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Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing Software http://www.ictbroadcast.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asteri...

...security, digital forensics and ethical hacking are my core areas of interest. Currently, I am working on a research project on live forensics of GPU and volatile memories like RAMs and Caches. I am looking forward your guidance to start my contribution for LLVM, thanks in anticipation. Regards Tahir Ramzan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20160306/b3e0999f/attachment.html>

Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, --------------------------------------------------------------------- Usman Tahir snom technology AG www.snom.com --------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051222/2f95ebcf/attachment.htm

...on='0x1'/> </source> </interface> The detail of the BroadCOm NIC are as follows. 02:00.1 Ethernet controller: Broadcom Limited NetXtreme II BCM5709 Gigabit Ethernet (rev 20) What would it take to add this NIC to VM as Passthrough Device? Regards, Tahir Masood Senior Engineer, xFlow Research Inc. +92-300-9779473 <mailto:tahir.masood@xflowresearch.com> tahir.masood@xflowresearch.com <http://www.xflowresearch.com/> www.xflowresearch.com

...s-at-website-with-Voice-broadcasting-Autodialer There will be restriction to call a number in off time accordingly to timezone of destination number automatically https://www.ictbroadcast.com/Time-Zone-based-restrictions-on-telemarketing-campaigns-ICTBroadcast-autodialer-scheduling Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20181219/44393cc4/attachment.html>

Why not to use Fail2ban https://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com http://www.ictbroadcast.com Leveraging open source in ICT On Tue, Feb 21, 2017 at 12:28 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Mon, Feb 20, 2017 at 11:36:24AM -0300, Victor Villarreal wrote: &gt...

...tion will check Asterisk for following 1. will check for Asterisk process. 2. will check Asterisk via AMI 3. will check Asterisk by sending a SIP request You simply need to install monit and place attached file on your server as /etc/monit.d/asterisk.conf and then restart monit service daemon *Tahir Almas* Managing Partner ICT Innovations http://www.ictbroadcast.com http://www.ictinnovations.com Leveraging open source in ICT On Thu, Feb 23, 2017 at 3:45 PM, Olivier <oza.4h07 at gmail.com> wrote: > > > 2017-02-21 14:09 GMT+01:00 Tahir Almas <tahir at ictinnovations.com&gt...

...n it went up to 8000s user time). This seems quite a bit slower than I expected. Maybe there's a better and faster way to add such variables to a data frame that are derived using some aggregation. Also, ddply seems to take twice as long as by. Are these two operations not equivalent? Thanks, Tahir

Interested to know a good wholesale sip providers for 15k concurrent calls regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20231120/75b0e652/attachment.html>

AHA! # 1 is the case! Seems the user was fooling around with the phone after the firmware upgrade. Shame that that setting couldn't be locked out. Thanks to Mr Tahir and Mr Stredicke for their spot on responses. -----Original Message----- From: Usman Tahir [mailto:Usman.Tahir@snom.de] Sent: Friday, September 23, 2005 12:34 AM To: ColinA@landmarkmasterbuilder.com Cc: asterisk-users@lists.digium.com Subject: Re: SNOM 190 '486/Busy here' after upgrade to...

On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote: > Interested to know good wholesale SIP providers for 15k concurrent calls You might want to specify a bit more detail, such as: - which country are you located in - do you require inbound DDIs (if so, in which region/s)? - which countries' Caller ID/s do you need to...

I'm trying to add MySQL CDR recording in Asterisk 1.4.6. I'm following the instructions posted here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql I have MySQL installed and it works fine - starts on stratup, I can create DBs, tables and so on and I can connect through php. rpm -qa indicates: MySQL-server-5.0.22-0 MySQL-devel-5.0.22-0 MySQL-client-5.0.22-0 However I still get XXX

Hi Valentine, I am currently trying out your inPlaceEditor and I am happy with so far as it helps with the CMS I am developing in Rails at the moment. However, when you click cancel and go into in-place editing right after again, the textfield does not switch into WYSIWYG mode anymore. That behaviour is correct, because you do not disconnect the editor instance form the DOM tree on cancel.

...t get it work. Any help will be highly appreciated. After compiling and running it I get error "Error: login_tcp_wrappers can't be used because Dovecot wasn't built with libwrap" Dovecot Version dovecot-1.1.5 OS CENTOS 5.1 Thankyou Tahir Riaz Assistant Director (Systems) COMSATS Internet Services COMSATS Headquarters Building 9, Shahrah-e-Jamhuriat, G-5/2 Islamabad.

...d, and then hit transfer and that call will be transferred If multiple calls are on hold, you can cycle thru which call you want to take off hold by pressing the big silver center key, or using the line keys on the right hand side of the phone. Thanks again for all the help!! On 11/21/06, Usman Tahir <Usman.Tahir@snom.de> wrote: > > Just don't put the 2nd call on hold if you want to transfer it to a third > one i.e. either through speed dial/extension or through manual input. > > Regards, > Usman. > > > ------------------------------ > > Message: 10 &g...

Hi, Watching Snom 870's video (http://www.youtube.com/watch?v=9e8hPxX0oDU), you can see a new sidecar (phone extension) which seem very interesting. Has someone details on this extension ? Any release date or/and data sheet ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone have a Snom they're happy with? How did you manage that? :) I have a system of: Asterisk 1.2.3 2 Wildcard TDM400P Rev I and E/F 1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841 ~15 Grandstream 101 Everything (currently) is on the same network, not a router to be seen between any two. Also everything, except the snom, is working sweetly. The main problem is ECHO.. awful echo and

where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) also anyone got a fix for the horrible speaker phone on the 200's

Hi everyone, Ive just installed a bunch of Snom 360s, and now having a NIGHTMARE of problems! Ive got a receponist phone with a extra sidecar on it. And when she gets 2+ calls and puts them on hold, when she goes to transfer them out the calls on hold get merged together. Somehow the calls on hold get merged and not to the extension needed!! Any help on this would be great guys, that would be

Thanks Buddy, I got it working. It was giving error when running make 'undefined reference to `request_init' etc' then I ran it with gmake LDFLAGS+=-lwrap and it worked and everything is working fine. Thanks again for your help Thankyou Tahir Riaz Assistant Director (Systems) COMSATS Internet Services COMSATS Headquarters Building 9, Shahrah-e-Jamhuriat, G-5/2 Islamabad. -----Original Message----- From: dovecot-bounces+tahir.riaz=comsats.net.pk at dovecot.org [mailto:dovecot-bounces+tahir.riaz=comsats.net.pk at dovecot.org] On Behalf O...