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:[email protected]:3072
<http://sip:[email protected]: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:[email protected] <mailto:sip%[email protected]>.*
From: <sip:[email protected]
<mailto:sip%[email protected]>>;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:[email protected]:5060
<http://sip:[email protected]: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:[email protected]
<mailto:sip%[email protected]>>;tag=134448175329440.
To: <sip:[email protected] <mailto:sip%[email protected]>>.
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, <[email protected]
<mailto:[email protected]>> wrote:
Very sure. Normal calls are working with clients behind the
OpenSIPS/SBC.
On , Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>> 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:[email protected]:3072;line=g2hfphrk SIP/2.0.
>
> Via: SIP/2.0/UDP 50.57.54.156;branch=z9hG4bKaab1.6c7ffd84.0.
>
> To: sip:[email protected] <mailto:sip%[email protected]>.
>
> From: sip:[email protected]
<mailto:sip%[email protected]>>;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:[email protected]:5060
<http://sip:[email protected]: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 [email protected]
<mailto:[email protected]>> 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, [email protected]
<mailto:[email protected]>
> 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 [email protected]
<mailto:[email protected]>>
> 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, [email protected]
<mailto:[email protected]>
> 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:[email protected]: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:[email protected] <mailto:sip%[email protected]>>.
>
> >
>
> >
>
> > Call-ID: [email protected]
<mailto:[email protected]>.
>
> >
>
> >
>
> > 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:[email protected] <mailto:sip%[email protected]>>.
>
> >
>
> >
>
> > Call-ID: [email protected].
>
> >
>
> >
>
> > 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 [email protected]
<mailto:[email protected]>>
> 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, [email protected]
<mailto:[email protected]>
> 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
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
