[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:all-tabpanel ]

Geert Van Pamel updated ASTERISK-17179:
---------------------------------------

          Description: 
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

I have solved the problem by patching chan_sip.c -- see rcsdiff -u below

Actually I found out that Asterisk 1.6.1.20 is indeed not conform to the RFC 3966 standard.

I have written a patch for Asterisk 1.6.1.20 chan-sip.c to support the TEL URI INVITE standard for incoming calls.

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.

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

****** ADDITIONAL INFORMATION ******

<code removed>
Please always add code as an attachment and mark as code.
</code removed>

  was:
This problem also exists in version 1.8!

Asterisk seems *not* to support RFC 3966 TEL URI for incoming INVITE. 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

I have solved the problem by patching chan_sip.c -- see rcsdiff -u below

Actually I found out that Asterisk 1.6.1.20 is indeed not conform to the RFC 3966 standard.

I have written a patch for Asterisk 1.6.1.20 chan-sip.c to support the TEL URI INVITE standard for incoming calls.

I have changed the following functions:

* parse_uri
* check_user_full
* get_destination

Now IMS and Asterisk are talking to each other without problems.

When there is no domain as with TEL INVITE we should set name to the calling number (I simply reversed dom and name in the original source for TEL INVITE leaving the domain blank). It was already suggested in the code that there is confusion with the RFC.



More information:

http://forums.digium.com/viewtopic.php?f=1&t=76432&sid=6d53062361c22079757c53ccc73d3132

****** ADDITIONAL INFORMATION ******

<code removed>
Please always add code as an attachment and mark as code.
</code removed>

    Affects Version/s: 11.5.1
    
> [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
>         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-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
> I have solved the problem by patching chan_sip.c -- see rcsdiff -u below
> Actually I found out that Asterisk 1.6.1.20 is indeed not conform to the RFC 3966 standard.
> I have written a patch for Asterisk 1.6.1.20 chan-sip.c to support the TEL URI INVITE standard for incoming calls.
> 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.
> 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
> ****** ADDITIONAL INFORMATION ******
> <code removed>
> Please always add code as an attachment and mark as code.
> </code removed>

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