I am looking at the pjsip.conf file shipped with asterisk, and trying to
understand it. For example, there are 3 transport-X sections as noted below.
Does this mean I could uncomment all 3? Must I uncomment 1? Is the -X portion
of [transport-X] arbitrary?
; Basic UDP transport
;
I have a custom voicemail script which reformats and forwards the attached
voicemail wav file to the recipient.
I would like to make use of a channel variable in my script; is there a way to
pass a channel variable to this voicemail script?
--
I'm building a CentOS 7 Asterisk and find my system log full of messages like
this:
Mar 5 17:07:01 pbx2 systemd: Started Session 823 of user asterisk.
Mar 5 17:07:01 pbx2 systemd: Starting Session 823 of user asterisk.
Mar 5 17:07:11 pbx2 systemd: Removed slice user-1001.slice.
Mar 5
I've ported an Asterisk 10 installation to Asterisk 13, and I've noticed that
whenever Asterisk plays my audio files it uses the slin format. I have not
converted ANY of my audio files, which means asterisk must be converting my wav
files to slin on the fly.
Is this the new standard for
I don't think you can do this natively within Asterisk, but take a look at
SecAst (from http://www.telium.cahttp://www.telium.ca/ ). There is a free
edition you can download right from the web site.
SecAst will monitor the rate at which a user/device places calls to detect
potential fraud.
Sent: Sunday, June 28, 2015 9:26 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Branch based on call volume
On 27Jun, 2015, at 15:34, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
Is there a simple way to get call volume from a particular trunk within the
dialplan
Is there a simple way to get call volume from a particular trunk within the
dialplan (for conditional branching)?
I suspect we will have to build an AGI script but I'm hoping something new in
Asterisk 13
--
_
-- Bandwidth
I think you are mixing up answers and general advice. FreePBX was intended to
get you over the dialplan creation hurdle (the biggest challenge for people new
to Asterisk).
In regards to the LinkSys they are compatible and you do find them in
enterprises, but admins are trying to get rid of
Take a look at the smartCID script available from
www.telium.cahttp://www.telium.ca/?
It does a web based CID lookup on incoming calls, you can at least use that as
a starting point for development...
From: asterisk-users-boun...@lists.digium.com
I'm guessing this is a small/home system? I suggest you install SecAst from
this site: www.telium.ca It's free for small office / home office and will
deal with these types of attacks and more. It can also block users based on
their Geographic location (based on the phone number it
The results of a security experiment were published this week, in which an
Asterisk PBX was set out in the wild to see who would attack it and how:
http://www.telium.ca/?honeypot1
What I find particularly interesting is that people/bots are scraping support
websites looking for valid IP's of
Dupuis
Cc: Asterisk Users List; byrn...@harte-lyne.ca
Subject: RE: [asterisk-users] Anonymous SIP calls
On Thu, March 26, 2015 22:29, Michelle Dupuis wrote:
You have to consider whether you really want anonymous calls, or you
just want to enable SIP calls from trusted companies/partners
You have to consider whether you really want anonymous calls, or you just
want to enable SIP calls from trusted companies/partners. The latter means
setting up routes to these companies and (ideally) registration between peers.
If you really want anonymous calls, then you will have to setup
As you've probably discovered, most of the API toolkits are half baked and
poorly maintained. The Java interface is not great for performance and is
suffering from the above too.
From our experience (including customer specific and commercial apps) using
the AMI directly is the best way to
Can someone tell me when the /proc/dahdi files are created for spans? Are they
created when asterisk starts (or the asterisk init script) - if not what script
creates them?
--
_
-- Bandwidth and Colocation Provided by
Do you have DISA setup? We're seeing lots of attackers running scripts that
send digits until they strike a DISA, misconfigured mailbox, etc. (Assuming it
wasn't a stupid employee forwarding an inbound call to a 9xxx number etc).
Have a look at SecAst (www.generationd.com) - it detects
I'm creating an app that needs to read the status of all dahdi spans and
channels, etc. (whatever is needed to tell a user the state of their DAHDI
connections).
What is the best way to do that? I see dahdi-tools available from the command
line, asterisk CLI commands, and AMI commands.
I'd suggest taking a look at the free edition of SecAst (www.generationd.com).
It handles these messages perfectly (and can also use AMI security events) - so
you don't need to constantly be updating fail2ban rules. It's a drop in
replacement for fail2ban.
-M-
P.S. My opinions are my own
When I issue the CLI command 'core show calls' I see how many calls have been
processed by Asterisk since it started; eg:
0 active calls
198 calls processed
Is there a way to reset the calls processed counter without having to shutdown
and restart asterisk?
