Asterisk Ready. *CLI> Jun 8 13:56:11 NOTICE[19327]: chan_sip.c:5473 sip_reregister: -- Re-registration for 37070070541@skambink.zebra.lt Jun 8 13:56:11 DEBUG[19327]: chan_sip.c:5638 transmit_register: Scheduled a registration timeout for skambink.zebra.lt id #10 REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 212.59.4.20:5060: REGISTER sip:skambink.zebra.lt SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK32f94d5d;rport From: ;tag=as6d9ead22 To: Call-ID: 780d992e252ba388789aea7a414c1f7f@192.168.1.104 CSeq: 102 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 212.59.4.20:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK32f94d5d;rport=5060 From: ;tag=as6d9ead22 To: ;tag=test_tag_0010355315 Call-ID: 780d992e252ba388789aea7a414c1f7f@192.168.1.104 CSeq: 102 REGISTER User-Agent: Asterisk PBX Event: registration Content-Length: 0 WWW-Authenticate: Digest realm="voip.telecom.lt", nonce="3222960750-a610ca217039dce7d64b1274830212c0", algorithm=MD5, qop="auth" --- (10 headers 0 lines) --- Jun 8 13:56:11 DEBUG[19327]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '780d992e252ba388789aea7a414c1f7f@192.168.1.104' of Request 102: Match Found Responding to challenge, registration to domain/host name skambink.zebra.lt REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 212.59.4.20:5060: REGISTER sip:skambink.zebra.lt SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK3e61acfe;rport From: ;tag=as4270b13f To: Call-ID: 780d992e252ba388789aea7a414c1f7f@192.168.1.104 CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="37070070541", realm="voip.telecom.lt", algorithm=MD5, uri="sip:skambink.zebra.lt", nonce="3222960750-a610ca217039dce7d64b1274830212c0", response="154254126f6967a7a4284f633d909dfd", opaque="", qop=auth, cnonce="1717db03", nc=00000001 Expires: 120 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK3e61acfe;rport=5060 From: ;tag=as4270b13f To: Call-ID: 780d992e252ba388789aea7a414c1f7f@192.168.1.104 CSeq: 103 REGISTER Contact: ;expires=30;audio Content-Length: 0 --- (8 headers 0 lines) --- Jun 8 13:56:11 DEBUG[19327]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '780d992e252ba388789aea7a414c1f7f@192.168.1.104' of Request 103: Match Found Jun 8 13:56:11 DEBUG[19327]: chan_sip.c:9957 handle_response_register: Registration successful Jun 8 13:56:11 DEBUG[19327]: chan_sip.c:9959 handle_response_register: Cancelling timeout 10 Scheduling destruction of call '780d992e252ba388789aea7a414c1f7f@192.168.1.104' in 32000 ms Jun 8 13:56:11 NOTICE[19327]: chan_sip.c:10009 handle_response_register: Outbound Registration: Expiry for skambink.zebra.lt is 30 sec (Scheduling reregistration in 23 s) <-- SIP read from 192.168.1.53:5060: REGISTER sip:192.168.1.104:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK20099190293237714176;rport From: 601 ;tag=3210115622 To: 601 Call-ID: 2720127995-1370317236@192.168.1.53 CSeq: 1 REGISTER Contact: Authorization: Digest username="601", realm="asterisk", nonce="36df6cab", uri="sip:192.168.1.104:5060", response="f770cb42bfdf936338fdd48d212c2ec3", algorithm=MD5 max-forwards: 70 expires: 60 user-agent: Voip Phone 1.0 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.53 : 5060 (NAT) Transmitting (no NAT) to 192.168.1.53:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK20099190293237714176;received=192.168.1.53;rport=5060 From: 601 ;tag=3210115622 To: 601 Call-ID: 2720127995-1370317236@192.168.1.53 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.1.53:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK20099190293237714176;received=192.168.1.53;rport=5060 From: 601 ;tag=3210115622 To: 601 ;tag=as247ad578 Call-ID: 2720127995-1370317236@192.168.1.53 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17131286" Content-Length: 0 --- Scheduling destruction of call '2720127995-1370317236@192.168.1.53' in 15000 ms <-- SIP read from 192.168.1.53:5060: REGISTER sip:192.168.1.104:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK60422322563411405;rport From: 601 ;tag=3210115622 To: 601 Call-ID: 2720127995-1370317236@192.168.1.53 CSeq: 2 REGISTER Contact: Authorization: Digest username="601", realm="asterisk", nonce="17131286", uri="sip:192.168.1.104:5060", response="9a811fdc2cd92e962187501fb9e6b543", algorithm=MD5 max-forwards: 70 expires: 60 user-agent: Voip Phone 1.0 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.1.53 : 5060 (NAT) Transmitting (no NAT) to 192.168.1.53:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK60422322563411405;received=192.168.1.53;rport=5060 From: 601 ;tag=3210115622 To: 601 Call-ID: 2720127995-1370317236@192.168.1.53 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Saved useragent "Voip Phone 1.0" for peer 601 Transmitting (no NAT) to 192.168.1.53:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK60422322563411405;received=192.168.1.53;rport=5060 From: 601 ;tag=3210115622 To: 601 ;tag=as247ad578 Call-ID: 2720127995-1370317236@192.168.1.53 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 60 Contact: ;expires=60 Date: Fri, 08 Jun 2007 10:56:15 GMT Content-Length: 0 --- Scheduling destruction of call '2720127995-1370317236@192.168.1.53' in 15000 ms <-- SIP read from 212.59.4.20:5060: INVITE sip:37070070541@192.168.1.104 SIP/2.0 Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK080q1610a8e0jakf85s0.1 To: From: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 1235 INVITE Contact: Accept-Language: en;q=0.0 Alert-Info: Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Fri,08 Jun 2007 07:59:03 GMT Max-Forwards: 68 Supported: timer Session-Expires: 1800;refresher=uac Min-SE: 1800 Content-Type: application/sdp Content-Length: 626 v=0 o=hiQ9200 1838020070508105901 1785725007 IN IP4 212.59.4.20 s=Phone Call via hiQ9200 SIPCA c=IN IP4 212.59.4.20 t=0 0 m=audio 15346 RTP/AVP 8 18 0 4 102 2 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:2 G726-32/8000 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 18 0 4 102 2 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=fmtp:8 vad=no a=cpar: a=rtpmap:18 G729/8000 a=cpar: a=rtpmap:0 PCMU/8000 a=cpar: a=rtpmap:4 G723/8000 a=cpar: a=rtpmap:102 G726-32/8000 a=cpar: a=rtpmap:2 G726-32/8000 a=cdsc: 7 image udptl t38 a=sendrecv --- (17 headers 24 lines) --- Using INVITE request as basis request - 3459821811-3393166743091811-11-863266 Sending to 212.59.4.20 : 5060 (non-NAT) Found peer 'teo' Jun 8 13:56:20 DEBUG[19327]: chan_sip.c:7393 check_user_full: Setting NAT on RTP to 524288 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 102 Found RTP audio format 2 Peer audio RTP is at port 212.59.4.20:15346 Jun 8 13:56:20 DEBUG[19327]: chan_sip.c:3687 process_sdp: Peer audio RTP is at port 212.59.4.20:15346 Found description format PCMA Found description format G729 Found description format PCMU Found description format G723 Found description format G726-32 Found description format G726-32 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x11d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Jun 8 13:56:20 DEBUG[19327]: chan_sip.c:10695 handle_request_invite: Checking SIP call limits for device 37070070541 Looking for 37070070541 in default (domain 192.168.1.104) Jun 8 13:56:20 DEBUG[19327]: chan_sip.c:6280 build_route: build_route: Contact hop: list_route: hop: Transmitting (NAT) to 212.59.4.20:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK080q1610a8e0jakf85s0.1;received=212.59.4.20 From: ;tag=test_tag_0010355350 To: Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 1235 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/37070070541-0819c9f8", "SIP/601") in new stack Jun 8 13:56:20 DEBUG[19335]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 0 Jun 8 13:56:20 DEBUG[19335]: chan_sip.c:2083 sip_call: Outgoing Call for 601 We're at 192.168.1.104 port 13558 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP 13 headers, 9 lines Reliably Transmitting (no NAT) to 192.168.1.53:5060: INVITE sip:601@192.168.1.53:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK6ea83c5a;rport From: "0037067198101" ;tag=as02eab0c9 To: Contact: Call-ID: 116a9be269793f0e443e5a3d7376eff7@192.168.1.104 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 08 Jun 2007 10:56:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 186 v=0 o=root 19317 19317 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 13558 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Called 601 <-- SIP read from 192.168.1.53:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK6ea83c5a;rport From: "0037067198101" ;tag=as02eab0c9 To: Call-ID: 116a9be269793f0e443e5a3d7376eff7@192.168.1.104 CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines) --- Jun 8 13:56:21 