[OpenSIPS-Users] Click to Dial example that comes with OpenSIPS
Vlad Paiu
vladpaiu at opensips.org
Thu Aug 9 17:04:43 CEST 2012
Hello,
The <> are only required if you want to have SIP header parameters for
the TO header.
Otherwise, if there are no <> , all the parameters are considered to be
SIP URI parameters.
So, from what I see, that TO header is correct.
Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 08/09/2012 06:13 AM, Duane Larson wrote:
> I changed the following in the ctd.sh script
> Changed the default of
> "`printf "v=0\r\no=click-to-dial 0 0 IN IP4
> 0.0.0.0\r\ns=session\r\nc=IN IP4 0.0.0.0\r\nb=CT:1000\r\nt=0
> 0\r\nm=audio 9 RTP/AVP 8 0\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:0
> PCMU/8000\r\n"`
> To
> "`printf "v=0\r\no=click2dial 0 0 IN IP4 50.XX.XX.156\r\ns=click2dial
> call\r\nc=IN IP4 173.XX.XX.111\r\nt=0 0\r\nm=audio 12790 RTP/AVP 0 8
> 18 3 4 97 98\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:18
> G729/8000\r\na=rtpmap:97 ilbc/8000\r\na=rtpmap:98 speex/8000\r\n"`
> And now it is making it into the OpenSIPS/SBC's main route. Not sure why.
> I noticed another issue now. My snom phone is receiving the INVITE
> but it is replying with a "404 Not Found" error. (If I test with a
> Jitsi client I don't have the 404 issue)
> This shouldn't happen since the TO header is the correct SIP URI.
> The only thing that can be wrong is that the To: URI is not in <>
> I think the TM MI function t_uac_dlg isn't placing the <> around the
> TO: header URI. Reading the RFC I am not 100% sure if the <> are
> required.
> U 2012/08/08 22:09:13.756976 192.168.88.1:5060
> <http://192.168.88.1:5060> -> 192.168.88.13:3072
> <http://192.168.88.13:3072>
> INVITE sip:9016XX6XX4 at 192.168.88.13:3072
> <http://sip:9016XX6XX4@192.168.88.13:3072> SIP/2.0.
> Max-Forwards: 10.
> Record-Route: <sip:192.168.88.1;r2=on;lr>.
> Record-Route: <sip:99.XX.XX.161;r2=on;lr>.
> Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.
> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.
> *To: sip:9016XX6XX4 at irck.com <mailto:sip%3A9016XX6XX4 at irck.com>.*
> From: <sip:controller at ae.com
> <mailto:sip%3Acontroller at ae.com>>;tag=134448175329440.
> CSeq: 1 INVITE.
> Call-ID: 134448175329440.fifouacctd.
> Content-Length: 226.
> User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).
> Contact: <sip:caller at 50.57.54.156:5060
> <http://sip:caller@50.57.54.156:5060>>.
> Content-Type: application/sdp.
> .
> v=0.
> o=click2dial 0 0 IN IP4 50.XX.XX.156.
> 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 2012/08/08 22:09:13.766974 192.168.88.13:3072
> <http://192.168.88.13:3072> -> 192.168.88.1:5060
> <http://192.168.88.1:5060>
> SIP/2.0 404 Not found.
> Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0.
> Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0.
> From: <sip:controller at ae.com
> <mailto:sip%3Acontroller at ae.com>>;tag=134448175329440.
> To: <sip:9016726924 at irock.com <mailto:sip%3A9016726924 at irock.com>>.
> Call-ID: 134448175329440.fifouacctd.
> CSeq: 1 INVITE.
> User-Agent: snom821/8.7.3.10 <http://8.7.3.10>.
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO, UPDATE.
> Allow-Events: talk, hold, refer, call-info.
> Supported: timer, replaces, from-change.
> Content-Length: 0.
>
> On Fri, Jul 27, 2012 at 3:24 PM, <duane.larson at gmail.com
> <mailto:duane.larson at gmail.com>> wrote:
>
> Very sure. Normal calls are working with clients behind the
> OpenSIPS/SBC.
>
>
>
>
> On , Bogdan-Andrei Iancu <bogdan at opensips.org
> <mailto:bogdan at opensips.org>> wrote:
> >
> >
> >
> >
> >
> >
> > Duane,some stupid question : are you sure your opensips is
> > listening on the given IP:port ? have you check with netstat ?
> > also have you checked with netstat also if there is traffic queued
> > on the sockets ?
> >
> >
> >
> > Regards,
> >
> >
> > Bogdan-Andrei Iancu
> > OpenSIPS Founder and Developer
> > http://www.opensips-solutions.com
> >
> >
> > On 07/27/2012 12:48 AM, Duane Larson wrote:
> > Oh yeah. My first email has the SIPTrace from the
> > OpenSIPS/SBC. So I am logged into the OpenSIPS/SBC and did the
> > NGREP. So I see it SIP invite (99.XX.XX.161 is the IP of the
> > OpenSIPS/SBC). I would even let you log into the OpenSIPS/SBC
> > and see it for yourself. Makes no sense.
> >
> >
> >
> > 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 <mailto:sip%3A9016XXXXX at irock.com>.
> >
> > From: sip:controller at ae.com
> <mailto:sip%3Acontroller at ae.com>>;tag=134274001013257.
> >
> > CSeq: 1 INVITE.
> >
> > Call-ID: 134274001013257.fifouacctd.
> >
> > Content-Length: 155.
> >
> > User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)).
> >
> > Contact: sip:caller at 50.57.54.156:5060
> <http://sip:caller@50.57.54.156:5060>>.
> >
> > 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.
> >
> >
> >
> > On Thu, Jul 26, 2012 at 4:35 PM,
> > Bogdan-Andrei Iancu bogdan at opensips.org
> <mailto:bogdan at opensips.org>> wrote:
> >
> > OK, so the OpenSIPS SBC is not receiving
> > the INVITE from the OpenSIPS proxy? If so, have you checked on
> > the opensips proxy, using network capture tool if the INVITE
> > is sent to the proper destination ??
> >
> >
> >
> >
> >
> > Regards,
> >
> >
> >
> > Bogdan-Andrei Iancu
> >
> > OpenSIPS Founder and Developer
> >
> > http://www.opensips-solutions.com
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > On 07/25/2012 07:58 AM, duane.larson at gmail.com
> <mailto:duane.larson at gmail.com>
> > wrote:
> >
> >
> > 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
> <mailto: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
> <mailto: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 <mailto:sip%3A9016XX6XX4 at irock.com>>.
> >
> > >
> >
> > >
> >
> > > Call-ID: 78ecbf1cd5542c12fe2293da14517b38 at 50.57.75.54
> <mailto: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 <mailto:sip%3A9016Xx6xx4 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
> <mailto: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
> <mailto: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.
> >
> > >
> >
> > >
> >
> > > >
> >
> > >
> >
> > >
> >
> > > >
> >
> > >
> >
> > >
> >
> > > >
> >
> > >
> >
> > >
> >
> > > >
> >
> > >
> >
> > >
> > <br /
>
>
>
>
> --
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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