On Sat, 18 Feb 2012 20:50:25 +0100, Andreas Sikkema h...@ramdyne.nl
wrote:
We're using a GSM gateway to send SMS messages from our network
monitoring system. Once you dig through some chipset specs it was
suprisingly easy to start sending SMS messages. While we didn't
investigate receiving
On Sat, 18 Feb 2012 12:21:31 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Not true. Some GWs have only a phone port that you connect to an ATA.
Good to know. What brands/models would you recommend?
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On Thu, 16 Feb 2012 19:41:16 +0100, Olivier oza_4...@yahoo.fr wrote:
You mean you can receive SMS on a landline in France (or the opposite) ?
Supposedly, but I never used it either.
www.google.fr/search?q=sms+ligne+fixe+asterisk
If a gateway has its own SIM card and GSM stuff, should it receive
On Wed, 8 Feb 2012 18:17:46 -0800, Chad Wallace
cwall...@lodgingcompany.com wrote:
Maybe the release announcements are what you're looking for. e.g.,
for 1.8:
http://www.asterisk.org/node/51444
And you can probably find the same for 1.4, 1.6.x, and 10 without too
much trouble.
Thanks. It's
On Wed, 08 Feb 2012 20:23:54 -0600 (CST), Richard Mudgett
rmudg...@digium.com wrote:
The CHANGES file is not just a dump. It is a manually created file that
documents each feature addition. There is a ChangeLog file that is a dump
of every single commit made to the source file.
Sorry about
On Thu, 9 Feb 2012 11:13:38 +, Steven Howes
steve-li...@geekinter.net wrote:
Why not just use the latest version?..
Because converting Asterisk to run on that non-x86 platform is quite
some work, so I need to know what I'm missing by staying with a 1.4.x
release.
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On Wed, 08 Feb 2012 15:58:43 -0600, Kevin P. Fleming
kpflem...@digium.com wrote:
No, unfortunately that's not quite correct. The UPGRADE files list
*important* changes that users need to know about because they are
changes in behavior of existing functionality. New features, even really
useful
On Mon, 06 Feb 2012 10:24:42 -0600 (CST), Richard Mudgett
rmudg...@digium.com wrote:
The UPGRADE.txt and CHANGES files do just that. They have been a part
of the Asterisk source files for a long time.
Thanks for the info. The problem is that the ChangeLog files
On Tue, 7 Feb 2012 14:31:31 +, Steven Howes
steve-li...@geekinter.net wrote:
The upgrade files may be more to your tastes than changes files.
Thanks. I downloaded and untarred asterisk-1.8.8.0.tar.gz, and it
looks like the UPGRADE*.txt files within tarballs are the closest
there is to knowing
On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com
wrote:
Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What
technology FreeSwitch is used and asterisk don't. I don't know it's the
right or wrong but this question come to my mind...
Provided Asterisk, even
On Tue, 07 Feb 2012 06:10:37 -1000, Jean-Denis Girard
jd.gir...@sysnux.pf wrote:
This link also presents changes between Asterisk versions:
http://linuxinnovations.com/applications1.4-1.6.2.html
Thanks for the link.
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Hello
Is there a document that sums up the major changes made to the four
main releases available (1.4, 1.6, 1.8, and 10), to check if it's
worth upgrading?
www.asterisk.org/downloads
Thank you.
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On Wed, 01 Feb 2012 18:47:49 -0500, James Sharp ja...@fivecats.org
wrote:
The Cisco DDR2200 that I just got from Centurylink for DSL appears to be
just that. I haven't tested the FXS ports on it yet, though.
Cisco announces the end-of-sale and end-of-life dates for the Cisco
DDR2200, DDR2201,
Hello
In case a NAT firewall prevents using STUN to open SIP/RTP ports, a
solution is to first connect the phone to the Asterisk server through
a tunnel, and then have data go through the tunnel.
Are there hardphones that support OpenVPN?
