Hello list,
I'm having some troubles with DTMF tones. When pressing numbers on a
Snom phone, the DTMF-signal takes too long.
I have the following in sip.conf :
dtmfmode = rfc2833
which works well for Grandstream, Yealink and Cisco phones. But not for
Snom.
Snom support tells me I should
Hi,
I am having issues sending and receiving fax on my asterisk setup.
Currently I have a server that has 2 x E1 TDM cards one is sangoma and the
other
one is openvox. Both support echo cancellation.
One of the e1 is connected to our telco provider via mfcr2 where all our
incoming calls
Hello,
I was wondering: What does Dial(Local/...) offer that a Goto()
doesn't?
For instance:
;exten = h,n,Goto(callback,start)
exten = h,n,Dial(Local/start@callback)
[callback]
exten = start,1,Verbose(In callback)
Thank you.
--
On Fri, 2011-02-18 at 08:10 +0100, Hans Witvliet wrote:
On Fri, 2011-02-18 at 00:51 +0100, Albert wrote:
On 18.02.2011 00:30, Andrew Joakimsen wrote:
On Sat, Feb 12, 2011 at 07:31, ast guy ast...@gmail.com wrote:
Hi,
I have been out of touch with asterisk for quit some time and
Well you simple use dtmfmode=info in peer configuration of Snome phone.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, February 18, 2011 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
The difference you will feel when using callback files or AMI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, February 18, 2011 1:31 PM
To: asterisk-users@lists.digium.com
Subject:
Dear Customer Support,
i connected the asterisk to a e1 interface of our hipath4000. outgoing
calls from a sip peer of my asterisk to an up0 telephone which iss
connected to the hipath4000 are working. If you want to dial from an up0
device to the e1 interface where asterisk is connected to,
On Fri, 18 Feb 2011 14:10:54 +0500, Faisal Hanif fai...@vopium.com
wrote:
The difference you will feel when using callback files or AMI.
Thanks Faisal.
Two of the most common areas where Local channels are used include
members configured for queues, and in use with callfiles. There are
also
This is not Digium's customer support address but free public emailing list
for asterisk user's contributed by community volunteers.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jan Zieher
Sent: Friday, February 18, 2011 2:19 PM
Dnia Fri, 18 Feb 2011 15:16:40 +0800
Malvin Rito mr...@mail.altcladding.com.ph napisał(a):
Were upgrading our network switches and need to create multiple
VLAN groups, but since our Squid Proxy (Transparent Proxy) Server
should be accessible to all VLAN groups we need to setup a trunk
If it is a E1/T1 trunk then all the channels are grouped in one group, once
done place your Asterisk server inside your squid network. I hope this
helps!
On Fri, Feb 18, 2011 at 12:46 PM, Malvin Rito mr...@mail.altcladding.com.ph
wrote:
Hi List,
Were upgrading our network switches and
Dean's link has references to Trixbox. TB has a bad, bad, very bad reputation
for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All
are Asterisk based and very easy to set up.
From: asterisk-users-boun...@lists.digium.com
i prefer to go with Elastix very easy to setup and maintain and reach UI
rather than freePBX
cheers
Dhaval
On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote:
Dean’s link has references to Trixbox. TB has a bad, bad, very bad
reputation for being very insecure.
Hello,
trying to load ael module in asterisk ver 1.6.2 got the following:
asterisk*CLI module load pbx_ael.so
Unable to load module pbx_ael.so
Command 'module load pbx_ael.so' failed.
[Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error
loading module 'pbx_ael.so':
Please could someone advise good manual for using lua for asterisk dialplan.
There is not much docu about it.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Hi,
I'm trying to automatically have the dialplan assign an extension to a
roaming phone on my network.
I tried the following without success:
exten = 3001,1(readop),BackGround(beep)
exten = 3001,n,Read(digito,vm-youhave,3)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Set(ROAM=${digito})
Yes, I use Elastix myself too. Funny that I didn't mention that one!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: Friday, February 18, 2011 6:11 AM
To: Asterisk Users Mailing List - Non-Commercial
Asterisk is open source and you can install in a normal PC itself and you
can avail all the features that proprietary system has.
If you want to integrate with any VoIP service then a PC with Asterisk is
enough or else if you want to integrate with PSTN lines then you need FXO
card to be
I'm VERY partial to Aastra's devices. Seriously, they don't take as long to
boot as Polycoms, they're relatively inexpensive but are not CHEAP (like a
certain brand beginning with a G, in my opinion), they have decent web
interfaces (also unlike the unnamed brand I non-mentioned a moment ago),
On Fri, Feb 18, 2011 at 2:44 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
I'm VERY partial to Aastra's devices. Seriously, they don't take as long to
boot as Polycoms, they're relatively inexpensive but are not CHEAP (like a
snip
Great info.
