[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...
> 
> Olle E Johansson wrote:
>     So it ends up in the SIPDOMAIN variable in the dial plan? It has to be reachable in the dial plan somehow.
> 
> Geert Van Pamel wrote:
>     The variable ${SIPDOMAIN} contains the local IP address of the Asterisk server.
>     The userinfo arrives in ${CALLERID} and is displayed on the display of the called device, and arrives in the CDR file.
>     Actually I do not know into which variable the incoming hostport info is copied to?
>     Could somebody else answer this question?

If I place a normal call to sip:geert at example.com to my Asterisk server. "geert" will be the extension I'm looking for, "example.com" ends up in SIPDOMAIN. It's not the local IP address, it's the domain/host part of the request URI in the INVITE.

I would prefer if phone context ended up in TELPHONECONTEXT so I could use it the same way as SIPDOMAIN in the dial plan. It should not end up in SIPDOMAIN as it is not a SIP uri. That way an extension in a local context can be routed differently than an extension in a global context.


- Olle E


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On March 22, 2014, 2:08 p.m., Geert Van Pamel wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> 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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