similar to: Pet project: one step Asterisk compile on Centos 7

Hello all, I created a set of Docker images running Asterisk and exposing AMI / ARI ports that i found to be quite useful for ARI / AMI development and regression. As they are based on Docker with whaleware, adding new configuration files to roll your own dialplan / queues / voicemail etc is pretty easy. And you can run quite a lot on the same box to simulate clusters. There is no SIP / RTP

Hi all, I have been playing with the AMI quite a bit lately - mostly debugging WombatDialer in production, but that's a different story - and I have been frustrated by the lack of a simple way to interact CLI-like with the AMI itself. So I have decided to write something myself to make my life easier, or at least a bit less miserable. The result is a little webapp that you can use as a sort

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a

Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Following new features are now supported by asterisk based telemarketing software Auto subscription / registration after call recipient press a key in voice broadcasting https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer There will be restriction to call a number in off time accordingly to timezone of

I need to make a Asterisk 18.0's offline compiling, SVN mp3 support sources downloading does't particulary works cause my asterisk is in an isolated network with NO network access whatsoever, I ve read this thread ( http://lists.digium.com/pipermail/asterisk-users/2013-June/279298.html) but I 'm not understading one thing, because I download the file and run the script but there is no

Hi all, I just prepared a little tutorial on installing Asterisk 13 on CentOS 6.5 64-bit. See http://astrecipes.net/index.php?n=668 Hope you like. :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com

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

Hi all, I'm running the following rsync job via cron: rsync --recursive --compress --human-readable --progress --update --perms --chmod=a-w -e "ssh -i /rsync-key" user@host:/source_dir/ /target_dir rsync: failed to set permissions on "/target_dir/.": Operation not permitted (1) rsync: failed to modify permissions on "/target_dir/.": Operation not permitted (1)

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

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

Hi, What determines the umask of sieve_pipe_bin_dir scripts ? The results from my script are always being set to 0600. My script is simple and shown below, even if I adjust the right line to add " && chmod 644", the actual resulting file still remains at 0600 ?!? #!/bin/bash # Usage: imapsieve_copy <email> <spam|ham> MSG_USER="$1" MSG_TYPE="$2"

Otherwise, automake will fail to process vivified/code/Makefile.am --- configure.ac | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/configure.ac b/configure.ac index a3e186b..0e28374 100644 --- a/configure.ac +++ b/configure.ac @@ -300,6 +300,7 @@ else AC_MSG_NOTICE([Vivified was not enabled.]) fi AM_CONDITIONAL(HAVE_VIVI, [test "x$HAVE_VIVI" = xyes])

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

Hi All; I am looking for a good Outbound Dialer and to be practical with possibility to do modification on it, the outbound dialer should send the calls to the agent when the agent is logged in as long the agent is belong to the queue (or let us say the skill group of this campaign). Any one can guide me? If I can build this using the AMI, so I appreciate if anyone did it before me so I can use

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

Hello list, I am trying to set a custom SIP header on all calls that are made by the app queue because I want to track a certain state at the SIP level. If I use the following code: exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID}) exten => s,n,Queue(myQueue) this works fine for the FIRST call made from the queue to an agent; but if that call does not go through, it's not repeated

rsync+ is ready to go onto HEAD. Could Jos or somebody else who's used the feature before please prepare a few paragraphs for the manpage explaining how they work? Either send a patch to the .yo files or just plain text. Thanks, -- Martin