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Summary:ASTERISK-26894: pjsip should support tel uri scheme
Reporter:Gergely Dömsödi (doome)Labels:patch pjsip
Date Opened:2017-03-23 08:45:06Date Closed:2022-09-13 04:51:06
Priority:MajorRegression?
Status:Closed/CompleteComponents:Resources/res_pjsip_session
Versions:Frequency of
Occurrence		Constant
Related
Issues:
is duplicated byASTERISK-27895 chan_pjsip: 'tel' URI is unsupported
Environment:Attachments:( 0) tel.patch
Description:When {{res_pjsip}} receives an INVITE with tel: request uris (as defined in rfc3966, rfc4694), it responds with {{416 Unsupported URI Scheme}} even though the underlying PJSIP stack supports it. chan_sip also supports it, the work was done at issue ASTERISK-17179.
Comments:By: Asterisk Team (asteriskteam) 2017-03-23 08:45:07.316-0500

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].
By: Joshua C. Colp (jcolp) 2017-03-23 08:48:52.850-0500

Features requests without patches are not accepted through the issue tracker. Features requests are openly discussed on the mailing lists, forums, and IRC [1]. Please see the Asterisk Issue Guidelines [2] for more information on feature request and patch submission.

[1] http://asterisk.org/community/discuss
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines


By: m0bius (m0bius) 2018-05-07 04:14:05.076-0500

This patch could help in bypassing the sip message filter in asterisk's res_pjsip blocking the tel: header from not working.

Might not be the best solution however. I am not sure if the tel: uri could contain the `x-ast-txp` parameter and if it should be sanitized
By: George Joseph (gjoseph) 2018-05-07 09:01:20.655-0500

While this patch removes the check for tel: URIs there is no support for tel: URIs in chan_pjsip at all so crashes will result.

You should bring this issue up on the asterisk-dev mailing list or in the #asterisk-dev IRC channel.


By: m0bius (m0bius) 2018-05-07 09:13:11.893-0500

I've looked into the PJSIP source code (2.7.2) and it appears that it does handle tel: uris, so I went ahead and tried the above patch to a server and so far I've not experienced any crashes due to it.

The only thing I've noticed is that it encodes the ascii characters in the P-Asserted-Identity and Remote-Party-Id responses if send rpid and send pai are enabled, but this might not be related to the above workaround

However, even if there is no intention on providing tel: support for the URIs, silently dropping the call without any NOTICE or WARNING is a major issue, since someone might be already affected by this, and not know it at all. I don't think there is any other similar case where asterisk drops sip calls without any warning

I'll connect to the IRC channel and bring it up.

By: Abhay Gupta (agupta) 2018-06-05 04:29:43.195-0500

Pls let me know why TEL URI is not supported by PJSIP asterisk . I am using asterisk 15.3 and on INVITE asterisk is sending

SIP/2.0 416 Unsupported URI Scheme

whereas RFC 3966 allows it and PJSIP 2.7.1 allows the same as well . Should i raise a new ticket for this BUG
By: Abhay Gupta (agupta) 2018-06-05 04:35:01.876-0500

<--- Received SIP request (1210 bytes) from UDP:10.232.130.170:5060 --->
INVITE sip:+911244310700@10.126.105.40:5060 SIP/2.0
Via: SIP/2.0/UDP 10.232.130.170:5060;branch=z9hG4bKuw6xxu92yr3uzzx25186w863w;Role=3;Hpt=8fb2_36;TRC=ffffffff-16b9
Call-ID: asbcjy71615rj11ffks55765ps15y5y8jf3s@B.5.203.ims.airtel.in
From: "08802809405"<tel:8802809405;noa=subscriber;srvattri=national;phone-context=+91>;tag=c589dd9l
To: <tel:+911244310700>
CSeq: 1 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE
Contact: <sip:10.232.130.170:5060;Dpt=eeba-200;Hpt=8fb2_16;CxtId=4;TRC=ffffffff-16b9>
Max-Forwards: 65
Supported: timer,100rel,histinfo
Session-Expires: 1800
Min-SE: 600
P-Asserted-Identity: <tel:08802809405>
P-Charging-Vector: icid-value=AE880F23FCFF6201865132030;orig-ioi=10.232.128.242;term-ioi=SIP_ZOMATO_4310700
Content-Length: 376
Content-Type: application/sdp

v=0
o=- 58669091 58669091 IN IP4 10.232.130.179
s=SBC call
c=IN IP4 10.232.130.179
t=0 0
m=audio 12422 RTP/AVP 108 102 8 0 18 97
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 telephone-event/8000
a=ptime:20
a=maxptime:20
a=3gOoBTC

<--- Transmitting SIP response (475 bytes) to UDP:10.232.130.170:5060 --->
SIP/2.0 416 Unsupported URI Scheme
Via: SIP/2.0/UDP 10.232.130.170:5060;received=10.232.130.170;branch=z9hG4bKuw6xxu92yr3uzzx25186w863w;Role=3;Hpt=8fb2_36;TRC=ffffffff-16b9
Call-ID: asbcjy71615rj11ffks55765ps15y5y8jf3s@B.5.203.ims.airtel.in
From: "08802809405" <tel:8802809405;phone-context=+91;noa=subscriber;srvattri=national>;tag=c589dd9l
To: <tel:+911244310700>;tag=z9hG4bKuw6xxu92yr3uzzx25186w863w
CSeq: 1 INVITE
Server: Asterisk PBX 15.3.0
Content-Length:  0
By: Abhay Gupta (agupta) 2018-06-05 05:51:53.486-0500

PJSIP supports PJSIP_URI_SCHEME_IS_TEL(uri) but from asterisk everywhere the check of TEL is removed

By: Friendly Automation (friendly-automation) 2022-09-13 04:51:08.934-0500

Change 19236 merged by Friendly Automation:
res_pjsip: Add TEL URI support for basic calls.

[https://gerrit.asterisk.org/c/asterisk/+/19236|https://gerrit.asterisk.org/c/asterisk/+/19236]
By: Friendly Automation (friendly-automation) 2022-09-13 04:51:21.556-0500

Change 19237 merged by Friendly Automation:
res_pjsip: Add TEL URI support for basic calls.

[https://gerrit.asterisk.org/c/asterisk/+/19237|https://gerrit.asterisk.org/c/asterisk/+/19237]
By: Friendly Automation (friendly-automation) 2022-09-13 04:51:40.636-0500

Change 18892 merged by Friendly Automation:
res_pjsip: Add TEL URI support for basic calls.

[https://gerrit.asterisk.org/c/asterisk/+/18892|https://gerrit.asterisk.org/c/asterisk/+/18892]
By: Friendly Automation (friendly-automation) 2022-09-13 04:51:44.685-0500

Change 19208 merged by Friendly Automation:
res_pjsip: Add TEL URI support for basic calls.

[https://gerrit.asterisk.org/c/asterisk/+/19208|https://gerrit.asterisk.org/c/asterisk/+/19208]
By: Friendly Automation (friendly-automation) 2022-09-13 04:51:54.366-0500

Change 19235 merged by Friendly Automation:
res_pjsip: Add TEL URI support for basic calls.

[https://gerrit.asterisk.org/c/asterisk/+/19235|https://gerrit.asterisk.org/c/asterisk/+/19235]