<-- SIP read from 203.169.36.145:5060: INVITE sip:500014808820711@213.211.134.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.9;branch=z9hG4bK432cf6a66ad2f64f From: "4001" ;tag=54c7a0200acc7f5f To: Contact: Supported: replaces Call-ID: 2d2ba7e7f449e25e@192.168.30.9 CSeq: 5162 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 211 v=0 o=4001 8000 8000 IN IP4 192.168.30.9 s=SIP Call c=IN IP4 192.168.30.9 t=0 0 m=audio 5004 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 0: INVITE sip:500014808820711@213.211.134.22 SIP/2.0 (49) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.30.9;branch=z9hG4bK432cf6a66ad2f64f (60) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 2: From: "4001" ;tag=54c7a0200acc7f5f (59) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 3: To: (40) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 4: Contact: (32) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 5: Supported: replaces (19) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 6: Call-ID: 2d2ba7e7f449e25e@192.168.30.9 (38) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 7: CSeq: 5162 INVITE (17) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 8: User-Agent: Grandstream BT100 1.0.6.7 (37) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 9: Max-Forwards: 70 (16) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 11: Content-Type: application/sdp (29) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 12: Content-Length: 211 (19) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 13: (0) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: v=0 (3) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: o=4001 8000 8000 IN IP4 192.168.30.9 (36) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: s=SIP Call (10) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: c=IN IP4 192.168.30.9 (21) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: t=0 0 (5) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: m=audio 5004 RTP/AVP 8 101 (26) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=sendrecv (10) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=ptime:20 (10) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) --- (13 headers 11 lines)--- Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3155 find_call: = Looking for Call ID: 2d2ba7e7f449e25e@192.168.30.9 (Checking From) --From tag 54c7a0200acc7f5f --To-tag Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3121 sip_alloc: Allocating new SIP dialog for 2d2ba7e7f449e25e@192.168.30.9 - INVITE (With RTP) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:10980 handle_request: **** Received INVITE (5) - Command in SIP INVITE Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:997 parse_sip_options: Begin: parsing SIP "Supported: replaces" Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:1009 parse_sip_options: Found SIP option: -replaces- Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:1015 parse_sip_options: Matched SIP option: replaces Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:1026 parse_sip_options: * SIP extension value: 1 for call 2d2ba7e7f449e25e@192.168.30.9 Using INVITE request as basis request - 2d2ba7e7f449e25e@192.168.30.9 Sending to 192.168.30.9 : 5060 (non-NAT) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:7042 check_user_full: Setting NAT on RTP to 524288 Found user '4001' Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.9:5004 Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:3521 process_sdp: Peer audio RTP is at port 192.168.30.9:5004 Found description format PCMA Found description format telephone-event Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:10354 handle_request_invite: Checking SIP call limits for device 4001 Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:2202 update_call_counter: Updating call counter for incoming call Looking for 500014808820711 in ESSHolland (domain 213.211.134.22) Jul 5 03:41:42 DEBUG[12580]: chan_sip.c:6047 build_route: build_route: Contact hop: list_route: hop: Transmitting (NAT) to 203.169.36.145:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.9;branch=z9hG4bK432cf6a66ad2f64f;received=203.169.36.145 From: "4001" ;tag=54c7a0200acc7f5f To: Call-ID: 2d2ba7e7f449e25e@192.168.30.9 CSeq: 5162 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 5 03:41:42 DEBUG[12570]: chan_sip.c:11502 sip_devicestate: Checking device state for peer 4001 Jul 5 03:41:42 DEBUG[12570]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/4001-381a' Jul 5 03:41:42 DEBUG[12605]: pbx.c:1674 pbx_extension_helper: Launching 'Dial' -- Executing Dial("SIP/4001-381a", "SIP/116302926/0014808820711|60|tT") in new stack Jul 5 03:41:42 DEBUG[12605]: chan_sip.c:3121 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Jul 5 03:41:42 DEBUG[12605]: chan_sip.c:1867 create_addr_from_peer: Setting NAT on RTP to 524288 Jul 5 03:41:42 DEBUG[12605]: channel.c:2813 ast_channel_inherit_variables: Not copying variable STACK-ESSHolland-500014808820711-1. Jul 5 03:41:42 DEBUG[12605]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPCALLID. Jul 5 03:41:42 DEBUG[12605]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. Jul 5 03:41:42 DEBUG[12605]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. Jul 5 03:41:42 DEBUG[12605]: channel.c:2813 ast_channel_inherit_variables: Not copying variable SIPURI. Jul 5 03:41:42 DEBUG[12605]: chan_sip.c:2061 sip_call: Outgoing Call for 0014808820711 Jul 5 03:41:42 DEBUG[12605]: chan_sip.c:2202 update_call_counter: Updating call counter for outgoing call We're at 