No - at least not that I've been able to figure out. These phone's were made to be used with Cisco's Call manager software (Skinny?) and the SIP firmware doesn't seem to allow this. Softkey buttons (like hold, transfer, conference), seem to be static and you can't change them. You could always use
All;
I am using Asterisk 1.8 and am running into some performance
bottlenecks. Right now I am sending upwards of 700 concurrent faxes. I have
no problem with that. The problems appear after the faxes complete. I was
thinking of using sqlite3 to log CDR's, thinking that would be faster than
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: Thursday, October 03, 2013 1:25 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Using sqlite3 for CDR logging
On 3/10/13 5:52 pm, Tech Support wrote:
I
;
John
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479 (Work/Fax)
supp...@voipbusiness.us mailto:f...@voipbusiness.us
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
Thanks for all of the hard work everyone put into this release. I think
sometimes we take some of these open-source projects for granted and don't
appreciate all the hours that people put into them. Is there a general
timeframe for when you think a stable 2.8.0 release will be available?
Have you thought of using the app_konference module? You can find it
here: http://sourceforge.net/projects/appkonference. You can configure many
of the options with the dialplan switches, there's a simple but functional
web page to monitor all of the conferences and attendees (mute, unmute,
What you want to use is Asterisk's dialplan Read command. Check it out
here. http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Regards;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine
Elharit
Sent: Thursday,
Hello;
Without trying to sound too commercialized, my company has created an
autodialer that does what you want. You can take a look at a non-functional
demo by going to http://demo.voipbusiness.us where we have several demos you
can look at. We have customers that make several dozen calls
You can have tens of thousands of phones as long as no one makes or receives
any calls J. The better question to ask is how many concurrent calls have
people been able to make. The quick answer is it depends on many things.
John
From: asterisk-users-boun...@lists.digium.com
How did the system behave with 244 calls? I've been able to make 1,024
concurrent faxes (which tend to use more resources than audio calls) in the
lab. The problem I had was after the faxes were transmitted, things couldn't
keep up and kept dumping core. Two things were going on, (1) the CDR
it was how we handled the deployment to get it up quickly, and may
have been able to prevent this, if tested better.
Keith
On Wed, Dec 18, 2013 at 10:46 AM, Tech Support aster...@voipbusiness.us
wrote:
How did the system behave with 244 calls? I've been able to make 1,024
concurrent faxes
kei...@vianet.ca
On Wed, Dec 18, 2013 at 12:05 PM, Tech Support aster...@voipbusiness.us
wrote:
Have you ever checked out the app_konference module? You can check it out
here. http://sourceforge.net/projects/appkonference. I have a customer who
routinely hosts 100+ users in a conference
=no
t38pt_udptl=yes,fec
t38pt_usertpsource=yes
Any help at all would be greatly appreciated.
Thanks;
John
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479
mailto:f...@voipbusiness.us supp...@voipbusiness.us
Although I haven't tried this for this particular example, instead of
using a .call file, you could probably originate a call using Ryan Bullock's
Asterisk::AMI PERL module
http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.8. It's one of the most
valuable tools that I have and I've written
to Asterisk. Can
someone point me in the right direction as far as documentation and examples
go? I would greatly appreciate it and will make it all available publically
if the implementation turns out well.
Thanks;
John
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479 (Work/Fax
All;
I'm running Asterisk 1.8.15-cert3 with the newest version of spandsp.
I've even tried unloading that and using Digium's FFA module but I receive
the same error on an outbound transmission:
[2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL
Rather than speculate, take a look at the output of top. If you're
running out of memory, shut down useless processes. You'd be surprised what
processes get started by default that you don't need. You should also check
the Asterisk logs and look at the last few things Asterisk did right before
You may want to check out the 3rd party Asterisk module app_konference.
You can find it at http://sourceforge.net/projects/appkonference. I have
customers using it for the last year or so with very few problems. One
customer is routinely running conferences with 80 - 100 users on a Pentium 4
Hello;
Check out this in cdr.conf. You may want to set it to yes. From
cdr.conf.sample:
; In brief, this option controls the reporting of unanswered calls which only
have an A
; party. Calls which get offered to an outgoing line, but are unanswered, are
still
; logged, and that is
a
context and an extension entry in voicemail.conf, it works the way it
should. Is there something that I'm missing here? Any insight at all would
be greatly appreciated.
Thanks;
John
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479 (Work/Fax)
mailto:f...@voipbusiness.us supp
to read the database, it would have to be modified.
On Wed, Jul 30, 2014 at 8:55 AM, Tech Support aster...@voipbusiness.us wrote:
All;
I’m currently running Asterisk 1.8.15-cert7 and am using realtime to store
my voicemail configuration. The voicemail application works fine
describing why it isn't working.
On Wed, Jul 30, 2014 at 10:32 AM, Tech Support aster...@voipbusiness.us wrote:
Scott;
I’m using Asterisk’s built-in application “Directory”, not the php script.
Thanks;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
What you may want to check out is the PlayTones and Ringing applications in
your dial plan. Asterisk will answer the call, but your users won't know that
because all they hear is the call still ringing. After a certain amount of time
passes, you can send them directly to voicemail, hangup,
All;
I have a customer who does some small, limited fax broadcasting. What he
wants to do is to be able to tell when a phone number is actually a human
rather than a fax machine so he can delete the number from his customer
list. Determining whether a call is a fax or not on the incoming is
Hello;
Just taking a quick glance at it, I think you have a syntax error in
your dial plan. Instead of ReceiveFax(${FAXDEST}/${tempfax}.tiff,f,d,
shouldn't it be ReceiveFax(${FAXDEST}/${tempfax}.tiff,fd) with no comma
between the f and d options?
Regards;
John
From:
All;
I'm currently using Asterisk 1.8 and I want to be able to have each user
be able to set as many of the voicemail options as possible. The
documentation calls voicemail options that can be overridden on a
per-mailbox basis advanced options. However, I've read conflicting
information as to
How about recording the call calling it whatever you want, and then using a
custom AGI script to append the call to the original one? That’s how I would do
it if it were me.
Regards;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Unless of course the database server is not running at all for some reason.
Regards;
JVC
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Tuesday, November 11, 2014 8:36 AM
To: Asterisk Users Mailing
to determine
that in the dial plan if I could. Any insight at all with this would be
extremely helpful.
Thanks;
John V.
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479 (Work/Fax)
mailto:f...@voipbusiness.us supp...@voipbusiness.us
One thing that concerns me is that this post is from 2009, even though the
newest version of the app on Sourceforge is up to date. I have a customer who
has been using a conference server that I built for him using app_konference
for several years now and he routinely runs conferences with
What you may want to consider is if you have a network management system
such as Nagios is create a service that checks the size of the binary every
5 minutes. You're notified if the size goes over a certain threshold. You
can also take the perf data and graph it using one of the many Nagios
Hello;
Did you remember to uncomment the dateformat in
/etc/asterisk/logger.conf? That's necessary for fail2ban to work.
Logger.conf
[general]
dateformat=%F %T
Regards;
John
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I didn't know that instant messaging was a feature with the Polycom's. Do
you have any documentation, how-to's, etc. that you can point me to? That
would be just way too cool.
Thanks;
John
-Original Message-
From: asterisk-users-boun...@lists.digium.com
If you take a look at the safe_asterisk shell script, usually located at
/usr/sbin/safe_asterisk (for CentOS at least), you'll be able to find where
the core files are located. If it's not located there, then you'll need to
look at the Asterisk init script for the scripts location. I hope this
All;
I'm running Asterisk 11.6-cert9 and am trying to play a pre-recorded
audio file to extensions using the Page() command. The dial plan looks like
this:
exten = s,n,Page(${AVAILCHANS},A(${AUDIOMSG})) and the paging by itself
works great. However, when I try it with the audio file, it
If I correctly understand what the problem is, what I did was write a
script that runs out of CRON every 15 minutes. It checks the outside IP address
by querying http://checkip.dyndns.org and compares it to the IP address stored
in the parameter “externip” in the [general] section of
I'm surprised that you didn't have to specify the full path to the 'touch'
command. When writing AGI scripts, I always do something like
$touch = which( 'touch' ). I guess it's over kill.
John
-Original Message-
From: asterisk-users-boun...@lists.digium.com
All;
I build a conference server using Asterisk 1.8 and the third party
module app_konference.so. I would ask on their forum, but the forum seems to
be pretty dead. The problem I am having is that when I have conferences that
have a lot of members, say 100+ users, the DTMF seems to not work.
All;
I have a problem that Ive been working on for a while now, but Im
stuck and cant see what the solution is. I have an Asterisk 1.11 server on
a public IP address and have two phones registered from behind a NAT. I can
send a page to/from each phone without a problem. My problem is that
Hey;
It seems to me that for what you want to do, it would be easier just to
email the user the voicemail audio file as an attachment. I believe that when
you choose to store voicemails using IMAP, it applies to all of your users
which may not be what you want to do.
Regards;
John
You should try it and find out if it works. If it does, let us know.