I do have a complaint about Aastra
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Friday, February 18, 2011 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Assigning an extension to a
Did you checked if you extension.ael doesn't have syntax error?
Did you upgraded anything after last compile?
Or
Try a clean recompile
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin
Sent: Friday, February
The only specific you need to do in extensions.lua is create a table to put
your extensions in like,
Extension{
}
Else all will be LUA code and all asterisk applications can be called as
app.application_name.
Regards,
Faisal Hanif
From:
On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif fai...@vopium.com wrote:
Did you checked if you extension.ael doesn’t have syntax error?
I think there is no error. I loaded the standard ael first (provided by
asterisk) then my test config, got the same result.
Did you upgraded anything after
Pls could you share some lua config which contains mysql quires
On Fri, Feb 18, 2011 at 3:38 PM, Faisal Hanif fai...@vopium.com wrote:
The only specific you need to do in extensions.lua is create a table to put
your extensions in like,
Extension{
}
Else all will be LUA code and
Are you on CentOS?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin
Sent: Friday, February 18, 2011 7:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pbx_ael.so: undefined symbol:
Hey Users,
I am using record application to record MeetMe conf. but look like its creating
individual files for every channel. What applucation is best to record MeetMe
conf ?
~ # ls -l /var/spool/asterisk/monitor/
total 489220
-rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42
Linux version 2.6.26-2-686 (Debian 2.6.26-24) (da...@debian.org) (gcc
version 4.1.3 20080704 (prerelease) (Debian 4.1.2-25))
On Fri, Feb 18, 2011 at 4:09 PM, Faisal Hanif fai...@vopium.com wrote:
Are you on CentOS?
*From:* asterisk-users-boun...@lists.digium.com [mailto:
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Friday, February 18, 2011 9:12 AM
To: asterisk-users
Subject: [asterisk-users] Meet me recording
Hey Users,
I am using record application to record MeetMe
The error appears only if I load module. There is no warning during
installation, so module pbx_ael.so is compiled and placed in modules dir of
asterisk
On Fri, Feb 18, 2011 at 4:15 PM, Borin katerin.bo...@gmail.com wrote:
Linux version 2.6.26-2-686 (Debian 2.6.26-24) (da...@debian.org) (gcc
Hello
I'm using an AGI script in Lua to make a callback through Zaptel.
For this to work, I must wait until the channel is idle, or I get this
kind of error, even after waiting over 10 seconds after the remote end
rings once and hangs up:
==
channel.c:2863 __ast_request_and_dial:
On Fri, 18 Feb 2011, Gilles wrote:
I'm using an AGI script in Lua to make a callback through Zaptel.
=== AGI script
#!/var/tmp/lua
for i=1,10 do
io.write(CHANNEL STATUS\n)
response=io.read()
_, _, key, value = string.find(response, (%a+)=(%d+))
On Fri, 18 Feb 2011 07:52:40 -0800 (PST), Steve Edwards
asterisk@sedwards.com wrote:
I've never written an AGI in lua, but don't you have to read the AGI
environment (from STDIN) before issuing requests?
Thanks for pointing it out. I forgot to prepend that part to that test
script.
Also,
Hi,
I'm thinking that _4XXX is an over-complication. _4XXX means you could
dial any number from 4000 through 4999 inclusive and get the extension at
SIP/${ROAM}.
Well it's kind of what I want.
I have a roaming phone that comes in. He dials 3001, sets his
extension to 123, so that he is assigned
On Fri, 18 Feb 2011 17:11:18 +0100, Gilles codecompl...@free.fr
wrote:
I'm guessing you would have better luck kicking off an external process
that checks the channel status via AMI.
Yes, it looks like it's not possible to reuse the FXO from either
extensions.conf or through an AGI script.
I'm
I think I have 3 PSTN lines because I can connect a normal telephone to them
all and make calls between each of them.
We have 5 normal telephones and 1 panasonic.
From what I got I need a PC and a of PCI card to interface to my 3 external
lines and my 6 internal lines.
For the PC I was planning
I think I have 3 PSTN lines because I can connect a normal telephone to them
all and make calls between each of them.
We have 5 normal telephones and 1 panasonic.
From what I got I need a PC and a of PCI card to interface to my 3 external
lines and my 6 internal lines.
For the PC I was planning
On Fri, 18 Feb 2011, Gilles wrote:
I'm not having much luck with AMI: After typing the right commands, it
just stays there, not replying to the Login action:
= Telnet to TCP5038
Action: Login
Username: admin
Secret: secret
just stays there, waiting
It's waiting for another
Yes, use a FXO device, like the AudioCodes MP-114. It is an external gateway
that will allow you to interface your PSTN lines to Asterisk via IP. There are
other brands out there but in my line of business we only use AudioCodes.
