12 headers, 0 lines Reliably Transmitting (NAT) to provider.sip.public.ip:5061: OPTIONS sip:provider.sip.public.ip:5061 SIP/2.0 Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK6d140a1f;rport From: "openpbx" ;tag=as3bcb6745 To: Contact: Call-ID: 06c255302379240f3e15459c053801b3@my.openpbx.hostname CSeq: 102 OPTIONS User-Agent: OpenPBX Max-Forwards: 70 Date: Tue, 10 Jul 2007 14:34:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- timur*CLI> <-- SIP read from provider.sip.public.ip:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK6d140a1f From: "openpbx" ;tag=as3bcb6745 To: ;tag=866787031-1184078055031 Call-ID: 06c255302379240f3e15459c053801b3@my.openpbx.hostname CSeq: 102 OPTIONS Supported: 100rel Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, NOTIFY, UPDATE Accept: application/sdp, application/broadsoft, text/plain Content-Length: 0 --- (10 headers 0 lines) --- Destroying call '06c255302379240f3e15459c053801b3@my.openpbx.hostname' 12 headers, 0 lines Reliably Transmitting (NAT) to coleague.public.ip:5060: OPTIONS sip:coleaguelogin@coleague.private.ip:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK41c7eb63;rport From: "openpbx" ;tag=as2ed3ed26 To: Contact: Call-ID: 29c8b3890b153673330f02076948d46d@my.openpbx.hostname CSeq: 102 OPTIONS User-Agent: OpenPBX Max-Forwards: 70 Date: Tue, 10 Jul 2007 14:34:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- timur*CLI> <-- SIP read from coleague.public.ip:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK41c7eb63;rport From: "openpbx" ;tag=as2ed3ed26 To: ;tag=83bb89632e00c8c2 Call-ID: 29c8b3890b153673330f02076948d46d@my.openpbx.hostname CSeq: 102 OPTIONS User-Agent: Grandstream BT200 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer Content-Length: 0 --- (11 headers 0 lines) --- Destroying call '29c8b3890b153673330f02076948d46d@my.openpbx.hostname' timur*CLI> <-- SIP read from my.sip.client.public.ip:5060: INVITE sip:MyMobilePhoneNumber@my.openpbx.hostname;user=phone SIP/2.0 v: SIP/2.0/UDP my.sip.client.public.ip;rport;branch=z9hG4bKzvynbzra Max-Forwards: 70 t: f: "Marek Greško" ;tag=cgaot i: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 352 INVITE m: c: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO k: replaces,norefersub,100rel User-Agent: Twinkle/1.0 l: 307 timur*CLI> v=0 o=mysiplogin 1393573505 1208221949 IN IP4 my.sip.client.public.ip s=- c=IN IP4 my.sip.client.public.ip t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --- (13 headers 14 lines) --- Using INVITE request as basis request - ixweqbnwvfkuiuo@my.sip.client.public.ip Sending to my.sip.client.public.ip : 5060 (NAT) Reliably Transmitting (NAT) to my.sip.client.public.ip:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP my.sip.client.public.ip;rport;branch=z9hG4bKzvynbzra;received=my.sip.client.public.ip From: "Marek Greško" ;tag=cgaot To: ;tag=as4c2cac79 Call-ID: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 352 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 WWW-Authenticate: Digest algorithm=MD5, realm="my.openpbx.hostname", nonce="265f3d03" Content-Length: 0 --- Scheduling destruction of call 'ixweqbnwvfkuiuo@my.sip.client.public.ip' in 15000 ms Found user 'mysiplogin' timur*CLI> <-- SIP read from my.sip.client.public.ip:5060: ACK sip:MyMobilePhoneNumber@my.openpbx.hostname;user=phone SIP/2.0 v: SIP/2.0/UDP my.sip.client.public.ip;rport;branch=z9hG4bKzvynbzra Max-Forwards: 70 t: ;tag=as4c2cac79 f: "Marek Greško" ;tag=cgaot i: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 352 ACK User-Agent: Twinkle/1.0 l: 0 --- (9 headers 0 lines) --- timur*CLI> <-- SIP read from my.sip.client.public.ip:5060: INVITE sip:MyMobilePhoneNumber@my.openpbx.hostname;user=phone SIP/2.0 v: SIP/2.0/UDP my.sip.client.public.ip;rport;branch=z9hG4bKrizmrtav Max-Forwards: 70 t: f: "Marek Greško" ;tag=cgaot i: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 353 INVITE m: c: application/sdp Authorization: Digest username="mysiplogin",realm="my.openpbx.hostname",nonce="265f3d03",uri="sip:MyMobilePhoneNumber@my.openpbx.hostname;user=phone",response="4bb6b9fb90cff1a4b78ffc89d60ad329",algorithm=MD5 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO k: replaces,norefersub,100rel User-Agent: Twinkle/1.0 l: 307 v=0 o=mysiplogin 1393573505 1208221949 IN IP4 my.sip.client.public.ip s=- c=IN IP4 my.sip.client.public.ip t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --- (14 headers 14 lines) --- Using INVITE request as basis request - ixweqbnwvfkuiuo@my.sip.client.public.ip Sending to my.sip.client.public.ip : 5060 (NAT) Found user 'mysiplogin' Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port my.sip.client.public.ip:8000 Found description format speex Found description format speex Found description format PCMA Found description format PCMU Found description format GSM Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for MyMobilePhoneNumber in internal (domain my.openpbx.hostname;user=phone) list_route: hop: Transmitting (NAT) to my.sip.client.public.ip:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP my.sip.client.public.ip;rport;branch=z9hG4bKrizmrtav;received=my.sip.client.public.ip From: "Marek Greško" ;tag=cgaot To: Call-ID: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 353 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- We're at my.openpbx.public.ip port 12056 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP 13 headers, 9 lines Reliably Transmitting (NAT) to provider.sip.public.ip:5061: INVITE sip:MyMobilePhoneNumber@provider.sip.public.ip:5061 SIP/2.0 Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK0b57d129;rport From: "Marek" ;tag=as45e75421 To: Contact: Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 102 INVITE User-Agent: OpenPBX Max-Forwards: 70 Date: Tue, 10 Jul 2007 14:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 190 v=0 o=root 17105 17105 IN IP4 my.openpbx.public.ip s=session c=IN IP4 my.openpbx.public.ip t=0 0 m=audio 12056 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- timur*CLI> <-- SIP read from provider.sip.public.ip:5061: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK0b57d129 From: "Marek" ;tag=as45e75421 To: Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 102 INVITE WWW-Authenticate: DIGEST realm="BroadWorks", nonce="1184078059466", algorithm=MD5 Content-Length: 0 --- (8 headers 0 lines) --- Transmitting (NAT) to provider.sip.public.ip:5061: ACK sip:MyMobilePhoneNumber@provider.sip.public.ip:5061 SIP/2.0 Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK0b57d129;rport From: "Marek" ;tag=as45e75421 To: Contact: Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 102 ACK User-Agent: OpenPBX Max-Forwards: 70 Content-Length: 0 --- We're at my.openpbx.public.ip port 12056 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (NAT) to provider.sip.public.ip:5061: INVITE sip:MyMobilePhoneNumber@provider.sip.public.ip:5061 SIP/2.0 Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK12246c66;rport From: "Marek" ;tag=as45e75421 To: Contact: Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 103 INVITE User-Agent: OpenPBX Max-Forwards: 70 Authorization: Digest username="mysiplogin", realm="BroadWorks", algorithm=MD5, uri="sip:MyMobilePhoneNumber@provider.sip.public.ip:5061", nonce="1184078059466", response="1de05d1acbff1298020b162ede08c645", opaque="" Date: Tue, 10 Jul 2007 14:34:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 190 v=0 o=root 17105 17106 IN IP4 my.openpbx.public.ip s=session c=IN IP4 my.openpbx.public.ip t=0 0 m=audio 12056 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- timur*CLI> <-- SIP read from provider.sip.public.ip:5061: SIP/2.0 100 Trying Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK12246c66 From: "Marek" ;tag=as45e75421 To: Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 103 INVITE Content-Length: 0 --- (7 headers 0 lines) --- timur*CLI> <-- SIP read from provider.sip.public.ip:5061: SIP/2.0 180 Ringing Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK12246c66 From: "Marek" ;tag=as45e75421 To: ;tag=921204486-1184078064138 Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 103 INVITE Contact: Supported: 100rel Remote-party-id: ;screen=yes;party=called;privacy=off;id-type=subscriber Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, UPDATE, NOTIFY Content-Type: application/sdp Content-Length: 00165 v=0 o=BroadWorks 37437305 1 IN IP4 195.146.137.187 s=- c=IN IP4 195.146.137.187 t=0 0 m=audio 40032 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=maxptime:20 --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 195.146.137.187:40032 Found description format PCMU Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Transmitting (NAT) to my.sip.client.public.ip:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP my.sip.client.public.ip;rport;branch=z9hG4bKrizmrtav;received=my.sip.client.public.ip