[asterisk-bugs] [JIRA] (ASTERISK-25414) CLONE - [patch] IMS TEL URI incoming INVITE RFC 3966 not recognized

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25414?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227647#comment-227647 ] 

Asterisk Team commented on ASTERISK-25414:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> CLONE - [patch] IMS TEL URI incoming INVITE RFC 3966 not recognized
> -------------------------------------------------------------------
>
>                 Key: ASTERISK-25414
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25414
>             Project: Asterisk
>          Issue Type: Improvement
>          Components: Channels/chan_sip/Interoperability
>    Affects Versions: 11.5.1, 12.0.0
>         Environment: All platforms
>            Reporter: Alexander Gonchiy
>              Labels: INVITE, PATCH, RFC3966, RFC5341, SIP, TEL, URI
>
> 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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