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Summary:ASTERISK-15470: [regression] CDR attended transfer missing
Reporter:Matteo (mpiazzatnetbug)Labels:
Date Opened:2010-01-19 08:46:33.000-0600Date Closed:2011-07-27 09:14:38
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:CDR/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I'm not sure if this issue is link with bug: 0011849.

I have some server in production running 1.4.15. Now we want to upgrade the servers with the latest stable version of 1.4 branch 1.4.29.

I'm doing some checks and I found an issue with attended transfer made by SIP phone.

The scenario is:
A (sip Phone 3998) --> B(PSTN mobile sip-idsn gateway)
A put in hold B
A --> C (other SIP phone 3996)
A make trasnfer beetween B and C
C talk with B


So this is what I have on the cdr:

"","3998","03493824114","users-advance","""Arnoldi Nicola"" <3998>","SIP/3998-00000024","","Dial","SIP/smartnode_tnnet/3493824114/n","2010-01-19 14:48:59","2010-01-19 14:49:06","2010-01-19 14:49:39",40,33,"ANSWERED","DOCUMENTATION","NOVAS-1263908939.36",""
"","3998","03493824114","dial-out","""PAT"" <3998>","SIP/3998-00000024","SIP/smartnode_tnnet-00000026","NoOp","Fine Verifica CallBack","2010-01-19 14:48:59","2010-01-19 14:49:06","2010-01-19 14:49:39",40,33,"ANSWERED","DOCUMENTATION","NOVAS-1263908939.36","BILLING/3493824114/pat/3998"
"","3998","3996","users-advance","""Arnoldi Nicola"" <3998>","SIP/3998-00000027","SIP/3996-00000028","Dial","SIP/3996|60|g","2010-01-19 14:49:28","2010-01-19 14:49:29","2010-01-19 14:49:49",21,20,"ANSWERED","DOCUMENTATION","NOVAS-1263908968.39",""


In the last record I don't have any information that I can use to link the two call in a post process operation, from the cdr it's not possible to see if the duration of the call between 3998 and 3996 it's only an internal call of it' the sum of the internal call plus the transfered call.

In the 1.4.15 version I'm using the same dialplan but in the last record was reported as destination channel the sip/channel between asterisk and the mediagateway ( SIP/smartnode_tnnet-01084630) (see below) in this way was possible with a post precess operation to link the two calls and bill the whole duration of the call.


"","3224","03355378482","dial-out","""Piazza Matteo"" <3224>","SIP/tnn3224-aff26cc0","","ForkCDR","","2010-01-19 10:54:31",,,0,0,"NO ANSWER","DOCUMENTATION","NOVAS-1263894871.14954",""
"","3224","03355378482","dial-out","3224","SIP/tnn3224-aff26cc0","SIP/smartnode_tnnet-01084630","Dial","SIP/smartnode_tnnet/3355378482","2010-01-19 10:54:31","2010-01-19 10:54:43","2010-01-19 10:56:44",133,121,"ANSWERED","DOCUMENTATION","NOVAS-1263894871.14954","BILLING/3355378482/tnn/3224"
"","3224","s","users-advance","3224","SIP/tnn3222-0105a540","SIP/smartnode_tnnet-01084630","","","2010-01-19 10:56:29","2010-01-19 10:56:33","2010-01-19 10:57:13",44,40,"ANSWERED","DOCUMENTATION","NOVAS-1263894989.14981",""




Comments:By: Russell Bryant (russell) 2011-07-27 09:14:32.426-0500

Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

If this is still an issue, please open a new issue so it can be re-triaged appropriately. Thanks!