Hi Duane,
That is really strange. Logically speaking, a change in SDP should have
no impact on receiving (or not) a message - the SDP is just payload and
is not even parsed by opensips (unless you ask it from script by using
rtpproxy/mediaproxy/sipmsgops modules). Anyhow, the request should make
it to the main route.
I rather suspect that something else is filtering your SIP traffic,
dropping the INVITEs in the first format :-/ .
Regarding the second issue:
1) the angle brackets are not mandatory by RFC - they should be used
only if you have URI params or a complex display name (which is not your
case here)
2) assuming that the phone does not like TO - it should reply with a 400
Bad request, not a 404 - a 404 represents a routing indication and
routing is done based RURI not TO.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and 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.
>
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