Asterisk Virtual SIP device: Is it possible?

Not sure whether this is the right section for this but here is my question;

I have a particular call center scenario below using Cisco CUCM and Asterisk where I would like to create multiple “virtual phones” in Asterisk which can auto-answer incoming calls.

Cisco CUCM ----SIP Trunk----Asterisk—(SIP/other)—VirtualPhones

  1. Cisco CUCM will create an active call to Asterisk Virtual Phone using Cisco Extend and Connect feature. It is a continuous call which will be active as long as Cisco Call Center agent is logged in.
  2. When a customer call is diverted into this active call, we will programmatically send a request to specific Asterisk virtual phone extension to conference that call with a specific number, in this case agent’s mobile number.


  1. How can we create multiple CTI port type of virtual devices in Asterisk? My guess is we need to work with PJSIP and create some Java-based phones/ports on the server which will register to Asterisk.
  2. How can we trigger a conference on an extension? I believe MeetMe command can be used but not sure about how to trigger the conference programmatically


There are two many “specifics” to be exactly sure what you are doing. I’m also not sure if you are using “extension” in the Asterisk sense.

You should probably read up on Local channels, as I suspect it will turn out that you need them. Also look at originate and channelredirect.

Local channels concept seems to be the right path, thanks for your recommendation.

Specific scenario is as follows;
We want Cisco call center agents to be able to login using mobile phone apps and retrieve customer calls from their mobile phone line. But Cisco call center application requires a Cisco Phone or a SIP Phone so that it can monitor the agent phone line. They also offer a feature called “Extend and Connect” if you want to use your analog home line. But the catch is; Cisco CUCM makes a permanently active call to your home phone as soon as you login as a Cisco call center agent and this line stays active until agent logs out. This can be “OK” for home phone lines but not an option for a mobile phone.

What we are trying to do is to terminate this permanently active call on a virtual SIP endpoint at Asterisk side. And when a customer call is transferred to this active call leg, we will programmatically invoke Asterisk SIP endpoint to conference the call with agent’s mobile phone number.