The error was in Mediaproxy. In rtphandler.py i changed
nonPublicNetworks = [ {'name': '0.0.0.0', 'value': 0x00000000L, 'mask': 0xff000000L}, {'name': '10.0.0.0', 'value': 0x0a000000L, 'mask': 0xff000000L}, {'name': '127.0.0.0', 'value': 0x7f000000L, 'mask': 0xff000000L}, {'name': '172.16.0.0', 'value': 0xac100000L, 'mask': 0xfff00000L}, {'name': '192.168.0.0', 'value': 0xc0a80000L, 'mask': 0xffff0000L}, {'name': '224.0.0.0', 'value': 0xe0000000L, 'mask': 0xf0000000L} ] To nonPublicNetworks = [ {'name': '0.0.0.0', 'value': 0x00000000L, 'mask': 0xff000000L}, {'name': '10.0.0.0', 'value': 0x0a000000L, 'mask': 0xff000000L}, {'name': '127.0.0.0', 'value': 0x7f000000L, 'mask': 0xff000000L}, {'name': '224.0.0.0', 'value': 0xe0000000L, 'mask': 0xf0000000L} ] I think Mediaproxy got confused with the RFC1918 IP's. In my setup there is no NAT between 172.17.0.0/16 and 192.168.0.0/24 - just a router. On 9/21/07, Norman Brandinger <[EMAIL PROTECTED]> wrote: > Is there a firewall in the picture ? You have two different subnets and > there probably is a box doing some (NAT) translation / routing between > them. Is is possible the RTP stream is being blocked at the firewall ? > > Norm > > > Morten Isaksen wrote: > > 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