[asterisk-dev] linking/loading external library

hi i am gooing to use libfftw3 library in asterisk application. compilation is ok, but when i am loading module i get error like this: dev*CLI> module load app_amdb.so Unable to load module app_amdb.so Command 'module load app_amdb.so' failed. [Nov 26 07:44:51] WARNING[17342]: loader.c:481 load_d

Re: [asterisk-dev] [Code Review] 2961: Add channel locking for calls that result in channel snapshot creation

> On Nov. 25, 2013, 6:54 p.m., rmudgett wrote: > > /branches/12/main/core_local.c, lines 682-685 > > > > > > Revert all the changes you've made to local_call() and reacquire the > > locks in publish_local_bridge_me

Re: [asterisk-dev] [Code Review] 2961: Add channel locking for calls that result in channel snapshot creation

> On Nov. 25, 2013, 6:54 p.m., rmudgett wrote: > > /branches/12/main/core_local.c, lines 682-685 > > > > > > Revert all the changes you've made to local_call() and reacquire the > > locks in publish_local_bridge_me

Re: [asterisk-dev] [Code Review] 3027: Valgrind support in TestSuite

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3027/ --- (Updated Nov. 25, 2013, 5:10 p.m.) Review request for Asterisk Developers.

[asterisk-dev] Asterisk 12.0.0-beta2 Now Available!

The Asterisk Development Team is pleased to announce the second beta release of Asterisk 12.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases All interested users of Asterisk are encouraged to participate in the Asterisk 12 test

Re: [asterisk-dev] [Code Review] 3023: Add MixMonitor() option to specify channel variable into which to store the recording filename

> On Nov. 21, 2013, 4:40 p.m., Tilghman Lesher wrote: > > Shouldn't something like this be a channel function from which you retrieve > > the value, instead of specifying a variable into which the name is placed? > > Seems like a significant regression in spitting things out to channel > > va

[asterisk-dev] Separate Audio/Video

Hello, I'm trying to separate the audio from video in asterisk, a device that gets the audio and video on another device. Can anyone help me where I can change in Asterisk?, Or is it better to create a new application, AGI, AMI, SIPSERVLET ...? Thanks! --

Re: [asterisk-dev] [Code Review] 2826: Debug threads: avoid double-initialization of lock tracking

> On Sept. 19, 2013, 10:06 p.m., rmudgett wrote: > > /branches/1.8/main/lock.c, lines 73-90 > > > > > > You could reduce the pain of the global lock here by: > > if *plt > > return *plt > > lock global

Re: [asterisk-dev] [Code Review] 2961: Add channel locking for calls that result in channel snapshot creation

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2961/#review10273 --- /branches/12/main/bridge.c

Re: [asterisk-dev] [Code Review] 2958: Testsuite - pjsip: AMI commands and events

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/2958/ --- (Updated Nov. 25, 2013, 12:31 p.m.) Review request for Asterisk Developers

[asterisk-dev] [Code Review] 3032: Testsuite: ARI device state

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3032/ --- Review request for Asterisk Developers. Bugs: ASTERISK-22838 https://i

Re: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ?

Oh thanks Olle, I'm on the wrong list. Sorry about that. Från: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] För Olle E. Johansson Skickat: den 25 november 2013 11:32 Till: Asterisk Developers Mailing List Kopia: Olle E Johansson Ämne: Re: [asterisk-dev] Ma

Re: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ?

Hi! The development mailing list is not a good list for asking general questions - please use the user's mailing list for that. There's currently no way of authenticating devices that are not peers or users of asterisk. Registration is unrelated to other requests. You could do this with Kamaili

Re: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ?

Did I catch you all at a bad time with the code reviews and all? I'm still hoping for some responses or ideas, or from someone with knowledge. Are there anyone out there? :) /Johan Från: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] För Johan Sandgren Ski