Zitat von A J Stiles asterisk_l...@earthshod.co.uk:
It will be in the /etc/asterisk/*.conf file for the appropriate calling
technology. So if the calls are going over a SIP trunk, it will be in
sip.conf . You want
disallow=all
allow=alaw
There probably will be some other allow= lines; just
hi list
can U help me
caller id in USTM if now working
-- Starting switch on '4211@4211-1' to 4203
-- Executing [4203@office:1] DumpChan(USTM/4211@4211-0x7f7ba4228fd0,
) in new stack
Dumping Info For Channel: USTM/4211@4211-0x7f7ba4228fd0:
On Monday 06 Jul 2015, Luca Bertoncello wrote:
Zitat von A J Stiles asterisk_l...@earthshod.co.uk:
Yes. You should definitely be using A-law for calls to the Outside
World.
Well, I wanted to change these settings, but I'm not sure, where I
have to do that...
I think in the users.conf,
On Monday 06 Jul 2015, Luca Bertoncello wrote:
Well, but for voice quality, which codec is better?
alaw or gsm?
A-law is better for voice quality (sorry, thought my original explanation was
obvious). But note that if the destination is a mobile phone, GSM will be
used anyway, at least for
Zitat von A J Stiles asterisk_l...@earthshod.co.uk:
Yes. You should definitely be using A-law for calls to the Outside World.
If you use a different codec, then your telephone company will
either transcode
it for you (if it is one they understand) or just block the call (if not).
Even
On Sunday 05 Jul 2015, Luca Bertoncello wrote:
Hi list!
I noticed that when the phone of my wife calls the gsm codec will be used,
but if someone calls the phone, alaw will be used:
Could someone explain me why?
Second question: I think, ulaw/alaw are better then gsm, isn't it?
If so, how
Zitat von A J Stiles asterisk_l...@earthshod.co.uk:
On Monday 06 Jul 2015, Luca Bertoncello wrote:
Well, but for voice quality, which codec is better?
alaw or gsm?
A-law is better for voice quality (sorry, thought my original
explanation was
obvious). But note that if the destination is
Zitat von A J Stiles asterisk_l...@earthshod.co.uk:
Yes. You should definitely be using A-law for calls to the Outside World.
Well, I wanted to change these settings, but I'm not sure, where I
have to do that...
I think in the users.conf, but I think, the allow keywords is for
the
Zitat von A J Stiles asterisk_l...@earthshod.co.uk:
Hi,
GSM is the native codec used for calls to mobile phones; it uses lossy
compression to achieve a low bit rate.
A-law is the native codec used by physical exchanges on the land line network
(PSTN and ISDN). It is non-lossy. It works by
On Monday 06 Jul 2015, Luca Bertoncello wrote:
So, I think, I should try to force the using of alaw for this phone,
is it right?
Usually we don't call mobile phones from our landline...
Yes. You should definitely be using A-law for calls to the Outside World.
If you use a different codec,
Hi.
I have a beginner conceptual question about Asterisk:
Let's suppose that there are 4 softphones registered in my Asterisk and all of
them are currently online. In addiction , there is no call.
Suddenly, one of these softphones sends a SIP message to the Asterisk. In this
case the
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.
The easiest solution may be to strip the leading zero's off your caller ID
before your caller enters the Voicemail app to leave you a message.
ExecIf(REGEX(^[0][0].
${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2}))
On Fri, Jul 3, 2015 at 10:53 PM, Luca Bertoncello lucab...@lucabert.de
Hi list!
I'd like to save all information about calls (CDR) in a MySQL-Database.
I created the DB and a user for Asterisk on a separate server, then I
configured my cdr_mysql.conf so:
[global]
hostname=192.168.10.3
dbname=asterisk
table=cdr
password=MYSECRET
user=asterisk
port=3306
and my
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis
Sent: Monday, July 06, 2015 4:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CDR in an
John Kiniston johnkinis...@gmail.com schrieb:
The easiest solution may be to strip the leading zero's off your caller ID
before your caller enters the Voicemail app to leave you a message.
ExecIf(REGEX(^[0][0].
${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2}))
Thanks!
I already
Hello everyone,
A few days ago I had a problem with a couple of extensions. I have about 12
Aastra 6731i phones, 6 are at our main office and 6 more on remote
branches. We use VPN to communicate to our branches so there's no NAT
involved any where.
The problem I had was that I couldn't call from
Hi list!
I'd like to save all information about calls (CDR) in a MySQL-Database.
I created the DB and a user for Asterisk on a separate server, then I
configured my cdr_mysql.conf so:
[global]
hostname=192.168.10.3
dbname=asterisk
table=cdr
password=MYSECRET
user=asterisk
port=3306
and my
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees
Sent: Monday, July 06, 2015 5:54 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF issue
Hello folks,
We have an issue with several Cisco SPA512G phones
The Authenticate application will do this for you.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate
You can either give it a single PIN to use for all calls, Authenticate
using a value in the Asterisk Database, Or use a plain text file for the
PIN's
On Mon, Jul
I've seen this before. It can be done by calling an AGI script when placing the
outgoing call. You'd then prompt and make sure the code matches and do your
billing logic, etc there. Then place the call if it's valid.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hello folks,
We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance.
I've tried changing the DTMF
Hello All,
I will like to configure Asterisk to use PIN Code for all outgoing
international calls.
Also, any suggestions as to when should I prompt users for code prior to
dialing the number or after dialing the number?
can someone provide with a example on how to accomplish this goal? I
Luca Bertoncello wrote:
Hi list!
I'd like to save all information about calls (CDR) in a MySQL-Database.
I created the DB and a user for Asterisk on a separate server, then I
configured my cdr_mysql.conf so:
[global]
hostname=192.168.10.3
dbname=asterisk
table=cdr
password=MYSECRET
On 7/6/15 5:53 PM, Jamie Rees wrote:
Hello folks,
We have an issue with several Cisco SPA512G phones connected to an
Asterisk platform where several users hear loud, random beeps during
calls to external recipients. The noises are akin to button press
tones, are very loud and a significant
I am currently, I create the VOIP server which enable the user to make the
call over the asterisk server, Additionally now I want the user to be able
to chat to each other too.
I found some suggestion of using the openfire with asterisk but not much
said on it, Anyway could you please share me
John Novack jnov...@stromberg-carlson.org schrieb:
Been doing this with MySql for the last 10 years, though not on an openWrt
machine MySql is on the Asterisk machine.
Also have additional database tables to block by callerId and name
Did have some issues with the dialplan syntax when moving
Good afteroon all,
First of all: thanks for everybody who is willing to think this through with me:
I'm having some issues regarding call quality between some calls. Let me try to
explain the situation first
We have a Asterisk 11.16 server based on the Xivo distribution. There are 2
servers
Hello,
I would like to setup a mechanism to trigger an alarm if user is deal
too many numbers within a very short period of time. Safeguard against
users hacked accounts.
can someone help?
Thanks,
--
_
-- Bandwidth and
Hi All
I have a problem with mixmonitor in 13.4.0 doing the following:
1. Caller phones in
2. Reception picks up
3. Talks to caller
4. Does attended transfer, talks to manager to screen the caller wanting to
speak to him
5. Complete the transfer by putting down her handset so the caller can
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