Giorgio, tcpdump and wireshark are your friends. Instead of guessing, capture a call with tcpdump then look at it with wireshark.
Darrick On 06/18/2009 08:58 AM, Giorgio Incantalupo wrote: > Hi Darrick, > > I always set canreinvite=no 'cause it gives a lot of problems if set to > yes (and the default is). > I made a call with rtp debug on and I noticed that normally, on the > asterisk CLI, I see one packet sent corresponding to one packet got > (made a test with a local call on our production server). On the other > server with the vpn, I get a bunch of sent followed by a group of > got...there is something in the way the RTP packets are sent/received by > Asterisk and maybe it can be correlated to the missing audio. > > Giorgio > > Darrick Hartman (lists) wrote: >> Do you have 'canreinvite=no' in your sip.conf entry for this phone? If >> not, you should. >> >> On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote: >> >>> Hi John, >>> >>> I already have the ccd dir with the iroute (mandatory for routing to >>> pc/phone connected to vpn client). During the last test I could register >>> and make a call but voice disappears after 1, 2 seconds. I'm trying to >>> understand if it is a bandwidth problem. At the moment I have my phone >>> connected to the openvpn client (which is its gateway) but I have to use >>> the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip >>> (192.168.1.12) is not working. I suppose it is a sip protocol problem: >>> I had to change the sip.conf setting nat=yes to make the phone dial and >>> domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds). >>> I keep on working on the vpn since it seems so little is missing to have >>> a clear conversation. Let me know if your tests are successfull. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