DEBUG[19327]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '116a9be269793f0e443e5a3d7376eff7@192.168.1.104' Request 102: Found <-- SIP read from 192.168.1.53:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK6ea83c5a;rport From: "0037067198101" ;tag=as02eab0c9 To: ;tag=146828229 Call-ID: 116a9be269793f0e443e5a3d7376eff7@192.168.1.104 CSeq: 102 INVITE Contact: Content-Length: 0 --- (8 headers 0 lines) --- Jun 8 13:56:21 DEBUG[19327]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '116a9be269793f0e443e5a3d7376eff7@192.168.1.104' Request 102: Found -- SIP/601-081a2c08 is ringing Transmitting (NAT) to 212.59.4.20:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK080q1610a8e0jakf85s0.1;received=212.59.4.20 From: ;tag=test_tag_0010355350 To: ;tag=as05bb784a Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 1235 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- <-- SIP read from 192.168.1.53:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK6ea83c5a;rport From: "0037067198101" ;tag=as02eab0c9 To: ;tag=146828229 Call-ID: 116a9be269793f0e443e5a3d7376eff7@192.168.1.104 CSeq: 102 INVITE Contact: supported: replaces Content-Type: application/sdp Content-Length: 169 v=0 o=553 53295792 25854605 IN IP4 192.168.1.53 s=A conversation c=IN IP4 192.168.1.53 t=0 0 m=audio 10096 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 --- (10 headers 8 lines) --- Jun 8 13:56:24 DEBUG[19327]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 Jun 8 13:56:24 DEBUG[19327]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '116a9be269793f0e443e5a3d7376eff7@192.168.1.104' of Request 102: Match Found Found RTP audio format 0 Found RTP audio format 8 Peer audio RTP is at port 192.168.1.53:10096 Jun 8 13:56:24 DEBUG[19327]: chan_sip.c:3687 process_sdp: Peer audio RTP is at port 192.168.1.53:10096 Found description format PCMU Found description format PCMA Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Jun 8 13:56:24 DEBUG[19327]: chan_sip.c:6280 build_route: build_route: Contact hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.53, port 5060 Transmitting (no NAT) to 192.168.1.53:5060: ACK sip:601@192.168.1.53:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK55d5318f;rport From: "0037067198101" ;tag=as02eab0c9 To: ;tag=146828229 Contact: Call-ID: 116a9be269793f0e443e5a3d7376eff7@192.168.1.104 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/601-081a2c08 answered SIP/37070070541-0819c9f8 We're at 192.168.1.104 port 16522 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (NAT) to 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK080q1610a8e0jakf85s0.1;received=212.59.4.20 From: ;tag=test_tag_0010355350 To: ;tag=as05bb784a Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 1235 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 186 v=0 o=root 19317 19317 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 16522 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/37070070541-0819c9f8 and SIP/601-081a2c08 <-- SIP read from 212.59.4.20:5060: ACK sip:37070070541@192.168.1.104 SIP/2.0 Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK080q1810agehha48r681.1 To: ;tag=as05bb784a From: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 1235 ACK Contact: Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Fri,08 Jun 2007 07:59:07 GMT Max-Forwards: 68 Content-Length: 0 --- (11 headers 0 lines) --- Jun 8 13:56:24 DEBUG[19327]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '3459821811-3393166743091811-11-863266' of Response 1235: Match Found <-- SIP read from 212.59.4.20:5060: INVITE sip:37070070541@192.168.1.104 SIP/2.0 Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK080q1810boa0ma4686g0.1 To: ;tag=as05bb784a From: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 1236 INVITE Contact: Accept-Language: en;q=0.0 Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Fri,08 Jun 2007 07:59:07 GMT Max-Forwards: 68 Supported: timer Session-Expires: 1800;refresher=uac Min-SE: 1800 Content-Type: application/sdp Content-Length: 186 v=0 o=hiQ9200 1838020070508105901 1785725008 IN IP4 212.59.4.20 s=Phone Call via hiQ9200 SIPCA c=IN IP4 212.59.4.20 t=0 0 m=audio 15346 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 --- (16 headers 8 lines) --- Using INVITE request as basis request - 3459821811-3393166743091811-11-863266 Sending to 212.59.4.20 : 5060 (NAT) Found RTP audio format 0 Peer audio RTP