similar to: ZAP to invalid SIP device call looping

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: --------- loadzone = es defaultzone=es fxsks=1 zapata.conf ---------- [channels]

As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing

Hi all, i use asterisk 1.2.7 and i have a problem with callerid. i use sangoma a200 cards. one fxo one fxs module i have these scenario - bob calls adam, where bob calls into my asterisk and adam picks up "from" my asterisk - bob and adam are speaking to each other - meanwhile eve calls adam, adam hears a beep, and knows there is an other caller on line. - bob and adam stop seaking

Hi , I have a problem with a X100, i do a external call to the asterisk server . The dialplan its simple answer and hangup.. when it's done , the telephone which i did the call , is in line but asterisk server is finish. I'll apreciate all your suggestion. Greetings, txus. The asterisk output: -- Executing Hangup("Zap/1-1", "") in new stack == Spawn

Dear all, Is there an R equivalent to SAS's proc format? Best regards J. Lamack _________________________________________________________________ O Windows Live Spaces ? seu espa?o na internet com fotos (500 por m?s), blog e agora com rede social http://spaces.live.com/

I have a system with two 4 port T1 cards, with 5 PRI's configured. Each PRI is configured as an individual PRI and belongs to it's own group (groups 1-5) This system is handling roll-over from another system, where any error in processing the call on that system results in it being sent here. This mainly results in all calls resulting in a busy being sent for retry here. I then

We have an asterisk box with the following configuration: - AMD Athlon XP 2400+ - 512 MB RAM - SUSE Linux 10.1 - a Digium card TDM400P with 3 FXO - another Digium card TDM400P with 4 FXS - asterisk 1.2.7.1 - zaptel 1.2.4 I already checked that those cards aren't sharing interrupts (by cat /proc/interrupts): 0: 14119786 XT-PIC timer 1: 10 XT-PIC i8042 2:

I'm running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9 and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports). Everything seems to work except threeway calling. I can establish a threeway call, but it uses up BOTH FXO lines. Note that I DO have threeway calling active with my Bell service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press

I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual Band Analoog FXO) working with Asterisk. I had a working FXO configuration to a analog port of a small home 1/4 ISDN pbx. I used this same configuration to connect a GSM Gateway that is supposed to be connected to the external(FXO) analog port of a pbx. I can get my configuration to dial the mobile number via the gateway, but

Dear all, Why the transform function does not accept two statistics functions? a = data.frame(matrix(rnorm(20),ncol=2)) transform(a,M.1=mean(X1),M.2=mean(X2)) # does not works #while: transform(a,M.1=mean(X1),M2=log(abs(X2))) #works Best regards JL _________________________________________________________________ O Windows Live Spaces ? seu espa?o na internet com fotos (500 por m?s), blog

Simple question that google hasn't helped much with (likely poor search terms) I just installed a Sangoma A200 with overall good results. Initial tests with both incoming and outgoing calls were very positive. Until I made a normal call that lasted more than 30 seconds. I setup the FXO with kewlstart signalling, and the outgoing call is never registered as 'Answered' and the dial

I am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone lines connected to the FXO ports one from telecom, another from vodaphone. He has set up the dialplan so that one of the trunks fails over to the other trunk. Everything seems to be working OK except for outgoing calls. He can call from

[18884732963 at from-fax-machine:... - your call is hitting the from-fax-machine context - yet your 'fax' exten is in the from-pstn-4 context. See the "[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no fax extension" line. When Asterisk detects an incoming fax tone - it tries to automagically route the call to the 'fax' extension in the SAME

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming

Hello friends, in aCentOS with a A400P01 OpenVox PCI I have a analog line connected. I am new in Linux and Asterisk, my steps are theese: 1. Install CentOS 4.4 (basic instalation). 2. Command line: yum -y update yum install gcc kernel-devel bison openssl-devel yum install openssl-devel 3. Download the source: wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

Hi, I have installed a quadBri card, with Asterisk-1.0.10 and the bristuff-0.2.0-RC8s (* 1.0.10) When calling 0207654321 the following happens: -- Executing Goto("Zap/1-1 ", " salsa-helpdesk-day|s|1 ") in new stack -- Goto (salsa-helpdesk-day,s,1) -- Executing Dial ("Zap/1-1 ", "Zap/g1/0201234567|30 ") in new stack -- Requested transfer capability:

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it