If none, what about SSH? Is this a good
On Tue, 31 Jan 2012 12:54:41 + (GMT), Arthur Stanfield
a...@dmcip.com wrote:
You can't tunnel UDP through SSH.
Some of the newer Grandstream handsets support OpenVPN and are a bit cheaper
than the Snom alternatives.
Thanks for the infos. So the only way to use SIP through locked-down
NAT
On Tue, 31 Jan 2012 07:57:22 -0500, bakko asannu...@gmail.com
wrote:
yeallink T26 and T28 support OpenVPN too
Thanks for the infos.
If someone tried the Snom, Grandstream, or Yeallink, how good is their
OpenVPN connection?
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Hello
To cut down on the number of hackers trying to break into an Asterisk
server, I'd like to simply move the SIP port from the standard UDP
5060 to something non-standard.
Since this server must be able to receive INVITEs from any SIP UA
(server or client), it appears that I must add an SRV
On Tue, 31 Jan 2012 10:03:46 -0600, Jeff LaCoursiere
j...@sunfone.com wrote:
Using Yealink T-28 with OpenVPN works fine - about three weeks now with
no issues. Bummed that it seems to only support one tunnel, though. I
asked their support team if they could make whatever changes necessary
to
On Tue, 31 Jan 2012 10:44:12 -0600, Jeff LaCoursiere
j...@sunfone.com wrote:
No - the phone allows you to register with multiple servers, and I would
like to reach each server over its own tunnel. It won't do that today.
Thanks for the info.
--
On Tue, 31 Jan 2012 16:25:37 -0600, Dale Noll dn...@wi.rr.com wrote:
You can also setup OpenVPN to connect a remote subnet (remote office)
and it will route all traffic between subnets. Configure the hard/soft
phones on the remote subnet to route through the OpenVPN. This works
pretty well for
On Tue, 31 Jan 2012 18:22:41 +0100, Daniel Pocock
dan...@readytechnology.co.uk wrote:
Something more appropriate for your goal might be a move to TLS, it is
definitely needed for any external connectivity
[...]
As a further safety measure, you could use something like repro or
Kamailio as a SIP
Hello
I don't understand how I should use the allowguest item: If set to
yes, callers from the Net should authenticate, but then, how can I
allow strangers to call extensions in my system?
allowguest
If set to no, this disallows guest SIP connections. The default is to
allow guest
On Tue, 24 Jan 2012 09:55:12 -0500, Jim DeVito
asterisk-users-mailing-l...@devito.cc wrote:
What they are talking about is SIP URI dialling. Let say you have
extension 1000 the rings a phone on your system. With allowguest=yes I
would be allowed to dial SIP:/1...@yourdomain.com and assuming the
On Tue, 24 Jan 2012 09:26:26 -0600, Kevin P. Fleming
kpflem...@digium.com wrote:
By definition this is impossible. If the caller is a 'stranger', that
means you have no knowledge of them prior to their INVITE request
arriving at your server. If you have no knowledge of them, then you
don't have
Hello
I just read this article about an Asterisk server that got hacked to
make free international calls through an ITSP:
www.rowetel.com/blog/?p=2210
I have a couple of questions:
1. Am I correct in understanding that SIP ALG on a router makes it
easier to host an Asterisk server on a
On Sat, 07 Jan 2012 09:27:29 -0500, sean darcy seandar...@gmail.com
wrote:
But what really made us choose linphone was you use it on android/iphone.
That has been a huge plus. As a bonus, you can use any degegistered
smartphone - that is, one not hooked up to the cellular network,only
wireless
On Sat, 7 Jan 2012 12:34:44 -0500, Sean Darcy seandar...@gmail.com
wrote:
Yes, I did mean de-registered. I meant a phone that no longer has the
ability to use the cellular network - only wifi. For instance, we have
a couple of Droids that used to be on Verizon. They work just fine as
sip-phones
On Thu, 1 Dec 2011 14:09:29 +0300, James Mutuku listmut...@gmail.com
wrote:
I have worked with bare asterisk + freepbx before. the mypbx was just
an example but my reference to appliances as a whole.
