Asterisk Ready. ]1;Asterisk]2;Asterisk Console on 'ClusterTestTwo' (pid 26128)*CLI> sip set debug SIP Debugging enabled *CLI> set debug 4 Core debug was 0 and is now 4 The 'set debug' command is deprecated and will be removed in a future release. Please use 'core set debug' instead. *CLI> <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK28d0db22;rport From: "100" ;tag=as2bc4a039 To: Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Dec 2008 07:58:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 26848 26848 IN IP4 192.168.128.51 s=session c=IN IP4 192.168.128.51 t=0 0 m=audio 12860 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK28d0db22;rport (65) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: (28) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 INVITE (16) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Date: Wed, 10 Dec 2008 07:58:53 GMT (35) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Supported: replaces (19) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 291 (19) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26848 IN IP4 192.168.128.51 (40) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.51 (23) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 12860 RTP/AVP 0 3 8 101 (31) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:2781 do_setnat: Setting NAT on RTP to Off [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4606 sip_alloc: Allocating new SIP dialog for 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 - INVITE (With RTP) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:1720 parse_sip_options: Begin: parsing SIP "Supported: replaces" [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:1728 parse_sip_options: Found SIP option: -replaces- [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:1734 parse_sip_options: Matched SIP option: replaces Sending to 192.168.128.51 : 5060 (NAT) Using INVITE request as basis request - 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 Found peer 'remote_server' [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:2781 do_setnat: Setting NAT on RTP to Off <--- Reliably Transmitting (no NAT) to 192.168.128.51:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK28d0db22;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as304370d1 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="611317db" Content-Length: 0 <------------> [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 407 Proxy Authentication Required (41) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK28d0db22;received=192.168.128.51;rport=5060 (94) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as304370d1 (43) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 102 INVITE (16) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="611317db" (76) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 Scheduling destruction of SIP dialog '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK28d0db22;rport From: "100" ;tag=as2bc4a039 To: ;tag=as304370d1 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK28d0db22;rport (65) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as304370d1 (43) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 ACK (13) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Response 102: Match Found <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK50f7afe0;rport From: "100" ;tag=as2bc4a039 To: Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="remote_server", realm="asterisk", algorithm=MD5, uri="sip:101@192.168.128.52", nonce="611317db", response="4577c8f0154702fc54301fb8f8275fb1" Date: Wed, 10 Dec 2008 07:58:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 26848 26849 IN IP4 192.168.128.51 s=session c=IN IP4 192.168.128.51 t=0 0 m=audio 12860 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK50f7afe0;rport (65) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: (28) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 103 INVITE (16) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Proxy-Authorization: Digest username="remote_server", realm="asterisk", algorithm=MD5, uri="sip:101@192.168.128.52", nonce="611317db", response="4577c8f0154702fc54301fb8f8275fb1" (178) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Date: Wed, 10 Dec 2008 07:58:53 GMT (35) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Supported: replaces (19) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Type: application/sdp (29) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: Content-Length: 291 (19) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 15: (0) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26849 IN IP4 192.168.128.51 (40) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.51 (23) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 12860 RTP/AVP 0 3 8 101 (31) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (15 headers 14 lines) --- [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.128.51 : 5060 (NAT) Using INVITE request as basis request - 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 Found peer 'remote_server' [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:2781 do_setnat: Setting NAT on RTP to Off Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.51:12860 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.51:12860 [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:14452 handle_request_invite: Checking SIP call limits for device remote_server [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:3244 update_call_counter: Updating call counter for incoming call Looking for 101 in bug12013 (domain 192.168.128.52) [Dec 10 02:58:55] DEBUG[26157]: frame.c:1288 ast_codec_choose: Could not find preferred codec - Going for the best codec [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4083 sip_new: *** Our native formats are 0x4 (ulaw) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4084 sip_new: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4085 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Dec 10 02:58:55] DEBUG[26157]: frame.c:1288 ast_codec_choose: Could not find preferred codec - Going for the best codec [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4086 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4109 sip_new: This channel will not be able to handle video. [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:8482 build_route: build_route: Contact hop: list_route: hop: [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:14535 handle_request_invite: SIP/remote_server-0f599f00: New call is still down.... Trying... <--- Transmitting (no NAT) to 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK50f7afe0;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK50f7afe0;received=192.168.128.51;rport=5060 (94) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: (28) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 103 INVITE (16) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:58:55] DEBUG[26157]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/remote_server [Dec 10 02:58:55] DEBUG[26159]: pbx.c:1842 pbx_extension_helper: Launching 'Dial' -- Executing [101@bug12013:1] Dial("SIP/remote_server-0f599f00", "SIP/101") in new stack [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:16549 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4606 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:2781 do_setnat: Setting NAT on RTP to Off [Dec 10 02:58:55] DEBUG[26159]: frame.c:1288 ast_codec_choose: Could not find preferred codec - Going for the best codec [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4083 sip_new: *** Our native formats are 0x80004 (ulaw|h263) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4084 sip_new: *** Joint capabilities are 0x0 (nothing) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4085 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Dec 10 02:58:55] DEBUG[26159]: frame.c:1288 ast_codec_choose: Could not find preferred codec - Going for the best codec [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4086 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4088 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4109 sip_new: This channel will not be able to handle video. [Dec 10 02:58:55] DEBUG[26159]: rtp.c:1633 ast_rtp_make_compatible: Seeded SDP of 'SIP/101-0f59faf0' with that of 'SIP/remote_server-0f599f00' [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable fromRemote. [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Dec 10 02:58:55] DEBUG[26159]: channel.c:3357 ast_channel_inherit_variables: Not copying variable SIPURI. [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:3044 sip_call: Outgoing Call for 101 [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:3244 update_call_counter: Updating call counter for outgoing call [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:3059 sip_call: Our T38 capability (0), joint T38 capability (0) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:6604 add_sdp: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:6622 add_sdp: This call needs video offers, but there's no video support enabled! Audio is at 192.168.128.52 port 11228 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (71) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK22eaaf62;rport (65) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: (62) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 INVITE (16) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Date: Wed, 10 Dec 2008 07:58:55 GMT (35) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: Supported: replaces (19) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 291 (19) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: o=root 26128 26128 IN IP4 192.168.128.52 (40) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.52 (23) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: m=audio 11228 RTP/AVP 0 3 8 101 (31) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.128.129:12538: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK22eaaf62;rport From: "100" ;tag=as7e836c3f To: Contact: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Dec 2008 07:58:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 26128 26128 IN IP4 192.168.128.52 s=session c=IN IP4 192.168.128.52 t=0 0 m=audio 11228 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (71) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK22eaaf62;rport (65) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: (62) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 INVITE (16) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Date: Wed, 10 Dec 2008 07:58:55 GMT (35) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: Supported: replaces (19) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 291 (19) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: o=root 26128 26128 IN IP4 192.168.128.52 (40) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.52 (23) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: m=audio 11228 RTP/AVP 0 3 8 101 (31) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 -- Called 101 [Dec 10 02:58:55] DEBUG[26132]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - remote_server [Dec 10 02:58:55] DEBUG[26132]: chan_sip.c:16473 sip_devicestate: Checking device state for peer remote_server [Dec 10 02:58:55] DEBUG[26132]: devicestate.c:287 do_state_change: Changing state for SIP/remote_server - state 1 (Not in use) [Dec 10 02:58:55] DEBUG[26136]: app_queue.c:671 handle_statechange: Device 'SIP/remote_server' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.128.129:12538 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK22eaaf62;rport=5060 To: From: "100" ;tag=as7e836c3f Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 102 INVITE Content-Length: 0 <-------------> [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK22eaaf62;rport=5060 (70) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: To: (62) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 102 INVITE (16) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: Content-Length: 0 (17) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:2249 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #9 - INVITE (got response) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:2257 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2245bc2a10c04be731e248df237d6a62@192.168.128.52' Request 102: Found [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:12273 handle_response_invite: SIP response 100 to standard invite <--- SIP read from 192.168.128.129:12538 