On Fri, Jun 16, 2017 at 1:43 PM, Jonathan H wrote:
> OK, thanks. That sort of makes sense. Is it case sensitive?
>
Is what case sensitive? Function names are case sensitive. Application
names have historically been not case sensitive.
>
> Bonus quickie while I'm here
OK, thanks. That sort of makes sense. Is it case sensitive?
Bonus quickie while I'm here (not worth own thread) - Asterisklint
complains that:
H_PAT_NON_CANONICAL: pattern '_#' is not in the canonical form '#'
for the line
exten => _#,1,Goto(s,1)
I'm sure I read somewhere it should be _#.
Am
> It was only when I ran AsteriskLint over my dialplan that I noticed this:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET
>
> Hmmm, they both seem to do the same thing. Or don't they?
In some
On Fri, Jun 16, 2017 at 12:41 PM, Jonathan H wrote:
> It was only when I ran AsteriskLint over my dialplan that I noticed this:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET
It looks like your database configuration is missing in Asterisk.
It is making up information about the connection using defaault values
as if it did not find any database configuration.
Ron
On 03/06/2013 10:49 AM, Olivier CALVANO wrote:
Hi
i have installed a new Asterisk server on
The database schema (table) is different in Asterisk 11.4 ?
because i have configured:
cdr_mysql.conf
extconfig.conf
res_config_mysql.conf
and on the mysql server, it's the old database of 1.6.x
i see:
[Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to
connect
Strange too, in the logs:
[Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No
database user found, using 'asterisk' as default.
[Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No
database password found, using 'asterisk' as default.
[Jun 3 17:09:49]
Fix this.
[Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No
database user found, using 'asterisk' as default.
[Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No
database password found, using 'asterisk' as default.
[Jun 3 17:09:49] WARNING[3464]
No other idea ?
2013/6/3 Olivier CALVANO o.calv...@gmail.com
Hi
i have installed a new Asterisk server on Fedora. My first server use
Asterisk 1.6.x with a MySQL CDR and
realtime.
I have a small problems, when i configure on the new server, the same
information in MySQL, we have a
Thanks for your help Ron,
Do you know where is the confirguration ?
Because i have put into res_config_mysql.conf:
[general]
dbhost = myhost.mydomain.net
dbname = MyDB
dbuser = MyUser
dbpass = MyPassword
dbport = 3306
dbsock = /tmp/mysql.sock
dbcharset = latin1
requirements = warn
after in
Hello,
are you sure MySQL socket is in /tmp directory?
dbsock = /tmp/mysql.sock
Regards
El 03/06/2013 12:16, Olivier CALVANO escribió:
Thanks for your help Ron,
Do you know where is the confirguration ?
Because i have put into res_config_mysql.conf:
[general]
dbhost = myhost.mydomain.net
on this server we don't have mysql.socket because he don't have mysql server
we want access to a mysql based on a other server
2013/6/3 Bakko asannu...@gmail.com
Hello,
are you sure MySQL socket is in /tmp directory?
dbsock = /tmp/mysql.sock
Regards
El 03/06/2013 12:16, Olivier
asterisk trying connect to mysql via socket remove that line from config
files.
1 check if port 3306 is open in iptables on both servers.
2 check permissions on db for user Asterisk.
On Mon, Jun 3, 2013 at 9:18 PM, Olivier CALVANO o.calv...@gmail.com wrote:
on this server we don't have
Do you have this problem in your conf file?
http://forums.digium.com/viewtopic.php?p=63736
The parser won't accept an ; (semicolon) for remarks! So he found at
the first the old remarks and tried to access my database with the false
data.
Ron
On 03/06/2013 3:18 PM, Olivier CALVANO
oh ron thanks for your help :
We have deleted all commented line, only put the configuration and now
that's work !
2013/6/3 Ron Wheeler rwhee...@artifact-software.com
Do you have this problem in your conf file?
http://forums.digium.com/viewtopic.php?p=63736
The parser won't accept an ;
grrr no in asterisk -d i have no error, but when i start normaly asterisk i
have :
[Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to
connect database server xxx on xxx.xxx.net (err 2003). Check debug for more
info.
[Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL
Well, at least you are making progress.
What is the error in the debug log?
Ron
On 03/06/2013 8:03 PM, Olivier CALVANO wrote:
grrr no in asterisk -d i have no error, but when i start normaly
asterisk i have :
[Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime:
Failed to
What is the best commande for put the debug ? because with core set debug,
i don't have a return.
voip-2*CLI realtime mysql status
Vop configured for m...@myhost.mydomain.net, port 3306 with username
Asterisk.
[Jun 4 06:48:25] ERROR[27879]: res_config_mysql.c:1577 mysql_reconnect:
MySQL
Alternatively, if you don't have that extension defined anywhere,
Asterisk will jump to the i extension, where you can then read the
actual entered digits from the INVALID_EXTEN variable and jump back to
the main part of the dialplan.
Note that if they enter digits that *could* match a defined
It sounds like you may want to use the READ command instead. This lets you
hard-set the number of digits to expect and then sets a variable which you
can use later in the dialplan. Generally you use the background command to
let them dial an extension or automated attendant option. Playback
First question - playback is not interruptable by DTMF, background is.