is at port 212.59.4.20:15346 Jun 8 13:56:24 DEBUG[19327]: chan_sip.c:3687 process_sdp: Peer audio RTP is at port 212.59.4.20:15346 Found description format PCMU Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) We're at 192.168.1.104 port 16522 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK080q1810boa0ma4686g0.1;received=212.59.4.20 From: ;tag=test_tag_0010355350 To: ;tag=as05bb784a Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 1236 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 162 v=0 o=root 19317 19318 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 16522 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- <-- SIP read from 212.59.4.20:5060: ACK sip:37070070541@192.168.1.104 SIP/2.0 Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK080q1810bgshpak8f6o0.1 To: ;tag=as05bb784a From: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 1236 ACK Contact: Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Fri,08 Jun 2007 07:59:07 GMT Max-Forwards: 68 Content-Length: 0 --- (11 headers 0 lines) --- Jun 8 13:56:24 DEBUG[19327]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '3459821811-3393166743091811-11-863266' of Response 1236: Match Found Jun 8 13:56:26 NOTICE[19335]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.1.53 Jun 8 13:56:26 NOTICE[19335]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 212.59.4.20 Jun 8 13:56:30 DEBUG[19327]: chan_sip.c:1336 __sip_autodestruct: Auto destroying call '2720127995-1370317236@192.168.1.53' Destroying call '2720127995-1370317236@192.168.1.53' <-- SIP read from 192.168.1.53:5060: BYE sip:0037067198101@192.168.1.104 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK58917246146878144;rport From: 601 ;tag=146828229 To: "0037067198101" ;tag=as02eab0c9 Call-ID: 116a9be269793f0e443e5a3d7376eff7@192.168.1.104 CSeq: 1 BYE max-forwards: 70 user-agent: Voip Phone 1.0 Content-Length: 0 --- (9 headers 0 lines) --- Sending to 192.168.1.53 : 5060 (NAT) Transmitting (NAT) to 192.168.1.53:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK58917246146878144;received=192.168.1.53;rport=5060 From: 601 ;tag=146828229 To: "0037067198101" ;tag=as02eab0c9 Call-ID: 116a9be269793f0e443e5a3d7376eff7@192.168.1.104 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Jun 8 13:56:34 DEBUG[19335]: channel.c:3375 ast_generic_bridge: Didn't get a frame from channel: SIP/601-081a2c08 Jun 8 13:56:34 DEBUG[19335]: channel.c:3664 ast_channel_bridge: Bridge stops bridging channels SIP/37070070541-0819c9f8 and SIP/601-081a2c08 Jun 8 13:56:34 DEBUG[19335]: chan_sip.c:2448 sip_hangup: update_call_counter(601) - decrement call limit counter Jun 8 13:56:34 DEBUG[19335]: app_dial.c:1661 dial_exec_full: Exiting with DIALSTATUS=ANSWER. == Spawn extension (default, 37070070541, 1) exited non-zero on 'SIP/37070070541-0819c9f8' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '0037067198101' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '0037067198101' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '37070070541' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'default' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/37070070541-0819c9f8' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/601-081a2c08' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'Dial' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/601' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-06-08 13:56:20' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-06-08 13:56:24' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-06-08 13:56:34' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '14' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '10' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '1181300180.0' Jun 8 13:56:34 DEBUG[19335]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)' Jun 8 13:56:34 DEBUG[19335]: chan_sip.c:2448 sip_hangup: update_call_counter(37070070541) - decrement call limit counter Scheduling destruction of call '3459821811-3393166743091811-11-863266' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 212.59.4.20, port 5060 Reliably Transmitting (NAT) to 212.59.4.20:5060: BYE sip:0037067198101@212.59.4.20:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK2c07712e;rport From: ;tag=as05bb784a To: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <-- SIP read from 