<-- SIP read from 212.59.4.20:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5061;received=82.135.246.157;branch=z9hG4bK7a26aa89;rport=5061 From: "370XXXXXXX" ;tag=as2c2153d5 To: ;tag=test_tag_0009975691 Call-ID: 748acb60474470b2281f7c550adae940@skambink.zebra.lt CSeq: 103 INVITE Contact: Accept-Language: en;q=0.0 Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Wed,30 May 2007 18:00:36 GMT Supported: timer Session-Expires: 1800;refresher=uas Content-Type: application/sdp Content-Length: 327 v=0 o=hiQ9200 4468220070430210112 1786249243 IN IP4 212.59.4.20 s=Phone Call via hiQ9200 SIPCA c=IN IP4 212.59.4.20 t=0 0 m=audio 15498 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 0 a=cpar: a=rtpmap:8 PCMA/8000 a=cpar: a=fmtp:8 vad=no a=cpar: a=rtpmap:0 PCMU/8000 a=sendrecv --- (14 headers 14 lines) --- Found RTP audio format 8 Peer audio RTP is at port 212.59.4.20:15498 Found description format PCMA Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) 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:867198101@212.59.4.20:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK55918215;rport From: "370XXXXXXX" ;tag=as2c2153d5 To: ;tag=test_tag_0009975691 Contact: Call-ID: 748acb60474470b2281f7c550adae940@skambink.zebra.lt CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/teo-09b48f18 answered SIP/602-09b439d8 We're at 192.168.1.104 port 14134 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.113:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK16582102572773723643;received=192.168.1.113 From: 602 ;tag=3126017097 To: 867198101 ;tag=as79333ef6 Call-ID: 57957819-58131525256@192.168.1.113 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 8628 8629 IN IP4 192.168.1.104 s=session c=IN IP4 192.168.1.104 t=0 0 m=audio 14134 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/602-09b439d8 and SIP/teo-09b48f18 <-- SIP read from 192.168.1.113:5060: ACK sip:867198101@192.168.1.104:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK2583825341197351216 From: 602 ;tag=3126017097 To: 867198101 ;tag=as79333ef6 Call-ID: 57957819-58131525256@192.168.1.113 CSeq: 2 ACK max-forwards: 70 user-agent: Voip Phone 1.0 Content-Length: 0 --- (9 headers 0 lines) --- May 30 20:59:02 NOTICE[8640]: chan_sip.c:5473 sip_reregister: -- Re-registration for 370XXXXXXX@skambink.zebra.lt REGISTER 12 headers, 0 lines REGISTER attempt 1 to 370XXXXXXX@skambink.zebra.lt Reliably Transmitting (no NAT) to 212.59.4.20:5060: REGISTER sip:skambink.zebra.lt SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK5f64266d;rport From: ;tag=as472cf00f To: Call-ID: 39c407d66e1ea91d551d267f2ced4b7f@127.0.0.1 CSeq: 103 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/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5061;received=82.135.246.157;branch=z9hG4bK5f64266d;rport=5061 From: ;tag=as472cf00f To: ;tag=aprqfbbc7i1-ficee530000e6 Call-ID: 39c407d66e1ea91d551d267f2ced4b7f@127.0.0.1 CSeq: 103 REGISTER Contact: ;expires=30 --- (7 headers 0 lines) --- Scheduling destruction of call '39c407d66e1ea91d551d267f2ced4b7f@127.0.0.1' in 32000 ms May 30 20:59:02 NOTICE[8640]: chan_sip.c:10009 handle_response_register: Outbound Registration: Expiry for skambink.zebra.lt is 30 sec (Scheduling reregistration in 23 s) Destroying call '1051226431-316095009@192.168.1.113' <-- SIP read from 192.168.1.113:5060: BYE sip:867198101@192.168.1.104:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK24979156341251630366 From: 602 ;tag=3126017097 To: 867198101 ;tag=as79333ef6 Call-ID: 57957819-58131525256@192.168.1.113 CSeq: 3 BYE max-forwards: 70 user-agent: Voip Phone 1.0 Content-Length: 0 --- (9 headers 0 lines) --- Sending to 192.168.1.113 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.1.113:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK24979156341251630366;received=192.168.1.113 From: 602 ;tag=3126017097 To: 867198101 ;tag=as79333ef6 Call-ID: 57957819-58131525256@192.168.1.113 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 --- Scheduling destruction of call '748acb60474470b2281f7c550adae940@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:867198101@212.59.4.20:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5061;branch=z9hG4bK524fdff0;rport From: "370XXXXXXX" ;tag=as2c2153d5 To: ;tag=test_tag_0009975691 Call-ID: 748acb60474470b2281f7c550adae940@skambink.zebra.lt CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="370XXXXXXX", realm="voip.telecom.lt", algorithm=MD5, uri="sip:867198101@212.59.4.20:5060", nonce="3215545596-a93477c41e3005250abbd7efc4222405", response="ce176fc726d6180339a188517950cd05", opaque="", qop=auth, cnonce="5f4ec035", nc=00000002 Content-Length: 0 --- == Spawn extension (default, 867198101, 2) exited non-zero on 'SIP/602-09b439d8' <-- SIP read from 212.59.4.20:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.104:5061;received=82.135.246.157;branch=z9hG4bK524fdff0;rport=5061 From: "370XXXXXXX" ;tag=as2c2153d5 To: ;tag=test_tag_0009975691 Call-ID: 748acb60474470b2281f7c550adae940@skambink.zebra.lt CSeq: 104 BYE User-Agent: Asterisk PBX Content-Length: 0 WWW-Authenticate: Digest realm="voip.telecom.lt", nonce="3215545822-13f65753e815b02fd80b4d3bff03d8c0", stale=true, algorithm=MD5, qop="auth" --- (9 headers 0 lines) --- May 30 20:59:06 WARNING[8640]: chan_sip.c:10163 handle_response: Got authentication request (401) on unknown BYE to ';tag=test_tag_0009975691' Destroying call '748acb60474470b2281f7c550adae940@skambink.zebra.lt' Destroying call '57957819-58131525256@192.168.1.113' Destroying call '39c407d66e1ea91d551d267f2ced4b7f@127.0.0.1' <-- SIP read from 212.59.4.20:5060: BYE sip:370XXXXXXX@192.168.1.104:5061 SIP/2.0 Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08nj5t10e0dgb1gul5g0.1 To: "370XXXXXXX" ;tag=as2c2153d5 From: ;tag=test_tag_0009975691 Call-ID: 748acb60474470b2281f7c550adae940@skambink.zebra.lt CSeq: 1 BYE Accept-Language: en;q=0.0 Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER Date: Wed,30 May 2007 18:00:51 GMT Max-Forwards: 68 Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (no NAT) to 212.59.4.20:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 212.59.4.20:5060;branch=z9hG4bK08nj5t10e0dgb1gul5g0.1;received=212.59.4.20 From: ;tag=test_tag_0009975691 To: "370XXXXXXX" ;tag=as2c2153d5 Call-ID: 748acb60474470b2281f7c550adae940@skambink.zebra.lt CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0