Call transfer & call forwarding using IAX PA168 Hard Ph

Hi.

I'm using AT-320 based on PA168 chipset hard phone with IAX2 protocol and I can't make chall transfer and call forwarding.

Does anybody know what can be wrong. Is this problem with IAX2
(I couldn't find if IAX2 supports this features) or mistakes in
implementation of the protocol.

Thanks

Tomash.

Hi Tomash,

I tried for over a week to get the same phone/chipset to call transfer, I had no luck and tried every config hack I could think of. Tried updating every CVS I had. Finally I tried another model of phone I have, and immediately I had transfer function. The new phone was SIP, so I am not yet sure if it was phone or protocol behind the problem with the AT-320. I have ordered the AT-402 IAX2 phone and will try that- if it works then it was just the 320 model. If not, I will probably try at least one more IAX2 phone to verify that it is protocol.

I will update you once I try the new phone.

-R

Hi -R

Thanks for answer my post.

I have solved "half" of that issue but it doesn't satisfy me.

You have to add "t" option in dial function.
Example:

exten => _9XX,1,Dial(IAX2/${EXTEN:1}||t)
exten => _9XX,2,hangup()

So, A dials for example "900" and gets connection to B (900 iax user), and during a call B dials "#901" then call from A is transfered to C (901 iax user).
But it is only blind transfer, it mean that after dialing "#901" B is hunged up and A is calling to C.

I'm looking for something like announced call.
Scenario is that:

A calls to B
B dials a number to C
A is hold
When C answers, B can announce A and after B
presses some button A gets connection to C.

I have read somewere that there is no way to implement such scenorio in asterisk, but some SIP phones (not IAX) can serve it - it is a feature of phones and SIP protocol but not asterisk.

I'm also considering usefulness of Transfer function, because it seems to do nothing except to trigger another Dial function - what is it for?

Another problem is with CDR which are dropped after hung up.
So when A calls to B and B transfers A to C, asterisk
drop CDR as if A calls to C - hmmmmm what if C is PSTN or mobile user, who will pay for that?

If somebody could answer my question and help me would be grateful.

Tomash.

Hi Tomash

CVS Head does support attended transfer as does 1.2, you have to add

Code:

[featuremap] atxfer => 1 blindxfer => #

to your features.conf

Then to do a blind txfer dial # then the extension number and hangup, and to do an attended transfer press 1 then the extn no. speak to the called exten and hang up.

Ian
www.cyber-cottage.co.uk

Tomash,
I am glad you were able to get the AT-320 to work with your server. I am using AMP so my extensions are setup with calls to various Macros and no matter what I try, I cant get the 't' variable to work. There is a call to DIAL_OPTIONS which contains the 't' variable but for some reason it is not translated with the AT-320. I have put 't' manually in every logical place I can think of, but no luck. We will be upgrading our AMP which contains Asterisk 1.2 once it is no longer BETA and hopefully the phone will be able to recognize the featuremap codes.

-R

Isn't 'atxfer' a new feature for Asterisk (ver 1.2 up), along with call recording?
I remember looking for the feature some months ago for 1.0.9

Sadly still no facility to bounce 'no reply on transfer' back to the referrer.

How and if transfer works (especially in AAH) depends heavily on the phone. My Budgetones take a dislike to the # but the 'transfer' key works (in AAH 1.5). A definite case of RTFM.
They also manage attended transfer too!

Hi All

Attended transfer has been in head for a long time, I sure all this year at least. but I have to say not very well documented.

Quote:

Sadly still no facility to bounce 'no reply on transfer' back to the referrer.

All I have to do is hang up and my phone rings me back and im connected again, But this method doesnt seem to work from an ATA

Ian