[asterisk-users] Asterisk issue

[Steve Totaro]

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