The appliances seem to have lower entry costs.
Appliances have less RAM + storage, so you'll
On Mon, 19 Sep 2011 12:12:54 +0200, Gilles codecompl...@free.fr
wrote:
Problem solved: Tried XLite 4.1 on another test, and sound is OK, so I
guess it's something in my work PC.
s/test/host/
An el cheapo $20 CMedia CMI8738 6CH solved the issue. Great sound
On Sun, 18 Sep 2011 22:28:32 +0200, Gilles codecompl...@free.fr
wrote:
For some reason, even through I'm using a headset, there's a lot of
echo and after a few seconds, it sounds like it enters a very fast
loop before the echo stops somewhat. IOW, unusable sound.
Problem solved: Tried XLite 4.1
Hello
I just set up Asterisk 1.6.2.9 through packages on a test host
running Ubuntu 11.04, configured sip.conf/extensions.conf, and
launched EyeBeam 1.5.20 to run the echo test.
For some reason, even through I'm using a headset, there's a lot of
echo and after a few seconds, it sounds
On Fri, 16 Sep 2011 11:13:16 +0500, Sam Govind govoi...@gmail.com
wrote:
The image you provided didn't open so I'm not sure about the design.
Sorry about that. It's a PNG file and it opens in the two browsers I
tried.
The reason I don't simply get a subscription with a VoIP provider and
must go
On Fri, 16 Sep 2011 12:49:51 +0200, Jeroen Eeuwes
jeroeneeu...@gmail.com wrote:
I think this is a very common situation, so I'm not really sure what
your problem is. Perhaps it's because I don't use an internal card,
but in my situation it works just fine. I dial a number on my SIP
phone, Asterisk
On Fri, 16 Sep 2011 10:54:48 -0500, Kevin P. Fleming
kpflem...@digium.com wrote:
This is true, but you already answered your own question in your
original post: since Asterisk cannot know whether the called party
(dialing out via an FXO port) has answered or not, it assumes the
outgoing call is
On Fri, 16 Sep 2011 19:35:19 -0400, Eric Wieling ewiel...@nyigc.com
wrote:
It does on PRI.
Unfortunately, this is for an ADSL modem, hence the connection to its
FXS port :-/
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Hello
My ISP provides an FXS port to plug a handset, which can be used to
make free calls to (GSM) cellphones, similar to the Billion ADSL
modems:
http://au.billion.com/product/voip.php
My plan is to install an SIP client on a smartphone, so that when I'm
travelling, I can connect to a
On Thu, 08 Sep 2011 14:52:06 -0400, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 08/09/11 02:19 PM, Cobra 2 wrote:
I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and
I've gotten asterisk to run on that just fine.
I think the question is, can you answer your incoming
On Fri, 02 Sep 2011 16:37:32 +0200, Tamer Higazi
th9...@googlemail.com wrote:
Do you want to run the entire PBX on the Android client or are you just
looking for a IAX programm to be installed for receiving calls?!
The entire PBX so I can have an IVR in the phone.
--
Hello,
Out of curiosity, has Asterisk been successfully compiled and ran
Asterisk on an Android smartphone?
I could use a small IVR on my smartphone to handle incoming calls.
Thank you.
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On Fri, 2 Sep 2011 13:23:18 +0100, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
TTBOMK it's been done; but without the necessary Zaptel / DAHDI drivers to
interface with the phone line, it's rather less useful than it sounds.
I'm looking for a way to an IVR in my smartphone to handle incoming
On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
cur...@telecomabmex.com wrote:
Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.
Thanks for the tip. It looks like the smallest option is 8 FXO ports:
www.xorcom.com/telephony-interfaces/astribank-models.html
--
On Sat, 27 Aug 2011 09:31:12 -0600, linux guy linuxguy...@gmail.com
wrote:
I'm looking for an FXO device to connect to a POTS line that communicates
via USB or Ethernet.
