[OpenSIPS-Users] Click to Dial example that comes with OpenSIPS
duane.larson at gmail.com
duane.larson at gmail.com
Wed Jul 25 06:58:02 CEST 2012
Hey Bogdan,
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.
Here is what I am doing
Client_on_LAN <-> OpenSIPS/SBC <-> Internet <-> OpenSIPS/Proxy
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.
On , Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:
> Hi Duane,
> 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.
> In your capture, I see that there is a 100 trying reply received after
> all from the destination - are there any other replies following ?
> Regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 07/23/2012 07:54 PM, duane.larson at gmail.com wrote:
> 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.
> #
> U 2012/07/23 11:49:04.398750 50.XX.XX.156:5060 -> 99.XX.XX.161:5060
> INVITE sip:9016XX6XX4 at 192.168.88.13:3072;line=996he62l SIP/2.0.
> Record-Route:
> 34533;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=38a.ff1a8d45>.
> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK10f3.c6fade97.0.
> Via: SIP/2.0/UDP
> 50.XX.XX.54:5065;received=50.XX.XX.54;rport=5065;branch=z9hG4bK185398.
> From: ;tag=34533.
> To: sip:9016XX6XX4 at irock.com>.
> Call-ID: 78ecbf1cd5542c12fe2293da14517b38 at 50.57.75.54.
> CSeq: 20 INVITE.
> Contact: .
> Content-Type: application/sdp.
> Max-Forwards: 69.
> User-Agent: PHP SIP.
> Subject: click2call.
> Content-Length: 225.
> P-hint: outbound->inbound .
> P-hint: Route[6]: mediaproxy .
> .
> v=0.
> o=click2dial 0 0 IN IP4 50.57.75.54.
> s=click2dial call.
> c=IN IP4 173.XX.XX.111.
> t=0 0.
> m=audio 12790 RTP/AVP 0 8 18 3 4 97 98.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:97 ilbc/8000.
> a=rtpmap:98 speex/8000.
> #
> U 2012/07/23 11:49:04.398750 99.XX.XX.161:5060 -> 50.XX.XX.156:5060
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK10f3.c6fade97.0.
> Via: SIP/2.0/UDP
> 50.XX.XX.54:5065;received=50.XX.XX.54;rport=5065;branch=z9hG4bK185398.
> From: ;tag=34533.
> To: sip:9016Xx6xx4 at irock.com>.
> Call-ID: 78ecbf1cd5542c12fe2293da14517b38 at 50.XX.xx.54.
> CSeq: 20 INVITE.
> Server: OpenSIPS (1.8.0-notls (x86_64/linux)).
> Content-Length: 0.
> .
> So I am not sure why the first sip trace INVITE I sent isn't being
> processed but the one above is. Very weird.
> I haven't tried to send it to another OpenSIPS server because I really
> don't have any other server to test with.
> On , Bogdan-Andrei Iancu bogdan at opensips.org> wrote:
> >
> >
> >
> >
> >
> >
> > Hi Duane,
> >
> >
> >
> > You mean there is no answer to that INVITE ? When you tried to
> > send the INVITE to another opensips, have you noticed errors in
> > the logs (like parsing errors)?
> >
> >
> >
> > Regards,
> >
> >
> > Bogdan-Andrei Iancu
> > OpenSIPS Founder and Developer
> > http://www.opensips-solutions.com
> >
> >
> > On 07/21/2012 07:45 AM, duane.larson at gmail.com wrote:
> > Has anyone used the ctd.sh example that comes with
> > Opensips in the "example" folder? I am trying to use it and the
> > INVITE gets sent out but nothing happens. I even tried with
> > sending the INVITE to an OpenSIPS server and the OpenSIPS server
> > never even sees it enter the main route even though I see that the
> > INVITE is making it to the server because an NGREP shows it making
> > it. It doesn't make much sense. I even did a debug and don't see
> > anything showing that OpenSIPS sees the INVITE.
> >
> >
> >
> >
> > Here is the INVITE that is generated from ctd.sh
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > U 2012/07/19 18:20:13.486847 50.XX.XX.156:5060 ->
> > 99.XX.XX.161:5060
> >
> >
> > INVITE sip:9016XXXXX at 192.168.88.13:3072;line=g2hfphrk SIP/2.0.
> >
> >
> > Via: SIP/2.0/UDP 50.57.54.156;branch=z9hG4bKaab1.6c7ffd84.0.
> >
> >
> > To: sip:9016XXXXX at irock.com.
> >
> >
> > From: ;tag=134274001013257.
> >
> >
> > CSeq: 1 INVITE.
> >
> >
> > Call-ID: 134274001013257.fifouacctd.
> >
> >
> > Content-Length: 155.
> >
> >
> > User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).
> >
> >
> > Contact: .
> >
> >
> > Content-Type: application/sdp.
> >
> >
> > .
> >
> >
> > v=0.
> >
> >
> > o=click-to-dial 0 0 IN IP4 0.0.0.0.
> >
> >
> > s=session.
> >
> >
> > c=IN IP4 0.0.0.0.
> >
> >
> > b=CT:1000.
> >
> >
> > t=0 0.
> >
> >
> > m=audio 9 RTP/AVP 8 0.
> >
> >
> > a=rtpmap:8 PCMA/8000.
> >
> >
> > a=rtpmap:0 PCMU/8000.
> >
> >
> >
> >
> >
> >
> >
> >
> > Is anyone else out there using anything else to do Click to
> > Dial????
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> >
> >
> >
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