Using multiple Broadvoice lines with Asterisk

[karthik]

Hi,

I am trying to setup an asterisk pbx with outbound calling driven through multiple BV lines. i want the first line to reflect as busy when a second or subsequent caller is automatically sent outbound from the second or subsequent line and so on. But the problem is that the broadvoice line never reflects as busy (maybe because of 3-way calling ability) and as a result any outbound call automatically goes through the first bv line which allows up to 2 individual calls to happen simultaneously but in the 3rd one goes all the way to bv and comes back with a message "Your call cannot be completed. Please hangup and try later." Instead i want this line to just give a busy tone so asterisk will automatically fallback to the next outbound line in the dial plan or something. Can anyone help?

I am using regular asterisk not a@h.

Here is what i am using.

exten => _1NXXNXXXXXX,1, Dial(SIP/${EXTEN}@account1server,60,rtT)
exten => _1NXXNXXXXXX,2, GotoIf(($[${DIALSTATUS}=CHANNELUNAVAIL]?7:8)

exten => _1NXXNXXXXXX,7, Dial(SIP/${EXTEN}@account2server,60,rtT)
exten => _1NXXNXXXXXX,8, GotoIf(($[${DIALSTATUS}=CHANNELUNAVAIL]?10:12)

exten => _1NXXNXXXXXX,10, Dial(${TRUNK}/${EXTEN},60,rtT)
exten => _1NXXNXXXXXX,12, Goto(102)

exten => _1NXXNXXXXXX,102, Congestion
exten => _1NXXNXXXXXX,103, Hangup

;=========

What happens is the first outbound call goes through account1server, so does a second outbound call from another user, and a third call instead of going to account2server also goes to account1server as there is no busy or fast busy tone indicating to the asterisk that the line is busy, tends to attempt through the account1server and gives a recorded message "your call cannot be completed. please hangup and try later".

Could anyone please help?

I have posted this in broadvoice forum also.

[mberlant]

This is an Asterisk situation, having nothing to do with BV.

It's not very elegant (there used to be an "outgoinglimit=" command for your [dial-broadvoice] block), but have a look at this posting over on voip-info.org.

You will need to set up each BV line independently with the SetGroup() command and enforce it in your Dial Plan with the CheckGroup() command.

[karthik]

Thank you very much.

The setgroup and checkgroup did it all.

I appreciate your help.

[columbo]

HI

I think I am trying to achieve something similar to this.

I have multiple sip lines and would like asterisk to try the first line for outgoing calls, if it's free to use it. If not free to try the next available free line and so on.

So for eg if all 10 lines are being used and say line two becomes free, then asterisk will route the next call made to this available free line ?

Can you provide guidance to help me do this?

Cheers

Columbo