For USB, AFAIK, there's only the one from Sangoma. All others are
Ethernet-based.
www.voip-info.org/wiki/view/VoIP+Gateways
On Thu, 28 Jul 2011 12:04:38 +0500, Faisal Hanif fai...@vopium.com
wrote:
I have tried asterisk on windows XP using Cygwin and it worked fine.
Would you mind explaining how to do this?
Thank you.
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On Thu, 28 Jul 2011 12:46:03 -0500, Danny Nicholas
da...@debsinc.com wrote:
Interrupting - you have to not use DAHDI (SIP Only) and make sure you have
the necessary libs downloaded in your Cygwin install.
It's OK, I don't mind using a VoIP gateway instead of a PCI card. Has
someone written an
On Thu, 28 Jul 2011 13:08:33 -0500, Danny Nicholas
da...@debsinc.com wrote:
If they have, it would probably be on www.nerdvittles.com
It looks like The Incredible PBX runs on CentOS
www.nerdvittles.com/index.php?p=740
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Hello,
Since Asterisk has been ported to exotic platforms like SOHO routers
(Linksys, Buffalo, etc.) and non-MMU CPUs (Blackfin, etc.), I was
wondering why the Windows port never really took off.
As far as I can tell, www.asteriskwin32.com is a one-man effort
(Patrick Deruel's) that is not going
On Tue, 26 Jul 2011 07:28:27 +, Soeren Malchow (MCon)
soeren.malc...@mcon.net wrote:
And asterisk just runs fine on linux why bother ?
Because I, for one, would like to run Asterisk on my Windows
workstation at home as an enhanced answering machine :-)
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On Tue, 26 Jul 2011 10:59:22 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Patches are welcomed.
Does someone know the kind of changes that were made by AsteriskWin32,
and how hard it'd be to apply them to more recent releases of
Asterisk?
--
On Tue, 26 Jul 2011 12:07:10 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
There were some later fixes at around 1.6.0 to try to get the code built
on cygwin. I would suggest you to try building it on cygwin and see
where things fail.
Also grep for CYGWIN or such in the source (especially
On Mon, 18 Jul 2011 20:59:02 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
/usr/lib/asterisk/modules/
Be sure to only include the ones you need. Finding which exactly may be
tricky.
Thanks Tzafrir. Actually, since the modules are the biggest files by
far, besides the obvious (SIP, Dahdi,
On Tue, 19 Jul 2011 09:27:41 -0500, Danny Nicholas
da...@debsinc.com wrote:
My .02 - FWIW, DAHDI will use almost as much space as the rest of Asterisk,
so you could save the space you don't have by forgoing that.
Thanks everyone for the feedback. I'll go through the list of modules
and see what I
Hello
I'd like to build a compact, affordable, fanless x86 solution to
handle my home landline.
I know about the following two platforms:
1. www.pcengines.ch/alix.htm
alix1d + case 100
Does Availability 500 mean that it's just not possible to buy just
one item?
2.
On Mon, 18 Jul 2011 08:04:31 -0400, John Novack
jnov...@stromberg-carlson.org wrote:
Just about any of the HP thin clients, either new or used off eBay, with
AstLinux installed do a wonderful job, especially if you are not going
to need a PCI card.
The older units will need a larger flash.
On Mon, 18 Jul 2011 09:03:52 -0400, John Novack
jnov...@stromberg-carlson.org wrote:
there are other low cost solutions around as well.
the ALIX boards I have seen do not impress me. I think they are somewhat
overpriced. Jut one opinion
Thanks for the feedback. I'll read what HP has to offer.
Hello,
I'd like to run Asterisk on an embedded device, where space is scarce.
It should be able to handle calls from a VoIP provider in SIP, calls
from the PSTN through Dahdi, and voicemail.