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK22eaaf62;rport=5060 Contact: To: ;tag=0a5d8a26 From: "100";tag=as7e836c3f Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 102 INVITE User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 0 <-------------> [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK22eaaf62;rport=5060 (70) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: Contact: (67) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: From: "100";tag=as7e836c3f (50) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 INVITE (16) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: X-Lite release 1014k stamp 47051 (44) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Content-Length: 0 (17) [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:2257 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2245bc2a10c04be731e248df237d6a62@192.168.128.52' Request 102: Found [Dec 10 02:58:55] DEBUG[26157]: chan_sip.c:12273 handle_response_invite: SIP response 180 to standard invite [Dec 10 02:58:55] DEBUG[26157]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/101 -- SIP/101-0f59faf0 is ringing [Dec 10 02:58:55] DEBUG[26159]: rtp.c:1562 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/remote_server-0f599f00' with that of 'SIP/101-0f59faf0' <--- Transmitting (no NAT) to 192.168.128.51:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK50f7afe0;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK50f7afe0;received=192.168.128.51;rport=5060 (94) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: CSeq: 103 INVITE (16) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:58:55] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:58:55] DEBUG[26132]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 101 [Dec 10 02:58:55] DEBUG[26132]: chan_sip.c:16473 sip_devicestate: Checking device state for peer 101 [Dec 10 02:58:55] DEBUG[26132]: devicestate.c:287 do_state_change: Changing state for SIP/101 - state 1 (Not in use) [Dec 10 02:58:55] DEBUG[26136]: app_queue.c:671 handle_statechange: Device 'SIP/101' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 10 02:58:58] DEBUG[26159]: rtp.c:879 ast_rtcp_read: Got RTCP report of 132 bytes <--- SIP read from 192.168.128.129:12538 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK22eaaf62;rport=5060 Contact: To: ;tag=0a5d8a26 From: "100";tag=as7e836c3f Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 193 v=0 o=- 0 2 IN IP4 192.168.128.129 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 0 3 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK22eaaf62;rport=5060 (70) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: Contact: (67) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: From: "100";tag=as7e836c3f (50) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 INVITE (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Content-Type: application/sdp (29) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: User-Agent: X-Lite release 1014k stamp 47051 (44) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 193 (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=- 0 2 IN IP4 192.168.128.129 (30) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=CounterPath X-Lite 3.0 (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 0 3 8 101 (31) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-15 (15) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (11 headers 9 lines) --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 102 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2245bc2a10c04be731e248df237d6a62@192.168.128.52' of Request 102: Match Found [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:12273 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:53806 Got unsupported a:fmtp in SDP offer Found audio description format telephone-event for ID 101 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/101-0f59faf0 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:53806 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:3244 update_call_counter: Updating call counter for outgoing call [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:8482 build_route: build_route: Contact hop: list_route: hop: [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2245bc2a10c04be731e248df237d6a62@192.168.128.52 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.129, port 12538 Transmitting (no NAT) to 192.168.128.129:12538: ACK sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK3c677c6f;rport From: "100" ;tag=as7e836c3f To: ;tag=0a5d8a26 Contact: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (68) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK3c677c6f;rport (65) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 ACK (13) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:58:58] DEBUG[26159]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/101 -- SIP/101-0f59faf0 answered SIP/remote_server-0f599f00 [Dec 10 02:58:58] DEBUG[26159]: rtp.c:1562 ast_rtp_early_bridge: Setting early bridge SDP of 'SIP/remote_server-0f599f00' with that of 'SIP/101-0f59faf0' [Dec 10 02:58:58] DEBUG[26159]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/remote_server [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:3718 sip_answer: SIP answering channel: SIP/remote_server-0f599f00 [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6840 transmit_response_with_sdp: Setting framing from config on incoming call [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) <--- Reliably Transmitting (no NAT) to 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK50f7afe0;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 291 v=0 o=root 26128 26128 IN IP4 192.168.128.52 s=session c=IN IP4 192.168.128.52 t=0 0 m=audio 16254 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK50f7afe0;received=192.168.128.51;rport=5060 (94) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: CSeq: 103 INVITE (16) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 291 (19) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: o=root 26128 26128 IN IP4 192.168.128.52 (40) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.52 (23) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: m=audio 16254 RTP/AVP 3 0 8 101 (31) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 -- Native bridging SIP/remote_server-0f599f00 and SIP/101-0f59faf0 [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:18258 sip_set_rtp_peer: Deferring reinvite on SIP '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' - It's audio will be redirected to IP 192.168.128.129 [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:18253 sip_set_rtp_peer: Sending reinvite on SIP '2245bc2a10c04be731e248df237d6a62@192.168.128.52' - It's audio soon redirected to IP 192.168.128.51 [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2245bc2a10c04be731e248df237d6a62@192.168.128.52 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.129, port 12538 [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x4 (ulaw) Audio is at 192.168.128.52 port 11228 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (presumably reinvite) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (71) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK317ae5d8;rport (65) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: CSeq: 103 INVITE (16) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 291 (19) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: o=root 26128 26129 IN IP4 192.168.128.51 (40) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.51 (23) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: m=audio 12860 RTP/AVP 0 3 8 101 (31) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.128.129:12538: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK317ae5d8;rport From: "100" ;tag=as7e836c3f To: ;tag=0a5d8a26 Contact: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 103 INVITE User-Agent: Asterisk PBX 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: 291 v=0 o=root 26128 26129 IN IP4 192.168.128.51 s=session c=IN IP4 192.168.128.51 t=0 0 m=audio 12860 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (71) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK317ae5d8;rport (65) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: CSeq: 103 INVITE (16) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 291 (19) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: o=root 26128 26129 IN IP4 192.168.128.51 (40) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.51 (23) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: m=audio 12860 RTP/AVP 0 3 8 101 (31) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:58:58] DEBUG[26132]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 101 [Dec 10 02:58:58] DEBUG[26132]: chan_sip.c:16473 sip_devicestate: Checking device state for peer 101 [Dec 10 02:58:58] DEBUG[26132]: devicestate.c:287 do_state_change: Changing state for SIP/101 - state 1 (Not in use) [Dec 10 02:58:58] DEBUG[26132]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - remote_server [Dec 10 02:58:58] DEBUG[26132]: chan_sip.c:16473 sip_devicestate: Checking device state for peer remote_server [Dec 10 02:58:58] DEBUG[26132]: devicestate.c:287 do_state_change: Changing state for SIP/remote_server - state 1 (Not in use) [Dec 10 02:58:58] DEBUG[26136]: app_queue.c:671 handle_statechange: Device 'SIP/101' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 10 02:58:58] DEBUG[26136]: app_queue.c:671 handle_statechange: Device 'SIP/remote_server' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1a216479;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1a216479;rport (65) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 103 ACK (13) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #11 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Response 103: Match Found [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:12228 check_pendings: Sending pending reinvite on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;rport (65) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 INVITE (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26129 IN IP4 192.168.128.129 (41) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26129 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;rport (65) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 INVITE (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26129 IN IP4 192.168.128.129 (41) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26850 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport (65) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 INVITE (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26850 IN IP4 192.168.128.129 (41) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE <--- Reliably Transmitting (no NAT) to 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;received=192.168.128.51;rport=5060 (94) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 104 INVITE (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:14242 handle_request_invite: Got INVITE on call where we already have pending INVITE, deferring that - 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 (94) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 102 INVITE (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 102 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #13 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 102: Match Found [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (no NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;rport (65) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 ACK (13) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 4 secs before retry <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport (65) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 ACK (13) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:58:58] DEBUG[26159]: rtp.c:2777 ast_rtp_write: Ooh, format changed from unknown to ulaw [Dec 10 02:58:58] DEBUG[26159]: rtp.c:2794 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 <--- SIP read from 192.168.128.129:12538 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK317ae5d8;rport=5060 To: ;tag=0a5d8a26 From: "100" ;tag=as7e836c3f Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 103 INVITE Content-Length: 0 <-------------> [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK317ae5d8;rport=5060 (70) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: To: ;tag=0a5d8a26 (75) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 103 INVITE (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: Content-Length: 0 (17) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2249 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #12 - INVITE (got response) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2257 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2245bc2a10c04be731e248df237d6a62@192.168.128.52' Request 103: Found [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 100 to RE-invite on outgoing call 2245bc2a10c04be731e248df237d6a62@192.168.128.52 [Dec 10 02:58:58] DEBUG[26159]: chan_sip.c:4380 sip_rtp_read: Oooh, format changed to 2 [Dec 10 02:58:58] DEBUG[26159]: channel.c:2849 set_format: Set channel SIP/remote_server-0f599f00 to read format ulaw [Dec 10 02:58:58] DEBUG[26159]: channel.c:2849 set_format: Set channel SIP/remote_server-0f599f00 to write format ulaw [Dec 10 02:58:58] DEBUG[26159]: rtp.c:2777 ast_rtp_write: Ooh, format changed from unknown to ulaw [Dec 10 02:58:58] DEBUG[26159]: rtp.c:2794 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Dec 10 02:58:58] DEBUG[26159]: rtp.c:2777 ast_rtp_write: Ooh, format changed from ulaw to gsm [Dec 10 02:58:58] DEBUG[26159]: rtp.c:2794 ast_rtp_write: Created smoother: format: 2 ms: 20 len: 33 <--- SIP read from 192.168.128.129:12538 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK317ae5d8;rport=5060 Contact: To: ;tag=0a5d8a26 From: "100";tag=as7e836c3f Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 193 v=0 o=- 0 2 IN IP4 192.168.128.129 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 0 3 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK317ae5d8;rport=5060 (70) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: Contact: (67) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: From: "100";tag=as7e836c3f (50) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 103 INVITE (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Content-Type: application/sdp (29) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: User-Agent: X-Lite release 1014k stamp 47051 (44) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 193 (19) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=- 0 2 IN IP4 192.168.128.129 (30) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=CounterPath X-Lite 3.0 (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 0 3 8 101 (31) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-15 (15) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (11 headers 9 lines) --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 103 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2245bc2a10c04be731e248df237d6a62@192.168.128.52' of Request 103: Match Found [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 200 to RE-invite on outgoing call 2245bc2a10c04be731e248df237d6a62@192.168.128.52 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:53806 Got unsupported a:fmtp in SDP offer Found audio description format telephone-event for ID 101 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/101-0f59faf0 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:53806 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:3244 update_call_counter: Updating call counter for outgoing call [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:12401 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:12406 handle_response_invite: T38 state changed to 0 on channel SIP [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:12409 handle_response_invite: T38 state changed to 0 on channel SIP/101-0f59faf0 [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2245bc2a10c04be731e248df237d6a62@192.168.128.52 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.129, port 12538 Transmitting (no NAT) to 192.168.128.129:12538: ACK sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK0e308699;rport From: "100" ;tag=as7e836c3f To: ;tag=0a5d8a26 Contact: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (68) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK0e308699;rport (65) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 103 ACK (13) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:58] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #14 (1) SIP/2.0 - 1 [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #14)) Retransmitting #1 (no NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport (65) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 ACK (13) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 (94) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 102 INVITE (16) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 102: Match Not Found [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (no NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;rport (65) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 ACK (13) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:58:59] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 5 secs before retry <--- SIP read from 192.168.128.129:12538 ---> <-------------> [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: (0) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: (0) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #14 (2) SIP/2.0 - 1 [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #14)) Retransmitting #2 (no NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport (65) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 ACK (13) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 102 INVITE (16) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 102: Match Not Found [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (no NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;rport (65) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 102 ACK (13) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:00] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 3 secs before retry [Dec 10 02:59:01] DEBUG[26159]: rtp.c:879 ast_rtcp_read: Got RTCP report of 100 bytes [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #14 (3) SIP/2.0 - 1 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #14)) Retransmitting #3 (no NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;rport (65) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 ACK (13) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12248 sip_reinvite_retry: Sending pending reinvite retry on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK69d5129a;rport (65) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 103 INVITE (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26130 IN IP4 192.168.128.129 (41) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK69d5129a;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 103 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26130 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK69d5129a;rport (65) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 103 INVITE (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26130 IN IP4 192.168.128.129 (41) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 102 INVITE (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK69d5129a;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK69d5129a;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 103 INVITE (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:2249 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #20 - INVITE (got response) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:2257 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' Request 103: Found [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 100 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK69d5129a;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26848 26851 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK69d5129a;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 103 INVITE (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26851 IN IP4 192.168.128.129 (41) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (12 headers 14 lines) --- [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 103 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 103: Match Found [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 200 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:10000 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/remote_server-0f599f00 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:10000 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:3244 update_call_counter: Updating call counter for incoming call [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12401 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12406 handle_response_invite: T38 state changed to 0 on channel SIP [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12409 handle_response_invite: T38 state changed to 0 on channel SIP/remote_server-0f599f00 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (no NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK7f04678b;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK7f04678b;rport (65) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 103 ACK (13) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:02] DEBUG[26159]: rtp.c:2944 bridge_native_loop: Oooh, 'SIP/remote_server-0f599f00' changed end address to 192.168.128.129:10000 (format 14) [Dec 10 02:59:02] DEBUG[26159]: rtp.c:2946 bridge_native_loop: Oooh, 'SIP/remote_server-0f599f00' was 192.168.128.51:12860/(format 14) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:18253 sip_set_rtp_peer: Sending reinvite on SIP '2245bc2a10c04be731e248df237d6a62@192.168.128.52' - It's audio soon redirected to IP 192.168.128.129 [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2245bc2a10c04be731e248df237d6a62@192.168.128.52 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.129, port 12538 [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x4 (ulaw) Audio is at 192.168.128.52 port 11228 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (presumably reinvite) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (71) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK4d758975;rport (65) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 INVITE (16) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: o=root 26128 26130 IN IP4 192.168.128.129 (41) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.128.129:12538: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK4d758975;rport From: "100" ;tag=as7e836c3f To: ;tag=0a5d8a26 Contact: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 104 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26130 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (71) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK4d758975;rport (65) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 INVITE (16) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: o=root 26128 26130 IN IP4 192.168.128.129 (41) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:02] DEBUG[26159]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 <--- SIP read from 192.168.128.129:12538 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK4d758975;rport=5060 Contact: To: ;tag=0a5d8a26 From: "100";tag=as7e836c3f Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 193 v=0 o=- 0 2 IN IP4 192.168.128.129 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 0 3 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK4d758975;rport=5060 (70) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: Contact: (67) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: From: "100";tag=as7e836c3f (50) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 INVITE (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Content-Type: application/sdp (29) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: User-Agent: X-Lite release 1014k stamp 47051 (44) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 193 (19) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=- 0 2 IN IP4 192.168.128.129 (30) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=CounterPath X-Lite 3.0 (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-15 (15) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (11 headers 9 lines) --- [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 104 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #21 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2245bc2a10c04be731e248df237d6a62@192.168.128.52' of Request 104: Match Found [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 200 to RE-invite on outgoing call 2245bc2a10c04be731e248df237d6a62@192.168.128.52 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:53806 Got unsupported a:fmtp in SDP offer Found audio description format telephone-event for ID 101 