[asterisk-bugs] [JIRA] (ASTERISK-28175) PJSIP support for TEL (RFC 3966)

[Joshua C. Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28175?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua C. Colp closed ASTERISK-28175.
-------------------------------------

    Resolution: Not A Bug

Feature requests are not accepted on the issue tracker[1] at this time.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Howtorequestafeature

> PJSIP support for TEL (RFC 3966)
> --------------------------------
>
>                 Key: ASTERISK-28175
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28175
>             Project: Asterisk
>          Issue Type: New Feature
>      Security Level: None
>          Components: Resources/res_pjsip_session
>    Affects Versions: 16.0.0, 16.0.1
>         Environment: Ubuntu 18.0.1 LTS
>            Reporter: ast
>            Severity: Critical
>              Labels: pjsip
>
> Essentially set up pjsip.conf to register to ITSP and made an endpoint to receive incoming calls.
> Unfortunately received response advising that TEL is unknown or at least not a ‘SIP’ URI, actual SIP error phrase is “416 Unsupported URI Scheme”.
> Chan_SIP is RFC 3966 compliant from what I can see but i would like to use PJSIP because there are more features, and after trialing running both together the system is unreliable. My preference is to just use PJSIP.
>     Optus NBN Australia
>     PJSIP Logging enabled
>     <— Received SIP request (1697 bytes) from UDP:210.49.225.101:5060 —>
>     INVITE sip:+61732103210 at 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 at anonymous.invalid;tag={weirdnumbers}
>     Call-ID: {weirdnumbers}@10.194.0.25
>     CSeq: 1 INVITE
>     Contact: sip:sgc_c at 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: {weirdnumber}@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 at 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 at 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:+61732103210 at 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:+61732103210;tag={weirdnumbers}
>     From: “Anonymous” sip:anonymous at anonymous.invalid;tag={weirdnumbers}
>     Call-ID: {weirdnumbers}@10.194.0.25
>     CSeq: 1 ACK
>     Content-Length: 0
>     *CLI> pjsip set logger off
>     PJSIP Logging disabled
> I’m hoping someone whos awsome and a wizard with Asterisk can help me out with a configuration setting or advise if its possible to patch the source code for Asterisk 16.0.1 to permit the correct handling of a tel: uri invite.



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