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Summary:ASTERISK-28174: PJSIP issue with TEL (RFC 3966)
Reporter:ast (ast@nashdl.com)Labels:pjsip
Date Opened:2018-11-20 06:21:51.000-0600Date Closed:2018-11-20 06:28:06.000-0600
Priority:CriticalRegression?Yes
Status:Closed/CompleteComponents:pjproject/pjsip Resources/res_pjsip_session
Versions:16.0.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Ubuntu 18.0.1 LTAttachments:
Description:Essentially set up pjsip.conf to register to ITSP and made an endpoint.

Received response advising that TEL is unknown or at least no SIP, actual SIP phrase is "416 Unsupported URI Scheme".


PJSIP Logging enabled
<--- Received SIP request (1697 bytes) from UDP:210.49.225.101:5060 --->
INVITE sip:+6173210321@192.168.1.100:5060;line={weirdnumbers} SIP/2.0
Max-Forwards: 65
Via: SIP/2.0/UDP 210.49.225.101:5060;branch={weirdnumbers}
To: "SIPLineUser SIPLineUser" <tel:+6173210321>
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag={weirdnumbers}
Call-ID: {weirdnumbers}@10.194.0.25
CSeq: 1 INVITE
Contact: <sip:sgc_c@210.49.225.101;transport=udp>
Record-Route: <sip:210.49.225.101;transport=udp;lr>
Min-Se: 900
Privacy: id
Session-Expires: 1800
Supported: com.nortelnetworks.firewall
Supported: p-3rdpartycontrol
Supported: nosec
Supported: join
Supported: x-nortel-sipvc
Supported: gin
Supported: com.nortelnetworks.im.encryption
Supported: 100rel
Supported: resource-priority
Supported: replaces
User-Agent: Nortel SESM 19.0.1.0
Content-Type: application/sdp
Content-Length: 574
X-Nt-Service: brdplayed=yes
X-Nt-Corr-Id: 40bc80137637c1f4413c5808d06ff1b42a3bacca3@10.194.0.25
X-Nortel-Profile: DEFAULT
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, INFO, SUBSCRIBE, REFER, NOTIFY, PRACK, UPDATE

v=0
o=- 3257871432 3257871432 IN IP4 210.49.225.101
s=-
e=unknown@invalid.net
c=IN IP4 210.49.123.41
t=0 0
m=audio 48426 RTP/AVP 8 0 18 101 110 111
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=maxptime:20
a=3gOoBTC
a=rtpmap:110 AMR/8000
a=fmtp:110 mode-change-period=2; mode-change-capability=2; mode-change-neighbor=1; max-red=0
a=rtpmap:111 AMR/8000
a=fmtp:111 octet-align=1; mode-change-period=2; mode-change-capability=2; mode-change-neighbor=1; max-red=0

<--- Transmitting SIP response (534 bytes) to UDP:210.49.225.101:5060 --->
SIP/2.0 416 Unsupported URI Scheme
Via: SIP/2.0/UDP 210.49.225.101:5060;received=210.49.225.101;branch={weirdnumbers}
Record-Route: <sip:210.49.225.101;transport=udp;lr>
Call-ID: {weirdnumbers}@10.194.0.25
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag={weirdnumbers}
To: "SIPLineUser SIPLineUser" <tel:+6173210321>;tag={weirdnumbers}
CSeq: 1 INVITE
Server: "{User_Agent}"
Content-Length:  0


<--- Received SIP request (489 bytes) from UDP:210.49.225.101:5060 --->
ACK sip:+6173210321@192.168.2.22:5060;line={weirdnumbers} SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 210.49.225.101:5060;received=210.49.225.101;branch={weirdnumbers}
To: "SIPLineUser SIPLineUser" <tel:+6173210321>;tag={weirdnumbers}
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag={weirdnumbers}
Call-ID: {weirdnumbers}@10.194.0.25
CSeq: 1 ACK
Content-Length: 0


lappy*CLI> pjsip set logger off
PJSIP Logging disabled
Comments:By: Asterisk Team (asteriskteam) 2018-11-20 06:21:53.919-0600

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) 2018-11-20 06:28:07.110-0600

The tel URI is not currently supported in PJSIP and noone is actively working on it. If you'd like to provide a fully tested patch with test coverage, then this could be reopened otherwise this is considered a new feature request.
By: ast (ast@nashdl.com) 2018-11-20 07:14:46.716-0600

Thank you, any suggestions on where to start in the code or where to look for some developer documentation so i can understand how the functions fit to gether?
By: Asterisk Team (asteriskteam) 2018-11-20 07:14:46.869-0600

This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable.
By: ast (ast@nashdl.com) 2018-11-20 07:26:34.103-0600

Raised as a new feature request.