--
There are lots of ways to solve this, and NOT to solve this. Don't start
adding lots of rules to iptables (or deep per packet inspection requirements)
as this will hurt capacity...and it doesn't really solve the problem
Take a look at
http://www.voip-info.org/wiki/view/Asterisk+security
If
You can also take a look at SecAst (www.generationd.com).The free version
is a drop-in replacement for fail2ban but also add a lot more intelligence (and
no need to update regex's etc). There's also geographic IP fencing so you can
block attacks by country / region / city etc., only allow
You might get a better response on the FreePBX forum. (FreePBX adds pre-built
dialplan elements onto standard asterisk. This forum is more for Asterisk)
But some suggestions:
SSH to your PBX
enter the Asterisk CLI
set verbose to 10
Call into the problematic number
...and watch where the call
If you have a small Asterisk installation install the free version of SecAst:
http://www.voip-info.org/wiki/view/SecAst+(Asterisk+Intrusion+Detection+and+Prevention)
For general Asterisk security info check this out:
http://www.voip-info.org/wiki/view/Asterisk+security
-=Michelle=-
All
After reading about the 2 major SSL (and TLS?) weaknesses discovered this
year, I was wondering how it affects asterisk.
Does the SIP authentication use TLS - or something that was recently broken?
Is there a risk of exposing passwords?
Thanks!
--
I have setup an Ast 11.6 host and I want to login via AJAM. I setup
manager.conf, http.conf described in the docs. When I login via the AMI it
works fine (see below), but when I login via AJAM the same credentials fail
(see further down)
Can someone tell me how to fix this?
---
: Friday, May 16, 2014 3:25 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Friday, May 16, 2014 2:43:30 PM
Subject: [asterisk
!
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis
mdup...@ocg.ca
Sent: Friday, May 16, 2014 3:39 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
You're
PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails
- Original Message -
From: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Friday, May 16, 2014 3:39:35 PM
Subject: Re: [asterisk-users] Login by AMI ok
actually rawman and manager are very different, and you don't need cookies just
to test login. However, I found the problem: I forgot quotes around the curl
command.
Thanks!
--
_
-- Bandwidth and Colocation Provided by
Another alternative is SecAst (Asterisk intrusion detection system). Grab the
free version from www.generationd.comhttp://www.generationd.com/?
It does everything fail2ban does, plus you have the option of blocking IP's
based on geograhic origin, detecting suspicious call patterns, etc.
These are at completely different levels of the ISO stack...question is making
sense to me.
(What does it mean to divert a call to a serial port). Do you mean route a
call over a link that is ppp/dialup and connected to another endpoint on the
other side of that link?
If so you would have to
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/
It does everything fail2ban does and more, including blocking users by
geography (we exclude all of Asia and Africa), detection of break-in patterns
(even if someone guessed your un/pw), detect changes in dial rates, etc.
To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk 1.6
On Friday 04 Apr 2014, Michelle Dupuis wrote:
Take a look a SecAst from www.generationd.comhttp://www.generationd.com/
It does everything fail2ban does and more, including blocking users by
geography (we exclude all of Asia and Africa
If you know your users are all from with your country, or state, or even city,
you could restrict geographic access in your secast.conf file like this:
ruledefault=deny
ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA
The above would:
- By default deny all source IP's anywhere
IMHO: If you're announcing a product, selling a product, etc. it belongs on the
commercial list. If you're asking/answering questions about Asterisk and the
ecosystem I think you can mention commercial products too. (We don't want to
pretend they don't exist, and then steer users to only
I (canadian) store has a deal on for the vera lite controller:
http://www.tigerdirect.ca/applications/searchtools/item-Details.asp?EdpNo=8930107sku=VEP-STARTER1
but this looks different than the vera lite green white:
?oops...wrong list :)
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis
mdup...@ocg.ca
Sent: Friday, March 28, 2014 5:43 PM
To: Asterisk Users List
Subject: [asterisk-users] Best zwave controller
: Michelle Dupuis mdup...@ocg.ca
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Thursday, March 27, 2014 12:55:21 AM
Subject: [asterisk-users] Security log format / content
I've noticed that the Asterisk (v11) security log captures attempts
do dial without first authenticating
I see a lot of attempts by hackers to call 00972595301123? or 011972595115207?
or variations but that same 972595 is often present.
Can someone break down that dial string with an explanation? The 011 look like
an overseas call (from Americas), while the 972595XX is unclear...
