[asterisk-bugs] [JIRA] (ASTERISK-21394) [patch] - Fundamental changes to CDR within single asterisk family (1.8) during externally initiated blind transfers with an h extension present
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Mon Apr 29 16:19:38 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-21394?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205966#comment-205966 ]
Rusty Newton edited comment on ASTERISK-21394 at 4/29/13 4:19 PM:
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Hi Matt,
I posted details on my other bug report here because the two issues were marked related and I was hopeful this patch would fix both (not because I was trying to hijack this bug report).
{quote}
1. A unique ID is not unique to a CDR. That is a channel unique ID. You can construct a full unique identifier based on the channel unique ID, the CDR sequence number, and - if you really need it - the linkedid.
{quote}
Fair enough, I will talk to the developers of the billing system.
{quote}
2. Asterisk has no concept of inbound versus outbound, nor will it ever. The concept of an inbound call versus an outbound call has to be performed by the dialplan writer. The fact that your billing system is flagging something as 'incoming' is a problem with your billing system.
{quote}
Fair enough, I will talk to the developers of the billing system.
'Since your issues are not related to this problem, and since it appears as if fixing the issues you are facing would require fundamental changes in the behavior of Asterisk's CDRs, it does not appear as if there is anything else required on this issue.'
I understand that my issue is no longer related to this bug report since they appear to be two separate issues.
You should remove the relationship then so there is no confusion.
was (Author: aragon):
Hi Matt,
I posted details on my other bug report here because the two issues were marked related and I was hopeful this patch would fix both (not because I was trying to hijack this bug report).
1. A unique ID is not unique to a CDR. That is a channel unique ID. You can construct a full unique identifier based on the channel unique ID, the CDR sequence number, and - if you really need it - the linkedid.
Fair enough, I will talk to the developers of the billing system.
2. Asterisk has no concept of inbound versus outbound, nor will it ever. The concept of an inbound call versus an outbound call has to be performed by the dialplan writer. The fact that your billing system is flagging something as 'incoming' is a problem with your billing system.
Fair enough, I will talk to the developers of the billing system.
'Since your issues are not related to this problem, and since it appears as if fixing the issues you are facing would require fundamental changes in the behavior of Asterisk's CDRs, it does not appear as if there is anything else required on this issue.'
I understand that my issue is no longer related to this bug report since they appear to be two separate issues.
You should remove the relationship then so there is no confusion.
> [patch] - Fundamental changes to CDR within single asterisk family (1.8) during externally initiated blind transfers with an h extension present
> ------------------------------------------------------------------------------------------------------------------------------------------------
>
> Key: ASTERISK-21394
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-21394
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: CDR/General
> Affects Versions: 1.8.18.0, 1.8.20.0, 1.8.21.0
> Environment: Centos 5
> Reporter: Ishfaq Malik
> Assignee: Matt Jordan
> Attachments: Basic SIP blind transfer debug with patch test results.txt, fix_missing_blindXfer_cdr, fix_missing_blindXfer_cdr2, Master(with patch).csv, queue cdr with patch.rar, Two recordings are identical in playback and bad call direction.pdf
>
>
> The way that asterisk handles the CDR for blind transfers has changed between version 1.8.7.0 and 1.8.18.0. The call path is: inbound call from SIP provider -> asterisk server -> SIP peer -> transfer to another SIP Peer. The transfer is done by using the dedicated keys on the hardware and not the features.conf method.
> In version 1.8.7.0 this would produce 2 entries in the CDR, 1 showing the time the inbound call started and which context it was handles by, the second showing the the context the peer belongs to and looking like an extension to extension call.
> We tried upgrading to 1.8.18.0 and did some regression testing and found that the CDR now only produces a single entry with the details all pertaining to the second leg of the call (bar the duration which is for the full life of the call). It has no reference to the inbound context that initially handled the call.
> This is a pretty big change to be inside a single branch and I'm guessing it wasn't intentional.
> Also, all our servers are installed using Puppet so the configuration on them all is the same and we use RealTime Architecture from a single database.
> Here is an example (with sanitised data):
> Here is how an inbound blind transferred call is recorded in 1.8.7.0
> {noformat}
> *************************** 1. row ***************************
> calldate: 2013-04-04 11:42:09
> clid:
> src: <my-mobile>
> dst: s
> dcontext: <inbound-number-context>
> channel: SIP/xxx.xxx.xxx.xxx-00002ba3
> dstchannel: SIP/501-00002ba7
> lastapp: Dial
> lastdata: SIP/501&SIP/511&SIP/531,10
> duration: 22
> billsec: 12
> disposition: ANSWERED
> amaflags: 3
> accountcode:
> userfield:
> *************************** 2. row ***************************
> calldate: 2013-04-04 11:42:31
> clid:
> src: <my-mobile>
> dst: 504
> dcontext: <peer-context>
> channel: SIP/xxx.xxx.xxx.xxx-00002ba3
> dstchannel: SIP/504-00002bbb
> lastapp: Dial
> lastdata: SIP/504,20
> duration: 7
> billsec: 5
> disposition: ANSWERED
> amaflags: 3
> accountcode:
> userfield:
> 2 rows in set (0.03 sec)
> {noformat}
> And here is how it is recorded in 1.8.18.0
> {noformat}
> *************************** 1. row ***************************
> calldate: 2013-04-04 11:39:24
> clid:
> src: <my-mobile>
> dst: 503
> dcontext: <peer-context>
> channel: SIP/xxx.xxx.xxx.xxx-000000ea
> dstchannel: SIP/503-000000ed
> lastapp: Dial
> lastdata: SIP/503,20
> duration: 34
> billsec: 9
> disposition: ANSWERED
> amaflags: 3
> accountcode:
> userfield:
> {noformat}
> In the newer version there is only one entry which has the combined call
> time in it but the dcontext corresponds with the second leg of the call
> which means we can't see what number was dialled.
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