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/101-0f59faf0 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:53806 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:3244 update_call_counter: Updating call counter for outgoing call [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12401 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12406 handle_response_invite: T38 state changed to 0 on channel SIP [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:12409 handle_response_invite: T38 state changed to 0 on channel SIP/101-0f59faf0 [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2245bc2a10c04be731e248df237d6a62@192.168.128.52 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.129, port 12538 Transmitting (no NAT) to 192.168.128.129:12538: ACK sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK33137a16;rport From: "100" ;tag=as7e836c3f To: ;tag=0a5d8a26 Contact: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (68) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK33137a16;rport (65) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 ACK (13) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:02] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26852 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport (65) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26852 IN IP4 192.168.128.129 (41) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.128.51 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:10000 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/remote_server-0f599f00 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:10000 [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:14511 handle_request_invite: Got a SIP re-invite for call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:14609 handle_request_invite: SIP/remote_server-0f599f00: This call is UP.... <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 105 INVITE (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6840 transmit_response_with_sdp: Setting framing from config on incoming call [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26131 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 105 INVITE (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26131 IN IP4 192.168.128.129 (41) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:12248 sip_reinvite_retry: Sending pending reinvite retry on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;rport (65) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 INVITE (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26132 IN IP4 192.168.128.129 (41) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26132 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;rport (65) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 INVITE (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26132 IN IP4 192.168.128.129 (41) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26853 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;rport (65) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 106 INVITE (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26853 IN IP4 192.168.128.129 (41) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 106 INVITE (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:14242 handle_request_invite: Got INVITE on call where we already have pending INVITE, deferring that - 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 104 INVITE (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 104 [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #23 [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 104: Match Found [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;rport (65) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 104 ACK (13) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 4 secs before retry <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;rport (65) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 106 ACK (13) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:03] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26852 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport (65) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26852 IN IP4 192.168.128.129 (41) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:15803 handle_request: Ignoring too old SIP packet packet 105 (expecting >= 106) <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <------------> [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 503 Server error (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 105 INVITE (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: (0) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:16030 sipsock_read: SIP message could not be handled, bad request: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:12248 sip_reinvite_retry: Sending pending reinvite retry on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK063c45c4;rport (65) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26133 IN IP4 192.168.128.129 (41) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK063c45c4;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26133 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK063c45c4;rport (65) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26133 IN IP4 192.168.128.129 (41) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #22 (1) SIP/2.0 - 1 [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #22)) Retransmitting #1 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26131 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK063c45c4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK063c45c4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 105 INVITE (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:2249 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #26 - INVITE (got response) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:2257 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' Request 105: Found [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 100 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #24 (1) SIP/2.0 - 1 [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #24)) Retransmitting #1 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK063c45c4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26848 26854 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK063c45c4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 105 INVITE (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26854 IN IP4 192.168.128.129 (41) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (12 headers 14 lines) --- [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 105 [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 105: Match Found [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 200 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:10000 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/remote_server-0f599f00 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:10000 [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:3244 update_call_counter: Updating call counter for incoming call [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:12401 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:12406 handle_response_invite: T38 state changed to 0 on channel SIP [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:12409 handle_response_invite: T38 state changed to 0 on channel SIP/remote_server-0f599f00 [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK7f77c9ff;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK7f77c9ff;rport (65) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 ACK (13) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 104 INVITE (16) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:04] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6ac7a336;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26855 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6ac7a336;rport (65) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 107 INVITE (16) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26855 IN IP4 192.168.128.129 (41) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.128.51 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:10000 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/remote_server-0f599f00 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:10000 [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:14511 handle_request_invite: Got a SIP re-invite for call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:14609 handle_request_invite: SIP/remote_server-0f599f00: This call is UP.... <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6ac7a336;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6ac7a336;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 107 INVITE (16) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:6840 transmit_response_with_sdp: Setting framing from config on incoming call [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6ac7a336;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26134 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6ac7a336;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 107 INVITE (16) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26134 IN IP4 192.168.128.129 (41) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #22 (2) SIP/2.0 - 1 [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #22)) Retransmitting #2 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26131 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26852 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport (65) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26852 IN IP4 192.168.128.129 (41) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:15803 handle_request: Ignoring too old SIP packet packet 105 (expecting >= 107) <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 503 Server error (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 105 INVITE (16) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: (0) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:16030 sipsock_read: SIP message could not be handled, bad request: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK213877ab;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK213877ab;rport (65) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 107 ACK (13) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #27 [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Response 107: Match Found [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #24 (2) SIP/2.0 - 1 [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #24)) Retransmitting #2 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 104 INVITE (16) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:05] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 102 INVITE (16) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #14 (4) SIP/2.0 - 1 [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #14)) Retransmitting #4 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK553218d4;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26856 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK553218d4;rport (65) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 108 INVITE (16) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26856 IN IP4 192.168.128.129 (41) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.128.51 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:10000 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/remote_server-0f599f00 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:10000 [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:14511 handle_request_invite: Got a SIP re-invite for call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:14609 handle_request_invite: SIP/remote_server-0f599f00: This call is UP.... <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK553218d4;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK553218d4;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:6840 transmit_response_with_sdp: Setting framing from config on incoming call [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK553218d4;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26135 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK553218d4;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26135 IN IP4 192.168.128.129 (41) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK46e9e7ca;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK46e9e7ca;rport (65) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 108 ACK (13) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #28 [Dec 10 02:59:06] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Response 108: Match Found <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26852 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport (65) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26852 IN IP4 192.168.128.129 (41) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:15803 handle_request: Ignoring too old SIP packet packet 105 (expecting >= 108) <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 503 Server error (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 105 INVITE (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: (0) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:16030 sipsock_read: SIP message could not be handled, bad request: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #22 (3) SIP/2.0 - 1 [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #22)) Retransmitting #3 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26131 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #24 (3) SIP/2.0 - 1 [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #24)) Retransmitting #3 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:12248 sip_reinvite_retry: Sending pending reinvite retry on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;rport (65) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 106 INVITE (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26136 IN IP4 192.168.128.129 (41) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26136 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;rport (65) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 106 INVITE (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26136 IN IP4 192.168.128.129 (41) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 104 INVITE (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26857 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport (65) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 109 INVITE (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26857 IN IP4 192.168.128.129 (41) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 109 INVITE (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:14242 handle_request_invite: Got INVITE on call where we already have pending INVITE, deferring that - 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 106 INVITE (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 106 [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #29 [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 106: Match Found [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;rport (65) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 106 ACK (13) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 5 secs before retry <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport (65) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 109 ACK (13) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:07] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #30 (1) SIP/2.0 - 1 [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #30)) Retransmitting #1 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport (65) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 109 ACK (13) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 106 INVITE (16) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 106: Match Not Found [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;rport (65) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 106 ACK (13) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:08] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 3 secs before retry <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 106 INVITE (16) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 106: Match Not Found [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;rport (65) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 106 ACK (13) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 3 secs before retry [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #30 (2) SIP/2.0 - 1 [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #30)) Retransmitting #2 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport (65) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 109 ACK (13) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:09] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 102 INVITE (16) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #14 (5) SIP/2.0 - 1 [Dec 10 02:59:10] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #14)) Retransmitting #5 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26852 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport (65) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26852 IN IP4 192.168.128.129 (41) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:15803 handle_request: Ignoring too old SIP packet packet 105 (expecting >= 109) <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <------------> [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 503 Server error (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 105 INVITE (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: (0) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:16030 sipsock_read: SIP message could not be handled, bad request: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #22 (4) SIP/2.0 - 1 [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #22)) Retransmitting #4 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26131 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #24 (4) SIP/2.0 - 1 [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #24)) Retransmitting #4 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 104 INVITE (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #30 (3) SIP/2.0 - 1 [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #30)) Retransmitting #3 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;rport (65) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 109 ACK (13) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:12248 sip_reinvite_retry: Sending pending reinvite retry on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;rport (65) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 107 INVITE (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26137 IN IP4 192.168.128.129 (41) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26137 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;rport (65) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 107 INVITE (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26137 IN IP4 192.168.128.129 (41) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 106 INVITE (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26858 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;rport (65) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 110 INVITE (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26858 IN IP4 192.168.128.129 (41) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <------------> [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 110 INVITE (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:14242 handle_request_invite: Got INVITE on call where we already have pending INVITE, deferring that - 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 107 INVITE (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 107 [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #34 [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 107: Match Found [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;rport (65) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 107 ACK (13) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 4 secs before retry <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;rport (65) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 110 ACK (13) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:11] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:12248 sip_reinvite_retry: Sending pending reinvite retry on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;rport (65) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 108 INVITE (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26138 IN IP4 192.168.128.129 (41) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26138 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;rport (65) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 108 INVITE (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26138 IN IP4 192.168.128.129 (41) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:12248 sip_reinvite_retry: Sending pending reinvite retry on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;rport (65) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 109 INVITE (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26139 IN IP4 192.168.128.129 (41) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26139 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;rport (65) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 109 INVITE (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26139 IN IP4 192.168.128.129 (41) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #35 (1) SIP/2.0 - 1 [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #35)) Retransmitting #1 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) --- (11 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26848 26859 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26859 IN IP4 192.168.128.129 (41) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (12 headers 14 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 107 INVITE (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 109 INVITE (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 109 [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #38 [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 109: Match Found [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;rport (65) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 109 ACK (13) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:12] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 5 secs before retry [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (1) INVITE - 5 [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #37)) Retransmitting #1 (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26138 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 503 Server error (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: (0) --- (13 headers 0 lines) --- [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #35 (2) SIP/2.0 - 1 [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #35)) Retransmitting #2 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2b791a4e;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 111 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26860 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2b791a4e;rport (65) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 111 INVITE (16) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26860 IN IP4 192.168.128.129 (41) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.128.51 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:10000 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/remote_server-0f599f00 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:10000 [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:14511 handle_request_invite: Got a SIP re-invite for call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:14609 handle_request_invite: SIP/remote_server-0f599f00: This call is UP.... <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2b791a4e;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 111 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2b791a4e;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 111 INVITE (16) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:6840 transmit_response_with_sdp: Setting framing from config on incoming call [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2b791a4e;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 111 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26140 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2b791a4e;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 111 INVITE (16) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26140 IN IP4 192.168.128.129 (41) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26848 26859 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26859 IN IP4 192.168.128.129 (41) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (12 headers 14 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 107 INVITE (16) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 