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

[Olle E Johansson]

> On March 22, 2014, 4:39 p.m., Olle E Johansson wrote:
> > I don't see what happens with the phone-context argument. Shouldn't we pass that on as a channel variable?
> 
> Geert Van Pamel wrote:
>     We return this into the hostport.
> 
> Geert Van Pamel wrote:
>     According to RFC 3966 phone-context is either a domain-name, or (part of) an international telephone number (indicated with +prefix).
>     It is used by a gateway to know how to dial the "local" number... the local number must be unique within its context...

So it ends up in the SIPDOMAIN variable in the dial plan? It has to be reachable in the dial plan somehow. 


- Olle E


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On March 22, 2014, 2:08 p.m., Geert Van Pamel wrote:
> 
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> https://reviewboard.asterisk.org/r/3349/
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> 
> (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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