From: "Marek Greško" ;tag=cgaot To: ;tag=as240568cf Call-ID: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 353 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- We're at my.openpbx.public.ip port 16040 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to my.sip.client.public.ip:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP my.sip.client.public.ip;rport;branch=z9hG4bKrizmrtav;received=my.sip.client.public.ip From: "Marek Greško" ;tag=cgaot To: ;tag=as240568cf Call-ID: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 353 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 269 v=0 o=root 17105 17105 IN IP4 my.openpbx.public.ip s=session c=IN IP4 my.openpbx.public.ip t=0 0 m=audio 16040 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- timur*CLI> timur*CLI> ****** I hung up my sip phone HERE ****** timur*CLI> timur*CLI> timur*CLI> <-- SIP read from my.sip.client.public.ip:5060: CANCEL sip:MyMobilePhoneNumber@my.openpbx.hostname;user=phone SIP/2.0 v: SIP/2.0/UDP my.sip.client.public.ip;rport;branch=z9hG4bKrizmrtav Max-Forwards: 70 t: f: "Marek Greško" ;tag=cgaot i: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 353 CANCEL User-Agent: Twinkle/1.0 l: 0 --- (9 headers 0 lines) --- Sending to my.sip.client.public.ip : 5060 (NAT) Reliably Transmitting (NAT) to my.sip.client.public.ip:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP my.sip.client.public.ip;branch=z9hG4bKrizmrtav;received=my.sip.client.public.ip;rport=5060 From: "Marek Greško" ;tag=cgaot To: ;tag=as240568cf Call-ID: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 353 INVITE User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Content-Length: 0 --- Transmitting (NAT) to my.sip.client.public.ip:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP my.sip.client.public.ip;branch=z9hG4bKrizmrtav;received=my.sip.client.public.ip;rport=5060 From: "Marek Greško" ;tag=cgaot To: ;tag=as240568cf Call-ID: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 353 CANCEL User-Agent: OpenPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- Scheduling destruction of call '7afd14137e2730980acfbf52747ef46a@provider.sip.hostname' in 32000 ms Reliably Transmitting (NAT) to provider.sip.public.ip:5061: CANCEL sip:MyMobilePhoneNumber@provider.sip.public.ip:5061 SIP/2.0 Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK12246c66;rport From: "Marek" ;tag=as45e75421 To: Contact: Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 103 CANCEL User-Agent: OpenPBX Max-Forwards: 70 Authorization: Digest username="mysiplogin", realm="BroadWorks", algorithm=MD5, uri="sip:MyMobilePhoneNumber@provider.sip.public.ip:5061", nonce="1184078059466", response="c8775dfd04278f1dd34917d178d474d4", opaque="" Content-Length: 0 --- timur*CLI> <-- SIP read from my.sip.client.public.ip:5060: ACK sip:MyMobilePhoneNumber@my.openpbx.hostname;user=phone SIP/2.0 v: SIP/2.0/UDP my.sip.client.public.ip;rport;branch=z9hG4bKrizmrtav Max-Forwards: 70 t: ;tag=as240568cf f: "Marek Greško" ;tag=cgaot i: ixweqbnwvfkuiuo@my.sip.client.public.ip CSeq: 353 ACK Authorization: Digest username="mysiplogin",realm="my.openpbx.hostname",nonce="265f3d03",uri="sip:MyMobilePhoneNumber@my.openpbx.hostname;user=phone",response="4bb6b9fb90cff1a4b78ffc89d60ad329",algorithm=MD5 User-Agent: Twinkle/1.0 l: 0 --- (10 headers 0 lines) --- Destroying call 'ixweqbnwvfkuiuo@my.sip.client.public.ip' timur*CLI> <-- SIP read from provider.sip.public.ip:5061: SIP/2.0 487 Request terminated Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK12246c66 From: "Marek" ;tag=as45e75421 To: ;tag=921204486-1184078064138 Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 103 INVITE Content-Length: 0 --- (7 headers 0 lines) --- Transmitting (NAT) to provider.sip.public.ip:5061: ACK sip:MyMobilePhoneNumber@provider.sip.public.ip:5061 SIP/2.0 Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK12246c66;rport From: "Marek" ;tag=as45e75421 To: ;tag=921204486-1184078064138 Contact: Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 103 ACK User-Agent: OpenPBX Max-Forwards: 70 Content-Length: 0 --- timur*CLI> <-- SIP read from provider.sip.public.ip:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP my.openpbx.public.ip:5060;branch=z9hG4bK12246c66 From: "Marek" ;tag=as45e75421 To: ;tag=921204486-1184078064138 Call-ID: 7afd14137e2730980acfbf52747ef46a@provider.sip.hostname CSeq: 103 CANCEL Content-Length: 0 --- (7 headers 0 lines) ---