[Ekiga-list] [Test site proposal]

[Daniel Huhardeaux]

Damien Sandras a écrit :
> Le dimanche 14 janvier 2007 à 16:41 +0100, Daniel Huhardeaux a écrit :
>   
>>> Just try...
>>> It is not possible.
>>>   
>>>       
>>  From our dialplan
>>
>> [IsFAX?]
>> ; For Outside
>> exten => _0.,1,NoOp(Fax for outside our network)
>> exten => _0.,n,Set(SIP_CODEC=ulaw)
>> ;or SetGlobalVar(SIP_CODEC=ulaw) for 1.2 Set(GLOBAL(SIP_CODEC)=ulaw) for 
>> 1.4 and trunk
>>
>> exten => [blablabla]
>>     
>
> I had tried that in the past too, and it didn't work.
>
> I just tried again, and it doesn't work. I have the feeling it makes the
> RTP stream switch to G.711 but only in one directly: which is logical
> because the SIP handshake has already taken place.
>   
Can be true as I always use it before starting a call.

If I well understand, the 500 at ekiga.net is accepted by Asterisk from the 
context=default part from [general] in sip.conf which allow anonymous 
calls? Perhaps you should create a [500] user with all accepted codecs 
and send incoming calls to context=echotest which is your actuel 500 
part in extension conf.

-- 
Daniel