--
To: Asterisk Users List
Subject: Re: [asterisk-users] Numbers hackers call
On 26 Mar 2014, at 15:05, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
I see a lot of attempts by hackers to call 00972595301123? or 011972595115207?
or variations but that same 972595 is often present.
Can someone
I've noticed that the Asterisk (v11) security log captures attempts do dial
without first authenticating, and places the number dialed into the accountid
field.
I'm trying to distinguish between failed attempts to register and attempts to
dial without registering, but the security log treats
After each line of text, please also dip the corner of your keyboard into your
ink well to ensure your writing can been seen.
Calling something natural because it used to be that way isn't always correct.
-MD-
P.S. Notice how little we see PS in posts...now that we can also edit our own
Some food for thought:
If you use DRBD, then you will mirror corruption from one system to another.
You also cannot selectively pick files in a folder to mirror (you will mirror a
lot!) As well, DRBD struggles as peers are set further apart (latency) or
number of changes increases.
A lot of
I'm looking for some beta testers to provide feedback on an Asterisk intrusion
detection prevention program we're releasing soon.
As a quick overview, the program provides:
- banning based on geographic location of source IP (Continent, country,
region, city, etc)
- detection and banning based
Markus,
We are developing an Asterisk intrusion detection prevention tool which will
allow you to limit connections by geographic region
(continent/country/region/city), and include/exclude IP subnets, etc.
If you are interested let me know off-list (we're looking for beta testers!).
Of Daniel Jenkins
[dan.jenkin...@gmail.com]
Sent: Thursday, January 23, 2014 9:03 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] AMI eventmask question
On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
Hi
I'm creating an AMI client and I
: [asterisk-users] AMI eventmask question
On Thu, Jan 23, 2014 at 3:06 PM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
That's an interesting link - I didn't know you could set a per user eventfilter
in the conf file
However, I'm hoping to do this in the AMI connection for more
Of Richard Mudgett
[rmudg...@digium.com]
Sent: Tuesday, January 21, 2014 6:12 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] core show channels truncates channel names?
On Tue, Jan 21, 2014 at 3:39 PM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
When I issue a 'core show
I'm looking at setting type=peer vs type=user (in both IAX and SIP conf
entries), and I found a comment attributed to digium
(http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that
type=user is depricated and that we should only use type=peer
Is that still correct? Will
I'm creating an AMI client and I only want to get newchannel events (as well as
responses to any actions I initiate). What would I set the eventmask to to
only get the newchannel events?
For anyone else looking...is there a table somewhere online that maps events to
their eventmask
Is there a mapping of AMI versions to Asterisk versions?
eg:
AMI 1.0 = Ast 1.4
AMI 1.1 = Ast 1.6
etc...
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
When I issue a 'core show channels' command I notice that long usernames (and
channel number) are truncated. For example, if the username is
FONEMITEL1234567890 for a trunk, then it will show
SIP
Privilege: Command
Channel Location State Application(Data)
: [asterisk-users] IAX2 bridge failing
Did you change your network switch recently? Some Digium IAX ATAs do not
behave well with Cisco equipment.
On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
meant to say restart didn't help either
Ok just restart
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Friday, December 13, 2013 11:46 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] IAX2 bridge failing
I tried
meant to say restart didn't help either..
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
[mdup...@ocg.ca]
Sent: Saturday, December 14, 2013 11:20 PM
To: Asterisk Users List
Subject: Re
: [asterisk-users] IAX2 bridge failing
Michelle Dupuis wrote:
Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx
? Or something I can fix through config?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
[mdup...@ocg.ca]
Sent: Thursday, December 12, 2013 5:08 PM
To: Asterisk Users List
Subject: [asterisk-users] IAX2
I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk
system has been stable for years, and has no trouble bridge SIP phone sets to
IAX trunks.
When I initiate a call from the IAX ATA, something goes wrong.One rare
occasion it works fine, but usually there is no
Is there a mapping of AMI versions to Asterisk versions somewhere? For
example, Asterisk 1.4 includes AMI version 1.0 (at least that's what I see when
I connect to Ast 1.4 via telnet to the AMI port)
Also, doe the AMI version changes reflect changes to the AMI commands? If so,
is there also
someone tries to use it during the 'off' time. no
need for anything as brutal as disabling it in sip.conf.
On 2013-10-23 12:37 AM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
I need to disable/enable a peer after hours automatically, and am thinking
about doing so via the AMI
I need to disable/enable a peer after hours automatically, and am thinking
about doing so via the AMI.
