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

[Tom Thompson (JIRA)]

Tom Thompson created ASTERISK-29447:
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             Summary: 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


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.



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