[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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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4400/
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> (Updated Feb. 3, 2015, 4:30 p.m.)
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>
> Review request for Asterisk Developers.
>
>
> Bugs: ASTERISK-24745
> https://issues.asterisk.org/jira/browse/ASTERISK-24745
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> Repository: Asterisk
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> Description
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>
> This change adds the new Hangup reason to ARI Channels Hangup, "no_answer".
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> 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.
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>
> Diffs
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> /trunk/rest-api/api-docs/channels.json 431537
> /trunk/res/ari/resource_channels.c 431537
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> Diff: https://reviewboard.asterisk.org/r/4400/diff/
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>
> Testing
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> 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
> ----
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>
> Thanks,
>
> Ben Merrills
>
>
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