Re: [OpenSER-Users] Missing RTP stream

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: >