Hot Transfer of Calls in Asterisk

Yes please, give some more info on exactly how it works. Thanks very much. Best Regards

Further to previous comments, the softphone is mozphone

Asterisks blind transfer -
Whiel talking to one person dial the blind transfer code and then the extension of the person. After the digit timeout (few seconds) the call is transferred immediatly.

Asterisk attended transfer  -
As above but you speak to the person you are transfering to after dialing and the original caller is put on hold. When you hang up the call is transferred.

SIP blind transfer is similar to asterisks but you normally get a specific key to do it. Not all phones support blind transfer.

SIP attended transfer can be done is a number of way.
On Aastra phones you press the XFER key and the number of the person you want to call. If you press the XFER key again it sort of does a blind transfer but there is a few seconds of music on hold while you enter the destination number etc... If you dont press the XFER button again and wait until the call is answered you can speak and the call is transferred when you press the XFER button. If you hang up you can reconnect to the call on hold.
Grandstream do have a blind transfer button. The way you do an attended transfer is to put the original caller on hold and call the new desination on a 2nd line. Then at any point you can switch between callers or at any time transfer the two lines together.

I could never get SIP transfers to work properly when I tested them with my Asterisk - it seems to be too dependent on the attached SIP devices (and possibly even on behaviour of remote SIP equipment too).

I had more success with Asterisk's built-in transfer mechanism in features.conf, which uses DTMF strings to signal requests. In addition to adding a [featuremap] section to features.conf, you also need to add option t or T to the Dial command in extensions.conf.

t allows the called party to transfer;
T allows the calling party to transfer (you can have both if you want).
http://www.voip-info.org/wiki-Asterisk+cmd+Dial

feptias is correct, This works perfectly.

http://www.voip-info.org/wiki-Asterisk+cmd+Dial

This page will give you complete info on How to Transfer,


exten => 1265,1,Dial(Phone/phone0,15)
exten => s,3,Dial(SIP/oej,20)
exten => _908.,1,Dial(Modem/ttyI0:${EXTEN:1})
exten => 233,1,Dial(SIP/4029&SIP/4027&Zap/4&IAX/jaz,15,tTr)
exten => 500,1,Dial(Zap/2r2,20,crh)
exten => 20,1,Dial(Zap/3/5551234)


see tTr in example

carchitect - thanks for concurring, but did you realise you have pasted exactly the same link as I gave in my answer or did you mean to paste a different link?

I'm sure you are trying to help, but can you please explain what is the significance of those 6 dial plan lines you've pasted in your answer (other than that they are examples of the Dial command) or in what way the r option might be relevant in call transfers.

John

Hi John

I just wanted to explain him that you can use options like tTr in the above mentioned manner.

http://dumbme.voipeye.com.au/trixbox/call_parking_and_transfer.htm

No intensions to rewrite what you wrote earlier.

Thank you all for help. But when I press # it does not give me time to dial the number but just after 1 sec or so it comes back and say "that is not a valid extension". Can I define a time period to wait until I dial the number? Moreover, can I transfer the call to another number outside the building also? Because my real requirement is that I want to transfer the call to another number outside the building.

Please paste the contents of features.conf here for us to check.

Have a look in features.conf. There is an option called featuredigittimeout which is the amount of time it waits between digits. After you change any of these settings you need to restart (a reload wont work) asterisk.

Yes you can transfer a call to any valid number that you can dial from the phone. So if that phone can dial an external number you can transfer the call to an external number aswell.

Thanks a lot grblades. The things seems to be working. Can we have conference call on this feature also because in this case the call is being transfered by droping me out. Does Asterisk supports conference calling also? I am increasing the number of points by 100 because this is another question I am asking. :)