Summary: | ASTERISK-26702: AMI/PJSIP: ExtensionStatus event incorrectly shows Unavailable | ||
Reporter: | Luke Escude (lukeescude) | Labels: | |
Date Opened: | 2017-01-07 10:11:59.000-0600 | Date Closed: | 2017-01-07 10:17:38.000-0600 |
Priority: | Minor | Regression? | |
Status: | Closed/Complete | Components: | |
Versions: | 13.13.1 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | x64 CentOS 7 | Attachments: | |
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 |