[asterisk-dev] [Code Review] 4400: Add no_answer to ARI hangup causes

Joshua Colp reviewboard at asterisk.org
Wed Feb 4 06:20:35 CST 2015


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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:
> 
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> https://reviewboard.asterisk.org/r/4400/
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> (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
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> 
> 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 at 192.168.3.201>;tag=66754982239395f0o3
> To: <sip:888 at 192.168.3.201>;tag=as258c3d5c
> Call-ID: 36c8966a-1bfa6e28 at 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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