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. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com _______________________________________________ -- 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
