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

[wdoekes]

> On Feb. 23, 2014, 8:04 p.m., Corey Farrell wrote:
> > /trunk/channels/sip/reqresp_parser.c, lines 103-118
> > <https://reviewboard.asterisk.org/r/3250/diff/1/?file=54390#file54390line103>
> >
> >     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.

Thanks for the speedy response :)

Let me forward your concerns.


- wdoekes


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3250/#review10929
-----------------------------------------------------------


On Feb. 23, 2014, 11:17 a.m., wdoekes wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3250/
> -----------------------------------------------------------
> 
> (Updated Feb. 23, 2014, 11:17 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-17179
>     https://issues.asterisk.org/jira/browse/ASTERISK-17179
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> 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.
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/sip/reqresp_parser.c 408868 
>   /trunk/channels/chan_sip.c 408868 
> 
> Diff: https://reviewboard.asterisk.org/r/3250/diff/
> 
> 
> Testing
> -------
> 
> Not by me. It compiles. I'm just filing it because Geert doesn't have an account and I understand his frustration.
> 
> 
> Thanks,
> 
> wdoekes
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140223/baffccd3/attachment-0001.html>