On Fri, Sep 3, 2010 at 11:50 AM, dave george <dgeo...@teletoneinc.com> wrote: > The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk > SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the > PSTN.
You don't say the percentage that are failing. However, people who have worked with SIP on asterisk have been known to do: exten => s,1,Playback(silence/1) exten => s,n,Whatever(is_next) And I don't know why, but this seems to make things better. If you're doing an Answer and then a receive_Fax, try putting a playback silence in between and see if that helps anything. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
