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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> 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/ _______________________________________________ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users