Hand off incoming call

Here is what I want to do. When a call comes in to a DID I would like to forward/transfer/handoff the call to another number without the call going through asterisk. Example. a call comes into 808-808-8008 which is a DID registered in my asterisk box, I do not want to handle this call with the DID or with asterisk, instead I want to hand the call off to 808-888-8888 and be done with it. Is this possible?

You’re talking as gui-user… Am I right?
Asterisk deals with calls as soon, as I has a corresponding record is extensions.conf

_8088088008 => Dial(IAX2/some_provider/${EXTEN},);

in this example you will dial out 8088088008, using “some_provider”…


Try this:

exten => _8088088008 => Dial(IAX2/some_provider/808-888-8888,)

this will work for you case.

Vikas Kumar

Firstly thank you for your response. I understand that setup will work and am using it now but from my experience it will still route the call through my box. My understanding is that when a call comes in from 777-777-7777 to 888-888-8888 and asterisk takes it, issues a Dial(SIP/999-999-9999@provider) the call will still be handled by asterisk incurring charges with the provider via the Dial command. Yes? My understading follows that I will have an inbound call with per minute charges from 7777 to 8888. I will also be handling an outbound call with per minute charges from 8888 to 9999 essentially being double billed for redirecting a call. I want to drop totally out of the picture (like say when I set up my cell phone to forward or a “regular” PTSN forward does) have the call running directly from 7777 to 9999 with the 8888 number non existant in the call so that I am not charged for creating and maintaining the bridge.
From what I have read now though here and elsewhere I don’t think it is possible.

You would need to use the Transfer application. I don’t know if it will work for your configuration; I’ve only used it for SIP.

what protocol/channels are you using?

Wouldn’t asterisk drop out of the audio stream once you dial out?

I was confused because you didn’t mention SIP in your initial question.

As you are using SIP, the Transfer application is the way to hand off calls completely, both signalling and speech.

However, any commercially savvy service provider would not pass the transfer request back to the originator, but would just bridge the call and still charge you for the outbound leg.

Re-inviting just the speech path out depends on whether the service provider cooperates, and you meet the other requirements for re-invites. I doubt that a commercially savvy provider would give you any credit for removing the speech path from their systems.