[asterisk-bugs] [JIRA] (ASTERISK-17179) [patch] IMS TEL URI incoming INVITE RFC 3966 not recognized

[Geert Van Pamel (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-17179?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=215473#comment-215473 ] 

Geert Van Pamel commented on ASTERISK-17179:
--------------------------------------------

Slashi, glad to hear that it works on your site.

Well, yes, the RFC 3966 TEL URI standard exists since 2004... time that the main Asterisk distro would implement it, no? -- standard exists since 10 years now... Technology adherence is not always that fast...

In 2012 I was in contact with Matt Jordan, and he proposed to put this path on the Asterisk Jira "Review board".

But I never could figure out how to do this... since then I implemented always a patch for every new release... a bit of embarrassing...

Now that we are already with at least 2 persons seeing the real benefit of this (standard, but missing) functionality do you know what steps needs to be taken to put this Issue on the review board?

See also https://twitter.com/asteriskpbx/status/434457042141859840
                
> [patch] IMS TEL URI incoming INVITE RFC 3966 not recognized
> -----------------------------------------------------------
>
>                 Key: ASTERISK-17179
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-17179
>             Project: Asterisk
>          Issue Type: Improvement
>          Components: Channels/chan_sip/Interoperability
>    Affects Versions: 11.5.1, 12.0.0
>         Environment: All platforms
>            Reporter: Geert Van Pamel
>              Labels: INVITE, PATCH, RFC3966, RFC5341, SIP, TEL, URI
>         Attachments: asterisk_10.1.3_chan_sip_diff.txt, asterisk_10.1.3_reqresp_parser_diff.txt, asterisk-11.5.1-chan_sip-diff.txt, asterisk-11.5.1-reqresp_parser-diff.txt, asterisk-12.0.0-chan_sip-RFC3966_patch.txt, asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt, asterisk-1.6.2.7-sip_chan.dif, asterisk-1.8.13.1-chan_sip-diff.txt, asterisk-1.8.13.1-reqresp_parser-diff.txt, chan_sip-asterisk_1.6.2.9-2ubuntu2.1-diff.txt, chan_sip-asterisk-1.6.2.9.txt, chan_sip-diff.txt
>
>
> This problem exists in ALL versions of Asterisk.
> Asterisk seems *not* to support RFC 3966 TEL URI for INCOMING INVITEs. X-Lite and other clients like Bria are compliant with RFC 3966.
> When an IMS server sends an incoming TEL URI INVITE I get the following errors, and the incoming call is disconnected (number busy).
> Here you find part of an (incoming) INVITE request and sip debug output:
> From: <*tel:0987654321;phone-context=+32987654321*>;tag=tag-etc
> CSeq: 1 INVITE
> P-Asserted-Identity: <tel:0987654321>
> P-Called-Party-ID: <sip:+3212345678 at ...>
> Diversion: <sip:+3212345678 at ...;user=phone>;reason="extension";privacy="off";counter=1
> Using INVITE request as basis request -
> 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*)?
> RDNIS is +3212345678
> SIP/2.0 404 Not Found
> Actually I found out that Asterisk is indeed not conform to the RFC 3966 standard.
> I have solved the problem by patching chan_sip.c and reqresp_parser.c -- see patch in code attachments.
> I have changed the following functions:
> * check_user_full
> * get_destination
> * parse_uri OR parse_uri_full (depending on the Asterisk version)
> When ;phone-context= is provided in the incoming tel:uri then we can extract the calling number for further call handling.
> Now IMS and Asterisk are talking to each other without problems.
> More information:
> http://forums.digium.com/viewtopic.php?f=1&t=76432&sid=6d53062361c22079757c53ccc73d3132

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