similar to: Dahdi incoming call detection and hangup detection durations.

Hello, I've got a test setup with 2 asterisk boxes: Asterisk1 with: asterisk 11.5.1 dahdi 2.7.0.1 Digium TDM400 with 2 FXO ports Asterisk2 with: asterisk 11.5.1 dahdi 2.7.0 Digium TDM400 with 2 FXS ports Asterisk1 has the following AEL Dialplan: context remote { s => { Answer(); Dial(DAHDI/g1/7005); }; }; When a call from Asterisk2 comes in, it is correctly

Hi all, I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk 11.14.2 and DAHDI 2.8.0. I try to set callwaiting = no AND callwaitingcallerid = no in chan_dahdi.conf. But I can't find native bridging information from CLI(opened debug mode in logger.conf). How can I test the dahdi_bridge in native bridge mode? I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from

Hi Richard, Thank you for your response. But after I remove the parameters of dial command (tTkK). The call was still not native bridge. Let me know if you have any suggestion. Best regards, Charles 2015-01-30 0:34 GMT+08:00 Richard Mudgett <rmudgett at digium.com>: > > > On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang <lazy.charles at gmail.com> > wrote: > >> Hi

Dear all, I am having an issue with CDR logging. What I want to do is log jitter variable from RTPAUDIOQOS module into Master.csv at the end of each call. I am using asterisk version 1.4.26. For CDR purposes, I am using cdr_custom, and the content of my cdr_custom.conf is the following: [mappings] Master.csv =>

On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang <lazy.charles at gmail.com> wrote: > Hi all, > > I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk > 11.14.2 and DAHDI 2.8.0. > > I try to set callwaiting = no AND callwaitingcallerid = no in > chan_dahdi.conf. > But I can't find native bridging information from CLI(opened debug mode in >

Hello-- I've just written and submitted a new module for asterisk, to the asterisk bug database. See http://bugs.digium.com/view.php?id=6021 There is a file there you can download, AEL2v0.3.patch.bz2 and I created a wiki page: http://www.voip-info.org/wiki/view/Asterisk+AEL2 Why did I do it? Because I was very impressed with AEL, but the current AEL compiler isn't real good at

Call to Action! For those who have the courage/ability, go grab an SVN copy of the asterisk release, the HEAD version, and my latest patch, from: http://bugs.digium.com/view.php?id=6021 Right now, the latest version of the patch is 0.10. apply it to the SVN head version, and do a "make". Read the Wiki on AEL2: http://www.voip-info.org/wiki/view/Asterisk+AEL2 Look at the examples

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

Hey all, I am very interested in using AEL2 (don't want to upgrade to 1.4 to get it though), but am having some problems upgrading/patching my asterisk system. I am following the instructions on the wiki: http://www.voip-info.org/wiki/view/Asterisk+AEL2#AEL2AnnouncementsandNews But get the following error:

Hello-- It's been a while since I wrote any updates about AEL/AEL2 to the users list, and I thought it might be worthwhile to update everyone on what is going on in respects to AEL. What the heck is AEL? The Asterisk Extension Language. A higher level language for extensions.conf, which will appear in the config file, extensions.ael, in the /etc/asterisk/ (or equiv) directory. It provides

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.7.0.1 DAHDI-Tools-v2.7.0.1 dahdi-linux-complete-2.7.0.1+2.7.0.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete Hotfix release to workaround

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.7.0.1 DAHDI-Tools-v2.7.0.1 dahdi-linux-complete-2.7.0.1+2.7.0.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete Hotfix release to workaround

I just downloaded and installed asterisk-1.2.13 Reading http://www.voip-info.org/wiki/view/Asterisk+AEL2 says I should be using AEL2, but what's downloaded is pbx_ael not pbx_ael2 as expected. Where and how do I get current release of AEL2 - Is there some 'How To' somewhere? TIA Bart

Hi, I am having the following issue after upgrading the kernel in a CentOS machine. [root at localhost dahdi-linux-complete-2.7.0.1+2.7.0.1]# make make -C linux all make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory

Hello, I am getting the following error when compiling dahdi : make[2]: Entering directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64' Building modules, stage 2. MODPOST 0 modules make[2]: Leaving directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64' make -C /lib/modules/2.6.32-431.1.2.0.1.el6.x86_64/build

that I should be coding my own queue system for AgentCallBackLogin using = AEL2 instead of using the AgentCallBackLogin command because it is buggy = and will no longer be supported. =20 Is this true? I don't seem to see too much literature on the Internet = about using AEL2 or are people still waiting until we are forced to use = AEL2? =20 ------_=_NextPart_001_01C8F242.AD0F4D37

Hello, during DAHDI 2.7.0.1 compilation on CentOS 6.5 64bit, I have this error: In file included from /usr/src/dahdi-linux-2.7.0.1/drivers/dahdi/dahdi-base.c:66: /usr/src/dahdi-linux-2.7.0.1/include/dahdi/kernel.h:1407: error: redefinici?n de 'PDE_DATA' include/linux/proc_fs.h:328: nota: la definici?n previa de 'PDE_DATA' estaba aqu? make[2]: ***

I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled from source on a Debian Lenny box, also running FreePBX 2.5.2. I also have an OpenVOX TDM400 card installed with an FXO port on port 1, and an FXS port on port 2. I have a POTS line installed and working on the FXO port. However I've encountered a weird problem that I can't seem to figure out. I have incoming POTS

Hi all, I've just upgraded to latest 1.6.0 SVN from a few days ago and my Gosubs have stopped working. This is from the verbose logs: -- Executing [03333407271 at incoming-aaisp:4] GotoIf("IAX2/aaisp-3802", "1?5:7") in new stack -- Goto (incoming-aaisp,03333407271,5) -- Executing [03333407271 at incoming-aaisp:5] Gosub("IAX2/aaisp-3802",

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);