[asterisk-bugs] [JIRA] (ASTERISK-29447) Is the SIP response code accessible through the AMI

[Tom Thompson (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29447?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=254992#comment-254992 ] 

Tom Thompson commented on ASTERISK-29447:
-----------------------------------------

Thank you Joshua
I had seen the HANGUPCAUSE details as a way of acquiring the 'tech' codes. I see too the UserEvent application could be used to stuff it back into the AMI. 
However we are originating calls in the AMI using Originate and so these don't pass control to the originating extension in the dialplan unless the calls connect. 
We had considered passing the outgoing leg to through a forwarding device in the dialplan that could relay the call as a dial, but we determined that this would be onerously convoluted and the performance overhead uncertain. Furthermore it would add a layer  of indirection to all calls, though we are needing the detailed SIP response info on a small % of calls only.  
We determined that as the information should be available to asterisk at the point of generating the Hangup event, this would be the simplest (and most obvious) source.
Our investigation of the source seems to bear this out. 
We have done some internal experimentation with the source. I am happy to turn that over to you if given authority. Are you interested in seeing it?
Tom 


> Is the SIP response code accessible through the AMI
> ---------------------------------------------------
>
>                 Key: ASTERISK-29447
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29447
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Core/General
>    Affects Versions: 18.4.0
>         Environment: any
>            Reporter: Tom Thompson
>            Assignee: Tom Thompson
>
> When a call is terminates only an ISDN cause code is visible in the Hangup event, even if the call is over SIP
> In the Asterisk mapping of SIP->ISDN there is no 1-to-1 relationship between ISDN codes and SIP response codes. Many given ISDN cause codes can result from a number number of SIP responses. The original SIP response that terminated the call is not available in the Hangup.
> Background:
> Many  SIP providers do not adhere well to simple protocol "standards" , and if a call passes through a number or originating, transit and terminating providers, the level and accuracy of information provided back to the originator may be determined by the lowest-common-denominator in the chain. The result may be variable or inaccurate response codes. 
> As a provider of telephony systems in over 30 countries over the past 25 years, I have seen some pitiful national environments for accurate ISDN information. When SIP replaces or layer-on to good ISDN environments there is usually a further loss of data. In the formerly bad ISDN environments It can result in a digital telephony network that is little better than analogue. 
>  However, any SIP environment, be it national or private, there are often idiosyncrasies  that can be accounted for and corrected if only the SIP response code was available.



--
This message was sent by Atlassian JIRA
(v6.2#6252)