0.20:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK0a768d5b;rport From: "anon" ;tag=as01a570eb To: ;tag=510742046 Contact: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 103 ACK User-Agent: blabla.ch 1.4PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 29 14:59:44] DEBUG[15102]: rtp.c:2825 bridge_native_loop: Oooh, 'SIP/172.16.0.20-081d1620' changed end address to 172.16.0.20:21346 (format 8) [Mar 29 14:59:44] DEBUG[15102]: rtp.c:2827 bridge_native_loop: Oooh, 'SIP/172.16.0.20-081d1620' changed end vaddress to 172.16.0.20:58039 (format 8) [Mar 29 14:59:44] DEBUG[15102]: rtp.c:2829 bridge_native_loop: Oooh, 'SIP/172.16.0.20-081d1620' was 172.16.0.20:21346/(format 8) [Mar 29 14:59:44] DEBUG[15102]: rtp.c:2831 bridge_native_loop: Oooh, 'SIP/172.16.0.20-081d1620' was 172.16.0.20:7688/(format 8) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:16912 sip_set_rtp_peer: Deferring reinvite on SIP '7cc2f01a-2b9d24d0@10.10.10.23' - It's audio will be redirected to IP 172.16.0.20 <--- SIP read from 10.10.10.23:5061 ---> SIP/2.0 491 Request Pending To: fax ;tag=31668792b665b3b8o1 From: ;tag=as6fa456c1 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 102 INVITE Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK71f92d17 Server: Linksys/SPA2102-3.3.6 Content-Length: 0 <-------------> [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: To: fax ;tag=31668792b665b3b8o1 (57) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: ;tag=as6fa456c1 (48) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 (40) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: CSeq: 102 INVITE (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK71f92d17 (55) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Server: Linksys/SPA2102-3.3.6 (29) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Content-Length: 0 (17) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: (0) --- (8 headers 0 lines) --- [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = No match Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag 510742046 Our tag: as01a570eb [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 7cc2f01a-2b9d24d0@10.10.10.23 Their Tag 31668792b665b3b8o1 Our tag: as6fa456c1 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2071 __sip_ack: Acked pending invite 102 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2079 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1348 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2089 __sip_ack: Stopping retransmission on '7cc2f01a-2b9d24d0@10.10.10.23' of Request 102: Match Not Found [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:11639 handle_response_invite: SIP response 491 to RE-invite on outgoing call 7cc2f01a-2b9d24d0@10.10.10.23 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:5643 reqprep: Strict routing enforced for session 7cc2f01a-2b9d24d0@10.10.10.23 set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.23, port 5061 Transmitting (NAT) to 10.10.10.23:5061: ACK sip:blabla@10.10.10.23:5061 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK71f92d17;rport From: ;tag=as6fa456c1 To: fax ;tag=31668792b665b3b8o1 Contact: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 102 ACK User-Agent: blabla.ch 1.4PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 29 14:59:44] DEBUG[15102]: rtp.c:2904 bridge_native_loop: Got a FRAME_CONTROL (8) frame on channel SIP/blabla-b7830da8 [Mar 29 14:59:44] DEBUG[15102]: channel.c:4048 ast_channel_bridge: Returning from native bridge, channels: SIP/blabla-b7830da8, SIP/172.16.0.20-081d1620 [Mar 29 14:59:44] DEBUG[15102]: channel.c:1693 ast_hangup: Hanging up channel 'SIP/172.16.0.20-081d1620' [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:3312 sip_hangup: Hangup call SIP/172.16.0.20-081d1620, SIP callid 1500a200028c8c5509988c94046c25f2@172.16.0.25) Scheduling destruction of SIP dialog '1500a200028c8c5509988c94046c25f2@172.16.0.25' in 32000 ms (Method: INVITE) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:5643 reqprep: Strict routing enforced for session 1500a200028c8c5509988c94046c25f2@172.16.0.25 set_destination: Parsing for address/port to send to set_destination: set destination to 172.16.0.20, port 5060 Reliably Transmitting (no NAT) to 172.16.0.20:5060: BYE sip:0123456789@172.16.0.20:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK3dc90afa;rport From: "anon" ;tag=as01a570eb To: ;tag=510742046 Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 104 BYE User-Agent: blabla.ch 1.4PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:1975 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #1350 [Mar 29 14:59:44] DEBUG[15102]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/172.16.0.20-081d1620 [Mar 29 14:59:44] DEBUG[15102]: rtp.c:1476 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Mar 29 14:59:44] DEBUG[15102]: app_dial.c:1670 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Mar 29 14:59:44] DEBUG[15102]: pbx.c:2393 __ast_pbx_run: Spawn extension (privileged-prod,0123456789,1) exited non-zero on 'SIP/blabla-b7830da8' == Spawn extension (privileged-prod, 0123456789, 1) exited non-zero on 'SIP/blabla-b7830da8' [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 172.16.0.20 [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '"anon" <0blabla>' [Mar 29 14:59:44] DEBUG[2711]: chan_sip.c:15244 sip_devicestate: Checking device state for peer 172.16.0.20 [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '0blabla' [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:287 do_state_change: Changing state for SIP/172.16.0.20 - state 1 (Not in