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:
<sip:50.XX.XX.156;lr;ftag=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: <sip:[email protected]>;tag=34533.
To: <sip:[email protected]>.
Call-ID: [email protected].
CSeq: 20 INVITE.
Contact: <sip:[email protected]:5065>.
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: <sip:[email protected]>;tag=34533.
To: <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]> 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] 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:[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].
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????
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