Re: [ATLSIP] How to generate "180 Ringing" with ATLSIP
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Re: [ATLSIP] How to generate "180 Ringing" with ATLSIP
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From: OSS F. A. <ope...@op...> - 2008-06-20 00:54:59
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You are right. I have created a ticket for this http://www.assembla.com/spaces/opensbc/tickets/24 Unfortunately there is currently no way of doing this in the appliation layer. But it will be there soon enough. Thanks for bringing it up. Joegen > {quote:title=optotronic wrote:}{quote} > I'm using OSSPhone to test ATLSIP. It detects incoming calls, but it doesn't send out "180 Ringing" packets so caller doesn't get rings, and proxy eventually routes call to voicemail. > > Is there a way to automatically or manually send the "180 Ringing" messages with ATLSIP? > > Here is a log from OSSPhone: > {quote}----------------26:34:32.350---------------- > Querying STUN server at stun01.sipphone.com. > This may take a while ... > > ----------------26:34:42.772---------------- > STUN server "stun01.sipphone.com" replies Cone NAT, external IP 72.240.235.122 > ----------------26:34:47.820---------------- > *** LISTENER STARTED *** [OPAL] 127.0.0.1:5060 > ----------------26:34:47.880---------------- > *** LISTENER STARTED *** [OPAL] 192.168.0.20:5060 [*** DEFAULT LISTENER ***] > ----------------26:34:47.936---------------- > >>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=629 > REGISTER sip:proxy01.sipphone.com SIP/2.0 > From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13 > To: sip:rec...@pr... > Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bKc3ce95d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport > CSeq: 1 REGISTER > Call-ID: c3c...@pr... > Contact: "recipient" <sip:recipient@72.240.235.122:22013;transport=udp> > User-Agent: OpenSIPStack v1.1.7-24 > Expires: 3600 > Max-Forwards: 70 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK > Content-Length: 0 > > ----------------26:34:48.069---------------- > <<< SIP/2.0 401 Unauthorized SRC: 198.65.166.131:5060:UDP enc=0 bytes=548 > SIP/2.0 401 Unauthorized > From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13 > To: sip:rec...@pr...;tag=21a483426c2cd5d9b85bffe6bba40a2e.76c8 > Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bKc3ce95d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport=22013 > CSeq: 1 REGISTER > Call-ID: c3c...@pr... > WWW-Authenticate: Digest realm="proxy01.sipphone.com", nonce="4859131f88a465b9f177c8041e68d1ee6d8d51e1" > Content-Length: 0 > > ----------------26:34:48.125---------------- > >>> REGISTER sip:proxy01.sipphone.com SIP/2.0 DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=858 > REGISTER sip:proxy01.sipphone.com SIP/2.0 > From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13 > To: sip:rec...@pr... > Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport > CSeq: 2 REGISTER > Call-ID: c3c...@pr... > Contact: "recipient" <sip:recipient@72.240.235.122:22013;transport=udp> > User-Agent: OpenSIPStack v1.1.7-24 > Expires: 3600 > Max-Forwards: 70 > Authorization: Digest username="recipient", realm="proxy01.sipphone.com", nonce="4859131f88a465b9f177c8041e68d1ee6d8d51e1", uri="sip:proxy01.sipphone.com", response="4a78a8e0712b92b26d17ba786a89a4eb", opaque="", algorithm=MD5 > Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS, PRACK > Content-Length: 0 > > ----------------26:34:48.226---------------- > <<< SIP/2.0 200 OK SRC: 198.65.166.131:5060:UDP enc=0 bytes=509 > SIP/2.0 200 OK > From: recipient <sip:rec...@pr...>;tag=c3ce95d9d4fb1810991aa33f609baf13 > To: sip:rec...@pr...;tag=21a483426c2cd5d9b85bffe6bba40a2e.90ab > Via: SIP/2.0/UDP 72.240.235.122:22013;iid=2493;branch=z9hG4bK2d1596d9d4fb1810991ba33f609baf13;uas-addr=198.65.166.131;rport=22013 > CSeq: 2 REGISTER > Call-ID: c3c...@pr... > Contact: <sip:recipient@72.240.235.122:22013;transport=udp>;expires=3600 > Content-Length: 0 > > ----------------26:34:58.666---------------- > <<< INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=1103 > INVITE sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...> > Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0 > Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060 > CSeq: 102 INVITE > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Contact: <sip:caller@66.54.140.46> > Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr> > Date: Wed, 18 Jun 2008 13:47:42 GMT > User-Agent: Asterisk PBX > Max-Forwards: 16 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > RemoteIP: 66.54.140.46 > Content-Type: application/sdp > Content-Length: 379 > > v=0 > o=root 16325 16325 IN IP4 66.54.140.46 > s=session > c=IN IP4 66.54.140.46 > t=0 0 > m=audio 14868 RTP/AVP 0 8 3 18 97 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:97 iLBC/8000 > a=fmtp:97 mode=30 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ----------------26:34:58.684---------------- > >>> SIP/2.0 100 Trying DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=411 > SIP/2.0 100 Trying > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...> > Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0 > Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060 > CSeq: 102 INVITE > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr> > Content-Length: 0 > > ----------------26:35:24.111---------------- > <<< CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=332 > CANCEL sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...> > Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0 > CSeq: 102 CANCEL > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Content-Length: 0 > > ----------------26:35:24.151---------------- > >>> SIP/2.0 200 OK DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=358 > SIP/2.0 200 OK > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...> > Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131 > CSeq: 102 CANCEL > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Server: OpenSIPStack v1.1.7-24 > Content-Length: 0 > > ----------------26:35:24.163---------------- > >>> SIP/2.0 487 Request Cancelled DST: 198.65.166.131:5060:UDP SRC: 72.240.235.122:22013 enc=0 bytes=535 > SIP/2.0 487 Request Cancelled > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...>;tag=a1f8ccd9d4fb1810991ba33f609baf13 > Via: SIP/2.0/UDP 198.65.166.131;iid=2494;branch=z9hG4bK86dc.5c1c1044.0;rport=5060;received=198.65.166.131 > Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK474b057b;rport=5060 > CSeq: 102 INVITE > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Record-Route: <sip:198.65.166.131;ftag=as47bd3a4c;lr> > Server: OpenSIPStack v1.1.7-24 > Content-Length: 0 > > ----------------26:35:24.263---------------- > <<< ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 SRC: 198.65.166.131:5060:UDP enc=0 bytes=364 > ACK sip:recipient@72.240.235.122:22013;transport=udp SIP/2.0 > From: "FLINT MI" <sip:caller@66.54.140.46>;tag=as47bd3a4c > To: <sip:rec...@pr...>;tag=a1f8ccd9d4fb1810991ba33f609baf13 > Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bK86dc.5c1c1044.0 > CSeq: 102 ACK > Call-ID: 2666fde21395b3ae59dc9d7836f06028@66.54.140.46 > Content-Length: 0{quote} > > Recipient OSSPhone was built using latest code in CVS on 2008-06-13. I tested by calling in through an ipKall phone number. No rings were heard from the caller phone system. Eventually the call was routed to voicemail. > > Finest regards, > Bill Root |
