[asterisk-dev] [Code Review] 4499: Support in dialog OPTIONS

[Matt Jordan]

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Ship it!


Thanks for the patch! I've clicked the Ship It button, although the same statement about requiring tests for things going into Asterisk 13 that I made on the DTMF review applies here as well.

In this particular case, a test for this patch should be done using SIPp, as it is pretty easy to construct an inbound INVITE request and put an OPTION request in-dialog with that INVITE request.

Most of the tests in channels/pjsip use SIPp to drive the tests, and so there is a lot of material to base a test on. We also have sample SIPp scenarios to use as a template in the contrib/sipp folder.

If you have any questions about where to start with that, please don't hesitate to ask on the asterisk-dev mailing list/#asterisk-dev.

- Matt Jordan


On March 18, 2015, 4:01 a.m., yaron nahum wrote:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4499/
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> (Updated March 18, 2015, 4:01 a.m.)
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> 
> Review request for Asterisk Developers.
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> Bugs: ASTERISK-24862
>     https://issues.asterisk.org/jira/browse/ASTERISK-24862
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> Repository: Asterisk
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> Description
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> Respond to OPTIONS message sent on an existing dialog with 200OK. 
> This feature is vital in order to interoperate with some switches that send OPTIONS message periodically per active call to make sure it is still alive. Not responding would cause the switch to disconnect the call. 
> This functionality used to work on the old SIP channel, but was not implemented on PJSIP.
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> Diffs
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>   /trunk/res/res_pjsip_dlg_options.c PRE-CREATION 
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> Diff: https://reviewboard.asterisk.org/r/4499/diff/
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> Testing
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> Thanks,
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> yaron nahum
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>

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