Caller ID is lost after an attended transfer or for directed call pickup

Did somebody test the callpickup / attended transfer caller ID function with Asterisk 1.6 ?

With Asterisk 1.4, the caller ID of the picked extension or the caller ID of the caller is lost during a transfer.

This is a terrible situation for european users, used to this function with traditionnal telephony hardware like Alcatel, Bosh, Siemens and similar.

This function (update caller ID during a call) is supported as least by Aastra and Snom phones.

For Aastra phones, it needs a specific setting in the server.cfg file, “sip update callerid:”

This work simply by sending a reinvite to the phone, with the new caller ID in the contact field. So it is not a terrible thing to implement.

Asterisk does not send a reinvite after an attended transfer, neither after a pickup. So the caller ID does not update.

When transfering, we get the operator caller ID, when picking up, we get **405 or something similar.

This is a difficult situation since year 2007 where i fisrt post a message about this problem here.

It would be nice if this was working now with asterisk 1.6 or with Freeswitch, or if we could have a workaround with FreePBX.

I’m surprised to see no follow up on this thread. Nobody here interested by a working CID presentation upon attended transfer ?

I’m having exactly the same issue.
Using GXP2000s and Asterisk 1.4.25.1

Can anyone confirm if this has been resolved in 1.6?
I keep hunting around for a solution of more information - but get nowhere

SipXecs did not suffer from this problem.

According to some readings i’ve done in the Asterisk track, this has been implemented inside Asterisk 1.6.1 and later. But i’ve not tested it yet.

Unfortunately, Asterisk 1.6.1 is not tested enough for compatibility issue with current distributions like Elastix and other commercial package for providers.

So i think we’ll still need to use 1.4 for about one year more for professional use.

We specialy have deep compatibility problems with Aastra XML scripts 2.2.1. They do not fully work with 1.6.1 (voicemail and parking not working).

This is something for the Freeswitch community :=)

Any news on this ? Has this been resolved in 1.6.2 or 3 ?
It’s really annoying we can’t do attended transfers without updating the caller id!

Also directed call pickup - just displays the ** picked up from.

Yes this is certainly the more important problem we have with opensource IP telephony.

We are saying this since two years now, and still no support.

This has been solved in commercial systems, but unfortunately, because implementation is not supported by all phone manufacturers, opensource community seems to be blind to the CID update problem.

I hope that somebody will change this fastly as it is not a big work.

Has anyone overcome this problem.
Still driving me nuts!

I don’t see why it hasn’t been fixed or implemented in 1.4

Does anyone have an idea when this functionality will be in the Asterisk?

Just saw this a few minutes ago in ticket 2563:

[quote]02/06/10 09:12:06 changed by jsmith ¶

This feature has been added in the trunk version of Asterisk, and will be available in Asterisk 1.8.[/quote]

Hope this helps! It seems to me to be a very necessary feature.

The link seems dead.

Anyway this would be a very nice news as many users are waiting for this.

Digium advertises Asterisk on their website as a “common PBX and key system replacement” and almost every single “common PBX” transfers the CallerID when you transfer a call.

More, the caller id update is not a complex function. Just send a reinvite to the phone with the new caller ID.

On Aastra phones, we just need this parameter to allow dynamic caller ID update : “sip update callerid”.

This is fully supported since may 2007, firmware release 1.4.2.!! More than three years ago…

I can’t understand this still does not work after many years. Perhaps because Asterisk was almost the only serious IPBX package and Digium has never been very open to propositions, as this closed ticket confirm :

https://issues.asterisk.org/view.php?id=7591

Now that Freeswitch is in the starting blocks, things will certainly change on the Asterisk side.

Sorry, fixed the link above. The # sign was screwing things up.

Additionally, here it is:

http://www.freepbx.org/trac/ticket/2563

:=)

This is a ticket i’ve opened three years ago, a few weeks after Aastra corrected Caller ID update inside firmware version 1.4.2 to make it really works.

It was present before this date but was buggy.

This behavior is still very annoying.
When the call is “attended transferred” from one ext to another we never know the link has been established with all the obvious consequences you guys can imagine.
As a work-around is there the way to ear a “beep”, an announce or some like this?