Summary:ASTERISK-08482: Attend transfer with internal Polycom tranfer method does not show original caller id
Reporter:kib (kibeki)Labels:
Date Opened:2007-01-04 08:32:36.000-0600Date Closed:2007-01-15 14:53:00.000-0600
Versions:Frequency of
Environment:Attachments:( 0) attended_transfer2.log
( 1) blind_transfer1.log
( 2) verbosedebug.txt
Description:Caller A (original CID information) calls B. B needs to transfer the call to C.

After a blind transfer the original CID information is available to the destination (C).

After a attended/consultion transfer of a call at the moment C can only view the CID infomation from B.

This happens with SIP firmware 1.6.7 an latest 2.x version.


We also open a case at Polycoms JIRA as Consultion/attended transfers with original CID information.

I attach the ethereal protocolls of a blind and attended transfer.

This is what Polycom told me latest:
The display of Caller ID for transferred calls is a little complex within SIP signalling. I have not yet been able to pin-point the precise difference for the Blind Transfer and Consultative Transfer cases you detail.

At a high level, the Caller-ID displayed on the phone is extracted either from SIP header "From" field or (if present) the ?P-Asserted-Identity? and ?Remote-Party-ID? fields.

Call flows for transfers differ a little from call server to call server.

Have you tried asking this question on one of the Asterisk user groups?

I will continue to work on this as time allows, so please keep me informed on what you are able to determine with this information.
Comments:By: Serge Vecher (serge-v) 2007-01-04 08:39:32.000-0600

so that you know, this is not an "Asterisk user group", but a bug-tracker.

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 amount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
  console => notice,warning,error,debug
3) restart Asterisk with the following command:
  'asterisk -Tvvvvvdddddngc | tee /tmp/verbosedebug.txt'
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
sip debug
5) Trim startup information and attach verbosedebug.txt to the issue.

By: kib (kibeki) 2007-01-04 10:21:50.000-0600

verbosedebug.txt is attached
client SIP/160 calls SIP/105, SIP/105 attended transfers the call to SIP/262

By: Serge Vecher (serge-v) 2007-01-04 12:34:09.000-0600

1. You have completely ommitted step 4
2. Do not presume we know your setup --> please explain who calls who and where is the callerid being lost...

By: Olle Johansson (oej) 2007-01-08 05:42:34.000-0600

When we transfer calls in the PBX, we can't update the caller ID display on the phone during a call. There's no SIp method for this and some phones have proprietary extensions in order to support this. Sorry.

By: Joshua C. Colp (jcolp) 2007-01-15 14:52:59.000-0600

Per oej's comment - not something we can easily accomplish and not a bug since we can't be expected to do it.