Regards;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
Saraiva
Sent: Tuesday, May 12, 2015 11:58 AM
To: Asterisk Users Mailing List -
Of A J Stiles
Sent: Wednesday, May 13, 2015 11:15 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Recommendations for IMAP Voicemail
On Wednesday 13 May 2015, Olivier wrote:
2015-05-06 17:51 GMT+02:00 Tech Support aster...@voipbusiness.us:
I believe that when you choose
Please keep the “me to” emails off the list.
Regards;
JV
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magno Guimarães
Sent: Monday, June 22, 2015 3:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Check out the “uniqueid” parameter in cdr.conf and cdr_adaptive_odbc.conf.
John V.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rui Mota
Sent: Friday, June 26, 2015 7:05 AM
To: asterisk-users@lists.digium.com
Subject:
Check out the externnotify parameter in voicemail.conf. What it does
is run an external program whenever a caller leaves a voicemail message for
a user. The way it works is basically any time that somebody leaves a
voicemail on the system (regardless of mailbox number), the command
specified
Hey;
Don't forget Perl. I'm not sure what everyone else calls it, but most
Perl programmers call it a fat comma.
From Chromatic's Modern Perl book: One of the simplest but most useful
examples of TIMTOWTDI in the design of Perl is the fat comma operator (=),
which acts like a regular
appreciate if you help me to solve it.
yours,
SAM
On Wed, May 20, 2015 at 7:11 PM, Tech Support aster...@voipbusiness.us wrote:
Hey;
Yes, I’ve also seen that 5 second delay with our fax server and it drove me
crazy. How I solved it was by doing a “core show channels concise|verbose”
and detect
Hey;
Yes, I’ve also seen that 5 second delay with our fax server and it drove me
crazy. How I solved it was by doing a “core show channels concise|verbose”
and detect if there was a fax transmission going on. Doing it this way shows
up instantaneously without any delay. Like so:
at
Digium ever think about things like this?
Thanks;
John V.
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479 x325
mailto:f...@voipbusiness.us supp...@voipbusiness.us
--
_
-- Bandwidth and Colocation
I believe that Asterisk 1.8 and older uses the BerkeleyDB for Asterisk's
internal database (AKA the Astdb) and in newer versions use SQLite. However,
the basic functionality is the same. Whether you use the Astdb or MySQL
really depends on what you want to do with it. The AstDB is not a
All;
I have a customer who is looking for a good speech to text solution,
either open source or reasonably priced commercial product, I'm open to
suggestions.
Thanks;
John V
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-- Bandwidth and Colocation Provided
I built a conference server for a customer using the app_konference Asterisk
module. He routinely has 75+ users in a conference and the load average
doesn't go above 1.00. Just a thought.
Regards;
John
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hey;
Is there a reason why you aren't using the standard FastAGI port 4573?
Regards;
John V.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
pierre.guy...@orange.com
Sent: Tuesday, December 15, 2015 12:18 PM
To:
will go
absolutely nowhere, will be deleted when finished, and I have no problem
anonymizing the data if needed. If anyone feels that they want to help out,
please shoot me an email at supp...@voipbusiness.us. Like I said, this is an
open source project.
Thanks in Advance;
John V.
Tech Support
Here you go… Found this in cdr.conf.sample.
; Define whether or not to log unanswered calls. Setting this to "yes" will
; report every attempt to ring a phone in dialing attempts, when it was not
; answered. For example, if you try to dial 3 extensions, and this option is
"yes",
; you will
not pick up,
it will log; as it currently does now.
With call files, it is not logging. There is no execution of a dialplan inside
my assigned context.
On Thu, Dec 31, 2015 at 9:29 AM, Tech Support <aster...@voipbusiness.us> wrote:
Here you go… Found this in cdr.conf.sample.
;
I don't think the original poster was asking about which OS is best. I think
he was asking which PBX manager people are using. Ex, PBX in a Flash,
Elastix, FreePBX, blah, blah, blah.
Thanks;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
V.
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479 x325
supp...@voipbusiness.us <mailto:f...@voipbusiness.us>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
All;
I have a customer running Asterisk 11.6-cert13. The server is solely
used for faxing. The problem he has is that Asterisk is dumping core on a
very regular basis, maybe half a dozen times a day at least. From the logs,
I see that the last commands executed before the dump were fax
One of the things you can do is google "app_konference". It doesn't require
a clock source and is a very good application. I've successfully been using
it for years and have had no problem with 100+ users in a single conference.
Regards;
John V.
