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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080812/6bb72833/attachment.htm
On Tue, Aug 12, 2008 at 9:29 AM, michel freiha <michofr at gmail.com> 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? > > RegardsThis sounds more like an OpenSER (or Kamailio) issue. How about posting SIP debug info and your relevant SIP configs? Thanks, Steve Totaro
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 <michofr at gmail.com> 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