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

[Richard Mudgett (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-17179?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227654#comment-227654 ] 

Richard Mudgett commented on ASTERISK-17179:
--------------------------------------------

If you look at the Source tab above you will see that this patch was committed to trunk on April 17, 2014.  The v13 branch has since been branched from trunk so it contains this patch.  Since it is a new feature, it will not go into Asterisk v11.

> [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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