Hey Bogdan,
<br />
<br />I think you might be a little confused from my emails. The last email that had a SIP trace and the 100 Trying was a click-to-dial generated from php-sip(http://code.google.com/p/php-sip/) and I wanted to show you that with php-sip the OpenSIPS server processes the INVITE and replies with a 100 Trying.
<br />
<br />Here is what I am doing
<br />
<br />Client_on_LAN <-> OpenSIPS/SBC <-> Internet <-> OpenSIPS/Proxy
<br />
<br />So I am executing the ctd.sh script on the OpenSIPS/Proxy server and the INVITE is going to the OpenSIPS/SBC server and that is where OpenSIPS isn't even seeing the INVITE.
<br />
<br />
<br />
<br />
<br />On , Bogdan-Andrei Iancu <bogdan@opensips.org> wrote:
<br />> Hi Duane,
<br />>
<br />>
<br />>
<br />>
<br />>
<br />> The INVITE generated by the opensips (triggered by the PHP script via MI) will not show up in the opensips script - it is directly sent out by opensips internals (without script interaction) to the destination from DURI / RURI - the only place where you can see it (on the opensips instance that generates that INVITE) is by using a local route.
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<br />>
<br />>
<br />>
<br />>
<br />> In your capture, I see that there is a 100 trying reply received after all from the destination - are there any other replies following ?
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<br />>
<br />>
<br />> Regards,
<br />>
<br />>
<br />>
<br />>
<br />>
<br />> Bogdan-Andrei Iancu
<br />>
<br />>
<br />> OpenSIPS Founder and Developer
<br />>
<br />>
<br />> http://www.opensips-solutions.com
<br />>
<br />>
<br />>
<br />>
<br />>
<br />>
<br />>
<br />>
<br />> On 07/23/2012 07:54 PM, duane.larson@gmail.com wrote:
<br />>
<br />>
<br />>
<br />>
<br />> That is correct there is no answer to the INVITE and it doesn't appear to even enter the main route of the OpenSIPS config. I have xlogs set up in the opensips script and I never see the INVITE enter. Here is another sip trace from the php-sip click to call program and for some reason this INVITE does go through the main route without issue.
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<br />>
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<br />>
<br />> #
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<br />>
<br />> U 2012/07/23 11:49:04.398750 50.XX.XX.156:5060 -> 99.XX.XX.161:5060
<br />>
<br />>
<br />> INVITE sip:9016XX6XX4@192.168.88.13:3072;line=996he62l SIP/2.0.
<br />>
<br />>
<br />> Record-Route: 34533;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=38a.ff1a8d45>.
<br />>
<br />>
<br />> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK10f3.c6fade97.0.
<br />>
<br />>
<br />> Via: SIP/2.0/UDP 50.XX.XX.54:5065;received=50.XX.XX.54;rport=5065;branch=z9hG4bK185398.
<br />>
<br />>
<br />> From: ;tag=34533.
<br />>
<br />>
<br />> To: sip:9016XX6XX4@irock.com>.
<br />>
<br />>
<br />> Call-ID: 78ecbf1cd5542c12fe2293da14517b38@50.57.75.54.
<br />>
<br />>
<br />> CSeq: 20 INVITE.
<br />>
<br />>
<br />> Contact: .
<br />>
<br />>
<br />> Content-Type: application/sdp.
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<br />>
<br />> Max-Forwards: 69.
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<br />> User-Agent: PHP SIP.
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<br />>
<br />> Subject: click2call.
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<br />>
<br />> Content-Length: 225.
<br />>
<br />>
<br />> P-hint: outbound->inbound .
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<br />>
<br />> P-hint: Route[6]: mediaproxy .
<br />>
<br />>
<br />> .
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<br />>
<br />> v=0.
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<br />>
<br />> o=click2dial 0 0 IN IP4 50.57.75.54.
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<br />> s=click2dial call.
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<br />>
<br />> c=IN IP4 173.XX.XX.111.
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<br />>
<br />> t=0 0.
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<br />> m=audio 12790 RTP/AVP 0 8 18 3 4 97 98.
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<br />> a=rtpmap:0 PCMU/8000.
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<br />> a=rtpmap:18 G729/8000.
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<br />> a=rtpmap:97 ilbc/8000.
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<br />> a=rtpmap:98 speex/8000.
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<br />> #
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<br />>
<br />> U 2012/07/23 11:49:04.398750 99.XX.XX.161:5060 -> 50.XX.XX.156:5060
<br />>
<br />>
<br />> SIP/2.0 100 Giving a try.
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<br />>
<br />> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK10f3.c6fade97.0.
<br />>
<br />>
<br />> Via: SIP/2.0/UDP 50.XX.XX.54:5065;received=50.XX.XX.54;rport=5065;branch=z9hG4bK185398.
