Did you set up OpenSER to properly statefully relay REGISTER and its replies?
On Tue, August 12, 2008 9:36 am, Steve Totaro wrote: > 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 > -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 _______________________________________________ -- 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
