Topic: Call return after failed transfer

[iam]

Hello!

Does anybody knows how set up call return after failed transfer: no answer and others

Thanks in advance
Kirill

[SARK devs]

It's automatic.  You just need to ensure that the target extension (the one which you want exhibit call-return behaviour) has its voicemail set to None.

If you blind transfer to it and it doesn't get answered then it will return.  If you also want it to return on "busy" then you will need to ensure that the phone only has a single SIP presence (i.e. call waiting turned off), otherwise the phone will never return a busy signal, it will just queue each new call.

Kind Regards
S

[iam]

Hello selintra!

Thank you for your answer.
With blind transfer its clear.

The thing is, that at the moment atxfer can also be used as blind i.e. we don't need to wait till called party answered.
It is much more convinient than to think during every call  if we need blind or attended transfer. I think everyone will agree with this.

But ...  atxfer doesn't support call return after failed transfer in asterisk 1.4. It can only be done through dialplan.

I've found several examples of this in forums.  (for ex. http://asterisk-support.ru/forum/topics/490/?page=1#32555)

So the first question: Is it possible to make changes into dialplan in sail to solve this problem with call return on atxfer?

On the other side. In asterisk 1.6 this problem solved with adding new parametres to features.conf: atxferdropcall, atxferloopdelay, atxfercallbackretries ....

And the second question is: What about plans to move selintra to asterisk 1.6 )))


Sincerely
Kirill

[SARK devs]

Hello Iam

The reason call return doesn't work for attended transfer is that Asterisk gives us no definitive state information during the creation of an AT channel.  We don't do anything in SARK to assist the transfer, we just pick up the pieces when it fails (i.e. - coming off the Dial).  Blind transfer DOES give us state information, which is how we achieve the bounce.

I'm afraid the link you posted is in Russian so it's difficult at short notice to work out what it is doing.  If you want to mimic it in SARK then simply transfer control to a custom app at dial time and handle the transfer yourself.

1.6 does offer bounce, as you say, but ONLY for in-band (i.e. *2 type) transfers, which is not very useful because most Asterisk users have SIP phones which use SIP reinvite to handle transfer.

We will move to 1.6 when it stabilises, there is very little real return because 1.6 does not have a great deal of new functionality. 

Kind Regards

S