Transfer Call -> Caller ID is not updated

Would you like to see this feature added?

  • Yes, it’s a must!
  • Well, nice to have
  • No, do not need that at all

0 voters

Hi!

We recently switched from our “regular” rented PBX to an Asterisk-based solution with 40+ SNOM 320 and 360 phones. The installation and configuration was done by a consultant and now everything works as it should, besides one single annoying thing. Before I start to whine, I must say that I am really amazed by this powerful and feature-rich piece of fantastic software.

Now on to my problem :smile:
When I transfer an incoming call from a customer to one of my colleagues, the colleague sees my extension as the caller ID. As long as I am talking to him and instruct him about who just called and what the topic is, this is perfectly OK for me. But when I then hang up and the customer is transferred to my colleague, he still sees my ID in the display and not the caller ID of the person he is talking to.

I used quite a lot of PBX systems so far and this is something which no other PBX did.

I mean, basically this is false information. As far as I know, the Caller ID in your phone display tells you who you are talking to, that’s what it is supposed to be used for.

Searching the Wiki, this Forum and other ressources (via Google), all I found is a closed bug report:
http://bugs.digium.com/view.php?id=7591

It is from July, 2006 and I was stunned to find out that this is not considered as a missing basic feature.

How can I get around that? Is there any way out for me (as a non-asterisk-developer)? Is there an unofficial patch available, maybe?

Thank you for any hints!

Best Regards,
Heiko

Good luck in your quest. I feel your pain.

The way I’ve learned to think of Asterisk is as a PBX for the old style touch tone phone with no display. As long as you think that way, things will be wonderful. Caller ID is not a problem in this situation, because there is no display to display it.

Now, just think of all the things you can do with Asterisk that you couldn’t do with those old-style PBX systems. VoIP, create your own attendent, Meet-Me conference rooms, easily add DIDs, some pretty powerful stuff.

[quote=“lacymoore”]Good luck in your quest. I feel your pain.

The way I’ve learned to think of Asterisk is as a PBX for the old style touch tone phone with no display. As long as you think that way, things will be wonderful. Caller ID is not a problem in this situation, because there is no display to display it.

Now, just think of all the things you can do with Asterisk that you couldn’t do with those old-style PBX systems. VoIP, create your own attendent, Meet-Me conference rooms, easily add DIDs, some pretty powerful stuff.[/quote]

Well, I would not go that far and compare * with an old-style analogue PBX :smile: Asterisk tries to be a full-blown PBX:

Digitium 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.

It is always good (IMHO) to try to look at a product with the eyes of an end-user and my end-users are really annoyed by this (and I suspect that there are a lot of others out there who feel the same).

Best Regards,
Heiko

Hi,

according to the SNOM FAQ at snom.com/wiki/index.php/Old/ … _a_call.3F this can be achieved by sending a SIP INFO message with the new caller id.

[code]Q: How to change the displayed source and destination data of a call?

A: This can be done via the following SIP INFO message. The message must be sent during the existing dialog.

INFO sip:123@192.168.4.185:2051;line=z6yxt6av SIP/2.0
v: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-c5036dd36491b35dba0953bcd912b2a7;rport
f: sip:*66@intern.snom.de;user=phone;tag=24433
t: sip:456@intern.snom.de;transport=udp;tag=ga2hjuhxot
i: 3c268afd0ea6-tk5ueq9zigfb@snom360-000413230077
CSeq: 1 INFO
Max-Forwards: 70
m: sip:*66@192.168.0.8:5060;transport=udp
c: message/sipfrag
l: 68

From: sip:456@intern.snom.de
To: “123” sip:123@intern.snom.de
[/code]

However, this is proprietary and Asterisk currently not implements that.
Anyone knows a workaround, how to send this message ?

Regards, tuger

there is a bug report on bugs.digium.com/ where someone is implementing this … they call it “called party id”. it might be worth you getting aquainted with what is happening there.

I assume you are talking about bug 0006643 ?

If yes, I am afraid this is another issue. They are dealing with called party identification, the example from Heiko would also need calling party identification being updated on transfers (such transfers are independant from the asterisk dialplan and therefore cannot be handled by an application started from the dialplan!) …

It can depend on what kind of transfer it was(blind or attended) for the CID to pass. Also you can look into the flag “o” on Dial.

[quote=“tuger”]I assume you are talking about bug 0006643 ?

If yes, I am afraid this is another issue. They are dealing with called party identification, the example from Heiko would also need calling party identification being updated on transfers (such transfers are independant from the asterisk dialplan and therefore cannot be handled by an application started from the dialplan!) …[/quote]

you’re quite right, sorry, was in a bit of a daze :smiley:

have you tried angler’s suggestion ? i use the ‘o’ option on my incoming calls … but can’t say i’ve noticed callerid problems.

Anything runs fine for blind transfers. However, the problem occurs on attended transfers only. Unfortunately angler’s suggestion doesn’t help here.

The problem on attended transfers is that you need to change the callerid as soon as the “transfer” button is pressed (while the call to the final destination user persists and there is no new Dial command issued).

was this every resolved?

Apologies for resurrecting an old thread, but I’ve just done my first installation and have already been asked a number of times why the caller id does not pass from the receptionist to the call center member when an attended transfer is performed.

Every legacy PBX I’ve used does this, but asterisk does not …?

Is this ever likely to be implemented?