212.59.4.20:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK2c07712e;rport=5060 From: ;tag=as05bb784a To: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 WWW-Authenticate: Digest realm="voip.telecom.lt", nonce="3222960978-ad853fc82755b18c53b674076ea33e69", algorithm=MD5, qop="auth" --- (9 headers 0 lines) --- Jun 8 13:56:34 DEBUG[19327]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '3459821811-3393166743091811-11-863266' of Request 102: Match Found set_destination: Parsing for address/port to send to set_destination: set destination to 212.59.4.20, port 5060 Reliably Transmitting (NAT) to 212.59.4.20:5060: BYE sip:0037067198101@212.59.4.20:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK70b10616;rport From: ;tag=as05bb784a To: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="37070070541", realm="voip.telecom.lt", algorithm=MD5, uri="sip:0037067198101@212.59.4.20:5060", nonce="3222960978-ad853fc82755b18c53b674076ea33e69", response="7fd0e1ef930bddf7456f33fd95c94f78", opaque="", qop=auth, cnonce="199831cc", nc=00000001 Date: Fri, 08 Jun 2007 10:56:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '116a9be269793f0e443e5a3d7376eff7@192.168.1.104' Retransmitting #1 (NAT) to 212.59.4.20:5060: BYE sip:0037067198101@212.59.4.20:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK70b10616;rport From: ;tag=as05bb784a To: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="37070070541", realm="voip.telecom.lt", algorithm=MD5, uri="sip:0037067198101@212.59.4.20:5060", nonce="3222960978-ad853fc82755b18c53b674076ea33e69", response="7fd0e1ef930bddf7456f33fd95c94f78", opaque="", qop=auth, cnonce="199831cc", nc=00000001 Date: Fri, 08 Jun 2007 10:56:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- <-- SIP read from 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK70b10616;rport=5060 From: ;tag=as05bb784a To: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 103 BYE Contact: Date: Fri,08 Jun 2007 07:59:17 GMT Content-Length: 0 --- (9 headers 0 lines) --- Jun 8 13:56:34 DEBUG[19327]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '3459821811-3393166743091811-11-863266' of Request 103: Match Found Destroying call '3459821811-3393166743091811-11-863266' <-- SIP read from 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK70b10616;rport=5060 From: ;tag=as05bb784a To: ;tag=test_tag_0010355350 Call-ID: 3459821811-3393166743091811-11-863266 CSeq: 103 BYE Contact: Date: Fri,08 Jun 2007 07:59:17 GMT Content-Length: 0 --- (9 headers 0 lines) --- Jun 8 13:56:35 NOTICE[19327]: chan_sip.c:5473 sip_reregister: -- Re-registration for 37070070541@skambink.zebra.lt Jun 8 13:56:35 DEBUG[19327]: chan_sip.c:5638 transmit_register: Scheduled a registration timeout for skambink.zebra.lt id #25 Jun 8 13:56:35 DEBUG[19327]: chan_sip.c:5697 transmit_register: >>> Re-using Auth data for 37070070541@skambink.zebra.lt REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 212.59.4.20:5060: REGISTER sip:skambink.zebra.lt SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK113458b2;rport From: ;tag=as33cdfb75 To: Call-ID: 780d992e252ba388789aea7a414c1f7f@192.168.1.104 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="37070070541", realm="voip.telecom.lt", algorithm=MD5, uri="sip:skambink.zebra.lt", nonce="3222960750-a610ca217039dce7d64b1274830212c0", response="567b3f1fed5f5f7a7f9658143af04bb1", opaque="", qop=auth, cnonce="51480152", nc=00000002 Expires: 120 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK113458b2;rport=5060 From: ;tag=as33cdfb75 To: ;tag=aprqfbbc7i1-cccqse10000g6 Call-ID: 780d992e252ba388789aea7a414c1f7f@192.168.1.104 CSeq: 104 REGISTER Contact: ;expires=30 --- (7 headers 0 lines) --- Jun 8 13:56:35 DEBUG[19327]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '780d992e252ba388789aea7a414c1f7f@192.168.1.104' of Request 104: Match Found Jun 8 13:56:35 DEBUG[19327]: chan_sip.c:9957 handle_response_register: Registration successful Jun 8 13:56:35 DEBUG[19327]: chan_sip.c:9959 handle_response_register: Cancelling timeout 25 Scheduling destruction of call '780d992e252ba388789aea7a414c1f7f@192.168.1.104' in 32000 ms Jun 8 13:56:35 NOTICE[19327]: chan_sip.c:10009 handle_response_register: Outbound Registration: Expiry for skambink.zebra.lt is 30 sec (Scheduling reregistration in 23 s) Beginning asterisk shutdown.... Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0).