[Asterisk-Users] Phoneserve SIP provider

[Lubomir Christov]

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

Sergey S. Stasyuk wrote:
> Hi all!
> 
> I use phoneserve provider with ATA-186 connected through * box. I need
> to use only one one connection to account at the same time. How can I
> switch to another if first is busy?
> 
> 
>>Phone1 |\                                  /| PhoneServe account 1
>>         \|                              |/
>>          | ATA-186 |-----| Asterisk Box |
>>         /|                      |       |\
>>Phone2 |/                        |         \| PhoneServe account 2
>>                                 |
>>                          Non-ATA users
> 
> 
> Is it possible to use * box in such way?
> 
> Best reagrds,
> Sergey Stasyuk
> 
> 
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