[asterisk-users] Asterisk issue

Alex Balashov abalashov at evaristesys.com
Tue Aug 12 09:23:33 CDT 2008


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
>
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-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
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