<--- SIP read from my.public.ip:5060 ---> INVITE sip:mymobilephonenumber@domain.sk;user=phone SIP/2.0 v: SIP/2.0/UDP my.public.ip;rport;branch=z9hG4bKawlnvpoi Max-Forwards: 70 t: f: "Me" ;tag=pxyuj i: byvmonogrnxlzlq@my.public.ip CSeq: 346 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: 306 v=0 o=client 2073724128 933010144 IN IP4 my.public.ip s=- c=IN IP4 my.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) --- Sending to my.public.ip : 5060 (NAT) Using INVITE request as basis request - byvmonogrnxlzlq@my.public.ip <--- Reliably Transmitting (no NAT) to my.public.ip:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP my.public.ip;branch=z9hG4bKawlnvpoi;received=my.public.ip;rport=5060 From: "Me" ;tag=pxyuj To: ;tag=as4b4e6f4b Call-ID: byvmonogrnxlzlq@my.public.ip CSeq: 346 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="domain.sk", nonce="4d046b33" Content-Length: 0 pbx*CLI> <------------> Scheduling destruction of SIP dialog 'byvmonogrnxlzlq@my.public.ip' in 32000 ms (Method: INVITE) Found user 'client' pbx*CLI> <--- SIP read from my.public.ip:5060 ---> ACK sip:mymobilephonenumber@domain.sk;user=phone SIP/2.0 v: SIP/2.0/UDP my.public.ip;rport;branch=z9hG4bKawlnvpoi Max-Forwards: 70 t: ;tag=as4b4e6f4b f: "Me" ;tag=pxyuj i: byvmonogrnxlzlq@my.public.ip CSeq: 346 ACK User-Agent: Twinkle/1.0 l: 0 <-------------> --- (9 headers 0 lines) --- pbx*CLI> <--- SIP read from my.public.ip:5060 ---> INVITE sip:mymobilephonenumber@domain.sk;user=phone SIP/2.0 v: SIP/2.0/UDP my.public.ip;rport;branch=z9hG4bKzaxkckcr Max-Forwards: 70 Proxy-Authorization: Digest username="client",realm="domain.sk",nonce="4d046b33",uri="sip:mymobilephonenumber@domain.sk;user=phone",response="8108c466c0f01ac02c801a442111681e",algorithm=MD5 t: f: "Me" ;tag=pxyuj i: byvmonogrnxlzlq@my.public.ip CSeq: 347 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: 306 v=0 o=client 2073724128 933010144 IN IP4 my.public.ip s=- c=IN IP4 my.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) --- Sending to my.public.ip : 5060 (NAT) Using INVITE request as basis request - byvmonogrnxlzlq@my.public.ip Found user 'client' 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.public.ip:8000 Found description format speex for ID 98 Found description format speex for ID 97 Found description format PCMA for ID 8 Found description format PCMU for ID 0 Found description format GSM for ID 3 Found description format telephone-event for ID 101 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 (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port my.public.ip:8000 Looking for mymobilephonenumber in internal (domain domain.sk) list_route: hop: <--- Transmitting (no NAT) to my.public.ip:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP my.public.ip;branch=z9hG4bKzaxkckcr;received=my.public.ip;rport=5060 From: "Me" ;tag=pxyuj To: Call-ID: byvmonogrnxlzlq@my.public.ip CSeq: 347 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [mymobilephonenumber@internal:1] Dial("SIP/client-009c65e0", "SIP/mymobilephonenumber@provider-provider") in new stack Audio is at pbx.public.ip port 16142 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to providerip1:5061: INVITE sip:mymobilephonenumber@providerip1:5061 SIP/2.0 Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK08b6b6c2;rport From: "Me" ;tag=as0a4685cd To: Contact: Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 23 Apr 2007 08:54:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 212 v=0 o=root 3150 3150 IN IP4 pbx.public.ip s=session c=IN IP4 pbx.public.ip t=0 0 m=audio 16142 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called mymobilephonenumber@provider-provider pbx*CLI> <--- SIP read from providerip1:5061 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK08b6b6c2 From: "Me" ;tag=as0a4685cd To: Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 102 INVITE WWW-Authenticate: DIGEST realm="BroadWorks", nonce="1177318442126", algorithm=MD5 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to providerip1:5061: ACK sip:mymobilephonenumber@providerip1:5061 SIP/2.0 Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK08b6b6c2;rport From: "Me" ;tag=as0a4685cd To: Contact: Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at pbx.public.ip port 16142 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to providerip1:5061: INVITE sip:mymobilephonenumber@providerip1:5061 SIP/2.0 Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK3b1ff5b6;rport From: "Me" ;tag=as0a4685cd To: Contact: Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="mylogin", realm="BroadWorks", algorithm=MD5, uri="sip:mymobilephonenumber@providerip1:5061", nonce="1177318442126", response="b6a15cddfbbbc28079815f9b23b7223f", opaque="" Date: Mon, 23 Apr 2007 08:54:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 212 v=0 o=root 3150 3151 IN IP4 pbx.public.ip s=session c=IN IP4 pbx.public.ip t=0 0 m=audio 16142 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- pbx*CLI> <--- SIP read from providerip1:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK3b1ff5b6 From: "Me" ;tag=as0a4685cd To: Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- pbx*CLI> <--- SIP read from providerip1:5061 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK3b1ff5b6 