----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/#review11227 -----------------------------------------------------------
Ship it! One very minor tweak, this looks good. Geert you had asked about how to ship it. The general agreement is we wait atleast 24 hours to commit after "Ship It". So if nobody objects by Monday afternoon I will commit this to trunk. I know 24 hours would be this afternoon but this is chan_sip, so I'm giving it longer. /trunk/channels/sip/reqresp_parser.c <https://reviewboard.asterisk.org/r/3349/#comment20859> scheme is the input parameter listing acceptable schemes, we don't need to see it here. The other ast_debug included scheme since the problem was a failure to match the uri with any scheme. - Corey Farrell On March 15, 2014, 8:42 a.m., Geert Van Pamel wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3349/ > ----------------------------------------------------------- > > (Updated March 15, 2014, 8:42 a.m.) > > > Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt > Jordan, and wdoekes. > > > 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 > ----- > > /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 > >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