Is there a command to enable/disable (or perhaps delete/add) a peer via the
AMI? I could create code to modify sip.conf and force a reload, but that seems
like the wrong approach...
--
Is there a recent survey of that Linux distro and version people are using for
the Asterisk installations? I recall seeing a pie chart over a year ago (I
think on a wiki but I can't find it again)also hoping for something more
current.
I suspect RH5 and RH6 are most popular...but I'm
Gareth:
Did you check if your message (or security) log recorded anything during these
attempts? If so, can you post the content of the logs during this attack?
M
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf
Is it possible to detect the failure of an agent to register with Asterisk via
the AMI ?
When I try to register with Asterisk 1.4 using an invalid password I don't see
any event in the AMI, but see this in the messages log:
[2013-10-05 22:05:03] NOTICE[24598] chan_sip.c: Registration from
Be careful with DRDB singe failing drive/corruption on one peers takes down the
other too...
Check out haast as well (at www.generationd.com) for a commercial asterisk
clustering solution.
Michelle
(GenerationD Systems)
From:
...
For redundant/failover of Asterisk checkout HAAST at
www.generationd.comhttp://www.generationd.com The HAAST product sits between
Linux and Asterisk, monitors for failures etc, and then fails over to another
Asterisk box. It effectively creates a low-cost cluster, moving IP's etc to
Check out smartCID on www.generationd.comhttp://www.generationd.com
This script allows lookup of incomming calls based on number and either Block
(no ring), endless ring (ignore), or pass through to asterisk. It allows
allows rewriting of CID name based on number. All numbers stored in a
take a look at AsteriskControl script at www.generationd.com
This is a free script that monitors, responds to IP address changes, etc. and
restarts asterisk.
You can also use HAAST (commercial) at same site - it can check for missing
registrations etc and restart asterisk too.
-=M=-
I want to track the number of calls up at any given time, through the AMI. I
found the Link and Unlink commands as the most likely candidates - is that the
right way?
Also, a comment on the wiki suggests that Link may be called several times for
a single bridge if transcoding is required.
channels verbose”
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Thursday, October 18, 2012 9:58 AM
To: Asterisk Users List
Subject: [asterisk-users] Counting calls in progress from AMI
I want to track the number
That's how we do it - write to a memory based (ramdisk) disk then write to HDD
upon call completion. We haven't tried a SSD but that may be necessary
depending on your call volumes.
From: asterisk-users-boun...@lists.digium.com
Does anyone know if Gigaset is for sale in the USA? Based on my assessment of
phones and features, i would like to try the N300IP base along with C610H
phones.
I can only find the handsets on ebay, no retailers in USA. And I suspect they
are using European frequencies.
--
We've deoplyed a number of pure VoIP wireless (wifi proprietary) phones, but
not dect.
Is there a simple overview of integrating DECT phones with Asterisk somewhere?
I assume the DECT basestation has a multi-account SIP VoIP interface, and the
handsets are just plain old dect?
Can you push
: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
[car...@televolve.com]
Sent: Friday, June 29, 2012 4:58 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Intro to DECT vs IP
On Fri, Jun 29, 2012 at 1:22 PM, Michelle Dupuis
mdup
-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
[aster...@lists.minotaur.cc]
Sent: Friday, June 29, 2012 6:27 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Intro to DECT vs IP
On 29/6/12 11:16 pm, Michelle Dupuis wrote:
Can you
...@lists.minotaur.cc]
Sent: Friday, June 29, 2012 8:22 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Intro to DECT vs IP
On 30/6/12 12:12 am, Michelle Dupuis wrote:
I like the look of the C610H. Is there a matching DECT base station by
Gigaset?
I use the N300IP. Supports 3 active SIP calls I
you,
Vladimir
On 6/5/2012 8:58 AM, Michelle Dupuis wrote:
We have an Ast 1.6 installation which is connected to an Avaya using ooh323.
Something is causing the log to fill with In ooEndCall call state is -
OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms. This causes the log
We have an Ast 1.6 installation which is connected to an Avaya using ooh323.
Something is causing the log to fill with In ooEndCall call state is -
OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms. This causes the log
to grow to 300MB in just 5 minutes, which eventually overloads the
I have an ATCOM ATA that is trying to connect to an asterisk server using IAX.
The ATA and Asterisk are on the same subnet, not firewall/nat etc.
Below is a a log excerpt, showing the REGREQ received, and then Asterisk goes
on to send lots of REGAUTH...and this continues for a while, but the
We have a multitenant Asterisk 1.4 installation for multiple small business,
and we need to report how many calls a single business has active at one time.