109 INVITE (16) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 109: Match Not Found SIP Response message for INCOMING dialog INVITE arrived -- Incoming call: Got SIP response 491 "Request Pending" back from 192.168.128.51 <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK665c6445;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 111 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK665c6445;rport (65) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 111 ACK (13) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #40 [Dec 10 02:59:13] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Response 111: Match Found <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK49b707c2;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 102 INVITE (16) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #14 (6) SIP/2.0 - 1 [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #14)) Retransmitting #6 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK6b65e634;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (2) INVITE - 5 [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #37)) Retransmitting #2 (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26138 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 503 Server error (24) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: (0) --- (13 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26848 26859 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26859 IN IP4 192.168.128.129 (41) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (12 headers 14 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 109 INVITE (16) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:14] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 109: Match Not Found SIP Response message for INCOMING dialog INVITE arrived -- Incoming call: Got SIP response 491 "Request Pending" back from 192.168.128.51 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #22 (5) SIP/2.0 - 1 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #22)) Retransmitting #5 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26131 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #24 (5) SIP/2.0 - 1 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #24)) Retransmitting #5 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 104 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #30 (4) SIP/2.0 - 1 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #30)) Retransmitting #4 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 106 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 112 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26861 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;rport (65) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 112 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26861 IN IP4 192.168.128.129 (41) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.128.51 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:10000 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/remote_server-0f599f00 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:10000 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:14511 handle_request_invite: Got a SIP re-invite for call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:14609 handle_request_invite: SIP/remote_server-0f599f00: This call is UP.... <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 112 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 112 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6840 transmit_response_with_sdp: Setting framing from config on incoming call [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 112 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26141 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 112 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26141 IN IP4 192.168.128.129 (41) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #35 (3) SIP/2.0 - 1 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #35)) Retransmitting #3 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 --- [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:12248 sip_reinvite_retry: Sending pending reinvite retry on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;rport (65) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 110 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26142 IN IP4 192.168.128.129 (41) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26142 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;rport (65) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 110 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26142 IN IP4 192.168.128.129 (41) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 107 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 113 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26862 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;rport (65) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 113 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26862 IN IP4 192.168.128.129 (41) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 113 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 113 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:14242 handle_request_invite: Got INVITE on call where we already have pending INVITE, deferring that - 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 110 INVITE (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 110 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #42 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 110: Match Found [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 491 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;rport (65) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 110 ACK (13) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:12572 handle_response_invite: Reinvite race. Waiting 4 secs before retry <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 113 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;rport (65) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 113 ACK (13) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:15] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (3) INVITE - 5 [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #37)) Retransmitting #3 (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26138 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2f35db39;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 114 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26863 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2f35db39;rport (65) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 114 INVITE (16) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26863 IN IP4 192.168.128.129 (41) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE Sending to 192.168.128.51 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:10000 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/remote_server-0f599f00 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:10000 [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:14511 handle_request_invite: Got a SIP re-invite for call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:14609 handle_request_invite: SIP/remote_server-0f599f00: This call is UP.... <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2f35db39;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 114 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2f35db39;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 114 INVITE (16) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:6840 transmit_response_with_sdp: Setting framing from config on incoming call [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) <--- Reliably Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2f35db39;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 114 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26143 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK2f35db39;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 114 INVITE (16) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26143 IN IP4 192.168.128.129 (41) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (1) SIP/2.0 - 1 [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #41)) Retransmitting #1 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 112 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26141 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 503 Server error (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: (0) --- (13 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26848 26859 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26859 IN IP4 192.168.128.129 (41) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (12 headers 14 lines) --- <--- SIP read from 192.168.128.51:5060 ---> ACK sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1aa36df8;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 114 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.52 SIP/2.0 (34) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1aa36df8;rport (65) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 114 ACK (13) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) --- (10 headers 0 lines) --- [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received ACK (6) - Command in SIP ACK [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #45 [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Response 114: Match Found [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #43 (1) SIP/2.0 - 1 [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #43)) Retransmitting #1 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 113 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 109 INVITE (16) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 110 INVITE (16) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:16] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 110: Match Not Found SIP Response message for INCOMING dialog INVITE arrived -- Incoming call: Got SIP response 491 "Request Pending" back from 192.168.128.51 [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (2) SIP/2.0 - 1 [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #41)) Retransmitting #2 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 112 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26141 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:12248 sip_reinvite_retry: Sending pending reinvite retry on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.128.52 port 16254 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (presumably reinvite) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK3201e6d4;rport (65) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 111 INVITE (16) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26144 IN IP4 192.168.128.129 (41) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK3201e6d4;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 111 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26144 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:100@192.168.128.51 SIP/2.0 (37) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK3201e6d4;rport (65) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 111 INVITE (16) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26128 26144 IN IP4 192.168.128.129 (41) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 3 0 8 101 (31) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #43 (2) SIP/2.0 - 1 [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #43)) Retransmitting #2 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 113 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK3201e6d4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 111 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 100 Trying (18) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK3201e6d4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 111 INVITE (16) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) --- (11 headers 0 lines) --- [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:2249 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #46 - INVITE (got response) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:2257 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' Request 111: Found [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 100 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK3201e6d4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 111 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26848 26864 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK3201e6d4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 