If someone's already done this, I'd like to know which
directories/files are required for a basic
On Tue, 12 Jul 2011 11:10:28 -0400, Steven Stromer
fil...@stevenstromer.com wrote:
A quick to implement open source network monitoring tool is smokeping:
http://oss.oetiker.ch/smokeping/index.en.html
Thanks guys for the tip on qualify=yes and SmokePing.
--
On Thu, 7 Jul 2011 09:32:08 +0500, Faisal Hanif fai...@vopium.com
wrote:
Community can help you better if you provide some details about you scenario
and requirement.
It's a very simple scenario: The Asterisk server is connected to a
VoIP provider for calls to the PSTN, and I'd like to have
Hello
I was wondering if Asterisk can be configured to monitor a
connection to a VoIP provider, whether someone is currently using it
for a call or the connection is idle?
FWIW, my VoIP provider doesn't run an iperf server on their side. I
don't know if ping/traceroute is a good enough
Hello
I just read this article about a kid in England who built a box with a
3G SIM card:
www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html
When someone rings your intercom, the box will call your cellphone so
On Tue, 7 Jun 2011 13:06:23 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Why bother when you can buy off the shelf stuff to do it for you.
The trick is that this connector must work with existing interphones,
such as this one at home:
On Wed, 11 May 2011 01:09:16 +0800, Scott Zhang
macromars...@gmail.com wrote:
So does this mean no solution when used ZAP/DAHDI with PSTN line?
If I installed an E1, will that work?
Before getting an E1, maybe ISDN provides call supervision?
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On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell sruff...@digium.com
wrote:
I know this thread is dead but: I do not believe this should go into the DAHDI
kernel modules.
I agree. It's just too bad Dahdi is unable to report how an outgoing
call is doing: Still ringing, busy, answered.
--
On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
beaasteriskg...@gmail.com wrote:
Anybody can explain me why asterisk is unable to detect ringback tone
from PSTN telco ? .
I guess it was a lot of work, and nobody bothered adding this to the
Zaptel driver.
--
On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali
beaasteriskg...@gmail.com wrote:
The problem here is that as soon as asterisk dialing on fxo lines it
sets channel status as answered although the chennel is getting
ring back tone from
other party.
Anyone can suggest me to solve this issue ?
The
On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Have a look at these:
Thanks much Gordon. I'll study the scripts you mentionned. It looks
like iptables is good enough and I won't have to install a second tool
to watch the logs and reconfigure iptables
On Tue, 5 Apr 2011 17:38:15 -0400, Paul Dugas
p...@dugasenterprises.com wrote:
First, this appears to be working for me though I'm not 100% sure of
that and cannot guarantee it will for you in any way, shape or form.
With the lawyering out of the way...
Thanks a lot, Paul.
--
Hello
I'm no expert of iptables, and it seems like it can handle banning
IP's that are trying to register and fail too many times.
I'd like to use this feature instead of having to install a second
tool such as SSHGuard or BFS that parses the logs and reconfigure
iptables on the fly.
Is
On Wed, 30 Mar 2011 01:45:20 +0300, Ioan Indreias indre...@gmail.com
wrote:
Just to provide an alternative to sshguard: you could use BFD[1]
Thanks Ioan. I'll give it a shot.
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On Wed, 30 Mar 2011 16:54:51 -0500, Darrick Hartman
dhart...@djhsolutions.com wrote:
One of our developers on the AstLinux team worked out a plugin for
Arno's firewall (iptables based) which performs similar to fail2ban, but
uses bash. He called it adaptive-ban. You might be able to adapt it
On Tue, 29 Mar 2011 07:48:08 +0200, magnu...@inputinterior.se wrote:
I was a little unclear, it is not the cell phone that does the call-back, it
is the cell-phone-network.
Makes more sense :-) Thank you.
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On Mon, 28 Mar 2011 08:20:23 -0400, vip killa vipki...@gmail.com
wrote:
Is anyone using asterisk with fail2ban?
