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 08:32:36.000-0600	Date Closed:	2007-01-15 14:53:00.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
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. ****** ADDITIONAL INFORMATION ****** 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.