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

[ast (JIRA)]

ast created ASTERISK-28175:
------------------------------

             Summary: 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
         Environment: Ubuntu 18.0.1 LTS
            Reporter: ast
            Severity: Critical


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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