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. >> > > -- Giorgio Incantalupo, mailto:[email protected] vo...@work - The Agile PBX http://www.voiceatwork.eu FG&A srl - http://www.fgasoftware.com Tel: 02 997663.14, Fax: 02 91390172 _______________________________________________ -- 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