111 INVITE (16) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26864 IN IP4 192.168.128.129 (41) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (12 headers 14 lines) --- [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 111 [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2691b8d3595a72147f9cdde66d8ba335@192.168.128.51' of Request 111: Match Found [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:12271 handle_response_invite: SIP response 200 to RE-invite on outgoing call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:10000 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel SIP/remote_server-0f599f00 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:10000 [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:5642 process_sdp: We have an owner, now see if we need to change this call [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:3244 update_call_counter: Updating call counter for incoming call [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:12401 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:12406 handle_response_invite: T38 state changed to 0 on channel SIP [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:12409 handle_response_invite: T38 state changed to 0 on channel SIP/remote_server-0f599f00 [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.51, port 5060 Transmitting (NAT) to 192.168.128.51:5060: ACK sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK64311eff;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 111 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:100@192.168.128.51 SIP/2.0 (34) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK64311eff;rport (65) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 111 ACK (13) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 110 INVITE (16) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:17] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:18] WARNING[26157]: chan_sip.c:1966 retrans_pkt: Maximum retries exceeded on transmission 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 for seqno 104 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:1673 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:18] WARNING[26157]: chan_sip.c:1988 retrans_pkt: Hanging up call 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 - no reply to our critical packet (see doc/sip-retransmit.txt). [Dec 10 02:59:18] DEBUG[26159]: rtp.c:2980 bridge_native_loop: Oooh, got a hangup [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:18253 sip_set_rtp_peer: Sending reinvite on SIP '2245bc2a10c04be731e248df237d6a62@192.168.128.52' - It's audio soon redirected to IP 192.168.128.52 [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2245bc2a10c04be731e248df237d6a62@192.168.128.52 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.129, port 12538 [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:6604 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:6605 add_sdp: ** Our prefcodec: 0x4 (ulaw) Audio is at 192.168.128.52 port 11228 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:6736 add_sdp: -- Done with adding codecs to SDP [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:6781 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:1661 initialize_initreq: Initializing already initialized SIP dialog 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (presumably reinvite) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (71) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK71b4d764;rport (65) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 291 (19) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: o=root 26128 26131 IN IP4 192.168.128.52 (40) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.52 (23) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: m=audio 11228 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 192.168.128.129:12538: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK71b4d764;rport From: "100" ;tag=as7e836c3f To: ;tag=0a5d8a26 Contact: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 105 INVITE User-Agent: Asterisk PBX 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: 291 v=0 o=root 26128 26131 IN IP4 192.168.128.52 s=session c=IN IP4 192.168.128.52 t=0 0 m=audio 11228 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (71) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK71b4d764;rport (65) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 291 (19) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: o=root 26128 26131 IN IP4 192.168.128.52 (40) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.52 (23) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: m=audio 11228 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Dec 10 02:59:18] DEBUG[26159]: channel.c:4111 ast_channel_bridge: Returning from native bridge, channels: SIP/remote_server-0f599f00, SIP/101-0f59faf0 [Dec 10 02:59:18] DEBUG[26159]: channel.c:1505 ast_hangup: Hanging up channel 'SIP/101-0f59faf0' [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:3557 sip_hangup: Hangup call SIP/101-0f59faf0, SIP callid 2245bc2a10c04be731e248df237d6a62@192.168.128.52) Scheduling destruction of SIP dialog '2245bc2a10c04be731e248df237d6a62@192.168.128.52' in 32000 ms (Method: INVITE) [Dec 10 02:59:18] DEBUG[26159]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/101 [Dec 10 02:59:18] DEBUG[26159]: rtp.c:1511 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Dec 10 02:59:18] DEBUG[26159]: app_dial.c:1862 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Dec 10 02:59:18] DEBUG[26159]: pbx.c:2425 __ast_pbx_run: Spawn extension (bug12013,101,1) exited non-zero on 'SIP/remote_server-0f599f00' == Spawn extension (bug12013, 101, 1) exited non-zero on 'SIP/remote_server-0f599f00' [Dec 10 02:59:18] DEBUG[26159]: channel.c:1412 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/remote_server-0f599f00' [Dec 10 02:59:18] DEBUG[26159]: channel.c:1505 ast_hangup: Hanging up channel 'SIP/remote_server-0f599f00' [Dec 10 02:59:18] DEBUG[26159]: chan_sip.c:3557 sip_hangup: Hangup call SIP/remote_server-0f599f00, SIP callid 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51) [Dec 10 02:59:18] DEBUG[26159]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/remote_server [Dec 10 02:59:18] DEBUG[26132]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 101 [Dec 10 02:59:18] DEBUG[26132]: chan_sip.c:16473 sip_devicestate: Checking device state for peer 101 [Dec 10 02:59:18] DEBUG[26132]: devicestate.c:287 do_state_change: Changing state for SIP/101 - state 1 (Not in use) [Dec 10 02:59:18] DEBUG[26132]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - remote_server [Dec 10 02:59:18] DEBUG[26132]: chan_sip.c:16473 sip_devicestate: Checking device state for peer remote_server [Dec 10 02:59:18] DEBUG[26132]: devicestate.c:287 do_state_change: Changing state for SIP/remote_server - state 1 (Not in use) [Dec 10 02:59:18] DEBUG[26136]: app_queue.c:671 handle_statechange: Device 'SIP/101' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 10 02:59:18] DEBUG[26136]: app_queue.c:671 handle_statechange: Device 'SIP/remote_server' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. <--- SIP read from 192.168.128.129:12538 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK71b4d764;rport=5060 Contact: To: ;tag=0a5d8a26 From: "100";tag=as7e836c3f Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 193 v=0 o=- 0 2 IN IP4 192.168.128.129 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 0 3 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK71b4d764;rport=5060 (70) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: Contact: (67) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: From: "100";tag=as7e836c3f (50) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Content-Type: application/sdp (29) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: User-Agent: X-Lite release 1014k stamp 47051 (44) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 193 (19) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=- 0 2 IN IP4 192.168.128.129 (30) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=CounterPath X-Lite 3.0 (24) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 53806 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-15 (15) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (11 headers 9 lines) --- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:2182 __sip_ack: Acked pending invite 105 [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #47 [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2245bc2a10c04be731e248df237d6a62@192.168.128.52' of Request 105: Match Found [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:12273 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.128.129:53806 Got unsupported a:fmtp in SDP offer Found audio description format telephone-event for ID 101 [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:5555 process_sdp: T38 state changed to 0 on channel Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.128.129:53806 [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:5635 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:3244 update_call_counter: Updating call counter for outgoing call [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:8420 build_route: build_route: Retaining previous route: [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2245bc2a10c04be731e248df237d6a62@192.168.128.52 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.129, port 12538 Transmitting (no NAT) to 192.168.128.129:12538: ACK sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK47848ccc;rport From: "100" ;tag=as7e836c3f To: ;tag=0a5d8a26 Contact: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 105 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: ACK sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (68) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK47848ccc;rport (65) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 ACK (13) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: (0) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:6048 reqprep: Strict routing enforced for session 2245bc2a10c04be731e248df237d6a62@192.168.128.52 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.128.129, port 12538 Reliably Transmitting (no NAT) to 192.168.128.129:12538: BYE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK0fcda930;rport From: "100" ;tag=as7e836c3f To: ;tag=0a5d8a26 Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 106 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: BYE sip:101@192.168.128.129:12538;rinstance=9a5db2b61c824463 SIP/2.0 (68) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK0fcda930;rport (65) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as7e836c3f (51) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 106 BYE (13) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Max-Forwards: 70 (16) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Content-Length: 0 (17) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: (0) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:2060 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 Scheduling destruction of SIP dialog '2245bc2a10c04be731e248df237d6a62@192.168.128.52' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.128.129:12538 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK0fcda930;rport=5060 Contact: To: ;tag=0a5d8a26 From: "100";tag=as7e836c3f Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 CSeq: 106 BYE User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 0 <-------------> [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK0fcda930;rport=5060 (70) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: Contact: (67) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=0a5d8a26 (75) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: From: "100";tag=as7e836c3f (50) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2245bc2a10c04be731e248df237d6a62@192.168.128.52 (56) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 106 BYE (13) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: X-Lite release 1014k stamp 47051 (44) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Content-Length: 0 (17) [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:2190 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #49 [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:2216 __sip_ack: Stopping retransmission on '2245bc2a10c04be731e248df237d6a62@192.168.128.52' of Request 106: Match Found Really destroying SIP dialog '2245bc2a10c04be731e248df237d6a62@192.168.128.52' Method: