[asterisk-dev] [Code Review] 3250: chan_sip: Add incoming tel: uri support (rfc3966)

[Corey Farrell]

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/trunk/channels/sip/reqresp_parser.c
<https://reviewboard.asterisk.org/r/3250/#comment20556>

    I feel this section should only apply when scheme is 'tel:'.  I'm concerned with changes to how sip URI's are handled.  For example:
    sip:example.com;phone-context=spoof.domain.com
    sip:+example.com
    
    The first URI should result in hostport="example.com", userinfo="".  This change causes it to be hostport="spoof.domain.com", userinfo="example.com".
    The second URI should result in the invalid hostport "+example.com", where this puts the value in userinfo.
    
    What happens to invalid tel: URI's?  For example "tel:10000" - no phone-context or + would cause 10000 to be used as hostport (like in SIP uri).
    
    I'd like to see test cases added to sip_parse_uri_full_test and/or sip_parse_uri_test.  At minimum the tests need to verify no change in results for URI scheme sip.


- Corey Farrell


On Feb. 23, 2014, 6:17 a.m., wdoekes wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
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> (Updated Feb. 23, 2014, 6:17 a.m.)
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> 
> Review request for Asterisk Developers.
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> Bugs: ASTERISK-17179
>     https://issues.asterisk.org/jira/browse/ASTERISK-17179
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> Repository: Asterisk
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> Description
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> This patch is filed on behalf of Geert Van Pamel as filed against Asterisk-12 on ASTERISK-17179. It was cleaned up by me.
> 
> The patch should allow incoming INVITEs with a tel: uri. An "IMS" server apparently uses it.
> 
> Geert would appreciate it if this was looked at and checked in, so he won't have to patch Asterisk 13. He has been patching this since Asterisk 1.6.2.x.
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> Diffs
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>   /trunk/channels/sip/reqresp_parser.c 408868 
>   /trunk/channels/chan_sip.c 408868 
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> Diff: https://reviewboard.asterisk.org/r/3250/diff/
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> 
> Testing
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> Not by me. It compiles. I'm just filing it because Geert doesn't have an account and I understand his frustration.
> 
> 
> Thanks,
> 
> wdoekes
> 
>

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