Asterisk Ready. *CLI> Jun 8 13:43:58 NOTICE[19296]: chan_sip.c:5473 sip_reregister: -- Re-registration for 37070070541@skambink.zebra.lt Jun 8 13:43:59 DEBUG[19296]: 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=z9hG4bK719303b2;rport From: ;tag=as0644a865 To: Call-ID: 37c8045f0a1d90830a9d291b7e88ab91@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=z9hG4bK719303b2;rport=5060 From: ;tag=as0644a865 To: ;tag=test_tag_0010353232 Call-ID: 37c8045f0a1d90830a9d291b7e88ab91@192.168.1.104 CSeq: 102 REGISTER User-Agent: Asterisk PBX Event: registration Content-Length: 0 WWW-Authenticate: Digest realm="voip.telecom.lt", nonce="3222953425-129767473364d13de3d63a9f96241681", algorithm=MD5, qop="auth" --- (10 headers 0 lines) --- Jun 8 13:43:59 DEBUG[19296]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '37c8045f0a1d90830a9d291b7e88ab91@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=z9hG4bK286790f3;rport From: ;tag=as4d366a1a To: Call-ID: 37c8045f0a1d90830a9d291b7e88ab91@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="3222953425-129767473364d13de3d63a9f96241681", response="79a42eaa7d773944c97120fb0aa2e9f1", opaque="", qop=auth, cnonce="2b43df99", 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=z9hG4bK286790f3;rport=5060 From: ;tag=as4d366a1a To: Call-ID: 37c8045f0a1d90830a9d291b7e88ab91@192.168.1.104 CSeq: 103 REGISTER Contact: ;expires=30;audio Content-Length: 0 --- (8 headers 0 lines) --- Jun 8 13:43:59 DEBUG[19296]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '37c8045f0a1d90830a9d291b7e88ab91@192.168.1.104' of Request 103: Match Found Jun 8 13:43:59 DEBUG[19296]: chan_sip.c:9957 handle_response_register: Registration successful Jun 8 13:43:59 DEBUG[19296]: chan_sip.c:9959 handle_response_register: Cancelling timeout 10 Scheduling destruction of call '37c8045f0a1d90830a9d291b7e88ab91@192.168.1.104' in 32000 ms Jun 8 13:43:59 NOTICE[19296]: 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: INVITE sip:117@192.168.1.104:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK2533899992141517645;rport From: 601 ;tag=15624288 To: 117 Call-ID: 520532611-346524340@192.168.1.53 CSeq: 1 INVITE Contact: max-forwards: 70 supported: 100rel user-agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 315 v=0 o=553 14450282 29058986 IN IP4 192.168.1.53 s=A conversation c=IN IP4 192.168.1.53 t=0 0 m=audio 10094 RTP/AVP 0 4 4 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 14 lines) --- Using INVITE request as basis request - 520532611-346524340@192.168.1.53 Sending to 192.168.1.53 : 5060 (NAT) Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:7304 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 192.168.1.53:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK2533899992141517645;received=192.168.1.53;rport=5060 From: 601 ;tag=15624288 To: 117 ;tag=as30160e8d Call-ID: 520532611-346524340@192.168.1.53 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="642ddbb0" Content-Length: 0 --- Scheduling destruction of call '520532611-346524340@192.168.1.53' in 15000 ms Found user '601' <-- SIP read from 192.168.1.53:5060: ACK sip:117@192.168.1.104:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK2533899992141517645;rport From: 601 ;tag=15624288 To: 117 ;tag=as30160e8d Call-ID: 520532611-346524340@192.168.1.53 CSeq: 1 ACK max-forwards: 70 Content-Length: 0 --- (8 headers 0 lines) --- Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '520532611-346524340@192.168.1.53' of Response 1: Match Found <-- SIP read from 192.168.1.53:5060: INVITE sip:117@192.168.1.104:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK69189958231996812;rport From: 601 ;tag=15624288 To: 117 Call-ID: 520532611-346524340@192.168.1.53 CSeq: 2 INVITE Contact: Proxy-Authorization: Digest username="601", realm="asterisk", nonce="642ddbb0", uri="sip:117@192.168.1.104:5060", response="e4f160b7111fbbcc7e7997c283817f39", algorithm=MD5 max-forwards: 70 supported: 100rel user-agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 315 v=0 o=553 14450282 29058986 IN IP4 192.168.1.53 s=A conversation c=IN IP4 192.168.1.53 t=0 0 m=audio 10094 RTP/AVP 0 4 4 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:4 G723high/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (14 headers 14 lines) --- Using INVITE request as basis request - 520532611-346524340@192.168.1.53 Sending to 192.168.1.53 : 5060 (NAT) Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:7304 check_user_full: Setting NAT on RTP to 0 Found user '601' Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.53:10094 Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:3687 process_sdp: Peer audio RTP is at port 192.168.1.53:10094 Found description format PCMU Found description format G723 Found description format G723high Found description format G729 Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing) Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:10695 handle_request_invite: Checking SIP call limits for device 601 Looking for 117 in default (domain 192.168.1.104) Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:6280 build_route: build_route: Contact hop: list_route: hop: 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=z9hG4bK69189958231996812;received=192.168.1.53;rport=5060 From: 601 ;tag=15624288 To: 117 Call-ID: 520532611-346524340@192.168.1.53 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/601-081992b0", "SIP/teo/117") in new stack Jun 8 13:44:03 DEBUG[19303]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 524288 Jun 8 13:44:03 DEBUG[19303]: chan_sip.c:2083 sip_call: Outgoing Call for 117 We're at 192.168.1.104 port 11174 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP 13 headers, 9 lines Reliably Transmitting (NAT) to 212.59.4.20:5060: INVITE sip:117@skambink.zebra.lt SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK1ee4bc5a;rport From: "601" ;tag=as5e638574 To: Contact: Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 08 Jun 2007 10:44:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 186 v=0 o=root 19286 19286 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 11174 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Called teo/117 <-- SIP read from 212.59.4.20:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK1ee4bc5a;rport=5060 From: "601" ;tag=as5e638574 To: Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 102 INVITE --- (6 headers 0 lines) --- Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4e781491226dd403274a086042ac3745@skambink.zebra.lt' Request 102: Found <-- 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=z9hG4bK1ee4bc5a;rport=5060 From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353247 Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 08 Jun 2007 10:44:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 WWW-Authenticate: Digest realm="voip.telecom.lt", nonce="3222953472-2c385bc18e93c3202e30afeda0d349e1", algorithm=MD5, qop="auth" --- (11 headers 0 lines) --- Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '4e781491226dd403274a086042ac3745@skambink.zebra.lt' of Request 102: Match Found Transmitting (NAT) to 212.59.4.20:5060: ACK sip:117@skambink.zebra.lt SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK1ee4bc5a;rport From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353247 Contact: Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 192.168.1.104 port 11174 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (NAT) to 212.59.4.20:5060: INVITE sip:117@skambink.zebra.lt SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK2a78d392;rport From: "601" ;tag=as5e638574 To: Contact: Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="37070070541", realm="voip.telecom.lt", algorithm=MD5, uri="sip:117@skambink.zebra.lt", nonce="3222953472-2c385bc18e93c3202e30afeda0d349e1", response="ed19927ee36e8da943da7757d2c554bd", opaque="", qop=auth, cnonce="69e5ce98", nc=00000001 Date: Fri, 08 Jun 2007 10:44:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 186 v=0 o=root 19286 19287 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 11174 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- <-- SIP read from 212.59.4.20:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK2a78d392;rport=5060 From: "601" ;tag=as5e638574 To: Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 103 INVITE --- (6 headers 0 lines) --- Jun 8 13:44:03 DEBUG[19296]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4e781491226dd403274a086042ac3745@skambink.zebra.lt' Request 103: Found <-- SIP