similar to: ignorepat doesn't work

Hi to all, I'm trying to make outside call in this way : ignorepat => 0 exten => _0.,1,Dial(CAPI/xxxxxxx:b${exten}) But the first number 0 is not ignored. I'm doing something wrong ? Bye

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type

Hi I have the following configuration at home one ZAPTEL interface connecting to an FXO card and two SIP UAs connecting to asterisk locally. I have configured extensions.conf such that dialing 9 on the SIP phones allows me to dial an outbound number via the FXO interface . Works fine. What's not working is that pressing 9 should causes either GS BT-100 phone to reacquire a dialtone

Dear All, Sorry if this question has been posted before but there's no enough answer to solved my problem and I'm going crazy because of it....:( I'm currently installed Samba-2.2.8a as logon server on My FreeBSD 5.0. For Win 9.X client there's no problem at all. But when I tried to joining W2K client into samba domain it's shown an error message such as : "the

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Hi, I have two asterisks connected via IAX2 trunk. The first * use dial prefix 2XX, the second one 3XX. Calls routing works OK. But I don't know how to get dialtone of remote asterisk pbx. I'd like to get dialtone of asterisk #2 after dialing 3 and dialtone of asterisk #1 after dialing 2. I know something about DISA but I'm not sure if it is a right way. Can you give me advice?

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for

> I have placed "ignorepat => 9" in just about every context I > can think of in my extensions.conf, but yet when I dial 9 > from my sip devices, the dialtone is broken. I even tried a > nearly untouched version of samples, and it stil doesn't > work. Is there something somewhere else that needs to be set > to make this work properly, like may in the sip

hello i have a mixed effect model which gives slope and intercept terms for 6 groups (diagnosis (3 levels) by risk group(2 levels)). the fixed part of the model is -- brain volume ~ Diagnosis + Risk Group + (Risk Group * age : Diagnosis) - 1 thus allowing risk group age/slope terms to vary within diagnosis and omitting a nonsignificant diagnosis by risk group intercept (age was centered)

HI, I'm using netmeeting to connect to an asterisk server and dial out. my extension looks like this exten => s,1,Dial,Zap/1/ Unfortunatelly the number that I have dialed in Netmeeting is lost ;-( If I hardcode the number on the line above, like ... exten => s,1,Dial,Zap/1/6642794 ... everything works fine What am I missing?

Hola, Alguien podría ayudarme en la siguiente cuestión: Trato de abrir múltiples archivos ncDF para dejarlos disponibles para trabajar luego con ellos. Intento hacerlo con un blucle pero no consigo lo que quiero. El código es el siguiente: setwd("/Users/jaime/Desktop/2008") a<-list.files() a library(ncdf) for (i in 1:length(a)) {

Hello all, I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my home PBX in such a way that whenever someone calls on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is

Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the

I have the following setup.. [extensions] ; all extensions defined here. exten => 1234,.... exten => 1235,.... [dial-out] ; PSTN dialout config ignorepat = 9 exten => _9,.... exten => h,.... [local] ; phone context in sip.conf is here.. include => extensions include => dialout The question is where will the "h" extension be active?? it appears to run for ALL,

Hi, I am trying to do the world's most simple install. I have a Wildcard TDM400P with 3 ports: 1 FXS on port 1 and 2 FXOs on ports 3 and 4. (i'm not using port 3 for now, put want it for expansion purposes) I simply (to start with) am looking to have the FXS phone ring when an FX0 port is dialed. I would also like to be able to place outgoing calls on the FXS through the FXO. Right

hi, I am new to * and setting up a test system. here my setup : - debian (from knoppix 3.3) - Asterisk 0.7.1 (from the debian package) - AVM Fritz card used with i4l - softphone I use for testing SJphone on windows - I can make great softphone - softphone calls - I can call from an outside line * and get connected to a softphone here my problem: I can not make outbound calls. I place a call

I've just started consulting for a SME client based in Reston Virginia. They don't know it yet but they are going to need a hosted asterisk service and some DID's. Email me if you are able to provide 10 DID's in Reston (must be able to be ported away!!) and hosted Asterisk with end user configurable IVR etc. Probably only 5-8 users at the moment BUT... they'll be

Hi, I was wondering if it's possible to separate elements in multiple rows that actually should appear in different columns. I have a file where in certain lines there are elements not separated, and they certainly should appear in different columns (an example of the file is attached). The point is that I do not want to manually add a space in the txt file, however, I did not

Hey guys, I''ve just built a sparc64 (Ultra/5) based firewall with ipp2p compiled as a module and I''m constantly getting the following message in my logs: Kernel unaligned access at TPC[100f8490] search_all_edk+0x20/0x4c [ipt_ipp2p] I''m running the following versions: - Kernel 2.6.22 - ipp2p 0.8.2-r4 - iptables 1.3.8-r1 Any thoughts?