On Tue, Aug 12, 2008 at 9:29 AM, michel freiha <[EMAIL PROTECTED]> wrote: > Dear All, > I have the below issue: > > I created an extension(5678) under extensions_custom.conf to record voice > messages and playback the voice as you can see below: > [custom-recordme] > > exten => 5678,1,Wait(2) > exten => 5678,2,Record(/tmp/asterisk-recording:g729) > exten => 5678,3,Wait(2) > exten => 5678,4,Playback(/tmp/asterisk-recording) > exten => 5678,5,Wait(2) > exten => 5678,6,Hangup > > When dialing this extension from another extension registered on the same > asterisk server everything works fine...The issue begins if I try to make a > call from an OpenSer server....The SIP authentication did not work... > > Can you please give me and step by step the configuration that i should do > in order to accomplish this task? > > Regards
This sounds more like an OpenSER (or Kamailio) issue. How about posting SIP debug info and your relevant SIP configs? Thanks, Steve Totaro _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
