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  1. FreePBX
  2. FREEPBX-6618

Support RFC 3966 TEL URI for incoming INVITE.

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    Details

    • Type: Bug
    • Status: Closed
    • Resolution: Not an issue
    • Affects Version/s: 2.10
    • Fix Version/s: None
    • Component/s: Asterisk SIP Settings
    • Labels:
      None
    • Distro:
      FreePBX Distro

      Description

      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@...>
      Diversion: <sip:+3212345678@...;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

      IT SEEMS there's a patch for Asterisk 1.8.13.1 to implement incoming TEL URI INVITE RFC 3966 functionality.

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              • Assignee:
                plindheimer PL
                Reporter:
                thecalle TheCalle
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                Dates

                • Created:
                  Updated:
                  Resolved:

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