Summary:	ASTERISK-25942: res_pjsip_caller_id: Transfer results in mixed ConnectedLine information
Reporter:	George Joseph (gjoseph)	Labels:
Date Opened:	2016-04-19 15:30:11	Date Closed:	2016-04-21 14:03:51
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Resources/res_pjsip_caller_id
Versions:	13.8.1	Frequency of Occurrence	Frequent
Related Issues:
Environment:		Attachments:
Description:	Alice calls Bob with no display name for caller id, only number. Bob does an attended transfer to Charlie. On the re-INVITE or UPDATE for COLP, Charlie gets Bob's display name and Alice's number. I'm pretty sure this is an issue in -res_pjsip_session- res_pjsip_caller_id when we're determining which values to set on the re-INVITE. This can be reproduced by running tests/channels/pjsip/transfers/attended_transfer/nominal/callee_local. Remove Alice's callerid display name from pjsip.conf and "sipp" from the uac_no_hangup.xml "From" lines. The test will hang (ignore it) but the Wireshark output will show "Bob" <sip:alice@127.0.0.1:xxxx> on the final INVITE to Charlie after Bob hangs up.
Comments:	By: Asterisk Team (asteriskteam) 2016-04-19 15:30:11.982-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].