[Asterisk-Users] Phoneserve SIP provider

[Lubomir Christov]

yes, just tray it :)

Sergey S. Stasyuk wrote:
> 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
> 
> 
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