similar to: reducing the number of extensions for every user

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Hi, I have to do a comparison among three groups of genetic transcription levels. I have a situation like this: group 0: baseline group 1: first treatment group 2: second treatment In the first group, I have only 2 samples, in the second one 4 samples and in the last group I have 10 samples. I would check if the trnascription of a gene increases from the baseline situation to the fist treatment

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

I have to design a dialplan for mulitple contexts (multiple companies) and I'm not sure how to go about it and I thought someone may offer help. Here is some background. There are three separate companies, let's say A, B and C. Each has their own context and each has their own set of numbers (these are just examples, not the actual config): [ContextA] exten =>

Could someone with inside knowledge comment on that? If the source code says "deprecated" but the CLI help does not mention that - whom do I trust? -------- Original message -------- Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions From: Philipp Kempgen <philipp.kempgen@amooma.de> Thomas Kenyon wrote: > Philipp Kempgen wrote: >> You might use

Estimada comunidad tengo un problema del que no encuentro datos que me ayuden mucho en la web. Estoy haciendo calculos por grupos con data,table. Tengo un archivo (zp.res) con tres columnas que clasifican los datos (sol, con, dia) y una columna de datos numericos (media), de la siguiente forma: sol con dia media 1: con 0 1 -22.6 2: con 0 1 -36.6 3: con 0 1 -35.6 y

I am attempting to write in incoming context for calls. 1. If the caller id is given and it is not black listed it will Playback a greeting and then right the phone or go to voicemail under busy or unavailable conditions 2. If no caller id is given, then Privacy Manager will ask for the number. I am testing 6145551212 to see if the black list will work 3. If a caller id is given, and it is

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

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Hi all, I'm looking for an easy way to lookup numbers from the database so I can fork calls from my daughters friends onto her IP phone/answering system. I'm looking for something very similar to LookupBlacklist, but I'm already using LookupBlackist to filter out telemarketers. What I'm doing now is adding multiple exten=> lines to my extensions.conf file to match those

Hi everyone, I'm running into a problem using JIT compilation on Linux x86-64. LLVM revision is 62079. I've installed a lazy function creator using InstallLazyFunctionCreator(). I return the value 0x5ce64e from my lazyFunctionCreator function. However, the disassembled JIT'd function looks like this: 0x00007f45ef2b6018: sub $0x8,%rsp 0x00007f45ef2b601c: mov

I want to attach two SIP phones to an asterisk server. The SIP phone I'm using is the software xlite for windows. The university LAN has a firewall and xlite says that a blocked firewall was detected. How can I get around this because they are not going to unblock UDP 5060 for me. Riz

Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.

Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0

I am trying out a new service from www.talkn.com. They use Sipura to terminate your service like most providers. They are looking at directly connecting into asterisk in the future. Right now my configuration is Talkn?Sipura?Asterisk/FXO Card. When someone calls in and get?s answered and when either party hangs up , * releases the port with out a problem. If someone calls in and * answers the

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 14th and 17th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral Symposium is to provide graduate

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 14th and 17th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral Symposium is to provide graduate

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 21th and 24th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 21th and 24th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 21th and 24th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral