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

Ben Merrills reviewboard at asterisk.org
Tue Feb 3 10:30:54 CST 2015


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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
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  /trunk/rest-api/api-docs/channels.json 431537 
  /trunk/res/ari/resource_channels.c 431537 

Diff: https://reviewboard.asterisk.org/r/4400/diff/


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

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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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