-Original Message-
From:
Do you mean the directory( ) application that’s used as a dial by name
directory service to match caller inputs to existing names? If not, then It's
not going to be possible to simply have the user enter a digit, say ‘2’, and
have Asterisk repeat a letter since the user could mean either
Hey;
I’ve used Camrivox in the past and it is an excellent product, the best
I’ve seen. However, it is commercial software, so you’ll have to determine if
it's within your budget or not. You can check it out at http://www.camrivox.com.
Regards;
John V.
Tech Support
Tech Support
VoIP
Hello;
If you are talking about the 'externnotify' parameter in voicemail.conf,
the variables are passed simply as @ARGV.
Regards;
John V.
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479 x325
<mailto:f...@voipbusiness.us> supp...@voipbusiness.us
Hello;
I have a question about Dahdi-Linux and Dahdi-Tools. If I'm using a
particular version of Dahdi-Linux, say x.y.z, does Dahdi-Tools have to be
the same version?
Thanks;
John V.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hello;
I ran into a similar problem not long ago. Always try the easiest (and
cheapest) solutions first. My solution was to replace the Ethernet cable and
then to change the network switch port. Did the trick. Switches with errors
tend to be due to faulty switch ports.
Regards;
John V.
All;
I was wondering what people are doing to bill customers for minutes. I
know that A2Billing is a popular option, but I was wondering if there are
other good alternatives. They don't necessarily have to be free, but they
need to be cost effective. Any insight at all would be greatly
I don't think you are going to be able to get that information using the
AMI. You should be able to figure it out though by looking at the voicemail
directory structure in /var/spool/asterisk/voicemail// or
in your database if you're using real time. It's probably just as easy to
write a
All;
What I want to do is create a way to easily send callers into a
conference room to have an N-way conference call. I created an extension
'100' that calls the MeetMe() command. Then all I need to do is transfer a
caller using a blind transfer (*2 in my case) to extension 100. Then I can
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Leave and re-enter a conference
On Sun, Aug 14, 2016 at 1:28 PM, Tech Support <aster...@voipbusiness.us> wrote:
> All;
>
> What I want to do is create a way to easily send callers into a
> confe
I agree, those are good ideas and we did eventually develop an SMS
feature later on. Another thing we did to cut down on the 'annoyance' factor
was to maximize the chance of sending the call to voicemail directly. We
were able to develop a feature to send the call to voicemail about 90% of
the
wanted you could leave it ringing for twenty minutes and it would still
have the same effect.
Kind regards,
Matt
On Feb 6, 2017, at 12:29 PM, Tech Support <aster...@voipbusiness.us> wrote:
That's the basics, but you have to nail the timing just right. The timing is
really important
We once developed a reminder system for a customer. He's a cleaning
company, cleaning homes and offices. He was spending two hours a day calling
his customers to remind them of their appointment the next day. Two hours a
day equates to 40 hours a month that he saved with that system. He's been
: [asterisk-users] Call List Campaign to an IVR
On Mon, 6 Feb 2017, Tech Support wrote:
> We were able to develop a feature to send the call to voicemail about
> 90% of the time. That way, an end user could (1) not be bothered by
> having to answer the call, (2) delete the message without
, 2017 12:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call List Campaign to an IVR
> On Mon, 6 Feb 2017, Tech Support wrote:
>
> We were able to develop a feature to send the call to voicemail
about 90% of the time. That way, an end u
Hello;
Over time, we’ve built a huge enterprise level monitoring system for our
internal and customer PBX’s. Using Nagios as the core, along with Grafana,
Graphite, Carbon, Whisper, etc. so we can also create custom dynamic
dashboards, we typically monitor over 1,000 different metrics for
All;
I am running Asterisk 11.6-cert16 and I have voicemail setup so
voicemail messages are sent as email attachments. That works fine. However,
the body of the email contains the CallerID(name), but is missing the
CallerID(num). For example, the email body looks like this:
Just
VM_DATE},\n${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?so:(originally sent
by ${ORIG_VM_CALLERID} on ${ORIG_VM_DATE})\nso)} you might want to check it
when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
On 18 February 2017 at 16:35, Tech Support <aster...@voipbusiness.us> wrote
For reconfiguring SIP phones? Can you give an example or short explanation?
