Summary:ASTERISK-07396: Wrong Caller ID after Attended Transfer
Reporter:Raphael Vallazza (endian)Labels:
Date Opened:2006-07-25 10:47:34Date Closed:2006-07-26 10:19:40
Versions:Frequency of
Environment:Attachments:( 0) astersik-atxfer.txt
( 1) Attended_Xfer1.txt
Description:We have the following problem with Snom phones (but i don't think it's Snom related):
1) Phone B recieves a call from Phone A
2) Phone B puts Phone A on hold and calls Phone C
3) Phone C answers and sees the Caller ID of Phone B
4) Phone B now just does a hangup to pass Phone A -> Phone C
5) Phone C still sees Phone B as Caller ID (instead of Phone A)

At point 5 it should show the Caller ID of Phone A instead of Phone B.

I don't think this is a correct behaviour so i contacted the Snom Support and they stated that it's an Asterisk related problem because it works with SER. They also sent me a SIP Traces of SER, i've attached them.

Maybe i'm doing something wrong...
Comments:By: Serge Vecher (serge-v) 2006-07-25 10:50:17

As per bug guidelines, you need to attach a SIP debug trace illustrating the problem. Please do the following:
1) Prepare test environment (reduce the ammount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterik.
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Save complete console log to file and _attach_ said file to the bug.

By: Joshua C. Colp (jcolp) 2006-07-25 11:15:36

This is just the way things work. Phone B is the phone that called Phone C, not Phone A. Now unless one could update callerid after a call has been established... then you could change it but we don't support doing that.

By: Raphael Vallazza (endian) 2006-07-25 11:35:38

I understand that that's the way things work, to be honest i don't have a problem with it, but our users do... :)
Everyone expects to see the original caller number, because all the time they hear from the caller "why are you asking my number, you should see it on your display" :-D

Well, i've attached the asterisk log. It's a call from 10 -> 11 then then 11 -> 13 and after hangup 10 -> 13.

The Snom support guy said that this problem is related to SIP REFER messages not being forwarded...

Sorry that i didn't attach the log earlier.

By: Joshua C. Colp (jcolp) 2006-07-25 12:59:52

Asterisk is not a SIP proxy, so it doesn't "forward" packets - thus why this entire situation is occuring. Each call is kept separate from the other, so when you do an attended transfer and finish it - the other side doesn't know it occured. Sorry, but that's the way it works and will continue to work... Asterisk has to remain a back to back user agent in order to keep being protocol independent.

By: Serge Vecher (serge-v) 2006-07-25 13:09:12

interesting line from debug ...

Jul 24 11:54:05 DEBUG[3885]: chan_sip.c:1030 parse_sip_options: Found no match for SIP option: callerid (Please file bug report!)

By: Joshua C. Colp (jcolp) 2006-07-25 13:16:29

'tis a SIP option we don't support, a cool thing would be trying to track down whether it has an RFC or not... and what the option is for exactly.

By: Raphael Vallazza (endian) 2006-07-25 15:05:55

Hmmm, ok, i understand :-/ The problem is that many users complain that and i also understand theri disappointment... is there any possible hack, workaround, whatever that could solve the problem?

By: Joshua C. Colp (jcolp) 2006-07-25 15:25:08

I can't think of any off the top of my head, sorry.

By: Serge Vecher (serge-v) 2006-07-25 15:29:53

you may want to consider posting a bounty for this support on wiki.

By: Joshua C. Colp (jcolp) 2006-07-26 10:19:39

And this bug comes to a close since it's not really a bug... if you do end up finding something endian, post back so that we can potentially help others who want this same thing. Thanks!