Sorry for hi-jacking the thread, but I was wondering if there were a
lighter alternative that I could run on appliances?
Python uses too much RAM, but I need to find a way to ban
On Tue, 29 Mar 2011 07:31:18 -0500 (CDT), Joe Greco
jgr...@ns.sol.net wrote:
sshguard is *extremely* lightweight compared to most things; it's a very
efficient compiled C application that doesn't have (m?)any dependencies.
Thanks much for the tip. I'll study how to install/configure iptable
and
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table for sip endpoints)
Thanks for the idea, but it's not possible, as the Asterisk must
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Remember guys, there's a LOT of IP blocks out there that are almost
definitely not going to be somewhere you expect to receive SIP traffic
from.
I agree. Is there a list I could use to check which blocks have
On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote:
On 03-29-2011 19:25, Steve Edwards wrote:
Really? How many callers are you expecting from North Korea, Libya, China,
Iran, etc?
after reviewing last week's log i'd say around 25-28k/min :)
So it looks like I should check out sshguard
On Sat, 26 Mar 2011 14:58:30 +0100, magnu...@inputinterior.se wrote:
Celluar Network - E1 - Avaya - OOH323 - Asterisk
Thanks for the tip.
So here's how it works:
1. The web app calls a script that uses AMI + Originate to send a call
to the Avaya PBX
2. Avaya is able to check that a number
On Mon, 28 Mar 2011 14:12:09 +0200, magnu...@inputinterior.se wrote:
Its not the Avaya that makes the call back, it is mobile.
I thought the way you handled things, is that Asterisk would call your
cellphone through the Avaya PBX just to check whether the cellphone is
in_use/busy. At what point
On Sat, 26 Mar 2011 10:50:19 +0100, magnu...@inputinterior.se wrote:
I am looking for a way to check the status of a cell phone. Found one way that
worked for me and would like to have some feedback or suggestion of
improvments.
I'd like to check I understood: Your Asterisk server is connected
On Fri, 18 Mar 2011 16:48:28 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
Somehow, I'm guessing that 'failed' means that something failed while
processing the call file or that the call failed to answer, not that
somebody terminated the call.
Thanks guys. After testing with a PCI
On Tue, 15 Mar 2011 14:54:53 +0100, Gilles codecompl...@free.fr
wrote:
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
For those
On Tue, 15 Mar 2011 11:44:20 -0500, Danny Nicholas
da...@debsinc.com wrote:
Don't depend on the tutorials you read to be 100% accurate or up-to-date.
The default action on a failure in Asterisk is usually going to be an s
jump, either to s,1 or s+100. Personally, I would replace failed,1 with
On Fri, 18 Mar 2011 10:14:37 -0500, Danny Nicholas
da...@debsinc.com wrote:
exten = start,n,Playback(manolo_camp-morning_coffee)
;exten = start,n,Hangup()
exten = start,n,Goto(${EXTEN}-${REASON})
;not run
;exten = failed,1,NoOp(Call ended with ${REASON})
;not run
;exten = s,1,NoOp(Call ended
On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
On Fri, 18 Mar 2011 17:56:12 -0500, Anthony Messina
amess...@messinet.com wrote:
You need to define the 'failed' extension in your context to have the
${REASON} variable set (I've found).
exten = failed,1,NoOp(Failure reason is: ${REASON})
Thanks but for some reason, after calling out through a
Hello
I'd like to install Asterisk and Dahdi on a Ubuntu host using packages
instead of compiling from the source.
Are the following packages enough for this?
==
asterisk - Open Source Private Branch Exchange (PBX)
asterisk-config - Configuration files for Asterisk
dahdi - utilities
On Thu, 17 Mar 2011 13:01:39 +0200, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
BTW, I notice dahdi-dkms: Does it mean that when I upgrade the
kernel, I'll also need to upgrade Dahdi?
Yes, basically.
Good to know. Thanks for the tip.