<br />>
<br />>
<br />> From: ;tag=34533.
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<br />>
<br />> To: sip:9016Xx6xx4@irock.com>.
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<br />> Call-ID: 78ecbf1cd5542c12fe2293da14517b38@50.XX.xx.54.
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<br />>
<br />> CSeq: 20 INVITE.
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<br />>
<br />> Server: OpenSIPS (1.8.0-notls (x86_64/linux)).
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<br />> Content-Length: 0.
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<br />>
<br />> .
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<br />>
<br />> So I am not sure why the first sip trace INVITE I sent isn't being processed but the one above is. Very weird.
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<br />> I haven't tried to send it to another OpenSIPS server because I really don't have any other server to test with.
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<br />>
<br />> On , Bogdan-Andrei Iancu bogdan@opensips.org> wrote:
<br />>
<br />>
<br />> >
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<br />> > Hi Duane,
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<br />> >
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<br />>
<br />> > You mean there is no answer to that INVITE ? When you tried to
<br />>
<br />>
<br />> > send the INVITE to another opensips, have you noticed errors in
<br />>
<br />>
<br />> > the logs (like parsing errors)?
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<br />> >
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<br />> >
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<br />> >
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<br />>
<br />> > Regards,
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> > Bogdan-Andrei Iancu
<br />>
<br />>
<br />> > OpenSIPS Founder and Developer
<br />>
<br />>
<br />> > http://www.opensips-solutions.com
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> > On 07/21/2012 07:45 AM, duane.larson@gmail.com wrote:
<br />>
<br />>
<br />> > Has anyone used the ctd.sh example that comes with
<br />>
<br />>
<br />> > Opensips in the "example" folder? I am trying to use it and the
<br />>
<br />>
<br />> > INVITE gets sent out but nothing happens. I even tried with
<br />>
<br />>
<br />> > sending the INVITE to an OpenSIPS server and the OpenSIPS server
<br />>
<br />>
<br />> > never even sees it enter the main route even though I see that the
<br />>
<br />>
<br />> > INVITE is making it to the server because an NGREP shows it making
<br />>
<br />>
<br />> > it. It doesn't make much sense. I even did a debug and don't see
<br />>
<br />>
<br />> > anything showing that OpenSIPS sees the INVITE.
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<br />> >
<br />>
<br />>
<br />> > Here is the INVITE that is generated from ctd.sh
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<br />>
<br />> > U 2012/07/19 18:20:13.486847 50.XX.XX.156:5060 ->
<br />>
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<br />> > 99.XX.XX.161:5060
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> > INVITE sip:9016XXXXX@192.168.88.13:3072;line=g2hfphrk SIP/2.0.
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<br />>
<br />> >
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> > Via: SIP/2.0/UDP 50.57.54.156;branch=z9hG4bKaab1.6c7ffd84.0.
<br />>
<br />>
<br />> >
<br />>
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<br />> >
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<br />> > To: sip:9016XXXXX@irock.com.
<br />>
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<br />> >
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<br />>
<br />> >
<br />>
<br />>
<br />> > From: ;tag=134274001013257.
<br />>
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<br />> >
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<br />>
<br />> >
<br />>
<br />>
<br />> > CSeq: 1 INVITE.
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<br />> >
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<br />> >
<br />>
<br />>
<br />> > Call-ID: 134274001013257.fifouacctd.
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<br />> >
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<br />> >
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<br />>
<br />> > Content-Length: 155.
<br />>
<br />>
<br />> >
<br />>
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<br />> >
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<br />>
<br />> > User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> > Contact: .
<br />>
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<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> > Content-Type: application/sdp.
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> >
<br />>
<br />>
<br />> > .
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<br />> >
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<br />> >
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<br />>
<br />> > v=0.
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<br />> >
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<br />> >
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<br />>
<br />> > o=click-to-dial 0 0 IN IP4 0.0.0.0.
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<br />> >
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<br />> >
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<br />> > s=session.
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<br />> >
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<br />> > c=IN IP4 0.0.0.0.
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<br />> >
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<br />> >
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<br />>
<br />> > b=CT:1000.
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<br />> > t=0 0.
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<br />> >
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<br />>
<br />> > m=audio 9 RTP/AVP 8 0.
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<br />> >
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<br />> >
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<br />>
<br />> > a=rtpmap:8 PCMA/8000.
<br />>
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<br />> >
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<br />> >
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<br />> > a=rtpmap:0 PCMU/8000.
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<br />> >
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<br />>
<br />> > Is anyone else out there using anything else to do Click to
<br />>
<br />>
<br />> > Dial????
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<br />> > _______________________________________________
<br />>
<br />>
<br />> > Users mailing list
<br />>
<br />>
<br />> > Users@lists.opensips.org
<br />>
<br />>
<br />> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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