On Wed, May 20, 2009 at 1:50 PM, Tim Nelson <[email protected]> wrote: > Could you elaborate a bit more? > What isn't 'working out to well'? > Are you getting failed calls? One way or no audio?
Sorry for the lack of information. I posted in a bit of haste. Initially it was failed calls, or not being able to register. I had a line similar to register => [email protected] in sip.conf and it was never able to successfully register. I would get a timeout after so long, and then it would send again. I then added the externalip and localnetwork configurations to sip.conf and set the proxy01.sipphone.com section to include the nat=yes, and this netted me one way audio, only after i swapped out the aging cisco router for a vyatta install. I mostly followed guides found on voip-info.org for gizmo and sip, and also the information on Gizmo's website. Another area that had issues with with having something like Dial(SIP/remotehost) would fail to connect to remotehost. -jonathan _______________________________________________ -- 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
