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