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Can Asterisk terminate and answer voip incoming calls without using any soft-phones?

I'm trying to provide an option to record incoming calls for my staff's office mobile calls

This works like a charm with asterisk for one voip number. Here is what i have done.

I have an asterisk server running. I have a voip number registered in sip.conf and when i recieve a call to my mobile number i bridge that call with my voip number on the mobile phone, the call to my voip number isi divert it to an internal extension say 2000 which is registerd to an xlite softphone installation in my LAN with the help of asterisk.  MixMonitor then records the call and saves it.

My problem is that as i have more than 50 mobile phones, and i want asterisk to perform just the recording mobile phone call function, i'm not sure how I can scale what i have done above to all 50 phones. Problems with scaling: one voip number can only handle one recording at a time. So i need at least 20 voip numbers. I can get that. Actually i spoke to my voip provider and they said they can provide me with a single voip number with 20 lines.  But now i'm stuck with the problem of having at least 20 different xlite softphones running on pcs in the lan.

1. Is there anyway to remove xlite component from this. Can asterisk directly terminate an incoming call to my voip number (registered in sip.conf) and still allow me to use mixmonitor?
2. If above doesn't work is there anyway possible for me to use appconference and mixmonitor together. I have installed appconference and i'm now able to create conference on the fly. This removes the xlite component, but somehow when i try mixmonitor with it...it only creates the header files for the recording but no content is inside it. (i can post the cli log and my extesnions.conf setup for this if needed)

My sip.conf
bindaddr =
context = sip

here is what i have in my extensions.conf
exten => 1001,1,Noop(from outside)
exten => 1001,n,MixMonitor(wav,filename,v(1))
exten => 1001,n,Dial(SIP/2000,1)
exten => 1001,n,Hangup()

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1 Solution
as i understand you,
you need something like forwarding calls comes to your SIP account to your VoIP provider

bongoloidAuthor Commented:
I answer calls on the mobile phone. If i need to record it then i bridge my gsm call with my voip number. My voip number forwards the call to a local asterisk extension where mixmonitor is workin.

I want to remove the local extension part.
Have you tried using Record instead of MixMonitor?

I'm unclear if the bridging is initiated by the mobile phone service to the VoIP number or if Asterisk is ringing both the mobile and the local extension. From your description and dial plan it looks like the first, but what does the X-Lite do during the call? Does it just ring?
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bongoloidAuthor Commented:
Hi Yes,

You are right. The mobile phone is dialing to the voip number, which is registered in asterisk. Xlite extension just serves the purpose of ringing. Hence i tried to remove the xlite part altogether.

Now i have tried another way of doing the same. Instead of dialing to another local extension, i created a conference room in meetme.conf and the used ConfBridge to form a conference. This way, i'm able to record the call without using any extensions.

I have a new problem however, with this method. Earlier, exten => 1001,n,Dial(SIP/2000,5,H) allowed me to press * from the mobile phone and disconnect the voip number from the call. ConfBridge does not have an option like that.


"Enters the user into a specified conference bridge. The user can exit the conference by hangup only."

So i'm kinda stuck here....
1. If i use Dial to a local extension and then record using MixMonitor, i'm forced to set-up several xlite soft phones so that i can record calls simultaneously.
2. If i use ConfBridge, i solve one, but i'm not sure how to disconnect the voip number alone from the conference using DTMF tone.

Also i tried using Record, but that recorded only what the asterisk extension recieved on its speaker. I don't need that. I only to record the incoming channel.

Thanks for your help and time. Happy New Year to both of you..
It might be possible to enable hangup detection using features.conf, but I'm not sure if it will work for a confbridge call:

Look for this line and remove the ; at the beginning. You may want to change *0 to just *:
;disconnect => *0               ; Disconnect
 please update status regarding this issue ..
 is it solved ??

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