From:
(Please don't top-post and please trim posts that are no longer relevant.)
On Fri, 18 Feb 2011, Francisco Javier Cintrón Olguín wrote:
I think I have 3 PSTN lines because I can connect a normal telephone to
them all and make calls between each of them. We have 5 normal
telephones and 1
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Friday, February 18, 2011 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assigning an extension to
On Fri, 18 Feb 2011 09:19:01 -0800 (PST), Steve Edwards
asterisk@sedwards.com wrote:
The action and username lines were followed by pressing ENTER.
The secret line was followed by pressing ENTERENTER.
Thanks for the tip. I figured this out after a while ;-)
I can now successfully log on,
Hi guys,
I'm trying to connect Asterisk to the MySQL, but I can't execute it. It
returns an error, as below:
-- Executing [200@teste:2] MYSQL(Console/dsp, Query resultid 1 SELECT/
ramal/ FROM/ colaboradores/ WHERE/ ramal=200) in new stack
[Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393
On Fri, 18 Feb 2011 09:19:01 -0800 (PST), Steve Edwards
The action and username lines were followed by pressing ENTER.
The secret line was followed by pressing ENTERENTER.
On Fri, 18 Feb 2011, Gilles wrote:
Thanks for the tip. I figured this out after a while ;-)
I can now successfully
-- Forwarded message --
From: Felipe Figueiredo felipe.figueired...@gmail.com
Date: Fri, Feb 18, 2011 at 4:03 PM
Subject: Re: [asterisk-users] cmd MySQL
To: Gerald A geraldabli...@gmail.com
- Executing [200@teste:2] MYSQL(Console/dsp, Query resultid 1 SELECT\
ramal\ FROM\
If you are using asterisk 1.8.x you don't need to type \ for spaces you can
write simple query and use spaces as normal it will work fine.
Faisal
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Sent: Friday,
Faisal,
yes, that's it!! I'm using 1.8.x , I didn't know about it!!!
Thank you so much, guy!!!
On Fri, Feb 18, 2011 at 4:26 PM, Faisal Hanif fai...@vopium.com wrote:
If you are using asterisk 1.8.x you don’t need to type \ for spaces you can
write simple query and use spaces as normal it will
On Friday 18 February 2011 05:29:56 Borin wrote:
Hello,
trying to load ael module in asterisk ver 1.6.2 got the following:
asterisk*CLI module load pbx_ael.so
Unable to load module pbx_ael.so
Command 'module load pbx_ael.so' failed.
[Feb 18 11:25:47] WARNING[7412]: loader.c:449
Thanks,
look like monitor application resolved my issue.
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 18 Feb 2011 09:16:36 -0600
Subject: Re: [asterisk-users] Meet me recording
From: asterisk-users-boun...@lists.digium.com
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as
Try to use Answer() in your dial plan. Not sure though but it had been
resoved my issue years ago.
--
Sent from my iPhone
On Feb 18, 2011, at 3:59 PM, Cassius Smith cass...@cassius.org wrote:
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with
VOIP
only trunks, and this
On 11-02-18 03:59 PM, Cassius Smith wrote:
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the
First of all, thank you for your help.
I was seing Cisco and Linsys web sites and I just came across this 2
devices:
Linksys SPA8000 8 phone ports, 1 port ethernet.
Cisco SPA8800 4 phone ports, 4 lines, 1 port ethernet.
I think they could work for us, because I need maximum 10 normal phones and
I have a sip trunk connecting to a huawei softx3000. At the moment, I can
register and dial in.
However, peer status shows not reachable
sip show peer as follow
* Name : cmphone
Secret : Set
MD5Secret: Not set
Remote Secret: Not set
Context : from-cmphone
Hi all,
I've got a perl agi script that exec()'s the FFA version of receivefax to...
receive a fax.
However, after the fax is received, the script seems to die.
This is what I have:
$main::agi-exec(receivefax,/tmp/${$}.tiff|fs);
$main::agi-verbose(FAX COMPLETE,1);
I never see the FAX COMPLETE
On Fri, 18 Feb 2011, Mike Diehl wrote:
I've got a perl agi script that exec()'s the FFA version of receivefax to...
receive a fax.
However, after the fax is received, the script seems to die.
This is what I have:
$main::agi-exec(receivefax,/tmp/${$}.tiff|fs);
$main::agi-verbose(FAX
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
--
Take care and have fun,
Mike Diehl.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi Satish,
You can Pass 'r' flag to meetme Application and file will be recorded nothin
to load mixmonitor and other Application on Channel, i think 'r' is better
than all options
Cheers
Dhaval
On Sat, Feb 19, 2011 at 1:37 AM, satish patel satish...@hotmail.com wrote:
Thanks,
look like
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