213.211.134.22 port 10354 Adding codec 0x8 (alaw) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jul 5 03:41:42 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 0: INVITE sip:0014808820711@213.61.187.150 SIP/2.0 (47) Jul 5 03:41:42 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 1: Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK71235abf;rport (65) Jul 5 03:41:42 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 2: From: "4001" ;tag=as6296c125 (58) Jul 5 03:41:42 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 3: To: (38) Jul 5 03:41:42 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 4: Contact: (39) Jul 5 03:41:45 DEBUG[12570]: devicestate.c:187 do_state_change: Changing state for SIP/4001 - state 2 (In use) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 5: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (56) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 7: User-Agent: Asterisk PBX (24) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 8: Max-Forwards: 70 (16) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 9: Date: Wed, 05 Jul 2006 10:41:42 GMT (35) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 11: Content-Type: application/sdp (29) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 12: Content-Length: 315 (19) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3337 parse_request: Header 13: (0) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: v=0 (3) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: o=root 12605 12605 IN IP4 213.211.134.22 (40) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: s=session (9) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: c=IN IP4 213.211.134.22 (23) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: t=0 0 (5) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: m=audio 10354 RTP/AVP 8 4 0 18 101 (34) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: a=rtpmap:4 G723/8000 (20) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: a=fmtp:18 annexb=no (19) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 5 03:41:45 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: a=fmtp:101 0-16 (15) Jul 5 03:41:47 DEBUG[12605]: chan_sip.c:3369 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 14 lines Reliably Transmitting (NAT) to 213.61.187.150:5060: INVITE sip:0014808820711@213.61.187.150 SIP/2.0 Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK71235abf;rport From: "4001" ;tag=as6296c125 To: Contact: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 05 Jul 2006 10:41:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 315 v=0 o=root 12605 12605 IN IP4 213.211.134.22 s=session c=IN IP4 213.211.134.22 t=0 0 m=audio 10354 RTP/AVP 8 4 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 5 03:41:47 DEBUG[12605]: chan_sip.c:1286 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #17 -- Called 116302926/0014808820711 Jul 5 03:41:47 WARNING[12605]: channel.c:2691 ast_channel_make_compatible: No path to translate from SIP/116302926-6ce0(1) to SIP/4001-381a(8) Jul 5 03:41:47 DEBUG[12607]: app_queue.c:471 changethread: Device 'SIP/4001' changed to state '2' (In use) <-- SIP read from 213.61.187.150:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK71235abf;received=213.211.134.22;rport=5060 From: "4001" ;tag=as6296c125 To: ;tag=as35b1a124 Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 102 INVITE User-Agent: SipProxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5a0fe198" Content-Length: 0 Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 0: SIP/2.0 407 Proxy Authentication Required (41) Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 1: Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK71235abf;received=213.211.134.22;rport=5060 (94) Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 2: From: "4001" ;tag=as6296c125 (58) Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 3: To: ;tag=as35b1a124 (53) Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 4: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (56) Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 6: User-Agent: SipProxy (20) Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY (55) Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 8: Contact: (43) Jul 5 03:41:47 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 9: Proxy-Authenticate: Digest realm="asterisk", nonce="5a0fe198" (62) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 10: Content-Length: 0 (17) --- (11 headers 0 lines)--- Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3155 find_call: = Looking for Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (Checking To) --From tag as6296c125 --To-tag as35b1a124 Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3169 find_call: = Found Their Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 Their Tag Our tag: as6296c125 Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:1372 __sip_ack: Acked pending invite 102 Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:1383 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #17 Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '15b2a2f22677a60817dd71d6245e5f14@213.211.134.22' of Request 102: Match Found Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:9459 handle_response_invite: SIP response 407 to standard invite Transmitting (NAT) to 213.61.187.150:5060: ACK sip:0014808820711@213.61.187.150 SIP/2.0 Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK71235abf;rport From: "4001" ;tag=as6296c125 To: ;tag=as35b1a124 