[asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE

[Geert Van Pamel]

> On March 17, 2014, 9:39 a.m., wdoekes wrote:
> > /trunk/channels/sip/reqresp_parser.c, lines 100-105
> > <https://reviewboard.asterisk.org/r/3349/diff/4/?file=56142#file56142line100>
> >
> >     I don't like it that we skip past the parameters.
> >     
> >     If we have:
> >     
> >       tel:123;param1=X;phone-context=Y;param2=Z
> >     
> >     Then *parameters will lose out on param1. Lose the uri=c+15.
> 
> wdoekes wrote:
>     But obviously that would break because of the *c='\0'.
>     
>     In that case:
>     
>     - does the tel uri ever get any parameters other than ;phone-context?
>     - if it doesn't, I'd rather drop all parameters than only take those that come *before* the phone-context.
>     
>     I don't mind a shortcut in this case, but at least the source should clearly document what we're silently ignoring.
>

Parameters before ";phone-context=" will currently make part of *userinfo. Normally they will be only used for local dialling (i.e. ";isub=" or ";ext=" are used for ISDN or PSTN subaddressing).

All other parameters should be encoded after the ";phone-context=" parameter, to arrive into *parameters.

We believe that RFC 3966 has been implemented correctly. For more info, see http://www.ietf.org/rfc/rfc3966.txt


- Geert


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On March 17, 2014, 8:01 p.m., Geert Van Pamel wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3349/
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> 
> (Updated March 17, 2014, 8:01 p.m.)
> 
> 
> Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes.
> 
> 
> Bugs: ASTERISK-17179
>     https://issues.asterisk.org/jira/browse/ASTERISK-17179
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Implements RFC-3966 TEL URI incoming INVITE.
> 
> See https://issues.asterisk.org/jira/browse/ASTERISK-17179 for a description of the original isssue.
> 
> I have been patching all versions since Asterisk 1.6. I would like to include the code into the main trunk for version 13.
> 
> Previously Asterisk was failing with error on incoming IMS call:
> 
> Nov 13 17:52:05 NOTICE[27459]: chan_sip.c:6973 check_user_full: From address missing 'sip:', using it anyway
> 
> Nov 13 17:52:05 WARNING[27459]: chan_sip.c:6525 get_destination: Huh? Not a SIP header (tel:0987654321;phone-context=+32987654321)?
> 
> Reason: tel: protocol was not recognized.
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/sip/reqresp_parser.c 410429 
>   /trunk/channels/chan_sip.c 410429 
> 
> Diff: https://reviewboard.asterisk.org/r/3349/diff/
> 
> 
> Testing
> -------
> 
> Executed an incoming TEL URI INVITE connection.
> CLI was present on the display and in the CDR file.
> No errors on SIP debug output.
> 
> 
> File Attachments
> ----------------
> 
> RFC-3966 tel URI patch
>   https://reviewboard.asterisk.org/media/uploaded/files/2014/03/13/cad7a996-88c1-47fe-a2a9-cc6987af3b75__rfc-3966-tel-uri-patch-diff.txt
> 
> 
> Thanks,
> 
> Geert Van Pamel
> 
>

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