[Asterisk-Users] Phoneserve SIP provider

[Sergey S. Stasyuk]

Lubomir Christov wrote:

> yes
> put something like this in your extension.conf
> it will route all calls started with 0 (it will send the numbers without 
> 0) to phoneserve accounts
> 
> exten => _0.,1,Dial(Sip/${EXTEN:1}@phoneserve1,,)
> exten => _0.,2,Dial(Sip/${EXTEN:1}@phoneserve2,,)
> 
> Lubo

Thanks, I'll try this.
But will this automatically switch to the second channel if first is
busy?

Best reagrds,
Sergey Stasyuk