> Before I start trying to get the bridge approach to work, > I'll try to finish the IP forwarding method. Currently > working only on outgoing calls. The destination phone rings, > but I hear no ringing tone in my SIP phone, and, once the > call is answered, I get no audio either way. > Any ideas about that? I know SIP with no audio often points > towards a NAT issue, but figuring out what is going on in > this scenario, and whether it qualifies as NAT, makes my head hurt.
OK, I guess this is to do with Asterisk taking itself out of the media path. I am seeing packet2packet bridging appearing on the console when I pick up the call, but the dialling handset has no route to the Berofix. I've tried adding canreinvite=no to both the handset and Berofix entries in sip.conf, and I've tried adding a route to the handset's gateway pointing requests for the Berofix at the Astlinux box, but I still have no audio. Any ideas? UPDATE: V odd. I changed everything from GSM to alaw, and it now seems to work. A puzzle for Monday, methinks. Thanks Tom ------------------------------------------------------------------------------ Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
