Semi Attended Transfer

Good morning,

Is there anyway i can achieve the following.

I call from extension A to an outside line and while i hear the progress that it is ringing (before the other end picks up) i want to transfer the call to extension B within the same network so he picks up where i left off from extension A.

Only typical way is to wait for the other end to answer (outside line) then i am given the standard options to do the transfer from extension A to extension B.

Phones being used are Fanvils.

Thank you.

Does dialing ## and then the extension to transfer to work at all?

No not at all

Hmm, yea not sure what options there are for transferring an outbound call mid ring.

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You might want to note that this is not the same as a normal half attended transfer, in which the final destination is the one that has the call completed whilst it is ringing.

Although even this relies on custom code, with FreePBX, you should look into Originate.

I think you would need to use ARI to do the full specification.

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Will have a look at what you are suggesting.

On the other hand with ARI i am not familiar with so i am not sure what i have to do. Will use our friend Google of course.

So what happens when you make a call, while it is ringing you hit the transfer button on the phone and transfer the call?

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When i hear the mid ring call i press transfer on the phone but the phone says transfer failed and the only option i get is the END CALL option, i hit transfer from the transfer button as no options appear on the dss keys. The options appear when the call is connected for blind attended and so on. I even tried adding a BLF key but with that method i keep seeing a TryingPlsWait and nothing happens.

Thank you

It has to do with IP phones (it doesn’t matter Fanvil, Yealink, Snom etc.) don’t enable call transfer until the communication with the other party is established (until pjsip channels are in “up” state rather than “ring” , see “pjsip show channels” in asterisk cli).
If you want to achieve a “select external number for me and transfer me before answer” service, probably an ARI application would be used as suggested by david55 (like also one of “dial by click” service integration) , but the real outside call starts after the ringing extension answers the internal call.

Yes and no. Asterisk doesn’t export a user initiated transfer to the phone, it is all done by internal manipulations.

I agree, I mean whatever trick you use, the manipulation is “internal” indeed, then the last sip dialog is between the end user/extension and the external party.

So it is mechanisms then, i have used an Avaya PBX system which is about 20 years old and somehow that is functioning and implemented with a basiif BRI line and a digital phone.
If you ask me it is more appropriate to connect to the other end before you transfer it (polite) but i find it strange that it cannot be achieved with IP Pbxs and it could be achieved with deprecated technologies.

There were useful and smart mechanisms on old analogue/isdn pbxes used succesfully for decades but hard or impossible to replicate with sip/VoIP technology…
I.e. a led for busy channel or all trunk channels busy (like for bri lines) is somewhat hard to replicate…

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Why not just use ChatGPT to help with the code?