> 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?
>
> Olle E Johansson wrote:
> If I place a normal call to sip:ge...@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 Johansson wrote:
> Just to make sure it's documented: The phone-context should NOT be
> matched to a context in the dial plan. From a security standpoint, that would
> be horrific. We need the information to be able to route correctly in the
> dial plan, but no automatic selection. (I am not suggesting you where going
> down that path, Geert. Just wanted to close that way of thinking since the
> word "context" could lead there.)
>
> Geert Van Pamel wrote:
> So I understand that Olle proposes to create a new variable
> ${TELPHONECONTEXT} that contains the TEL URI phone-context which can be
> either the global number, or a global number prefix, or the related routing
> domain or any other unique routing identifier, or would be empty in case
> there is just a local number (as specified in RFC 3966).
>
> Currently this variable is not available yet in Asterisk. In the dialplan
> treating incoming calls currently I do not use nor do not need this
> information, as the local number in ${CALLERID} is sufficient (for the time
> being)... Still this phone context identifier could be needed for subsequent
> outgoing calls (return calls, callbacks, etc.).
>
> I agree that ${SIPDOMAIN} will remain reserved for SIP invites, and is
> untouched for TEL URI invites.
>
> I perfectly understand that this TEL URI context has nothing to do with
> dialplan context.
>
> Who could us further advise how to create the new variable
> ${TELPHONECONTEXT} ?
Just grep for SIPDOMAIN in the chan_sip source code :-)
- Olle E
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3349/#review11323
-----------------------------------------------------------
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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