> On Feb. 23, 2014, 9:04 p.m., Corey Farrell wrote: > > /trunk/channels/sip/reqresp_parser.c, lines 103-118 > > <https://reviewboard.asterisk.org/r/3250/diff/1/?file=54390#file54390line103> > > > > I feel this section should only apply when scheme is 'tel:'. I'm > > concerned with changes to how sip URI's are handled. For example: > > sip:example.com;phone-context=spoof.domain.com > > sip:+example.com > > > > The first URI should result in hostport="example.com", userinfo="". > > This change causes it to be hostport="spoof.domain.com", > > userinfo="example.com". > > The second URI should result in the invalid hostport "+example.com", > > where this puts the value in userinfo. > > > > What happens to invalid tel: URI's? For example "tel:10000" - no > > phone-context or + would cause 10000 to be used as hostport (like in SIP > > uri). > > > > I'd like to see test cases added to sip_parse_uri_full_test and/or > > sip_parse_uri_test. At minimum the tests need to verify no change in > > results for URI scheme sip. > > wdoekes wrote: > Thanks for the speedy response :) > > Let me forward your concerns.
I have uploaded a new patch for TEL URI reqresp_parser.c to https://issues.asterisk.org/jira/browse/ASTERISK-17179 see asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt. Taking into account the remarks from Corey Farrell. Sorry, but I do not know (yet) how to use the Review board... If there would be an invalid TEL URI (e.g. no phone-context, nor +global-number) then an error indicating such failure is issued. - Geert ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3250/#review10929 ----------------------------------------------------------- On Feb. 23, 2014, 12:17 p.m., wdoekes wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3250/ > ----------------------------------------------------------- > > (Updated Feb. 23, 2014, 12:17 p.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-17179 > https://issues.asterisk.org/jira/browse/ASTERISK-17179 > > > Repository: Asterisk > > > Description > ------- > > This patch is filed on behalf of Geert Van Pamel as filed against Asterisk-12 > on ASTERISK-17179. It was cleaned up by me. > > The patch should allow incoming INVITEs with a tel: uri. An "IMS" server > apparently uses it. > > Geert would appreciate it if this was looked at and checked in, so he won't > have to patch Asterisk 13. He has been patching this since Asterisk 1.6.2.x. > > > Diffs > ----- > > /trunk/channels/sip/reqresp_parser.c 408868 > /trunk/channels/chan_sip.c 408868 > > Diff: https://reviewboard.asterisk.org/r/3250/diff/ > > > Testing > ------- > > Not by me. It compiles. I'm just filing it because Geert doesn't have an > account and I understand his frustration. > > > Thanks, > > wdoekes > >
-- _____________________________________________________________________ -- 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
