Hi.
I'm using asterisk as a PSTN -> SIP gateway, so that you can call to any of the 4 PSTN lines connected to the asterisk box from and dial your number, and asterisk will dial out through one of the 4 sip accounts I have on a SIP -> PSTN provider. I think of something like this in the extensions.conf
[incoming] exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer ; Answer the line exten => s,3,DigitTimeout,2 ; Set Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,5 ; Set Response Timeout to 10 seconds exten => s,5,BackGround(welcome_and_dial_your_number) ;
exten => _.,1,Dial(SIP/[EMAIL PROTECTED]) ;*******
I dont know what to write instead of the line marked with *******. A multiple dial like following is not the solution I think.
exten => _.,1,Dial(SIP/[EMAIL PROTECTED]&SIP/[EMAIL PROTECTED]&SIP/[EMAIL PROTECTED])
How can I know the free (or busy, is the same to me) SIP channels at any moment? Is there any built-in var?
Thanks in advance.
RODOLFO
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