use) [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '0123456789' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is 'privileged-prod' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is 'SIP/blabla-b7830da8' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is 'SIP/172.16.0.20-081d1620' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is 'Dial' [Mar 29 14:59:44] DEBUG[15107]: app_queue.c:546 changethread: Device 'SIP/172.16.0.20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is 'SIP/0123456789@172.16.0.20' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '2007-03-29 14:59:35' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '2007-03-29 14:59:44' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '2007-03-29 14:59:44' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '9' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '0' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '1175173175.20' [Mar 29 14:59:44] DEBUG[15102]: pbx.c:1648 pbx_substitute_variables_helper_full: Function result is '' [Mar 29 14:59:44] DEBUG[15102]: channel.c:1693 ast_hangup: Hanging up channel 'SIP/blabla-b7830da8' [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:3312 sip_hangup: Hangup call SIP/blabla-b7830da8, SIP callid 7cc2f01a-2b9d24d0@10.10.10.23) Scheduling destruction of SIP dialog '7cc2f01a-2b9d24d0@10.10.10.23' in 32000 ms (Method: INVITE) [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:5643 reqprep: Strict routing enforced for session 7cc2f01a-2b9d24d0@10.10.10.23 set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.23, port 5061 Reliably Transmitting (NAT) to 10.10.10.23:5061: BYE sip:blabla@10.10.10.23:5061 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK257a7070;rport From: ;tag=as6fa456c1 To: fax ;tag=31668792b665b3b8o1 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 103 BYE User-Agent: blabla.ch 1.4PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 29 14:59:44] DEBUG[15102]: chan_sip.c:1975 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #1352 [Mar 29 14:59:44] DEBUG[15102]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/blabla-b7830da8 [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - blabla [Mar 29 14:59:44] DEBUG[2711]: chan_sip.c:15244 sip_devicestate: Checking device state for peer blabla [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:287 do_state_change: Changing state for SIP/blabla - state 1 (Not in use) [Mar 29 14:59:44] DEBUG[2711]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - blabla [Mar 29 14:59:44] DEBUG[15108]: app_queue.c:546 changethread: Device 'SIP/blabla' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Mar 29 14:59:44] DEBUG[2711]: chan_sip.c:15244 sip_devicestate: Checking device state for peer blabla <--- SIP read from 10.10.10.23:5061 ---> ACK sip:0123456789@172.16.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-8df10e6d From: fax ;tag=31668792b665b3b8o1 To: ;tag=as6fa456c1 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 103 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="blabla",realm="blabla.ch",nonce="03e45a7d",uri="sip:0123456789@172.16.0.25",algorithm=MD5,response="e6d0efd56e7c77914e4a809a229ace58" Contact: fax User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 <-------------> [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: ACK sip:0123456789@172.16.0.25 SIP/2.0 (37) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-8df10e6d (59) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: fax ;tag=31668792b665b3b8o1 (59) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=as6fa456c1 (46) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 (40) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 103 ACK (13) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Max-Forwards: 70 (16) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Proxy-Authorization: Digest username="blabla",realm="blabla.ch",nonce="03e45a7d",uri="sip:0123456789@172.16.0.25",algorithm=MD5,response="e6d0efd56e7c77914e4a809a229ace58" (171) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Contact: fax (47) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: User-Agent: Linksys/SPA2102-3.3.6 (33) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: Content-Length: 0 (17) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = No match Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag 510742046 Our tag: as01a570eb [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 7cc2f01a-2b9d24d0@10.10.10.23 Their Tag 31668792b665b3b8o1 Our tag: as6fa456c1 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:14633 handle_request: **** Received ACK (6) - Command in SIP ACK <--- SIP read from 10.10.10.23:5061 ---> SIP/2.0 200 OK To: fax ;tag=31668792b665b3b8o1 From: ;tag=as6fa456c1 Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 CSeq: 103 BYE Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK257a7070 Server: Linksys/SPA2102-3.3.6 P-RTP-Stat: PS=401,OS=64160,PR=134,OR=21440,PL=0,JI=1,LA=3137,DU=0,EN=G711a,DE=G711a Content-Length: 0 <-------------> [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: SIP/2.0 200 OK (14) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: To: fax ;tag=31668792b665b3b8o1 (57) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: ;tag=as6fa456c1 (48) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: Call-ID: 7cc2f01a-2b9d24d0@10.10.10.23 (40) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: CSeq: 103 BYE (13) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bK257a7070 (55) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Server: Linksys/SPA2102-3.3.6 (29) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: P-RTP-Stat: PS=401,OS=64160,PR=134,OR=21440,PL=0,JI=1,LA=3137,DU=0,EN=G711a,DE=G711a (84) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: Content-Length: 0 (17) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = No match Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag 510742046 Our tag: as01a570eb [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 7cc2f01a-2b9d24d0@10.10.10.23 Their Tag 31668792b665b3b8o1 Our tag: as6fa456c1 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2079 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1352 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2089 __sip_ack: Stopping retransmission on '7cc2f01a-2b9d24d0@10.10.10.23' of Request 103: Match Not Found Really destroying SIP dialog '7cc2f01a-2b9d24d0@10.10.10.23' Method: ACK anarki*CLI> <--- SIP read from 172.16.0.20:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK3dc90afa;rport=5060 From: "anon" ;tag=as01a570eb To: ;tag=510742046 Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 CSeq: 104 BYE Content-Length: 0 <-------------> [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: SIP/2.0 200 Ok (14) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.25:5060;received=172.16.0.25;branch=z9hG4bK3dc90afa;rport=5060 (86) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: "anon" ;tag=as01a570eb (55) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: ;tag=510742046 (45) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 (52) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 104 BYE (13) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Content-Length: 0 (17) [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:4361 find_call: = Found Their Call ID: 1500a200028c8c5509988c94046c25f2@172.16.0.25 Their Tag 510742046 Our tag: as01a570eb [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2079 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1350 [Mar 29 14:59:44] DEBUG[2735]: chan_sip.c:2089 __sip_ack: Stopping retransmission on '1500a200028c8c5509988c94046c25f2@172.16.0.25' of Request 104: Match Not Found Really destroying SIP dialog '1500a200028c8c5509988c94046c25f2@172.16.0.25' Method: INVITE anarki*CLI> <--- SIP read from 10.10.10.23:5060 ---> NOTIFY sip:172.16.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.23:5060;branch=z9hG4bK-44102cdb From: alx ;tag=82a91f60f0afa1ado0 To: Call-ID: edc685a6-17ad004d@10.10.10.23 CSeq: 728 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 <-------------> [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: NOTIFY sip:172.16.0.25 SIP/2.0 (29) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.23:5060;branch=z9hG4bK-44102cdb (59) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: alx ;tag=82a91f60f0afa1ado0 (59) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: (20) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: edc685a6-17ad004d@10.10.10.23 (40) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 728 NOTIFY (16) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Max-Forwards: 70 (16) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Event: keep-alive (17) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: User-Agent: Linksys/SPA2102-3.3.6 (33) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: Content-Length: 0 (17) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: (0) --- (10 headers 0 lines) --- <--- Transmitting (no NAT) to 10.10.10.23:5060 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 10.10.10.23:5060;branch=z9hG4bK-44102cdb;received=10.10.10.23 From: alx ;tag=82a91f60f0afa1ado0 To: ;tag=as1008f030 Call-ID: edc685a6-17ad004d@10.10.10.23 CSeq: 728 NOTIFY User-Agent: blabla.ch 1.4PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces ontent-Length: 0 <------------> [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:14817 sipsock_read: Invalid SIP message - rejected , no callid, len 328 anarki*CLI> <--- SIP read from 10.10.10.23:5061 ---> NOTIFY sip:172.16.0.25 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-40aeaab0 From: fax ;tag=d2bd39c8aad00a65o1 To: Call-ID: 3258edd2-a37bd611@10.10.10.23 CSeq: 728 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA2102-3.3.6 Content-Length: 0 <-------------> [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 0: NOTIFY sip:172.16.0.25 SIP/2.0 (29) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 1: Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-40aeaab0 (59) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 2: From: fax ;tag=d2bd39c8aad00a65o1 (59) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 3: To: (20) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 4: Call-ID: 3258edd2-a37bd611@10.10.10.23 (40) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 5: CSeq: 728 NOTIFY (16) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 6: Max-Forwards: 70 (16) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 7: Event: keep-alive (17) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 8: User-Agent: Linksys/SPA2102-3.3.6 (33) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 9: Content-Length: 0 (17) [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:4573 parse_request: Header 10: (0) --- (10 headers 0 lines) --- anarki*CLI> <--- Transmitting (no NAT) to 10.10.10.23:5061 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 10.10.10.23:5061;branch=z9hG4bK-40aeaab0;received=10.10.10.23 From: fax ;tag=d2bd39c8aad00a65o1 To: ;tag=as7d7f624a Call-ID: 3258edd2-a37bd611@10.10.10.23 CSeq: 728 NOTIFY User-Agent: blabla.ch 1.4PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Mar 29 14:59:45] DEBUG[2735]: chan_sip.c:14817 sipsock_read: Invalid SIP message - rejected , no callid, len 328