Is there a way to VIEW how many calls are up in a single context? (Or some
other way to accomplish the same)?
Thanks
--
Wow - nice! A few quick questions:
1. How long can the recording be for translation?
2. Any limitation on how much text the return (transcribed) variable can hold?
3. Any commercial / terms of use limitations?
From: asterisk-users-boun...@lists.digium.com
1. I checked the log and I don't see any registration attempt, so I *assume*
they simply send an invite, and so they are in the external/outside context of
my dialplan. So they are trying to reach extensions which don't exist. If
they succesfully registered they would be on the internal
...@lists.digium.com] On Behalf Of Andrew Furey
[andrew.fu...@gmail.com]
Sent: Wednesday, December 28, 2011 11:37 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Interesting attack tonight fail2ban them
On 29 December 2011 12:07, Michelle Dupuis mdup...@ocg.ca wrote:
I thought that it might be worth
Here is more of a SIP debug log:
As you can see Asterisk retries four times but I assume the softphone is not
responding?
---
Really destroying SIP dialog
'637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4'mailto:'637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4'
Method: OPTIONS
Reliably
The BB is using wifi, on the same subnet as the asterisk server so no need for
NAT.
There is no keep alive option on the softphone (very simplistic settings)
Thanks
--
_
-- Bandwidth and Colocation Provided by
I happened to be in the cli tonight as some (208.122.57.58) initiated a simple
attack - just trying to make long distance calls from outside context.
Although harmless, this went on for several minutes as the idiot just used up
my bandwidth with SIP messages. Here's and example:
[2011-12-28
On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis
mdup...@ocg.camailto:mdup...@ocg.ca wrote:
I happened to be in the cli tonight as some (208.122.57.58) initiated a simple
attack - just trying to make long distance calls from outside context.
Although harmless, this went on for several minutes
I have a softphone I'm trying on a blackberry, that registers on my Asterisk,
can make outgoing calls, but can't receive calls.
There is very little traffic with this phone (see debug below - as the phone
registers), and sip show peers confirms it is unreachable.
Any suggestions? Is this just
There is a script on www.generationd.com designed for Asterisk. It will
convert the Wav49 to mp3, add call info into MP3 tags, add a company logo, etc.
and then email the message.
It's a one line change to add to asterisk - very handy. (We use it for Android
phones, nice to see call info
] On Behalf Of jon pounder
[j...@inline.net]
Sent: Friday, November 25, 2011 8:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] android won't play wav49: how to change format
On 11/25/2011 06:39 PM, Michelle Dupuis wrote:
There is a script on www.generationd.com designed for Asterisk
Although you say SIMPLE...not all virtualization hosts allow software
installation. On VMware the host has become an appliance you can't really mess
with...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On
VMware is moving all server products to their ESXi engine. (The old VMware
server and ESX products are moving to legacy status - with these you could
actually do stuff on the kernel). ESXi is no longer a kernel you can mess
with, can't install drivers, etc. ESXi is being treated as an
If one server is supposed to carry the full load of the other during failure,
then you have to size each server to handle 100% load - so load balancing is
pointless.
Checkout haast at www.generationd.comhttp://www.generationd.com and read the
docs on how it does failover...certainly good for
Has anyone written a C wrapper to ease development with the AMI? I found a
couple of c++ ones, but not C.
Thanks!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
We are building an app that will initiate outbound calls using .call files, and
each call can be a different duration (eg: 1min to 5min). These calls will go
through an Asterisk service with other calls/apps running.
I need to control the MAX number of channels in use so I don't overload this
We ran into this a few years ago. Polycoms and Grandstreams worked fine with
#xxx extensions, but Aastra's would not. Could not dial extensions beginning
with #
We chased Aastra tech support for 2 weeks. They acknowledge the bug, and we
were told they would fix this in their next firmware
If you check the archives you might find the original messages on this topic
from a few years ago...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
[davies...@gmail.com]
Sent: Wednesday,
Cool topic!
Our company (generationD) developed some CID scripts for free use, and we would
be interested in building and hosting this service.
On the spec side, how do we avoid users claiming numbers belonging to others?
(Could be an admin nightmare)
Do we allow number ranges?
Do we require
I'm looking for recommendations for standalond PRI to SIP converters. (Needs
to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy features...
Thanks!
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I think the OP's point was that open source should mean:
Free to modify
Free to contribute code
Free to use.
Leaving the first two but taking away the free to use really takes the F out
of FOSS. There have been other posts discussing Digium's license requirements,
code ownership, etc. I
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