similar to: dialplan reload: not working with large dialplans

Hi all, We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to any newer releases: We use the following cli command to feed a wave/mp3 file into an existing conference on an other serve: /opt/asterisk/sbin/asterisk -r -x "channel originate Local/ConfGongAdmin at XY_Features extension ConfGongPlay at XY_Features" The corresponding extensions.conf part looks like

I have a problem with my internal extensions, I'm using Asterisk 1.6.2.5 and freePBX 2.6. When I call betwen extensions these don't connect. There is a long silence and finally hang up. I have an E1 whit r2 and I use openr2, but I don't have problems to do calls to the PSTN......my problem it only with internal extension. Please help!!!!...it's an Asterisk bug??..... Thanks :S

I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and installed dahdi-2.2.1. kernel modules loaded. lsmod | grep wctdm wctdm 37233 0 dahdi 194985 1 wctdm lsmod | grep dahdi dahdi 194985 1 wctdm crc_ccitt 1549 2 dahdi,isdnhdlc dmesg: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.1

I'm running Asterisk 1.4.18 and having a problem with the clearglobalvars option. I have a NIGHT_SERVICE variable which I initially set equal to off. I then have an extension they can dial which will toggle that variable. My problem is when you enter the CLI and type "reload", it resets to "off" again. I've tried setting the clearglobalvars=no as well as just

"Request For Comments on a Feature Suggestion" -- just wondering if others would find this useful. Frequently, when something really doesn't make sense, I like to bump up the console logging by editing logger.conf and changing console = error to console = debug,dtmf,error,event,info,notice,verbose,warning and entering logger reload When done, I

Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI> reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found

Hi I'm lately bothered with the need to provide a set of Asterisk configuration files in a package that will be good for a wide range of Asterisk users. Asterisk configuration files support #include and a number of other interesting tricks, as mentioned in http://svn.digium.com/svn/asterisk/branches/1.4/doc/configuration.txt [0]. Let's start with manager.conf . Let's start with

Hi, I am working on Trixbox. I want to create my own dial() function (named specificdial()) and I want to know how I can create a module and integrate the module in the trixbox plateform. thanks a lot Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091006/b69d4103/attachment.htm

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

I have an issue where a particular dialplan works but another doesn't. I'm not sure why. To me they look identical and it has me stumped. This works: [to-test] exten => _X., 1, SetCallerPres(allowed) exten => _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb) exten => _X., 3, Ringing exten => _X., 4, Dial(SIP/9330 at a-test,20,ro) exten => _X., 5,

Hello ! I am having issues since the upgrade in debian from zaptel 1.4.7.1 to 1.4.8. I did m-a a-i zaptel to upgrade what i still get is: cat /sys/module/zaptel/version 1.4.7.1 It doesn't seem that it wants to load the new driver, any ideas ? Kind regards, Jan -- Jan Prunk <janprunk AT SPAMFREE gmail DOT com> http://www.prunk.si GnuPG: 1024D/00E80E86 77C5 156E 29A4 EB6C 1C4A 5EBA

Is there some magic to compiling asterisk-addons-1.6.0 under Debian 2.6.18 and mysql 5.0? I am unable to get configure to recognize the existance of mysqlclient. Imparticular, when it gets to: checking for mysql_init in -lmysqlclient... it returns "no". For the past several releases, I've had to hack or otherwise coerce this to work, but this time, no amount of fiddling with

Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no answer, the extension hangs up, but the dial status is busy instead of no answer. How do I get this to work -- do I need to update dahdi? The card is an X400p using its FXS module. Thanks in advance for any ideas on this. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?

Folks, I'm trying to implement a simple loop in a dialplan. The object is to set a counter, run through some IVR options, increment the counter, return to the start, then finally fall through to an operator or voicemail. Am using 1.4.10 and have reviewed doc/ exten => s,1,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=20) exten => s,n,Set(loop = 0) exten =>

Hy guys i am having so much hard time to setup asterisk on a virtual machine that i got , i just want to know if i really need to use Dahdi and libpri on a complete Digital PBX i just gonna use sip and iax. I will never use any kind of analog line on this machine. Wait for a feed back. Daniel Abreu.

Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]:

Trying to build asterisk 1.6.0-rc3, it fails with this message: make[2]: Entering directory `/home/fhimpe/rpm/BUILD/asterisk-1.6.1.0-rc3/ main/editline' /bin/sh makelist -h common.c > common.h /bin/sh makelist -h emacs.c> emacs.h /bin/sh makelist -h vi.c > vi.h /bin/sh makelist -fh common.h emacs.h vi.h > fcns.h /bin/sh makelist -fc common.h emacs.h vi.h > fcns.c if uname -s |

Hi, It's not clear to me which asterisk version drops support for zaptel in favor of dahdi. Dahdi and zaptel can "coexist" in some 1.4 versions but it seems that from 1.4.22 onward, chan_zap.so is not built. Documentation within the 1.4.23.1 tarball indicates that one can keep using the zap*.conf files by switching an option in asterisk.conf. Is this true or should I forcefully

Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. > dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any guide for configuring

Hi All, For the most part, the PBX works as it should. Occasionally people complain that they call and the PBX doesn't pick up. Other times it looks like the call is answered by Asterisk but I still hear ringing and I start listening to the IVR menu a few seconds into it. As for Asterisk not picking up, I see the following in the logs: [Mar 27 13:32:29] NOTICE[13197]: chan_zap.c:7071