Re: [OpenSER-Users] Missing RTP stream - sr-users

20 Sep 2007

Hi!

I can see in the mediaproxy log the it is initialized to proxy the
call, but I newer get a "session xxxxx: called signed in from xxx"
from Asterisk.

session.py shows that the the connection between mediaproxy and
Asterisk is missing.

I will try to take a look at the sip debug from asterisk and try to
change the NAT settings in Asterisk.

Thanks for your input.

On 9/20/07, Norman Brandinger &lt;norm(a)goes.com&gt; wrote:
...
  You stated that you've forced every call through
mediaproxy.  Are you
 positive ?

 Have you taken a look at the mediaproxy logs (and/or sessions.py  when
 the call is up) ?  They might provide some useful information to you.

 Ditto for Asterisk "sip set debug on" (note that the sip debug command
 format is a moving target).

 Have you looked at the "nat=" settings in sip.conf as well ?  At times,
 they tie closely with "canreinvite=".

 Norm

 Morten Isaksen wrote:
  Hi!

 canreinvite is set to no and the OpenSER/mediaproxy is "hiding" the
 clients IP-addresses from Asterisk, so I am pretty sure that this is
 not the issue.

 On 9/20/07, Norman Brandinger &lt;norm(a)goes.com&gt; wrote:

  Hi Morten,

 Admittedly, I haven't looked closely at your trace.  However, based on
 the description you gave, the first place to look is at the "canrevite"
 setting in Asterisk sip.conf.  You might want to try "canreinvite=no"
 after reading up on this particular setting.

 Regards,
 Norm

 Morten Isaksen wrote:

> Hi!
>
> I have a strange problem with a missing RTP stream between OpenSER and
> Asterisk. I am not sure if it is OpenSER og Asterisk related.
>
> I have this setup
>
> Phone A (172.17.96.17) --
>                                       \      Openser    --    Asterisk
>       --    PSTN
>                                       /      (192.168.0.6)   (192.168.0.3)
> Phone B (172.17.96.10) --        (172.17.64.1)
>
> I also have a Mediaproxy running on OpenSER and I force every call to
> use the Mediaproxy.
>
> I call from Phone A or B to the PSTN works fine and from PSTN to Phone
> A or B it also works.
>
> I have the dialplan logic on my Asterisk server so I want calls from
> Phone A to Phone B to pass the Asterisk server. And this is were I
> have the problem. When the call is established the RTP stream is
> missing between Mediaproxy and Asterisk. I only have a RTP stream
> between the phones and Mediaproxy. As far as I can see the SIP
> signalling is correct.
>
> The SIP traces is listed below. Can you spot the problem in this?
>
> I will buy a beer (or 5) at OpenSER training in Rome to anyone who can
> help me solve this problem.
>
> SIP trace between the phones and OpenSER:
>
> 

-- 
Morten Isaksen
http://www.misak.dk/blog/