similar to: Asterisk Crash 1.8.13.0

The Asterisk Development Team has announced the release of Asterisk 1.8.13.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.13.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

All, I am experiencing this same issue. it seems that you were able to resolve it offline. Could you by any chance post the solution. telephonics1*CLI> module load app_swift.so Unable to load module app_swift.so Command 'module load app_swift.so' failed. [Aug 1 05:01:01] WARNING[28635]: loader.c:458 load_dynamic_module: Error loading module 'app_swift.so':

Hi, I am trying now for over 4 hours setting up cdr-logging via odbc into a mysql database. But with no success. Do you have any hint for me? cat /etc/odbc.ini ------------------ [MySQL-asterisk] Description = MySQL ODBC Driver Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = localhost User = xxx Password = xxx Database = asterisk Option = 3 Port = and /etc/odbcinst.ini

I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site, and I have installed via the "make linux install" command. I can execute lua scripts via the command line, but asterisk configure script is unable to

Hello All, I am trying to install Asterisk 1.8.13.0 & dahdi-complete 2.5.1 & libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. I am using "SunOS solaris-server 5.11 11.1 i86pc i386 i86pc" on Virtual Box. -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was

Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using "meetme list" command it shows Minus in activity column. Any Idea. >meetme list Conf Num Parties Marked Activity Creation Locked 54682 0002 N/A

Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts

asterisk-1.8.13.0 iksemel-1.4 I have a client who setup a gvoice account using their domain in the login name: username=client at theirdomain@gmail.com This appears to have caused a problem with authentication. I've tried escaping the @ and quoting the login string, etc. but it simply won't authenticate. I don't believe my configuration is bad as the same server /

Hi, I just ran into what seems to be an issue on re-invites. I'm not sure if it's a bug or as designed, so I thought I'd ask the question. Here's my setup: - Asterisk 1.8.13.0 - Phone A: Polycom ip331, only allowed to use ulaw, canreinvite=yes - Phone B: Polycom ip330, only allowed to use alaw, canreinvite=yes Phone A calls the extension of phone B. After the normal call setup

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

Hello, a person trying to call me by my phone number is getting the error 488 Not acceptable here. I googled that error, seems like this error is normally caused by a failed codec negotation, though I have no clue how I could have read this out of the logs. Anyway, my setup is as follows: Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider The user calling me is also using Sipgate and is calling my

Hi all, i have had sip TLS with an own signed certificate (using the ast_tls_cert script) running on asterisk-1.8.8 - i then have updated to 1.8.9.3 - and now i get the message "FILE * open failed!" I have already recreated the certificates with the script - but still no luck... Does anyone here know the source of the problem ? best regards, Wolfgang Pichler

________________________________ From: Jody Gugelhupf <knueffle at yahoo.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: Thursday, November 29, 2012 10:23:01 PM Subject: audio trouble with asterisk, help very much appreciated Hi there :) first about my setup, running centos 6.2, asterisk 1.8.13.0, freepbx 2.9.0.12.? I have a

Hi, I think I encountered a bug in the qualify timers for IAX on asterisk 1.8 but I'd like to check if I'm not messing up in my config somewhere before reporting a bug. In my IAX peer configuration I have this: [remote-host] type=friend host=172.16.6.45 username=remote-host secret=test notransfer=yes qualify=16000 qualifyfreqnotok=30000 disallow=all allow=alaw allow=ulaw allow=ilbc

Hello list, We're currently facing some issues concerning T.38 gateway faxing. This is a device used almost exclusively for receiving faxes. Calls are incoming to asterisk on a SIP trunk (sangoma netborder) using G711A. Gateway mode is activated in the asterisk dialplan towards a Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0 with the T.38 gateway patch applied (I know I

Hello, I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like "number 12345 not available" I was only hearing "345 not available". Verbose level 5 on the asterisk console didn't give me any hint on this, it only shows that playback of the prompt started