read from 212.59.4.20:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK2a78d392;rport=5060 From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353248 Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 103 INVITE Contact: Date: Fri,08 Jun 2007 07:46:46 GMT Content-Length: 0 --- (9 headers 0 lines) --- Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4e781491226dd403274a086042ac3745@skambink.zebra.lt' Request 103: Found <-- SIP read from 212.59.4.20:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK2a78d392;rport=5060 From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353248 Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 103 INVITE Contact: Date: Fri,08 Jun 2007 07:46:46 GMT Content-Type: application/sdp Content-Length: 310 v=0 o=hiQ9200 1792520070508104644 1785724987 IN IP4 212.59.4.20 s=Phone Call via hiQ9200 SIPCA c=IN IP4 212.59.4.20 t=0 0 m=audio 15342 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 0 8 a=cpar: a=rtpmap:0 PCMU/8000 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=fmtp:8 vad=no a=sendrecv --- (10 headers 13 lines) --- Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4e781491226dd403274a086042ac3745@skambink.zebra.lt' Request 103: Found Found RTP audio format 0 Peer audio RTP is at port 212.59.4.20:15342 Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:3687 process_sdp: Peer audio RTP is at port 212.59.4.20:15342 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) -- SIP/teo-0819f8c8 is making progress passing it to SIP/601-081992b0 We're at 192.168.1.104 port 16494 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Transmitting (no NAT) to 192.168.1.53:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK69189958231996812;received=192.168.1.53;rport=5060 From: 601 ;tag=15624288 To: 117 ;tag=as2095b85f Call-ID: 520532611-346524340@192.168.1.53 CSeq: 2 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 19286 19286 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 16494 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- Jun 8 13:44:04 NOTICE[19303]: 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 <-- SIP read from 212.59.4.20:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK2a78d392;rport=5060 From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353248 Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 103 INVITE Contact: Date: Fri,08 Jun 2007 07:46:47 GMT Content-Type: application/sdp Content-Length: 310 v=0 o=hiQ9200 1792520070508104644 1785724987 IN IP4 212.59.4.20 s=Phone Call via hiQ9200 SIPCA c=IN IP4 212.59.4.20 t=0 0 m=audio 15342 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 0 8 a=cpar: a=rtpmap:0 PCMU/8000 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=fmtp:8 vad=no a=sendrecv --- (10 headers 13 lines) --- Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4e781491226dd403274a086042ac3745@skambink.zebra.lt' Request 103: Found Found RTP audio format 0 Peer audio RTP is at port 212.59.4.20:15342 Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:3687 process_sdp: Peer audio RTP is at port 212.59.4.20:15342 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) -- SIP/teo-0819f8c8 is making progress passing it to SIP/601-081992b0 <-- SIP read from 212.59.4.20:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.104:5060;received=82.135.246.157;branch=z9hG4bK2a78d392;rport=5060 From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353248 Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 103 INVITE Contact: Date: Fri,08 Jun 2007 07:46:47 GMT Content-Type: application/sdp Content-Length: 310 v=0 o=hiQ9200 1792520070508104644 1785724987 IN IP4 212.59.4.20 s=Phone Call via hiQ9200 SIPCA c=IN IP4 212.59.4.20 t=0 0 m=audio 15342 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 0 8 a=cpar: a=rtpmap:0 PCMU/8000 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=fmtp:8 vad=no a=sendrecv --- (10 headers 13 lines) --- Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4e781491226dd403274a086042ac3745@skambink.zebra.lt' Request 103: Found Found RTP audio format 0 Peer audio RTP is at port 212.59.4.20:15342 Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:3687 process_sdp: Peer audio RTP is at port 212.59.4.20:15342 