[asterisk-dev] delaying few seconds for FAS problem

[Nic Bellamy]

robert santos wrote:
> hi jean,
>
> tnx for the reply, already tried grandstream, sipura and a NO brand 
> ATA from china.
> got the same problem. do u have a specific ATA that u can recommend?
>
> if theres none im down to last option (hope not the last) hehehe:
> where in the asterisk code do the SIP 200 are receive?
>
> sip <---> asterisk1 <---> asterisk2 <---> ATA/fxs/fxo <---> PSTN
Ahh, he's talking VoIP <-> FXO adapters, not VoIP <-> FXS adapters, 
which is what I'm sure we all thought when "ATA" was mentioned.

Robert - there are ways around this on analogue lines, involving things 
like line polarity reversals on answer, call progress detection, etc., 
but they're all pretty flaky - you're better of with digital interfaces, 
eg. BRI or PRI.

Ask asterisk-users about them, not asterisk-dev - this is not a problem 
with the code, but with your configuration.

Cheers,
    Nic.

-- 
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/