Re: [asterisk-dev] [asterisk-users] Fwd: Asterisk pjsip auto dtmf mode

Thank you Matthew, I will start working on the contribution. I have a question for the developers, but before that I will explain what I did. I have been able to add auto dtmf mode on PJSIP channel. The basic idea is to setup the dtmf mode just like RFC4733, but when going over the codecs in res_

[asterisk-dev] [Code Review] 4095: Add a prompt to be read to the "winner" when a call is connected through app_followme

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4095/ --- Review request for Asterisk Developers. Bugs: ASTERISK-24372 https://i

Re: [asterisk-dev] [Code Review] 4356: realtime: Updates fail to work due to update fields being passed over for lookup fields

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4356/ --- (Updated Jan. 19, 2015, 9:10 p.m.) Review request for Asterisk Developers

[asterisk-dev] [Code Review] 4356: realtime: Updates fail to work due to update fields being passed over for lookup fields

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4356/ --- Review request for Asterisk Developers and Joshua Colp. Bugs: ASTERISK-246

Re: [asterisk-dev] [Code Review] 4354: Bridge core: Pass a ref with the swap channel when joining a bridge.

> On Jan. 19, 2015, 4:24 p.m., Matt Jordan wrote: > > /branches/13/main/bridge.c, line 1601 > > > > > > I don't think the join actually failed here. This should just get > > called when bridge_channel_internal_joi

Re: [asterisk-dev] [Code Review] 4355: chan_sip: Fix leak of SIP registrations

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4355/ --- (Updated Jan. 19, 2015, 7:50 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 4355: chan_sip: Fix leak of SIP registrations

> On Jan. 19, 2015, 6:56 p.m., rmudgett wrote: > > cleanup_all_regs() should just be the ao2_callback() line and the original > > guts should be put into a cleanup_registration() ao2_callback function. I'm not sure I understand your comment. cleanup_all_regs is being called by the ao2_callback

Re: [asterisk-dev] [Code Review] 4355: chan_sip: Fix leak of SIP registrations

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4355/#review14236 --- cleanup_all_regs() should just be the ao2_callback() line and t

Re: [asterisk-dev] [Code Review] 4354: Bridge core: Pass a ref with the swap channel when joining a bridge.

> On Jan. 19, 2015, 4:24 p.m., Matt Jordan wrote: > > /branches/13/include/asterisk/bridge.h, lines 460-461 > > > > > > Why is it not a good idea for the caller to have their own reference, > > if they bump the ref

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

> On Jan. 16, 2015, 6:25 p.m., rmudgett wrote: > > /branches/13/res/stasis/stasis_bridge.c, lines 136-137 > > > > > > May need to add this call here to ensure that the swap channel leaves > > the bridge: > > as

[asterisk-dev] [Code Review] 4355: chan_sip: Fix leak of SIP registrations

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4355/ --- Review request for Asterisk Developers. Bugs: ASTERISK-24640 and ASTERISK-

Re: [asterisk-dev] [Code Review] 4354: Bridge core: Pass a ref with the swap channel when joining a bridge.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4354/#review14233 --- /branches/13/include/asterisk/bridge.h

[asterisk-dev] [Code Review] 4354: Bridge core: Pass a ref with the swap channel when joining a bridge.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4354/ --- Review request for Asterisk Developers. Bugs: ASTERISK-24649 https://i

Re: [asterisk-dev] [Code Review] 4316: ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis dialplan application to another system; improve and fix PJSIP's transfer ability

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4316/ --- (Updated Jan. 19, 2015, 1:17 p.m.) Review request for Asterisk Developers

Re: [asterisk-dev] [Code Review] 4316: ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis dialplan application to another system; improve and fix PJSIP's transfer ability

> On Jan. 19, 2015, 10:25 a.m., Joshua Colp wrote: > > /branches/13/include/asterisk/stasis_app.h, line 489 > > > > > > Everything else has doxygen. Why not this? Bad eyesight? > On Jan. 19, 2015, 10:25 a.m., Jos

Re: [asterisk-dev] [Code Review] 4351: ARI: improve wiki documentation

> On Jan. 19, 2015, 12:17 p.m., rmudgett wrote: > > /branches/13/res/ari/resource_bridges.h, line 309 > > > > > > This line needs to be wrapped to at most 90 columns. Prefereable to 75 > > columns for a comment bl

Re: [asterisk-dev] [Code Review] 4351: ARI: improve wiki documentation

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4351/#review14230 --- /branches/13/res/ari/resource_bridges.h

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/#review14229 --- Ship it! Ship It! - rmudgett On Jan. 16, 2015, 9:03 p.m., M

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/#review14228 --- branches/13/res/parking/parking_applications.c

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

> On Jan. 19, 2015, 4:52 p.m., Joshua Colp wrote: > > I may have broken your patch slightly but anyway! I'll ask this: > > > > Why can we not construct a proper URI from the beginning, why is a module > > required to alter it afterwards? RFC 3261 section 8.1.1.8 applies to *all* UAC communicat

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

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4345/#review14226 --- I may have broken your patch slightly but anyway! I'll ask this

Re: [asterisk-dev] [Code Review] 4350: AMI: Add documentation for the missing Cdr/CEL events.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4350/#review14225 --- Ship it! Ship It! - Joshua Colp On Jan. 17, 2015, 5:32 p.m.

Re: [asterisk-dev] [Code Review] 4316: ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis dialplan application to another system; improve and fix PJSIP's transfer ability

> On Jan. 19, 2015, 4:25 p.m., Joshua Colp wrote: > > /branches/13/res/res_pjsip_multihomed.c, line 155 > > > > > > There's no guarantee that the host will be NULL terminated. If you're looking for how we typically

Re: [asterisk-dev] [Code Review] 4316: ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis dialplan application to another system; improve and fix PJSIP's transfer ability

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4316/#review14223 --- /branches/13/channels/chan_pjsip.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. 19, 2015, 3:49 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 4342: CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4342/#review14222 --- Ship it! Ship It! - Matt Jordan On Jan. 15, 2015, 12:33 p.m

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

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4330/#review14221 --- Ship it! /asterisk/trunk/tests/rest_api/external_interaction/

Re: [asterisk-dev] [Code Review] 4348: Testsuite: Creating a named ARI bridge twice causes a crash

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4348/#review14220 --- Ship it! Ship It! - Matt Jordan On Jan. 15, 2015, 4:44 p.m.

Re: [asterisk-dev] [Code Review] 4349: Asterisk: Creating a named ARI bridge twice causes a crash

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4349/#review14219 --- Ship it! Ship It! - Matt Jordan On Jan. 18, 2015, 5:18 p.m.

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/ --- (Updated Jan. 19, 2015, 7:19 a.m.) Status -- This change has been mar

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. 19, 2015, 7:18 a.m.) Status -- This change has been mar