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) -- SIP/teo-0819f8c8 is ringing Transmitting (no NAT) to 192.168.1.53:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK69189958231996812;received=192.168.1.53;rport=5060 From: 601 ;tag=15624288 To: 117 ;tag=as2095b85f Call-ID: 520532611-346524340@192.168.1.53 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- SIP/teo-0819f8c8 is making progress passing it to SIP/601-081992b0 <-- 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=z9hG4bK2a78d392;rport=5060 From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353248 Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 103 INVITE Contact: Accept-Language: en;q=0.0 Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Fri,08 Jun 2007 07:46:47 GMT Supported: timer Session-Expires: 1800;refresher=uas Content-Type: application/sdp Content-Length: 310 v=0 o=hiQ9200 1792520070508104644 1785724987 IN IP4 212.59.4.20 s=Phone Call via hiQ9200 SIPCA c=IN IP4 212.59.4.20 t=0 0 m=audio 15342 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 0 8 a=cpar: a=rtpmap:0 PCMU/8000 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=fmtp:8 vad=no a=sendrecv --- (14 headers 13 lines) --- Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:1392 __sip_ack: Acked pending invite 103 Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '4e781491226dd403274a086042ac3745@skambink.zebra.lt' of Request 103: Match Found Found RTP audio format 0 Peer audio RTP is at port 212.59.4.20:15342 Jun 8 13:44:04 DEBUG[19296]: chan_sip.c:3687 process_sdp: Peer audio RTP is at port 212.59.4.20:15342 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) Jun 8 13:44:04 DEBUG[19296]: 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 212.59.4.20, port 5060 Transmitting (NAT) to 212.59.4.20:5060: ACK sip:117@212.59.4.20:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK5952d80b;rport From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353248 Contact: Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/teo-0819f8c8 answered SIP/601-081992b0 We're at 192.168.1.104 port 16494 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably 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=z9hG4bK69189958231996812;received=192.168.1.53;rport=5060 From: 601 ;tag=15624288 To: 117 ;tag=as2095b85f Call-ID: 520532611-346524340@192.168.1.53 CSeq: 2 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 19286 19287 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 16494 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/601-081992b0 and SIP/teo-0819f8c8 <-- SIP read from 192.168.1.53:5060: ACK sip:117@192.168.1.104 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK9119 From: 601 ;tag=15624288 To: 117 ;tag=as2095b85f Call-ID: 520532611-346524340@192.168.1.53 CSeq: 2 ACK max-forwards: 70 user-agent: Voip Phone 1.0 Content-Length: 0 --- (9 headers 0 lines) --- Jun 8 13:44:05 DEBUG[19296]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '520532611-346524340@192.168.1.53' of Response 2: Match Found <-- 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=z9hG4bK93661387143965066;rport From: 601 ;tag=3210115622 To: 601 Call-ID: 2965513143-1545317990@192.168.1.53 CSeq: 7 REGISTER Contact: Authorization: Digest username="601", realm="asterisk", nonce="00d12e6e", uri="sip:192.168.1.104:5060", response="3577fe39da1500cc527f2b06824f5fe2", 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=z9hG4bK93661387143965066;received=192.168.1.53;rport=5060 From: 601 ;tag=3210115622 To: 601 Call-ID: 2965513143-1545317990@192.168.1.53 CSeq: 7 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=z9hG4bK93661387143965066;received=192.168.1.53;rport=5060 From: 601 ;tag=3210115622 To: 601 ;tag=as7ba4ab03 Call-ID: 2965513143-1545317990@192.168.1.53 CSeq: 7 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="36df6cab" Content-Length: 0 --- Scheduling destruction of call '2965513143-1545317990@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=z9hG4bK1197235461947813336;rport From: 601 ;tag=3210115622 To: 601 Call-ID: 2965513143-1545317990@192.168.1.53 CSeq: 8 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=z9hG4bK1197235461947813336;received=192.168.1.53;rport=5060 From: 601 ;tag=3210115622 To: 601 Call-ID: 2965513143-1545317990@192.168.1.53 