INVITE [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11295 sip_dump_history: ---------- SIP HISTORY for '2245bc2a10c04be731e248df237d6a62@192.168.128.52' [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11299 sip_dump_history: * SIP Call [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 001. NewChan Channel SIP/101-0f59faf0 - from 2245bc2a10c04be731e248df237d6a6 [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 002. TxReqRel INVITE / 102 INVITE - -UNKNOWN- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 003. Rx SIP/2.0 / 102 INVITE / 100 Trying [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 004. Rx SIP/2.0 / 102 INVITE / 180 Ringing [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 005. Rx SIP/2.0 / 102 INVITE / 200 OK [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 006. TxReq ACK / 102 ACK - -UNKNOWN- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 007. ReInv Re-invite sent [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 008. TxReqRel INVITE / 103 INVITE - -UNKNOWN- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 009. Rx SIP/2.0 / 103 INVITE / 100 Trying [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 010. Rx SIP/2.0 / 103 INVITE / 200 OK [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 011. TxReq ACK / 103 ACK - -UNKNOWN- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 012. ReInv Re-invite sent [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 013. TxReqRel INVITE / 104 INVITE - -UNKNOWN- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 014. Rx SIP/2.0 / 104 INVITE / 200 OK [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 015. TxReq ACK / 104 ACK - -UNKNOWN- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 016. ReInv Re-invite sent [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 017. TxReqRel INVITE / 105 INVITE - -UNKNOWN- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 018. Hangup Cause Normal Clearing [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 019. SchedDestroy 32000 ms [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 020. CancelDestroy [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 021. Rx SIP/2.0 / 105 INVITE / 200 OK [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 022. TxReq ACK / 105 ACK - -UNKNOWN- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 023. TxReqRel BYE / 106 BYE - -UNKNOWN- [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 024. SchedDestroy 32000 ms [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11302 sip_dump_history: 025. Rx SIP/2.0 / 106 BYE / 200 OK [Dec 10 02:59:18] DEBUG[26157]: chan_sip.c:11305 sip_dump_history: ---------- END SIP HISTORY for '2245bc2a10c04be731e248df237d6a62@192.168.128.52' <--- SIP read from 192.168.128.51:5060 ---> INVITE sip:101@192.168.128.52 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26848 26852 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: INVITE sip:101@192.168.128.52 SIP/2.0 (37) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;rport (65) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Contact: (33) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: CSeq: 105 INVITE (16) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Max-Forwards: 70 (16) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Supported: replaces (19) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: Content-Type: application/sdp (29) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 13: Content-Length: 293 (19) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 14: (0) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26852 IN IP4 192.168.128.129 (41) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (14 headers 14 lines) --- [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:15795 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:15803 handle_request: Ignoring too old SIP packet packet 105 (expecting >= 114) <--- Transmitting (NAT) to 192.168.128.51:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 503 Server error (24) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 (94) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: "100" ;tag=as2bc4a039 (51) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: ;tag=as1191f6e9 (43) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 105 INVITE (16) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:16030 sipsock_read: SIP message could not be handled, bad request: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #22 (6) SIP/2.0 - 1 [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #22)) Retransmitting #6 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK7a5fc595;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26131 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK684c47bc;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 104 INVITE (16) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #24 (6) SIP/2.0 - 1 [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #24)) Retransmitting #6 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1206461b;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #30 (5) SIP/2.0 - 1 [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #30)) Retransmitting #5 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 106 INVITE (16) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 107 INVITE (16) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #35 (4) SIP/2.0 - 1 [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #35)) Retransmitting #4 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 --- [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (3) SIP/2.0 - 1 [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #41)) Retransmitting #3 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 112 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26141 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #43 (3) SIP/2.0 - 1 [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #43)) Retransmitting #3 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 113 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 110 INVITE (16) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:19] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #37 (4) INVITE - 5 [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 8000 ms (t1 500 ms (Retrans id #37)) Retransmitting #4 (NAT) to 192.168.128.51:5060: INVITE sip:100@192.168.128.51 SIP/2.0 Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;rport From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Contact: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX 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: 293 v=0 o=root 26128 26138 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <-------------> [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 503 Server error (24) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Length: 0 (17) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: (0) --- (11 headers 0 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26848 26859 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 10000 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 200 OK (14) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK275e4308;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 108 INVITE (16) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Contact: (33) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: Content-Type: application/sdp (29) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: Content-Length: 293 (19) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: v=0 (3) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: o=root 26848 26859 IN IP4 192.168.128.129 (41) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: s=session (9) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: c=IN IP4 192.168.128.129 (24) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: t=0 0 (5) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: m=audio 10000 RTP/AVP 0 3 8 101 (31) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=fmtp:101 0-16 (15) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=silenceSupp:off - - - - (25) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=ptime:20 (10) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4927 parse_request: Line: a=sendrecv (10) --- (12 headers 14 lines) --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK50915a3a;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 109 INVITE (16) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:20] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- stop n[Dec 10 02:59:23] WARNING[26157]: chan_sip.c:1966 retrans_pkt: Maximum retries exceeded on transmission 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 for seqno 105 (Non-critical Response) -- See doc/sip-retransmit.txt. [Dec 10 02:59:23] WARNING[26157]: chan_sip.c:1966 retrans_pkt: Maximum retries exceeded on transmission 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 for seqno 106 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:1673 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #30 (6) SIP/2.0 - 1 [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #30)) Retransmitting #6 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK551d2c85;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 106 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK09043cf4;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 106 INVITE (16) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #35 (5) SIP/2.0 - 1 [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #35)) Retransmitting #5 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK1dfd804a;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 --- [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #41 (4) SIP/2.0 - 1 [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #41)) Retransmitting #4 (NAT) to 192.168.128.51:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0ec1a09e;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 112 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 26128 26141 IN IP4 192.168.128.129 s=session c=IN IP4 192.168.128.129 t=0 0 m=audio 53806 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 107 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 <-------------> [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK21e33487;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 107 INVITE (16) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Interworking, unspecified (49) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 127 (31) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:1930 retrans_pkt: SIP TIMER: Rescheduling retransmission #43 (4) SIP/2.0 - 1 [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:1944 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #43)) Retransmitting #4 (NAT) to 192.168.128.51:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.51:5060;branch=z9hG4bK0a64d576;received=192.168.128.51;rport=5060 From: "100" ;tag=as2bc4a039 To: ;tag=as1191f6e9 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 113 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Interworking, unspecified X-Asterisk-HangupCauseCode: 127 --- <--- SIP read from 192.168.128.51:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 From: ;tag=as1191f6e9 To: "100" ;tag=as2bc4a039 Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <-------------> [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 0: SIP/2.0 491 Request Pending (27) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.128.52:5060;branch=z9hG4bK546b2175;received=192.168.128.52;rport=5060 (94) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 2: From: ;tag=as1191f6e9 (45) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 3: To: "100" ;tag=as2bc4a039 (49) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 4: Call-ID: 2691b8d3595a72147f9cdde66d8ba335@192.168.128.51 (56) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 5: CSeq: 110 INVITE (16) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 6: User-Agent: Asterisk PBX (24) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 8: Supported: replaces (19) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 9: Content-Length: 0 (17) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 11: X-Asterisk-HangupCauseCode: 16 (30) [Dec 10 02:59:23] DEBUG[26157]: chan_sip.c:4891 parse_request: Header 12: (0) --- (12 headers 0 lines) --- ow Beginning asterisk shutdown.... Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). [Dec 10 02:59:23] DEBUG[26128]: asterisk.c:1310 quit_handler: Asterisk ending (0).