Summary:	ASTERISK-26415: cdr: CDR merged to first call on attended transfer
Reporter:	Ross Beer (rossbeer)	Labels:
Date Opened:	2016-09-28 09:51:43	Date Closed:	2016-10-03 09:24:28
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	CDR/General
Versions:	13.11.2 14.0.1	Frequency of Occurrence	Constant
Related Issues:
Environment:	Fedora 23	Attachments:	( 0) cdr_example.txt
Description:	When receiving a call to a handset, then placing the call on hold to perform an attended transfer. The resulting CDR is merged with the original CDR. For example: A -- > B -- > Hold --> Dial Number --> Transfer Call The resulting CDR is a single entry with the original call's information. In my opinion, this should create a second CDR for the second call, containing data relating to that call. This should not be merged with the original call. This works fine if the transfer context is used for a blind transfer as it's possible to use the /n in a local channel call which does create two CDRs. It is possible to split the two CDRs as this is what happens in previous asterisk releases such as 1.8 Its worth noting that on an attended transfer, the destination channel is updated, however, the 'dst' or the 'lastdata' column isn't updated.
Comments:	By: Asterisk Team (asteriskteam) 2016-09-28 09:51:43.595-0500 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]. By: Ross Beer (rossbeer) 2016-10-04 06:26:40.055-0500 From further investigation, the Attended part of the call before the transfer is accounted for, but then after the transfer, the third CDR does not update with the 'dst' and 'dcontext'. According to the documentation, this is incorrect: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification#Asterisk12CDRSpecification-SIPProtocolAttendedTransfer The 1st record is the portion of the call between A and B until it was transferred to C (including hold time). The 2nd record is the portion of the call between B and C until B completed the transfer (i.e hangup). The 3rd record is the portion of the call between A and C since the transfer was completed until hangup. The total time of the call for the Party A is 1st record' duration + 3rd record' duration. This can be replicated easily. By: Asterisk Team (asteriskteam) 2016-10-04 06:26:40.233-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: Joshua C. Colp (jcolp) 2016-10-04 12:05:17.979-0500 Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. The specific steps or actions you took that caused you to encounter the problem. 2. The behavior you expected and the location of documentation that led you to that expectation. 3. The behavior you actually encountered. To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines [2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Ross Beer (rossbeer) 2016-10-05 04:24:48.030-0500 Basically, when performing an attended transfer, the 'dst', 'dcontext', 'lastapp' and 'lastdata' is not updated to the bridged channel. Please see the attached example of the current CDR output, this is easy to replicate by making an attended transfer and referring the CDR specification you will see that it is incorrect. Using a physical phone this can be reproduced: 1) Make an inbound call to a handset such as a SNOM 2) Answer the call, place the call on hold 3) Make a third call, talk to the far end 4) Transfer the held call to the far end (hangup on the SNOM) 5) Let the two calls stay connected for 60 seconds 6) Hangup When looking at the 3 CDR entries you will see that the last record is incorrect. By: Rusty Newton (rnewton) 2016-10-12 20:19:29.408-0500 I'm not able to reproduce this like you are seeing it. I think for clarity we'll need you to provide a debug log including the SIP trace so we can see the messaging alongside the Asterisk debug messages where the CDR entries are made. I'd say, include , warning, error, notice, verbose, debug. With verbose and debug turned up to 5 each. With "pjsip set logger on" of course. Grab a separate log simultaneously *without* the additional DEBUG messages if possible. By: Asterisk Team (asteriskteam) 2016-10-27 12:00:01.087-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: Ross Beer (rossbeer) 2016-10-31 05:15:40.210-0500 I will try to provide this I think this is related to ASTERISK-20068 as the same fields are incorrect By: Asterisk Team (asteriskteam) 2016-10-31 05:15:40.637-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: Rusty Newton (rnewton) 2016-11-04 09:56:13.932-0500 Great, thanks. Awaiting the debug. I'll put this in Waiting for Feedback. By: Asterisk Team (asteriskteam) 2016-11-18 12:00:01.693-0600 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines