[asterisk-dev] [Code Review] 2614: chan_pjsip: Anonymous Support

[Mark Michelson]

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Ship it!


Looks good!

I'm noticing some repeated patterns in SIP code, such as
* Finding an endpoint based on name at domain and falling back to just name
* Finding a transport based on the rdata
It may be worth making some helper functions in res_sip that do these tasks centrally. No need to address that in this review though.

- Mark Michelson


On June 11, 2013, 11:03 a.m., Joshua Colp wrote:
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> https://reviewboard.asterisk.org/r/2614/
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> (Updated June 11, 2013, 11:03 a.m.)
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> 
> Review request for Asterisk Developers.
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> 
> Bugs: ASTERISK-21434
>     https://issues.asterisk.org/jira/browse/ASTERISK-21434
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> Repository: Asterisk
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> Description
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> This change adds an anonymous endpoint identifier which is loaded after all other endpoints, allowing anonymous calling (if configured).
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> The module works by initially searching for an endpoint named anonymous at domain, followed by anonymous. If either endpoint exists the endpoint is returned. If neither endpoint exists no endpoint is returned and anonymous calling is not permitted. The module can also be unloaded to allow no possibility of anonymous calling.
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> Diffs
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>   /team/group/pimp_my_sip/res/res_sip.c 391193 
>   /team/group/pimp_my_sip/res/res_sip_endpoint_identifier_anonymous.c PRE-CREATION 
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> Diff: https://reviewboard.asterisk.org/r/2614/diff/
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> Testing
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> Configured a softphone with a dummy username, placed call to chan_pjsip, confirmed anonymous used.
> Called from phone using valid username, confirmed valid endpoint used.
> Unloaded anonymous endpoint identifier, confirmed call rejected.
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> 
> Thanks,
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> Joshua Colp
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>

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