From: "Me" ;tag=as0a4685cd To: ;tag=483288967-1177318448422 Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 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: 00164 v=0 o=BroadWorks 6157210 1 IN IP4 providerip2 s=- c=IN IP4 providerip2 t=0 0 m=audio 28038 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 providerip2:28038 Found description format PCMU for ID 0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port providerip2:28038 -- SIP/provider-provider-009c33d0 is ringing <--- Transmitting (no NAT) to my.public.ip:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP my.public.ip;branch=z9hG4bKzaxkckcr;received=my.public.ip;rport=5060 From: "Me" ;tag=pxyuj To: ;tag=as393e5190 Call-ID: byvmonogrnxlzlq@my.public.ip CSeq: 347 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Call on SIP/provider-provider-009c33d0 left from hold -- SIP/provider-provider-009c33d0 is making progress passing it to SIP/client-009c65e0 Audio is at pbx.public.ip port 18350 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 (no NAT) to my.public.ip:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP my.public.ip;branch=z9hG4bKzaxkckcr;received=my.public.ip;rport=5060 From: "Me" ;tag=pxyuj To: ;tag=as393e5190 Call-ID: byvmonogrnxlzlq@my.public.ip CSeq: 347 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 291 v=0 o=root 3150 3150 IN IP4 pbx.public.ip s=session c=IN IP4 pbx.public.ip t=0 0 =audio 18350 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 - - - - a=ptime:20 a=sendrecv <------------> pbx*CLI> <--- SIP read from my.public.ip:5060 ---> CANCEL sip:mymobilephonenumber@domain.sk;user=phone SIP/2.0 v: SIP/2.0/UDP my.public.ip;rport;branch=z9hG4bKzaxkckcr Max-Forwards: 70 t: f: "Me" ;tag=pxyuj i: byvmonogrnxlzlq@my.public.ip CSeq: 347 CANCEL User-Agent: Twinkle/1.0 l: 0 <-------------> --- (9 headers 0 lines) --- Sending to my.public.ip : 5060 (NAT) <--- Reliably Transmitting (NAT) to my.public.ip:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP my.public.ip;branch=z9hG4bKzaxkckcr;received=my.public.ip;rport=5060 From: "Me" ;tag=pxyuj To: ;tag=as393e5190 Call-ID: byvmonogrnxlzlq@my.public.ip CSeq: 347 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (NAT) to my.public.ip:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP my.public.ip;branch=z9hG4bKzaxkckcr;received=my.public.ip;rport=5060 From: "Me" ;tag=pxyuj To: ;tag=as393e5190 Call-ID: byvmonogrnxlzlq@my.public.ip CSeq: 347 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1' in 6400 ms (Method: INVITE) Reliably Transmitting (no NAT) to providerip1:5061: CANCEL sip:mymobilephonenumber@providerip1:5061 SIP/2.0 Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK3b1ff5b6;rport From: "Me" ;tag=as0a4685cd To: Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 103 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1' in 6400 ms (Method: INVITE) == Spawn extension (internal, mymobilephonenumber, 1) exited non-zero on 'SIP/client-009c65e0' pbx*CLI> <--- SIP read from my.public.ip:5060 ---> ACK sip:mymobilephonenumber@domain.sk;user=phone SIP/2.0 v: SIP/2.0/UDP my.public.ip;rport;branch=z9hG4bKzaxkckcr Max-Forwards: 70 Proxy-Authorization: Digest username="client",realm="domain.sk",nonce="4d046b33",uri="sip:mymobilephonenumber@domain.sk;user=phone",response="8108c466c0f01ac02c801a442111681e",algorithm=MD5 t: ;tag=as393e5190 f: "Me" ;tag=pxyuj i: byvmonogrnxlzlq@my.public.ip CSeq: 347 ACK User-Agent: Twinkle/1.0 l: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog 'byvmonogrnxlzlq@my.public.ip' Method: ACK pbx*CLI> <--- SIP read from providerip1:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK3b1ff5b6 From: "Me" ;tag=as0a4685cd To: Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 103 CANCEL Content-Length: 0 <-------------> --- (7 headers 0 lines) --- pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> Really destroying SIP dialog '5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1' Method: INVITE pbx*CLI> ************************************************************** Here I have hang up my IP phone. My mobile phone still ringing. ************************************************************** <--- SIP read from providerip1:5061 ---> SIP/2.0 486 Busy here Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK3b1ff5b6 From: "Me" ;tag=as0a4685cd To: ;tag=483288967-1177318448422 Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- pbx*CLI> <--- SIP read from providerip1:5061 ---> SIP/2.0 486 Busy here Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK3b1ff5b6 From: "Me" ;tag=as0a4685cd To: ;tag=483288967-1177318448422 Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- pbx*CLI> <--- SIP read from providerip1:5061 ---> SIP/2.0 486 Busy here Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK3b1ff5b6 From: "Me" ;tag=as0a4685cd To: ;tag=483288967-1177318448422 Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- pbx*CLI> <--- SIP read from providerip1:5061 ---> SIP/2.0 486 Busy here Via: SIP/2.0/UDP pbx.public.ip:5060;branch=z9hG4bK3b1ff5b6 From: "Me" ;tag=as0a4685cd To: ;tag=483288967-1177318448422 Call-ID: 5bddd1094ffe6f937eabb4e37b832f87@providerdomainname1 CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) ---