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Summary:ASTERISK-26702: AMI/PJSIP: ExtensionStatus event incorrectly shows Unavailable
Reporter:Luke Escude (lukeescude)Labels:
Date Opened:2017-01-07 10:11:59.000-0600Date Closed:2017-01-07 10:17:38.000-0600
Priority:MinorRegression?
Status:Closed/CompleteComponents:
Versions:13.13.1 Frequency of
Occurrence
Related
Issues:
Environment:x64 CentOS 7Attachments:
Description:The following event is sent by the AMI:

{quote}
   [Event] => ExtensionStatus
   [Exten] => 100
   [Context] => from-internal
   [Hint] => SIP/100
   [Status] => 4
   [StatusText] => Unavailable
{quote}

When in fact extension 100 is registered:

{quote}
testbed3*CLI> pjsip show endpoints
...
Endpoint:  100/100                                              Not in use    0 of inf
   OutAuth:  100-auth/100
    InAuth:  100-auth/100
       Aor:  100                                                3
     Contact:  100/sip:100@72.190.32.250:55048            82f89a927d Avail        13.618
{quote}

Actually, Core Show Hints also shows the extensions "Unavailable"... It is probably a bug in my dial plan at this point, right?
Comments:By: Asterisk Team (asteriskteam) 2017-01-07 10:12:00.961-0600

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: Joshua C. Colp (jcolp) 2017-01-07 10:17:39.049-0600

You are using "SIP/100" as the hint. That's for chan_sip. You need to use "PJSIP/100".
By: Luke Escude (lukeescude) 2017-01-07 10:18:58.634-0600

Resolved - Hints were added to the dial plan as SIP/ext instead of PJSIP/ext. Sorry, we thought we had already gotten rid of all old SIP code.

EDIT: Thanks Josh!
By: Asterisk Team (asteriskteam) 2017-01-07 10:18:58.757-0600

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: Luke Escude (lukeescude) 2017-01-07 10:19:21.123-0600

Not a bug