similar to: Every one Busy Problem

HelloI have the following system Asterisk 1.6.1dahdi 2.2.0.2 TE420P card Centos I have noticed that all the four lights are blinking(ie coming red and then off so on)... Previously I also noted that when dahdi drivers are not installed lights blink but one by one in sequence(like in marriage cermonies :P) and after dahdi installation lights get off ... but this time all at same time

Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Hello I have recently configured TDM400P with four FXO ports. My next requirement is to configure for E1 line. which contain 30 phone lines and 2 for signalling information. The problem is I dont want to go for E1 line directly .....Is it possible to get simulation for E1 line ... so that i can develop a system for an E1 line. -- Best Regards Shakeel Abbas

Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection ....now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a

Hello, What i am trying to do is ..... Dail a number and ask if you wana talk to XXX press 1 and if you dont wana talk press any other key. For this purpose i am using this link<http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial> . *I am using this option :- * *M(**x**)*: Executes the macro (x) upon connect of the call (i.e. when the called party answers). IMPORTANT - The CDR

Hello I have installed cepstral .... It works woderfull using an agi script but ..... when i try to use Swift("say this") is Dial plan .... I get the error [Aug 26 12:30:18] WARNING[7420]: pbx.c:3167 pbx_extension_helper: No application 'Swift' for extension (actdemo, 123, 2) Now i come to know to install app_swift Here is the issue... when i try to execute make command

Hello Community, I have a question , I have been working with asterisk and developed some successful applications. I am facing an issue of security i.e. We deploy servers to client end. Now i dont want the client to see my configuration files (Of course copy and distribute or replicate the logic with out permission). Now the configuration files are stored in /etc/asterisk/* (Of course we can

Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these

Hi Shakeel, I had the same problem building app_swift (1.6..) myself and searched the web far-and-wide for a solution. I eventually contacted Darren Sessions -- who was maintaining that plug-in -- about a month ago. He was involved in another project and said he might be able get to it after a few weeks. But, since then, his website http://www.darrensessions.com/ has gone out of comission. I

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Hello I have configured TDM400P with asterisk . The problem is that when i make a call to server. and while going on it dont detects call hang up. ie i called the Asterisk server and it start playing files that i indicated to do so in extensions.conf i suddenly put down the phone. now the server must detect that phone is hangup but it dont. How can i make server to detect this -- Best Regards

Can someone point me to a page about writing a text file to call an external number and play a TTS with cepstral? I know it includes the creation of a .call file but beyond that im a bit lost.

Hello I have two questions ! 1. What is the best speech recognition engine for asterisk? I have searched and asked on forums and found that lumen vox is best for asterisk bala bla bla 2. For TTS (text to speech) which TTS engine will be better to use ? I have tested Flite , cepstral (i have not buyed lisence for it trial only) but still thinking may be i have a good option ? -- Best Regards

Hello I need some advice regarding the Asterisk server that are located at different locations. Three asterisk servers are here each at different location. Suppose A,B,C be the three servers respectively. Server A is connected to server B and server C through a VPN. I have a developed an IVR service on server B and server C where users come and record their voice. On the same servers B and C

Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are

Hello Please advice how can i configure a sip phone that is not on my local network. ie i have Xlite far some where in America and my Asterisk server is at Sahara desert . Now how can i make a call to that sip phone? Please advice what keywords to carry on?? -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Community, I have installed Dahdi on Centos on many system and succesfully used that.. But today i have bad luck... This is the error that i am facing [root at localhost dahdi-linux-complete-2.2.1+2.2.1]# make all make -C linux all make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.2.1+2.2.1/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory

Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

@Steffan: This is the error I see in log file: sieve: info: started log at Nov 27 14:30:15. main script: line 42: error: total number of actions exceeds policy limit (33 > 32). Regards... Shakeel On 11/27/2015 06:09 PM, Steffen Kaiser wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Fri, 27 Nov 2015, Shakeel Shahzad wrote: > >> I have configured

Hi Shakeel, Assuming that you are starting with a bunch of dates: # make a vector of character strings that can be converted to dates rep_dates<-paste(sample(1:30,500,TRUE),sample(1:12,500,TRUE), sample(2013:2017,500,TRUE),sep="/") # if this isn't your format, change it date_format<-"%d/%m/%Y" # create a data frame with a column of dates