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)
bindaddr = 0.0.0.0
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()