[asterisk-dev] [Code Review] 4010: General option to musiconhold.conf, to make applications MOH override channels musicclass.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4010/ --- Review request for Asterisk Developers. Bugs: ASTERISK-24276

Re: [asterisk-dev] [Code Review] 4010: General option to musiconhold.conf, to make applications MOH override channels musicclass.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4010/#review13370 --- Your explanation wasn't immediately obvious to me, but

[asterisk-dev] [Code Review] 4012: testsuite: Allow multiple -t options to be passed.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4012/ --- Review request for Asterisk Developers. Repository: testsuite

Re: [asterisk-dev] [Code Review] 4006: Test to validate 503 not generated on INVITE retransmissions

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4006/#review13371 --- Thanks for supplying a working test. I wouldn't commit it

[asterisk-dev] ICE failure with Chrome, multi-homed Asterisk

This appears to be an Asterisk issue but I'm cc'ing discuss-webrtc because similar things have been discussed there too, please reply on asterisk-dev I've observed that calls are failing badly (appears to answer, no audio) when calling from Chrome to Asterisk and when my Asterisk server has

Re: [asterisk-dev] [Code Review] 4006: Test to validate 503 not generated on INVITE retransmissions

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4006/ --- (Updated Sept. 22, 2014, 12:10 p.m.) Review request for Asterisk

Re: [asterisk-dev] [Code Review] 4010: General option to musiconhold.conf, to make applications MOH override channels musicclass.

On Sept. 22, 2014, 8:48 a.m., wdoekes wrote: /trunk/res/res_musiconhold.c, lines 1428-1433 https://reviewboard.asterisk.org/r/4010/diff/1/?file=67441#file67441line1428 (A) If !MOH_APPOVERRIDECHANNEL, then you're loading this twice. Don't do that, especially not when there is

Re: [asterisk-dev] [Code Review] 4010: General option to musiconhold.conf, to make applications MOH override channels musicclass.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4010/ --- (Updated Sept. 22, 2014, 12:23 p.m.) Review request for Asterisk

Re: [asterisk-dev] [Code Review] 4006: Test to validate 503 not generated on INVITE retransmissions

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4006/ --- (Updated Sept. 22, 2014, 12:50 p.m.) Review request for Asterisk

Re: [asterisk-dev] [Code Review] 4010: General option to musiconhold.conf, to make applications MOH override channels musicclass.

On Sept. 22, 2014, 8:48 a.m., wdoekes wrote: /trunk/res/res_musiconhold.c, lines 1428-1433 https://reviewboard.asterisk.org/r/4010/diff/1/?file=67441#file67441line1428 (A) If !MOH_APPOVERRIDECHANNEL, then you're loading this twice. Don't do that, especially not when there is

Re: [asterisk-dev] [patch] ICE failure with Chrome, multi-homed Asterisk

On 22/09/14 13:59, Daniel Pocock wrote: This appears to be an Asterisk issue but I'm cc'ing discuss-webrtc because similar things have been discussed there too, please reply on asterisk-dev I've observed that calls are failing badly (appears to answer, no audio) when calling from Chrome to

[asterisk-dev] beta2 compile failure

Asterisk 13 beta2 compile fails: . . . [CC] chan_pjsip.c - chan_pjsip.o [CC] pjsip/dialplan_functions.c - pjsip/dialplan_functions.o [LD] chan_pjsip.o pjsip/dialplan_functions.o - chan_pjsip.so /usr/lib/gcc/x86_64-pc-linux-gnu/4.5.3/../../../../x86_64-pc-linux-gnu/bin/ld:

[asterisk-dev] [Code Review] 4013: Alembic: Add 'outgoing' enum value to sippeers directmedia enumerator

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4013/ --- Review request for Asterisk Developers and Matt Jordan. Bugs:

[asterisk-dev] [Code Review] 4014: Changes to CDR and CEL unit tests to prevent FRACKs.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4014/ --- Review request for Asterisk Developers and Matt Jordan. Repository:

Re: [asterisk-dev] [Code Review] 4014: Changes to CDR and CEL unit tests to prevent FRACKs.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4014/#review13374 --- Ship it! Ship It! - opticron On Sept. 22, 2014, 1:10 p.m.,

Re: [asterisk-dev] [Code Review] 4013: Alembic: Add 'outgoing' enum value to sippeers directmedia enumerator

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4013/ --- (Updated Sept. 22, 2014, 1:24 p.m.) Review request for Asterisk

[asterisk-dev] [Code Review] 4015: Get rid of most old libc free/malloc/realloc and replace with ast_free and friends.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4015/ --- Review request for Asterisk Developers. Bugs: ASTERISK-24348

Re: [asterisk-dev] [patch] ICE failure with Chrome, multi-homed Asterisk

On 22/09/14 15:28, Daniel Pocock wrote: On 22/09/14 13:59, Daniel Pocock wrote: This appears to be an Asterisk issue but I'm cc'ing discuss-webrtc because similar things have been discussed there too, please reply on asterisk-dev I've observed that calls are failing badly (appears to

[asterisk-dev] DAHDI-Linux and DAHDI-Tools 2.10.0.1 Now Available

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.10.0.1 DAHDI-Tools-v2.10.0.1 dahdi-linux-complete-2.10.0.1+2.10.0.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux

Re: [asterisk-dev] [Code Review] 4006: Test to validate 503 not generated on INVITE retransmissions

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4006/ --- (Updated Sept. 22, 2014, 2:38 p.m.) Status -- This change has been

Re: [asterisk-dev] [Code Review] 4006: Test to validate 503 not generated on INVITE retransmissions

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4006/#review13375 --- Ship it! I've managed to force the re-transmit and drop the

[asterisk-dev] [Code Review] 4016: chan_sip: Unref outbound proxy structure on dialog(pvt) struct

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4016/ --- Review request for Asterisk Developers. Bugs: ASTERISK-24063

Re: [asterisk-dev] [Code Review] 3948: Asterisk does not respect outbound proxy when sending qualify requests

On Aug. 29, 2014, 11:04 p.m., Damian Ivereigh wrote: Thanks for all that info Matt. In answer to the question how should outboundproxy behave, perhaps it might be useful to detail my setup. I have a number of Asterisk servers on an internal network with a kamailio server and a media

Re: [asterisk-dev] beta2 compile failure

Paul Albrecht wrote: Asterisk 13 beta2 compile fails: . . . [CC] chan_pjsip.c - chan_pjsip.o [CC] pjsip/dialplan_functions.c - pjsip/dialplan_functions.o [LD] chan_pjsip.o pjsip/dialplan_functions.o - chan_pjsip.so /usr/lib/gcc/x86_64-pc-linux-gnu/4.5.3/../../../../x86_64-pc-linux-gnu/bin/ld:

[asterisk-dev] [Code Review] 4017: chan_pjsip: Don't attempt to apply formats if there aren't any capabilities defined when creating a new channel

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4017/ --- Review request for Asterisk Developers, Joshua Colp and Mark Michelson.