similar to: PRI/E1: machine freeze/dies after a few calls

Hello all, I have some questions about the libpri and zap lib. I've sent mail to Mark for help. Also I wonder whether anyone else would be interested and helpful in them. I'm going to migrate a program to Wildcard E400P card. Our original program is based on C API similar to the functions provided in libpri.h and zap.h, so I want to use the E400P card in the following way: 1. install and

I am getting this: make gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pritest.o -MF .pritest.o.d -MP -c -o pritest.o pritest.c pritest.c:50:28: fatal error: dahdi/tonezone.h: No such file or directory #include <dahdi/tonezone.h> I am using dahdi-linux-complete-2.11.0+2.11.0 I did "find . | grep timezone.h" from the dahdi root and its not

Hi all, i 've written a little Callgen script for generating calls through the outgoing spool directory. The calls goes over 8 ttyI devices to another pbx and come in through other 8 ttyI devices. But when i generate the calls, sometimes * register the calls but never initiate them. Especially when the files come to fast into the outgoing dir. What can be wrong ? Is it possible that the

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in

We are using DISA with local SIP users. The user enters in a 2 digit code then they get a dialtone and the phone dials out. The problem is that the calls waits 10 seconds after the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten =>

Hi all, I replaced a TE410P Rev C 1st Generation Firmware with a TE411P without any cfg changes (zaptel/zapata). As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls: Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

Hi all, Please help me in installing Asterisk. I am getting the following error when trying to install Libpri [treepr at CDC5S3FB-174202 Asterisk]$ cd libpri-1.4.2 [treepr at CDC5S3FB-174202 libpri-1.4.2]$ make clean rm -f *.o *.so *.lo *.so.1 *.so.1.0 rm -f testprilib libpri.a libpri.so.1.0 rm -f pritest pridump rm -f .depend [treepr at CDC5S3FB-174202 libpri-1.4.2]$ make install gcc -Wall

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

My cdrs are missing accountcodes for incoming calls from other asterisk servers.. I've seen a few people mentioning this on the list and the solution seems to be setting up a dialplan for incoming calls from a particular sip peer.. in my opinion this does not scale well at all and I am looking for a solution to correct this problem. example sip peer: [asterisk_gw] type=friend

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

I am using a2billing as billing software ,and I make an 800 call service which means that the destination extension should be build I put this code at extensions.conf exten => 99909994,1,SetAccount(2704714849) exten => 99909994,2,Wait,2 exten => 99909994,3,DeadAGI(a2billingp.php) exten => 99909994,4,Wait,2 exten => 99909994,5,Hangup its not stable ,its works for 3 times

Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA

Hi, I want to assign different accountcodes (for billing) according to the IP address and or the H.323 name (chan_oh323). I tried in extensions.conf something like setVar(ACCOUNTCODE=userid) but in cdr I find the accountcode set in oh323.conf. Howto change it in extensions.conf? Roger.

Is it possible to modify CDR variables before insert into MySQL or CVS? how? thanks all; Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: alexm.vcf Type: text/x-vcard Size: 334 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060116/5a095cdc/alexm.vcf

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Greetings all, I'm not real good with dial plan programming and need some help. I've looked at the 2nd edition of the Asterisk book about GotoIf() and have a basic idea what I need to do but not sure about the correct way or the best way, to set it up. I need to branch based on whether the dialed number is long distance (international or not) or not. I have branch offices on SIP and IAX

I'm using Asterisk 1.4 and I'm wanting to set an account code for incoming calls. In the extensions.conf file I have the following: exten => s,1,SetAccount(1234) exten => s,n,Dial(SIP/1234) Then when I dial the extension the following error message pops up in the CLI: [Jun 6 19:12:40] WARNING[28167]: pbx.c:1783 pbx_extension_helper: No application 'SetAccount' for

> I'm using Asterisk 1.4 and I'm wanting to set an > account code for incoming calls. In the > extensions.conf file I have the following: > > exten => s,1,SetAccount(1234) > exten => s,n,Dial(SIP/1234) > > Then when I dial the extension the following error > message pops up in the CLI: > > [Jun 6 19:12:40] WARNING[28167]: pbx.c:1783 >