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Ship it! This can go into 13. It's backwards compatible. - Joshua Colp On Feb. 3, 2015, 4:30 p.m., Ben Merrills wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4400/ > ----------------------------------------------------------- > > (Updated Feb. 3, 2015, 4:30 p.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24745 > https://issues.asterisk.org/jira/browse/ASTERISK-24745 > > > Repository: Asterisk > > > Description > ------- > > This change adds the new Hangup reason to ARI Channels Hangup, "no_answer". > > Currently the only supported hangup reasons are : normal, busy and > congestion. > > I've amended 'res/ari/resource_channels.c' to include the new hangup reason, > "no_answer" which maps to AST_CAUSE_NOANSWER alias. > I've amended 'rest-api/api-docs/channels.json' to include the new value > "no_answer" as part of the swagger definition of Channels/Hangup(Delete). > > *Note* I created this against trunk, was unsure what to put in branch field. > This could be applied to both 12 and 13 however. > > > Diffs > ----- > > /trunk/rest-api/api-docs/channels.json 431537 > /trunk/res/ari/resource_channels.c 431537 > > Diff: https://reviewboard.asterisk.org/r/4400/diff/ > > > Testing > ------- > > 1. The code has been compiled. > 2. The compiled version of asterisk was run and a test ari application loaded > 3. The Swagger UI exposed the new hangup reason "no_answer" under the > accepted values for 'reason' when pointed at the running instance of asterisk > 4. A channel was created and passed to the ari application using cmd Stasis > 5. The channel was then hangup via ari with a hangup cause of "no_answer" > 6. SIP debug was used to confirm the correct cause was being returned by > asterisk > > ---- > SIP/2.0 480 Temporarily unavailable > Via: SIP/2.0/UDP > 192.168.3.14:5063;branch=z9hG4bK-a17e20bc;received=192.168.3.14 > From: "test" <sip:test@192.168.3.201>;tag=66754982239395f0o3 > To: <sip:888@192.168.3.201>;tag=as258c3d5c > Call-ID: 36c8966a-1bfa6e28@192.168.3.14 > CSeq: 102 INVITE > Server: Asterisk PBX SVN-trunk-r431522M > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Length: 0 > ---- > > > Thanks, > > Ben Merrills > >
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