Re: [asterisk-dev] [Code Review] 4336: app_dial: Don't publish DialEnd events twice if GOSUB_RESULT or MACRO_RESULT return an unexpected value

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4336/ --- (Updated Jan. 16, 2015, 9:03 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 4341: stasis transfers: fix race condition on set replace channel

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4341/#review14214 --- I think the changes to res/stasis/app.c need to be reverted. T

Re: [asterisk-dev] [Code Review] 4335: res_pjsip_multihomed: Test to ensure correct message contents when multiple transports exist

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4335/#review14213 --- Ship it! /asterisk/trunk/tests/channels/pjsip/incoming_call_o

Re: [asterisk-dev] [Code Review] 4339: PJSIP: Prevent hung channel on a blind transfer

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4339/ --- (Updated Jan. 16, 2015, 4:12 p.m.) Status -- This change has been mar

Re: [asterisk-dev] [Code Review] 4341: stasis transfers: fix race condition on set replace channel

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4341/ --- (Updated Jan. 16, 2015, 3:58 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 4344: Add capath support to res_pjsip (new version of /r/4230)

Thanks for getting this finished! -JimC -- James Cloos OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options

Re: [asterisk-dev] [Code Review] 4328: res_pjsip: Document transport selection process

> "OEJ" == Olle E Johansson writes: OEJ> For dual stack support, we need to open two TCP or TLS connections OEJ> at the same time. I assume that means for a new call origination. If it is the result of an incoming sip session, and the rtp is expected to avoid the asterisk box, it is best in

Re: [asterisk-dev] [Code Review] 4330: Testsuite: Add external bridging tests for Stasis bridge (one channel) interactions

> On Jan. 14, 2015, 4:45 p.m., opticron wrote: > > /asterisk/trunk/tests/rest_api/external_interaction/ami_bridge/stasis_bridge/tests.yaml, > > line 2 > > > > > > The files for this test seem to be missing. > > jbig

Re: [asterisk-dev] [Code Review] 4344: Add capath support to res_pjsip (new version of /r/4230)

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4344/ --- (Updated Jan. 16, 2015, 11:45 a.m.) Status -- This change has been ma

Re: [asterisk-dev] [Code Review] 4331: res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4331/ --- (Updated Jan. 16, 2015, 5:05 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 4347: Investigate and fix memory leaks in Asterisk

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4347/#review14211 --- branches/13/channels/chan_iax2.c

Re: [asterisk-dev] [Code Review] 4346: Use SIPS Contact headers as prescribed by RFC 3261 (chan_sip)

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4346/ --- (Updated Jan. 16, 2015, 4:35 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 4345: Use SIPS Contact headers as prescribed by RFC 3261 (res_pjsip)

It turns out that Blink does not INVITE to SIPS URIs (and I should have known that based on saghul's comments [1]) for. [1] http://lists.digium.com/pipermail/asterisk-dev/2013-September/062561.html On 01/16/2015 09:21 AM, Mark Michelson wrote: On the reported issue, CSipSimple was a SIPS-usin

Re: [asterisk-dev] [Code Review] 4345: Use SIPS Contact headers as prescribed by RFC 3261 (res_pjsip)

On the reported issue, CSipSimple was a SIPS-using client that the reporter used. CSipSimple uses PJSua under the hood, so it may be common for PJSua-based clients (e.g. Blink) to use SIPS for secure calls. I'm in a different environment today and I might be able to test with Blink myself. O

Re: [asterisk-dev] [Code Review] 4346: Use SIPS Contact headers as prescribed by RFC 3261 (chan_sip)

> On Jan. 16, 2015, 8:19 a.m., wdoekes wrote: > > Thanks for the quick feedback! > On Jan. 16, 2015, 8:19 a.m., wdoekes wrote: > > /branches/11/channels/chan_sip.c, lines 13752-13755 > > > > > > I'd rather see

Re: [asterisk-dev] [Code Review] 4328: res_pjsip: Document transport selection process

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4328/ --- (Updated Jan. 16, 2015, 3:13 p.m.) Status -- This change has been mar

Re: [asterisk-dev] [Code Review] 4346: Use SIPS Contact headers as prescribed by RFC 3261 (chan_sip)

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4346/#review14209 --- /branches/11/channels/chan_sip.c