similar to: Realtime goto problem

Hello everybody, I have setup asterisk real time extensions and its working pretty well. But the problem is when I am jumping between the contexts using the Goto statement in the database. I am getting a error = Parsing '/etc/asterisk/sip_notify.conf': Found -- SIP Seeding peers from Astdb: 'ezzibpo4' at ezzibpo4@210.211.246.47:5061 for 60

Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk??

Hello all! We have a PSTN line with four numbers calling into it. There is distinctive ring on these lines. They are are follows: 1. standard ring 2. short ring 3. long ring 4. short ring, long ring, short ring Based on the information I have been able to find, I have created the following entries in my zapata.conf file, to try and weed out some of the timings: dring1=95,0,0

I am using a combo of static files and Asterisk Realtime configuration. This section works fine when a static file: --------------------------- [from_pstn] ;Voipgate exten => 4507,1,Goto(from_pstn,s,1) exten => s,1,Macro(dial-ext) exten => s,2,Hangup --------------------------- But, when I drop it in the database and try it in Realtime mode I get this error: ---------------------------

I'm a newbie to Asterisk, but I'm moderately knowledgeable about phone systems. Right now, I'm most certainly confused. I have a TDM-04B (four FXO) and four analog FXO lines running into it from an AdTran 616. I have Asterisk working internally, although I could use some help getting incoming calls to answer properly and configuring my outbound dialplan. Here's where I'm

I'm trying the latest bri 0.1.0 RC2 drivers. In announce I see implementation of so long waited Transfer feature. But I can't make it work. When the person who is making transfer after talking with second party press "R" second time to establish 3 way call the person to which call supposed to be transfered being disconnected. Any ideas whats wrong? Thanks, Dmitry

Thanks a lot for all the suggestions! Unfortunately, it still gives problems. Most common error message is "ast_realaudio_callback Failed to write frame" after "paying the beep". Then it says "User disconnected". Also, it doesn't react to any extension entered and doesn't do any forwarding (as it should in "exten =>

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks

hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>

Hello group I just update to the newest CVS now I'm not able to get asterisk to start. No error during the make or make install I did a make clean before the make;make install Any help would be great!!!! Here is the output asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding realtime_ext to mysql/realtime/extensions_table == Binding

Hi, Greatly appreciate if some one help me with the application read(). asterisk*CLI> show application read asterisk*CLI> -= Info about application 'Read' =- [Synopsis]: Read a variable [Description]: Read(variable[|filename]): Reads a '#' terminated string of digits from the user, optionally playing a given filename first. Returns -1 on hangup or error and 0

Looking at the error I tried moving chan_modem* out of the modules folder and asterisk started and its working again... Not sure what changed in the chan_modem_i4l.so but removing it from the folder fixed my problem. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Hall Sent: Sunday, January 23, 2005

Hi, Sorry, in my last mail I wrote "wcfxs" instead of what I actually used, "wcfxo." I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and

Hi all, I have two sipgate accounts (numbers), if I have both accounts register only one will work for incoming calls (which is all i'm interested in). However if I disable either account the other account will work perfectly. Am I missing something obvious? Thanks in advance, Ray. Excerpts from sip.conf - [general] 8<---- SNIP! ---->8 Register => 1212121:aaaaaaaa at

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

Here is my file to parse and load extensions. No wise cracks about my code.... DB.php is the Pear DB module. (pear.php.net) <?php include('DB.php'); $db_host = ''; $db_name = ''; $db_login = ''; $db_pass = ''; $db_table = 'extensions_table'; define(DBINFO,"mysql://$db_login:$db_pass@$db_host/$db_name"); $db =

After many hours of hitting with my head on the wall I discover i had the worng ip address. Wine works as a swiss clock Many thanks to Dan Kegel ---------- Forwarded message ---------- From: Pedro Ferreira <pedro.m.ferreira at gmail.com> Date: Dec 7, 2007 4:45 PM Subject: Wine Problem To: wine-users at winehq.org I dont know if this is the wright mailing list but here it goes I wanted

All - this is probably a simple problem, but I've been pulling my hair out trying to figure out what I'm doing wrong. I'm building a *simple* IVR menu. Here it is: [main-menu] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout(5) exten => s,4,ResponseTimeout(30) exten => s,5,Background(logic-main) exten =>

I posted a message a little while ago but got no response (that I can recall), I've also seen other people mention this issue. Basically, when you have music on hold, it doesn't play the music on hold, the debug info shows it is starting and then stops straight away.. # My extensions.conf ... exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten

I ran into a couple of odd problems with a new Asterisk box I have set up... I have a T100P connected to a channelized T-1 (24ch) to the PSTN. The T-1 is set up properly so far as I know and seems to work perfectly with one exception: When a call comes in, asterisk starts a 'simple switch' for it, goes through my 's' extension progression until it gets to the first Playback