CSeq: 8 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=z9hG4bK1197235461947813336;received=192.168.1.53;rport=5060 From: 601 ;tag=3210115622 To: 601 ;tag=as7ba4ab03 Call-ID: 2965513143-1545317990@192.168.1.53 CSeq: 8 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 60 Contact: ;expires=60 Date: Fri, 08 Jun 2007 10:44:06 GMT Content-Length: 0 --- Scheduling destruction of call '2965513143-1545317990@192.168.1.53' in 15000 ms Jun 8 13:44:08 NOTICE[19303]: 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 <-- SIP read from 192.168.1.53:5060: BYE sip:117@192.168.1.104 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.53:5060;branch=z9hG4bK841778651570415181;rport From: 601 ;tag=15624288 To: 117 ;tag=as2095b85f Call-ID: 520532611-346524340@192.168.1.53 CSeq: 3 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=z9hG4bK841778651570415181;received=192.168.1.53;rport=5060 From: 601 ;tag=15624288 To: 117 ;tag=as2095b85f Call-ID: 520532611-346524340@192.168.1.53 CSeq: 3 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 13:44:08 DEBUG[19303]: channel.c:3375 ast_generic_bridge: Didn't get a frame from channel: SIP/601-081992b0 Jun 8 13:44:08 DEBUG[19303]: channel.c:3664 ast_channel_bridge: Bridge stops bridging channels SIP/601-081992b0 and SIP/teo-0819f8c8 Jun 8 13:44:08 DEBUG[19303]: chan_sip.c:2448 sip_hangup: update_call_counter(117) - decrement call limit counter Scheduling destruction of call '4e781491226dd403274a086042ac3745@skambink.zebra.lt' 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:117@212.59.4.20:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK0d07eb89;rport From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353248 Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="37070070541", realm="voip.telecom.lt", algorithm=MD5, uri="sip:117@212.59.4.20:5060", nonce="3222953472-2c385bc18e93c3202e30afeda0d349e1", response="c143f907591c361c3426695dcb888cf7", opaque="", qop=auth, cnonce="065e59c9", nc=00000002 Content-Length: 0 --- Jun 8 13:44:08 DEBUG[19303]: app_dial.c:1661 dial_exec_full: Exiting with DIALSTATUS=ANSWER. == Spawn extension (default, 117, 1) exited non-zero on 'SIP/601-081992b0' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '"601" <601>' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '601' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '117' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'default' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/601-081992b0' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/teo-0819f8c8' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'Dial' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'SIP/teo/117' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-06-08 13:44:03' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-06-08 13:44:04' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '2007-06-08 13:44:08' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '5' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '4' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '1181299443.0' Jun 8 13:44:08 DEBUG[19303]: pbx.c:1542 pbx_substitute_variables_helper_full: Function result is '(null)' Jun 8 13:44:08 DEBUG[19303]: chan_sip.c:2448 sip_hangup: update_call_counter(601) - decrement call limit counter <-- 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=z9hG4bK0d07eb89;rport=5060 From: "601" ;tag=as5e638574 To: ;tag=test_tag_0010353248 Call-ID: 4e781491226dd403274a086042ac3745@skambink.zebra.lt CSeq: 104 BYE User-Agent: Asterisk PBX Content-Length: 0 WWW-Authenticate: Digest realm="voip.telecom.lt", nonce="3222953517-d289245bb07a236bd97663afa326b398", stale=true, algorithm=MD5, qop="auth" --- (9 headers 0 lines) --- Jun 8 13:44:08 DEBUG[19296]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '4e781491226dd403274a086042ac3745@skambink.zebra.lt' of Request 104: Match Found Jun 8 13:44:08 WARNING[19296]: chan_sip.c:10163 handle_response: Got authentication request (401) on unknown BYE to ';tag=test_tag_0010353248' Destroying call '4e781491226dd403274a086042ac3745@skambink.zebra.lt' Destroying call '520532611-346524340@192.168.1.53' Beginning asterisk shutdown.... Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0). drakonas:~#