----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3250/#review10929 -----------------------------------------------------------
/trunk/channels/sip/reqresp_parser.c <https://reviewboard.asterisk.org/r/3250/#comment20556> 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. - Corey Farrell On Feb. 23, 2014, 6:17 a.m., wdoekes wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3250/ > ----------------------------------------------------------- > > (Updated Feb. 23, 2014, 6:17 a.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
