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...k 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/asterisk-users/attachments/20120805/61bc444c/attachment.htm>

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: > > Hi, Oliver. > > > > Maybe something like this (a...

...l 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>

...eck 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>: > >> Why not to use Fail2ban https://www.voip-info.org/wiki >> /view/Fail...

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>

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

...ug. Regards On Tue, Oct 2, 2018, 13:03 Olivier <oza.4h07 at gmail.com> wrote: > @Nasir: > Thanks for replying here. > > Did you met in your deployments, the kind of stability issues Carlos > reported earlier ? > > Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal <nasir at ictinnovations.com> a > écrit : > >> Hi Olivior, >> >> We have recently worked on a WebRTC based agent panel. As based on my >> experience I think that WebRTC based phones are far better and cheaper then >> those soft / sip phone. the big plus is that they are easy to custom...

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is there an undocumented way to set Content-Type? -------------- next part -------------- An HTML

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 present? Antony. -- These clients are often

Hi everybody, >From asterisk CLI we can add extensions in dial-plan dynamically using "dialplan add extension" command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal

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

Can asterisk log the registration date/time in a database? Is there a standard option to do this? I know it being logged in the asterisks 'full' (debug) log and we are probably able to script something with the API interface but there might be somewhat easier if there is a option to make asterisk log this information directly into a database. Thanks in advance, Bram

Greetings list, Whilst running through a routine check of some CDRs, I've noticed that the originating channel's accountcode isn't preserved on creating a local channel. For example, if we start with: exten => 123,1,Set(CDR(accountcode)=foo) exten => 123,n,Queue(bar,nrtw,,,) And the queue 'bar' is defined as follows: [bar] member => Local/456 at outbound member

Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo.

Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 00003ms SCall: 00130 DCall: 00000 [64.229.229.111:64823]* * USERNAME : 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type:

On 7/8/2023 5:32 PM, Federico wrote: > > I am using Asterisk 16.30 inside Freepbx, with commercial modules, > purchased from Sangoma and Symphony. After a few hours my memory usage > reaches 900 GB, no kidding, in a box with 1 TB of RAM.  The question > is: how can I determine what is causing the memory leak? Can somebody > send me instructions to find out what module is

Hi, I have xlite registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==>1500,1,AGI("localhost//hello.agi" So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard as this is very urgent.

Right, I'm baffled. I have: exten => s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2) exten => s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\ (caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\ VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12)) exten => s,n,MYSQL(Query RESULT1 ${DB1} SELECT\

Hi Olivior, We have recently worked on a WebRTC based agent panel. As based on my experience I think that WebRTC based phones are far better and cheaper then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call