> On March 21, 2014, 7:01 p.m., Corey Farrell wrote:
> > /trunk/channels/sip/reqresp_parser.c, line 130
> > <https://reviewboard.asterisk.org/r/3349/diff/7-8/?file=56285#file56285line130>
> >
> > This needs to blank both variables:
> > userinfo = uri = "";
>
> Geert Van Pamel wrote:
> We return the local number anyway when an incoming RFC 3966 TEL URI
> INVITE call
> does not contain a global number nor a phone-context.
>
> Corey Farrell wrote:
> First sentence of 3rd paragraph of section 5.1.5:
> Local numbers MUST have a 'phone-context' parameter that identifies the
> scope of their validity.
>
> Note the word "MUST", this has specific meaning in RFC's. I will not
> approve this review if it's going to contradict the RFC it's claiming to
> implement.
>
> Olle E Johansson wrote:
> You have to be strict in what you send, but open for receiving stuff that
> doesn't always follow the RFC. We can add an option that sets strictness. I
> haven't seen many implementations of Tel: uri's sadly, but many of the few
> did not follow the RFC.
>
>
>
> Corey Farrell wrote:
> If that is the case then should we not return error = -1? As for
> optional strictness maybe use sip_settings.pedanticsipchecking?
I do return both the local number, and throw the error -1:
ast_debug(1, "No RFC 3966 global number or context found in '%s';
returning local number anyway\n", uri);
userinfo = uri; /* Return local number anyway */
error = -1;
This would take care of both alerting the non RFC-compliance, and allowing some
openness for receiving stuff that doesn't follow strictly the RFC...
- Geert
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3349/#review11314
-----------------------------------------------------------
On March 22, 2014, 2:08 p.m., Geert Van Pamel wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3349/
> -----------------------------------------------------------
>
> (Updated March 22, 2014, 2:08 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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