Simulate SIP phone in asterisk

Question

I've installed GoAutoDial Call Center Software. Due to our ancient phone equipment - until we upgrade it - I'm trying to use it only as a call management software for outbound campaigns. So I can present the phone number to agent , etc. without actually dialing it. agent will still do that by hand.

The problem is that agent disconnects in approx. 20 seconds (no sip phone present). I've tracked down this to asterisk.

From the asterisk log :

NOTICE[15426] channel.c : Unable to request channel SIP/8001

Is there a way to simulate a dummy sip phone , so it doesn't throw the client out ? Or are there any obscure settings to prevent this (otherwise desirable) behavior ? I was looking at /etc/conf/asterisk/sip.conf

Answer 1

You can use local channel to do call via dialplan. After that you can call client, if fail wait, call again etc. As needed

exten => s,1,dial(Local/exten@context/n,,)

You have setup nat correctly so your agents always online and setup at agents phone re-registration inteval to 20 sec or less.