Contact: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:8903 do_proxy_auth: Auth attempt 1 on INVITE We're at 213.211.134.22 port 10354 Adding codec 0x8 (alaw) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 213.61.187.150:5060: INVITE sip:0014808820711@213.61.187.150 SIP/2.0 Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;rport From: "4001" ;tag=as6296c125 To: Contact: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="116302926", realm="asterisk", algorithm=MD5, uri="sip:0014808820711@213.61.187.150", nonce="5a0fe198", response="df8184c5be442a047947b4058b364587", opaque="" Date: Wed, 05 Jul 2006 10:41:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 315 v=0 o=root 12605 12606 IN IP4 213.211.134.22 s=session c=IN IP4 213.211.134.22 t=0 0 m=audio 10354 RTP/AVP 8 4 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:1286 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #18 <-- SIP read from 213.61.187.150:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK71235abf;received=213.211.134.22;rport=5060 From: "4001" ;tag=as6296c125 To: ;tag=as35b1a124 Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 102 INVITE User-Agent: SipProxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5a0fe198" Content-Length: 0 Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 0: SIP/2.0 407 Proxy Authentication Required (41) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 1: Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK71235abf;received=213.211.134.22;rport=5060 (94) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 2: From: "4001" ;tag=as6296c125 (58) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 3: To: ;tag=as35b1a124 (53) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 4: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (56) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 6: User-Agent: SipProxy (20) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY (55) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 8: Contact: (43) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 9: Proxy-Authenticate: Digest realm="asterisk", nonce="5a0fe198" (62) Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 10: Content-Length: 0 (17) --- (11 headers 0 lines)--- Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3155 find_call: = Looking for Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (Checking To) --From tag as6296c125 --To-tag as35b1a124 Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:3169 find_call: = Found Their Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 Their Tag Our tag: as6296c125 Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '15b2a2f22677a60817dd71d6245e5f14@213.211.134.22' of Request 102: Match Not Found Jul 5 03:41:50 DEBUG[12580]: chan_sip.c:9459 handle_response_invite: SIP response 407 to standard invite Transmitting (NAT) to 213.61.187.150:5060: ACK sip:0014808820711@213.61.187.150 SIP/2.0 Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;rport From: "4001" ;tag=as6296c125 To: ;tag=as35b1a124 Contact: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:1177 retrans_pkt: SIP TIMER: Rescheduling retransmission #18 (1) INVITE - 5 Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:1191 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #18)) Retransmitting #1 (NAT) to 213.61.187.150:5060: INVITE sip:0014808820711@213.61.187.150 SIP/2.0 Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;rport From: "4001" ;tag=as6296c125 To: Contact: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="116302926", realm="asterisk", algorithm=MD5, uri="sip:0014808820711@213.61.187.150", nonce="5a0fe198", response="df8184c5be442a047947b4058b364587", opaque="" Date: Wed, 05 Jul 2006 10:41:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 315 v=0 o=root 12605 12606 IN IP4 213.211.134.22 s=session c=IN IP4 213.211.134.22 t=0 0 m=audio 10354 RTP/AVP 8 4 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- <-- SIP read from 213.61.187.150:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;received=213.211.134.22;rport=5060 From: "4001" ;tag=as6296c125 To: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 103 INVITE User-Agent: SipProxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 0: SIP/2.0 100 Trying (18) Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 1: Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;received=213.211.134.22;rport=5060 (94) Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 2: From: "4001" ;tag=as6296c125 (58) Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 3: To: (38) Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 4: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (56) Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 5: CSeq: 103 INVITE (16) Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 6: User-Agent: SipProxy (20) Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY (55) Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 8: Contact: (43) Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 9: Content-Length: 0 (17) --- (10 headers 0 lines)--- Jul 5 03:41:51 DEBUG[12580]: chan_sip.c:3155 find_call: = Looking for Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (Checking To) --From tag as6296c125 --To-tag Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3169 find_call: = Found Their Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 Their Tag Our tag: as6296c125 Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:1438 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #18 - INVITE (got response) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:1447 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '15b2a2f22677a60817dd71d6245e5f14@213.211.134.22' Request 103: Found Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:9459 handle_response_invite: SIP response 100 to standard invite <-- SIP read from 213.61.187.150:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;received=213.211.134.22;rport=5060 From: "4001" ;tag=as6296c125 To: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 103 INVITE User-Agent: SipProxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 0: SIP/2.0 100 Trying (18) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 1: Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;received=213.211.134.22;rport=5060 (94) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 2: From: "4001" ;tag=as6296c125 (58) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 3: To: (38) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 4: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (56) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 5: CSeq: 103 INVITE (16) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 6: User-Agent: SipProxy (20) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY (55) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 8: Contact: (43) Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 9: Content-Length: 0 (17) --- (10 headers 0 lines)--- Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3155 find_call: = Looking for Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (Checking To) --From tag as6296c125 --To-tag Jul 5 03:41:52 DEBUG[12580]: chan_sip.c:3169 find_call: = Found Their Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 Their Tag Our tag: as6296c125 Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:1447 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '15b2a2f22677a60817dd71d6245e5f14@213.211.134.22' Request 103: Found Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:9459 handle_response_invite: SIP response 100 to standard invite <-- SIP read from 213.61.187.150:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;received=213.211.134.22;rport=5060 From: "4001" ;tag=as6296c125 To: ;tag=as2708f8cd Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 103 INVITE User-Agent: SipProxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 365 v=0 o=root 7986 7986 IN IP4 213.61.187.150 s=session c=IN IP4 213.61.187.150 t=0 0 m=audio 35990 RTP/AVP 8 0 18 3 4 2 7 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:7 LPC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 0: SIP/2.0 183 Session Progress (28) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 1: Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;received=213.211.134.22;rport=5060 (94) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 2: From: "4001" ;tag=as6296c125 (58) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 3: To: ;tag=as2708f8cd (53) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 4: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (56) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 5: CSeq: 103 INVITE (16) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 6: User-Agent: SipProxy (20) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY (55) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 8: Contact: (43) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 9: Content-Type: application/sdp (29) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 10: Content-Length: 365 (19) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 11: (0) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: v=0 (3) Jul 5 03:41:56 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: o=root 7986 7986 IN IP4 213.61.187.150 (38) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: s=session (9) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: c=IN IP4 213.61.187.150 (23) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: t=0 0 (5) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: m=audio 35990 RTP/AVP 8 0 18 3 4 2 7 101 (40) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:4 G723/8000 (20) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:7 LPC/8000 (19) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=fmtp:101 0-16 (15) --- (11 headers 16 lines)--- Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3155 find_call: = Looking for Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (Checking To) --From tag as6296c125 --To-tag as2708f8cd Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3169 find_call: = Found Their Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 Their Tag Our tag: as6296c125 Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:1447 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '15b2a2f22677a60817dd71d6245e5f14@213.211.134.22' Request 103: Found Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:9459 handle_response_invite: SIP response 183 to standard invite Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 4 Found RTP audio format 2 Found RTP audio format 7 Found RTP audio format 101 Peer audio RTP is at port 213.61.187.150:35990 Jul 5 03:41:57 DEBUG[12580]: chan_sip.c:3521 process_sdp: Peer audio RTP is at port 213.61.187.150:35990 Found description format PCMA Found description format PCMU Found description format G729 Found description format GSM Found description format G723 Found description format G726-32 Found description format LPC Found description format telephone-event Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x19f (g723|gsm|ulaw|alaw|g726|lpc10|g729)/video=0x0 (nothing), combined - 0x10d (g723|ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) <-- SIP read from 203.169.36.145:5060: --- (0 headers 0 lines) Nat keepalive --- -- SIP/116302926-6ce0 is making progress passing it to SIP/4001-381a We're at 213.211.134.22 port 11724 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to 203.169.36.145:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.30.9;branch=z9hG4bK432cf6a66ad2f64f;received=203.169.36.145 From: "4001" ;tag=54c7a0200acc7f5f To: ;tag=as6268fd8c Call-ID: 2d2ba7e7f449e25e@192.168.30.9 CSeq: 5162 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 220 v=0 o=root 12605 12605 IN IP4 213.211.134.22 s=session c=IN IP4 213.211.134.22 t=0 0 m=audio 11724 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 5 03:42:00 DEBUG[12605]: chan_sip.c:2981 sip_rtp_read: Oooh, format changed to 8 Jul 5 03:42:00 WARNING[12605]: channel.c:2326 set_format: Unable to find a codec translation path from alaw to g723 Jul 5 03:42:00 WARNING[12605]: channel.c:2326 set_format: Unable to find a codec translation path from alaw to g723 Jul 5 03:42:00 DEBUG[12605]: rtp.c:1341 ast_rtp_write: Ooh, format changed from unknown to alaw <-- SIP read from 203.169.36.145:5060: REGISTER sip:213.211.134.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.9;branch=z9hG4bK4c95e91dd17d2289 From: "4001" ;tag=b09d22b7329d878d To: Contact: Call-ID: 44f70f2438349e2b@192.168.30.9 CSeq: 104 REGISTER Expires: 60 User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 0: REGISTER sip:213.211.134.22 SIP/2.0 (35) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.30.9;branch=z9hG4bK4c95e91dd17d2289 (60) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 2: From: "4001" ;tag=b09d22b7329d878d (59) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 3: To: (29) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 4: Contact: (32) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 5: Call-ID: 44f70f2438349e2b@192.168.30.9 (38) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 6: CSeq: 104 REGISTER (18) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 7: Expires: 60 (11) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 8: User-Agent: Grandstream BT100 1.0.6.7 (37) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 9: Max-Forwards: 70 (16) Jul 5 03:42:01 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 11: Content-Length: 0 (17) --- (12 headers 0 lines)--- Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3155 find_call: = Looking for Call ID: 44f70f2438349e2b@192.168.30.9 (Checking From) --From tag b09d22b7329d878d --To-tag Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3169 find_call: = No match Their Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 Their Tag as2708f8cd Our tag: as6296c125 Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3169 find_call: = No match Their Call ID: 2d2ba7e7f449e25e@192.168.30.9 Their Tag 54c7a0200acc7f5f Our tag: as6268fd8c Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3121 sip_alloc: Allocating new SIP dialog for 44f70f2438349e2b@192.168.30.9 - REGISTER (No RTP) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:10980 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.30.9 : 5060 (non-NAT) Transmitting (NAT) to 203.169.36.145:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.9;branch=z9hG4bK4c95e91dd17d2289;received=203.169.36.145 From: "4001" ;tag=b09d22b7329d878d To: Call-ID: 44f70f2438349e2b@192.168.30.9 CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 5 03:42:04 DEBUG[12605]: rtp.c:477 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 213.61.187.133:17580 Jul 5 03:42:04 NOTICE[12605]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 213.61.187.133 Jul 5 03:42:04 DEBUG[12605]: rtp.c:477 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 203.169.36.145:5004 Jul 5 03:42:04 DEBUG[12605]: rtp.c:1341 ast_rtp_write: Ooh, format changed from unknown to alaw Transmitting (NAT) to 203.169.36.145:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.9;branch=z9hG4bK4c95e91dd17d2289;received=203.169.36.145 From: "4001" ;tag=b09d22b7329d878d To: ;tag=as7da78f7a Call-ID: 44f70f2438349e2b@192.168.30.9 CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 60 Contact: ;expires=60 Date: Wed, 05 Jul 2006 10:42:04 GMT Content-Length: 0 --- Scheduling destruction of call '44f70f2438349e2b@192.168.30.9' in 15000 ms <-- SIP read from 213.61.187.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;received=213.211.134.22;rport=5060 From: "4001" ;tag=as6296c125 To: ;tag=as2708f8cd Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 103 INVITE User-Agent: SipProxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 365 v=0 o=root 7986 7987 IN IP4 213.61.187.150 s=session c=IN IP4 213.61.187.150 t=0 0 m=audio 35990 RTP/AVP 8 0 18 3 4 2 7 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:7 LPC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 0: SIP/2.0 200 OK (14) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 1: Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK2dba3faf;received=213.211.134.22;rport=5060 (94) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 2: From: "4001" ;tag=as6296c125 (58) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 3: To: ;tag=as2708f8cd (53) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 4: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (56) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 5: CSeq: 103 INVITE (16) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 6: User-Agent: SipProxy (20) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY (55) Jul 5 03:42:04 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 8: Contact: (43) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 9: Content-Type: application/sdp (29) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 10: Content-Length: 365 (19) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3337 parse_request: Header 11: (0) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: v=0 (3) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: o=root 7986 7987 IN IP4 213.61.187.150 (38) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: s=session (9) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: c=IN IP4 213.61.187.150 (23) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: t=0 0 (5) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: m=audio 35990 RTP/AVP 8 0 18 3 4 2 7 101 (40) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:3 GSM/8000 (19) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:4 G723/8000 (20) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:7 LPC/8000 (19) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3369 parse_request: Line: a=fmtp:101 0-16 (15) --- (11 headers 16 lines)--- Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3155 find_call: = Looking for Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (Checking To) --From tag as6296c125 --To-tag as2708f8cd Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3169 find_call: = No match Their Call ID: 44f70f2438349e2b@192.168.30.9 Their Tag b09d22b7329d878d Our tag: as7da78f7a Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3169 find_call: = Found Their Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 Their Tag as2708f8cd Our tag: as6296c125 Jul 5 03:42:34 DEBUG[12579]: manager.c:1249 process_message: Manager received command 'Command' Jul 5 03:42:34 DEBUG[12579]: manager.c:1249 process_message: Manager received command 'Command' Jul 5 03:42:34 DEBUG[12570]: chan_sip.c:11502 sip_devicestate: Checking device state for peer 4001 Jul 5 03:42:34 DEBUG[12570]: devicestate.c:187 do_state_change: Changing state for SIP/4001 - state 2 (In use) Jul 5 03:42:34 DEBUG[12605]: rtp.c:406 ast_rtcp_read: RTCP NAT: Got RTCP from other end. Now sending to address 213.61.187.133:17581 Jul 5 03:42:34 DEBUG[12605]: rtp.c:410 ast_rtcp_read: Got RTCP report of 136 bytes Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:11155 sipsock_read: Failed to grab lock, trying again... Jul 5 03:42:34 DEBUG[12605]: rtp.c:410 ast_rtcp_read: Got RTCP report of 112 bytes Jul 5 03:42:34 DEBUG[12605]: rtp.c:410 ast_rtcp_read: Got RTCP report of 112 bytes Jul 5 03:42:34 DEBUG[12605]: rtp.c:410 ast_rtcp_read: Got RTCP report of 48 bytes Jul 5 03:42:34 DEBUG[12605]: rtp.c:410 ast_rtcp_read: Got RTCP report of 48 bytes Jul 5 03:42:34 DEBUG[12608]: app_queue.c:471 changethread: Device 'SIP/4001' changed to state '2' (In use) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3155 find_call: = Looking for Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 (Checking To) --From tag as6296c125 --To-tag as2708f8cd Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3169 find_call: = No match Their Call ID: 44f70f2438349e2b@192.168.30.9 Their Tag b09d22b7329d878d Our tag: as7da78f7a Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3169 find_call: = Found Their Call ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 Their Tag as2708f8cd Our tag: as6296c125 Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:1372 __sip_ack: Acked pending invite 103 Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:1394 __sip_ack: Stopping retransmission on '15b2a2f22677a60817dd71d6245e5f14@213.211.134.22' of Request 103: Match Found Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:9459 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 3 Found RTP audio format 4 Found RTP audio format 2 Found RTP audio format 7 Found RTP audio format 101 Peer audio RTP is at port 213.61.187.150:35990 Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:3521 process_sdp: Peer audio RTP is at port 213.61.187.150:35990 Found description format PCMA Found description format PCMU Found description format G729 Found description format GSM Found description format G723 Found description format G726-32 Found description format LPC Found description format telephone-event Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x19f (g723|gsm|ulaw|alaw|g726|lpc10|g729)/video=0x0 (nothing), combined - 0x10d (g723|ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:6047 build_route: build_route: Contact hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.61.187.150, port 5060 Transmitting (NAT) to 213.61.187.150:5060: ACK sip:0014808820711@213.61.187.150 SIP/2.0 Via: SIP/2.0/UDP 213.211.134.22:5060;branch=z9hG4bK3b586f2f;rport From: "4001" ;tag=as6296c125 To: ;tag=as2708f8cd Contact: Call-ID: 15b2a2f22677a60817dd71d6245e5f14@213.211.134.22 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 5 03:42:34 DEBUG[12580]: chan_sip.c:1316 __sip_autodestruct: Auto destroying call '44f70f2438349e2b@192.168.30.9' Destroying call '44f70f2438349e2b@192.168.30.9' <-- SIP read from 203.169.36.145:5060: REGISTER sip:213.211.134.22 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.9;branch=z9hG4bK4c95e91dd17d2289 From: "4001" ;tag=b09d22b7329d878d To: Contact: Call-ID: 44f70f2438349e2b@192.168.30.9 CSeq: 104 REGISTER Expires: 60 User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0