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(Updated March 17, 2014, 8:01 p.m.) Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes. Changes ------- Moved the test section for invalid TEL URI to sip_uri_parse_test (solving the remark from Corey). Parameters before ";phone-context=" will currently make part of *userinfo. Normally they will be only used for local dialling (i.e. ";isub=" or ";ext=" are used for ISDN or PSTN subaddressing). All other parameters should be encoded after the ";phone-context=" parameter, to arrive into *parameters. We believe that RFC 3966 has been implemented correctly. For more info, see http://www.ietf.org/rfc/rfc3966.txt Bugs: ASTERISK-17179 https://issues.asterisk.org/jira/browse/ASTERISK-17179 Repository: Asterisk Description ------- Implements RFC-3966 TEL URI incoming INVITE. See https://issues.asterisk.org/jira/browse/ASTERISK-17179 for a description of the original isssue. I have been patching all versions since Asterisk 1.6. I would like to include the code into the main trunk for version 13. Previously Asterisk was failing with error on incoming IMS call: 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)? Reason: tel: protocol was not recognized. Diffs (updated) ----- /trunk/channels/sip/reqresp_parser.c 410429 /trunk/channels/chan_sip.c 410429 Diff: https://reviewboard.asterisk.org/r/3349/diff/ Testing ------- Executed an incoming TEL URI INVITE connection. CLI was present on the display and in the CDR file. No errors on SIP debug output. File Attachments ---------------- RFC-3966 tel URI patch https://reviewboard.asterisk.org/media/uploaded/files/2014/03/13/cad7a996-88c1-47fe-a2a9-cc6987af3b75__rfc-3966-tel-uri-patch-diff.txt Thanks, Geert Van Pamel
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