Thanks;
John V.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, January 18, 2017 08:54 AM
To: Asterisk Users Mailing List -
Hello;
We’ve been using Nagios and a lot of customizations for the plugins for
several years now to monitor over 1,000 metrics on each of our PBX’s. We’re in
the process of GPL’ing the Asterisk plugins now. That gives us our core
monitoring, notifications, event handlers, etc. To put it
All;
When I transfer a call to another extension, I can simply press *2 and
then the extension number, say 101. No big deal. The problem I am having is
in programming a speed dial key to dial *2101, which is failing. The only
thing I can think of is that the speed dial key is dialing the
All;
I have a problem with regards to "re-parking" calls and I was hoping
someone could shed some light on the topic. Consider this scenario:
(1) An inbound call comes in and the attendant answers it
(2) The attendant places the call on hold and the caller is sent to
extension 701
(3)
All;
I have a problem with regards to "re-parking" calls and I was hoping
someone could shed some light on the topic. Consider this scenario:
(1) An inbound call comes in and the attendant answers it
(2) The attendant places the call on hold and the caller is sent to
extension 701
(3)
Hello;
Way down at the bottom of your post you're getting an error that says
“-bash: /usr/sbin/asterisk: No such file or directory”. Where is the asterisk
binary located? There's a high likelihood that that’s the problem.
Regards;
John V.
From: asterisk-users-boun...@lists.digium.com
is the asterisk
binary located? There's a high likelihood that that’s the problem. where is
itlocated how do i find it and what do i need to do. i tried it on centos 6.8
and i had same error. where am i going wrong?
chris
On Sat, Dec 10, 2016 at 10:48 PM, Tech Support <as
Just an FYI, in the dialplan below, The ReceiveFax() application
receives the fax document and then automatically hangs up the call when it
is finished. That means Asterisk will then jump to the hangup extension in
the same context (if it exists) without executing any lines of code after
the
.
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479 x325
<mailto:f...@voipbusiness.us> supp...@voipbusiness.us
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Che
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16
I don't - it just seems to.. work!
Try a reboot - it always comes up OK for me. Are you doing "make install"?
On 19 April 2017 at 14:19, Tech Support <aster...@vo
t.o] Error 1
Makefile:402: recipe for target 'res' failed
make: *** [res] Error 2
Has anyone seen this error before? Any insight at all would be greatly
appreciated.
Thanks;
John V.
Tech Support
Tech Support
VoIP Business Solutions
240-215-3479 x325
<mailto:f...@voipbusiness.us
ow you get on.
On 18 April 2017 at 13:41, Tech Support <aster...@voipbusiness.us> wrote:
> All;
>
> I am trying to build and install certified Asterisk 13.13 cert3 on
> a Ubuntu 16.04.2 LTS host without much success. I am getting the
> following errors when I try to com
It's possible that you need to increase the value of 'findtime' to
something greater than 300 secs. You also may want to set "timestamp = yes"
in asterisk.conf so each line in the CLI will be time stamped. Time stamping
it will be the definitive determination on whether or not the 'findtime'
Is ** also defined in features.conf?
Thanks;
John
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, April 26, 2017 05:41 PM
To: Asterisk Users Mailing List - Non-Commercial
On a similar note, does anyone have any idea as to the total number of Asterisk
installations out there?
Thanks;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Klein
Sent: Tuesday, April 25, 2017 10:00 AM
To:
All;
I'm trying to install certified asterisk 11.6 cert16 on a Ubuntu 16
server. However, when I try to compile it, I'm getting hundreds and hundreds
of errors. Here is a sample of the output.
make[1]: Leaving directory
'/usr/src/asterisk-certified-11.6-cert16/menuselect'
[LD]
All;
We have always tried to avoid charging customers for minutes simply
because we didn't want the hassle of doing the accounting. I was wondering
what software packages or services people are using for this.
Best Regards;
John V.
--
All;
I have an application that dials a list of numbers and then plays a
recorded message. My customer uses it to dial a list of customers to confirm
their appointment for the next day. No biggie, maybe 25 - 30 calls per day
for customers who want the confirmation call. What they need now is
HA machine in front of the two,
so that writes can go to either server using only a single IP address
configured in Asterisk.
Then, if one fails, you can still write to (and read from) the other, repair
the failed one, and restore replication.
Antony
> > On Jun 19, 2017, at 17:47, Tec
All;
I know that there are probably several solutions to this problem, but
what I am trying to do is provide some redundancy for my customers CDR data.
I know that doing simple backups of MySQL is probably the easiest way to go,
but I'm thinking that there may be some benefit to
he extension to pick
up. Simply placing the AMD command after the SendDTMF() wasn’t the answer I
don’t know how to approach this problem.
Thanks;
John V.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent
] Automatically dial a number, then an extension
On Tuesday 23 May 2017 at 19:20:25, Tech Support wrote:
Isn't it safe to assume that if you've been given an extension number to dial
after the initial call is answered, then it wasn't answered by an answering
machine?
The extensi
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Automatically dial a number, then an extension
On Tuesday 23 May 2017 at 20:01:14, Tech Support wrote:
> Ok, the purpose of the answering machine detection (AMD) is to
> determine when the audio file
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