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...rts. 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

...canpark=yes cancallforward=yes callreturn=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 busydetect=yes busycount=3 ;busypattern=500,500 ; DEFINING CHANNEL context = incoming_context_for_ptcl signalling=fxs_ks channel=> 1 -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090907/20147c6d/attachment.htm

.....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 direction Best Regards Shakeel Abbas

Hello list I have a problem with a dataset (see toy example below) where I am trying to find the difference between two (or more numbers) and discard those observations which fall outside a set interval. An example and further explanation: values ind 1 2655 7A5 2 3028 7A5 3 689 ABBA-1 4 1336 ABBA-1 5 1560 ABBA-1 6 2820 ABLIM1 7 3339 ABLIM1 8

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

Anyone besides me totally throw (most) caution into the wind and yum update any mission critical production 4.x boxen to 4.5 without any problems? 8-p - rh -- Abba Communications Spokane, WA www.abbacomm.net

...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 (blinking) Is this a problem ?? -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090909/4a68c3bf/attachment.htm

Hey. Perhaps someone can help me with the following (OpenSSH 6.7): I have a host reachable via miscellaneous interfaces (and network addresses) running SSH. Some specific users should be only reachable from the inside, so e.g. though something like this would do the job in sshd_config: #general config #... Match User foo LocalAddress 10.0.0.1,fe80:abba::0 PasswordAuthentication

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.

.... 2. Recored in Wav and send to central server. In this case we may face Network Bandwidth problem.(Even we create VPN). QUESTION IS: Is there any other format in which we can record using the record application provided its is small in size and directly playable on WEB. Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091102/8aaef28f/attachment.htm

...hen a user records a voice on Server B . The file will be send to Server A and Server A will send it to all other servers ie C and others if exists. But if alot of user start to record their voices then sockets may fail ??? DO any one have idea to do it in better way ??? -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091020/afed1da6/attachment.htm

...E) exten => s,n(no),System(/bin/rm ${ARG1}) THE PROBLEM IS when the called party picks up the phone The call get connected between the caller and callee. But I want it must ask the callee if (s)he wana talk if press yes then connect else not. Please shed some light on it -- Kind Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091203/c9bf2932/attachment.htm

...ast_log(LOG_NOTICE, "Failed to load config\n"); } return res; } char *description(void) { return tdesc; } #define AST_MODULE "app_swift" AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Cepstral Swift TTS Application"); -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090826/765439e5/attachment.htm

...we can specify a different location but at end we specify this in a configuration file). Is there a way that the configuration files get encrypted or some thing else so that some one who have system access can not copy the configuration files data or look into that files. -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100706/1373c712/attachment.htm

On a Compaq DL360 G1 w/ dual PIII 1.27's I installed Centos 5 minimal and then a yum update When I do a "uname -a" I get this Linux tstsrvr.abbacomm.net 2.6.18-8.1.3.el5 #1 SMP Mon Apr 30 19:55:44 EDT 2007 i686 i686 i386 GNU/Linux When I do a rpm -qa | grep kern* I get this kernel-devel-2.6.18-8.1.3.el5 kernel-2.6.18-8.1.3.el5 kernel-headers-2.6.18-8.1.3.el5 It is interesting

I am stuck on a simple problem where an example works fine but the real one does not. I have a data.frame where I wish to sum up some values across the rows and create a new data.frame with some of old data.frame variables and the new summed variable. It works fine in my simple example but I am doing something wrong in the real world. In the real world I am loading a labeled data.frame. The

...oing 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 Shakeel Abbas

...k Dialplan Functions XXX chan_h323 PBX Modules [*] chan_iax2 Resource Modules XXX chan_jingle Test Modules [*] chan_local the XXX makes me worry..... how to remove this ?? include chan_gtalk also -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090903/68731113/attachment.htm

Hi fellows, Pretty new to CentOS. I was trying to find a way to boot CentOS into command prompt instead of GUI (or without loading any services). Tried using 'Crl+Alt+F1' at the boot process, but, that holds the screen at mounting and doing fstab and doesn't proceed further. Is there anyother way to boot CentOS into command prompt without using Rescue option from the installation CD?

...d 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 Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090818/dcd21697/attachment.htm