similar to: Documentation for Asterisk AMI Events?

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Dear R-community, I'm analysing some noise using the nlme-package. I'm writing in order to get my usage of lme verified. In practise, a number of samples have been processed by a machine measuring the same signal at four different channels. I want to model the noise. I have taken the noise (the signal is from position 1 to 3500, and after that there is only noise). My data looks like

I am working on a script that takes numeric performance indicators and runs them against a series of regressors (dummy regressors, yes\no stuff via 0 and 1, e.g. Was is Christmas this week 0=no, 1=yes). The script is as follows (Written as a function): -- Begin Script -- doEnv <- function(HOUR,ENVNAME,REPORTNAME) { library(RODBC) library(forecast) library("geneplotter")

Hello, I'm wondering how I can merge two featuresets into one. My dataset is two sets of microarray data and it looks like followings: > rawData $v1 TilingFeatureSet (storageMode: lockedEnvironment) assayData: 2197815 features, 59 samples element names: channel1, channel2 protocolData rowNames: LT290677RU_D1_2011-02-16 LT286300LU_D1_2010-07-24 ... LT003990RU_D1_2010-11-04 (59

Hi All, I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 separate offices as follows, Toshiba Strata dk28 Toshiba Strata dk280 Toshiba Strata dk8 I need to install 3 Asterisk servers in these 3 locations and integrate them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the

Hi All I am trying to setup : PSTN E1 ---> Asterisk------>Legacy PBX------->Legacy Analog extensions. I've followed steps using : http://www.voipinfo.org/wiki/view/Asterisk-Panasonic i get the green light (sync) on both the 2nd span of digium TE420P (that is cnnected to the legacy pbx pri card) and the pri card of the legacy pbx. but when i try to make a call to asterisk so that

I want to build my own calling card system on Asterisk. I looked at this page - http://www.voipinfo.org/wiki/view/CallingCard+Applications and it has listed some applications that I thought could help speed up the development process though the link down the bottom doesn't work. Does anyone know of any AGI etc applications to build a Calling Card system on Asterisk? Michael

I have an * installation which will refuse incoming callers once a max (5 callers) is reached. Caller 6 and up should be notified of congestion...without network load on my trunk. How would I do this? The voipinfo wiki shows playing a congestion tone to the caller, but that seems stupid since I'm consuming bandwidth to send a tone. I also tried just responding with the congestion

Markus Weiler <markus_weiler at mailworks.org> schrieb: Hi > from voipinfo... > > If an Asterisk command specifies a sound file in a*subdirectory*, > Asterisk looks in that subdirectory for the language subdirectory. For > example, theSayDigits > <http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigits>command may > play the sound file

I just want to pdf and email faxes coming in over pstn on a TDM400P. Outgoing faxes would just go out over pstn, not through asterisk. All the voipinfo , etc, howto's are quite complicated. And most use third party apps like Hylafax. I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm now using 1.4.22, but I'd go to 1.6 if it made this easier. But I've

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I

Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : "Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP

Hi List, I used to access a Paradox database using RODBC, but since last week I am not able anymore to fetch any table which has a name longer than 11 or 12 characters. Here is the the pattern of my code, nothing spectacular: library(RODBC) channel2<-odbcDriverConnect('DSN=xxx') #table names with up to 11 characters still work sqlFetch(channel2, 'abcdefghijk') #table names

Hi all, I'm going to buy E1 digium110P ,any one knows how i can get faxtomail working for three different channels? I mean: channel1-->FaxtoMAil1@XXX.com channel2-->FaxtoMail2@xxx.com?? for 1 channel is not dificult with*@home and NVfaxdetect Using *@home 2.5 faxToPDFmail works, but i always get error opening pdf in outlook. i've solved this with a mimeconstruct update explained

I'm looking for ideas here. This probably will wind up being more of an alsa problem, but I thought the people on this list might have been in similar circumstances and have ideas for me. I work at a radio station with both a webcast and FM broadcast. We want to use icecast both for our webcast and as a backup studio to transmitter link. However, the webcast audio needs to be

Dear List, I have a problem with RODBC on a Paradox-DB, sqlQuery, and special characters in table names. Unfortunately, some of the latter include the underscore <_>. And I am not able to change them. That's not a problem, when I quote the table name: > sqlQuery(channel2, 'SELECT * FROM "anmeldung-alt"') Btw, if I swap ' and " it does not work: >