--
Hello
I thought I had things set OK to have Asterisk play FR files for
prompts and MOH, but for some reason, it still can't find them:
ll /var/lib/asterisk/sounds/
drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/
drwxr-xr-x 10 root root61440
On Thu, 17 Mar 2011 09:23:35 -0500, Danny Nicholas
da...@debsinc.com wrote:
Moh should be in /var/lib/asterisk/moh not /var/lib/asterisk/sounds or in
this case /var/lib/asterisk/moh/custom.
Thanks for the tip, but after moving the MOH files to the right
location, and even restarting Asterisk, it
On Thu, 17 Mar 2011 15:09:18 +, Ishfaq Malik i...@pack-net.co.uk
wrote:
MusicOnHold() doesn't take a file name as a parameter, it takes a class
name or if left blank, plays from the default class
Yes, thanks for the tip.
Found it: Turns out the Ubuntu package expects sound files to be
On Wed, 16 Mar 2011 22:45:35 +1100, John Kosmas
batc...@optusnet.com.au wrote:
i have the same problem but it doesnt always happen tho from the same
caller.
im using Asterisk 1.4 - maybe newer version updates have
had bug fixes. maybe this could rectify it.
Thanks John, but I still get the
Hello
The Ubuntu Asterisk package doesn't install
/etc/modprobe.d/dahdi.conf, so I was wondering where to put the
following line:
===
options wctdm opermode=FRANCE
===
Should it be in /etc/dahdi/modules?
===
options wctdm opermode=FRANCE
===
Thank you.
--
On Thu, 17 Mar 2011 10:48:07 -0500, Danny Nicholas
da...@debsinc.com wrote:
You should manually create /etc/modprobe.d/dahdi.conf since
/etc/init.d/dahdi start is going to do a modprobe and that's the only way
you're going to get this option started correctly (subject to correction).
Thanks for
On Tue, 15 Mar 2011 13:45:00 -0400, Paul Belanger
pabelan...@digium.com wrote:
Is this an analog line? If so, is your CO providing a disconnect tone?
Yes, it's an analog line, but it's actually VoIP provided by an RJ11
on an ADSL modem, not a real landline.
Is there a way to check how the
Hello
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
== extensions.conf
;Play MoH for a few seconds, hang up, and
;check
On Tue, 15 Mar 2011 14:54:53 +0100, Gilles codecompl...@free.fr
wrote:
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
It looks like
Hello
For some reason, when dialing out through a call file and the remote
line is busy, Asterisk doesn't jump to the failed extension in the
context used by the call file:
== call file
Channel: Zap/1/5551234
Context: callbacktest
Extension: start
Priority: 1
MaxRetries: 1
==
On Tue, 08 Mar 2011 13:22:18 +0100, Gilles codecompl...@free.fr
wrote:
I need to write a script which prompts the callee to type a number,
and then read it back to them as confirmation:
Apparently, the right way to read a phone number back to the user is
not to use SayNumber() (which might be OK
On Thu, 10 Mar 2011 13:18:41 +0100, Dave Cotton
dcot...@linuxautrement.com wrote:
Look at the GotoIf statement for example
Thanks Dave for the tip, but I found that I needed to change a pattern
that was already in say.conf:
===
[fr](date-base,digit-base) ;BAD _[n]um:0. = num:${SAY:1}
On Thu, 10 Mar 2011 15:30:51 +0200, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
I think you're missing SayDigits().
say.conf does use the syntax of the extensions.conf, but it's not a
dialplan.
Thanks for the input, but SayDigit() isn't right for what I want to
do, since it simply reads a
On Thu, 10 Mar 2011 14:37:45 +0100, Gilles codecompl...@free.fr
wrote:
I figured out how extensions.conf and say.conf work and posted my
results in the reply to Dave.
Noticed something strange, though: 0800123456 is played OK (ie.
0.800.12.34.56) , but 092123456 is played digit by digit (0.8.9.2
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