You have two options here
Option 1
Use Read
[getnum]
Exten = start,1,read(mydigit,prompt,4,skip,1,2)
.. verification stuff
Option 2
Use WaitExten with Background
[getnum]
Exten = start,1,background(prompt)
Exten =
On Mon, 21 Nov 2011, Danny Nicholas wrote:
Option 2
Use WaitExten with Background
[getnum]
Exten = start,1,background(prompt)
Exten = start,n,waitexten(2)
Exten = ,1,noop(user pressed )
Exten = I,1,playback(invalid)
For option 2 you have to define each valid 4 digit entry in the
Hi James,
Thanks I give me the clear view and now I am able to describe that. again
thank a lot
2011/5/23 James zhu zhulizh...@live.com
hello:
please refer this link:
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi,
Hi James,
My question was related to CLI commands not sip.conf type
value(friends,peer,user)
main question was difference below commands
*sip show peers* Vs *sip show user*
On Mon, May 23, 2011 at 1:28 PM, virendra bhati virbh...@gmail.com wrote:
Hi James,
Thanks I give me the clear view
hello:
please refer this link:
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Sat, 21 May 2011 17:49:37 +0530
From: virbh...@gmail.com
To:
Discussion
*Subject:* Re: [asterisk-users] Difference
Is there any major architectural difference between 1.4 and 1.8?
The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from
, to |) and the AGI structure is enhanced. If you don’t use AGI’s, a
qualified “not really
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Wednesday, October 06, 2010 7:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Difference
Hi All,
Please can
Is there any major architectural difference between 1.4 and 1.8?
On Wed, Oct 6, 2010 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan
: Wednesday, October 06, 2010 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Difference
Is there any major architectural difference between 1.4 and 1.8?
The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from ,
to |) and the AGI
On Wed, 6 Oct 2010, Rizwan Hisham wrote:
Is there any major architectural difference between 1.4 and 1.8?
Nope. The developer's just got tired of typing .4
Of course, the joke's on them -- 1.8 is only .4 better than 1.4.
--
Thanks in advance,
To: Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Difference
Is there any major architectural difference between 1.4 and 1.8?
The dialplan uses the 1.6 nomenclature (delimiter in dialplan changes from ,
to |) and the AGI structure is enhanced. If you don’t use
Back in the days i heard that they have changed the architecture in 1.6 and
its a lot better than 1.4 (6 times better call handling and robust
architecture, someone told me). If they have decided to take the 1.6
architecture to the next level in the new 1.8 version then its a good thing.
On Wed,
Here is a presentation from Kevin P. Fleming, Director of Software
Technologies at Digium. Information might be old by now still gives a good
overview of what is new in 1.6:
http://www.asterisk-tag.org/2008/slides/Kevin-Fleming-Asterisk-Tag-2008.pdf
Summary of his presentation is as follows:
–
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Rizwan Hisham
*Sent:* Wednesday, October 06, 2010 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Difference
Is there any major
On Sat, 7 Nov 2009, jonas kellens wrote:
vps*CLI iax2 show channels
Channel Peer UsernameID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
0 active IAX channels
vps*CLI core show channels
Channel Location State
On Mon, Aug 3, 2009 at 4:29 PM, Michael
Cunninghammichael.cunningham@gmail.com wrote:
Forgive me if this is a FAQ question but I didnt see anything on the website
of forum spelling out the difference between 1.4.x and 1.6.x
Obviously 1.6.x is in development. Is it stable enough for
For changes between 1.4 and 1.6 you might find useful this one:
http://svn.digium.com/svn/asterisk/tags/1.6.0/CHANGES
For changes between 1.6 branches:
http://svn.digium.com/svn/asterisk/tags/1.6.1.0/CHANGES
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta1/CHANGES
Regards
Jose
2009/8/4
1.6.0 is stable
1.6.1 is stable
1.6.2 is release candidate
See the files Changelog* and UPDATE* in this distributions for changes.
regards
klaus
Michael Cunningham schrieb:
Forgive me if this is a FAQ question but I didnt see anything on the
website
of forum spelling out the difference
Klaus Darilion wrote:
1.6.0 is stable
1.6.1 is stable
1.6.2 is release candidate
See the files Changelog* and UPDATE* in this distributions for changes.
Actually, almost...
1.6.0 is in release status
1.6.1 is in release status
1.6.2 is in beta status (soon release candidate)
That's a more
, August 04, 2009 8:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Difference between 1.4.x and 1.6.x?
Klaus Darilion wrote:
1.6.0 is stable
1.6.1 is stable
1.6.2 is release candidate
See the files Changelog* and UPDATE* in this distributions
On Tuesday 04 August 2009 08:22:52 Danny Nicholas wrote:
Ok. What's the difference between stable and in release status? Does
the hierarchy go beta, stable, release candidate, release status?
We don't actually use the word stable because it's a loaded word and leads
to poor expectations. In
On Tue, Aug 04, 2009 at 08:45:13AM -0500, Tilghman Lesher wrote:
On Tuesday 04 August 2009 08:22:52 Danny Nicholas wrote:
Ok. What's the difference between stable and in release status? Does
the hierarchy go beta, stable, release candidate, release status?
We don't actually use the word
Michael Cunningham wrote:
Forgive me if this is a FAQ question but I didnt see anything on the
website
of forum spelling out the difference between 1.4.x and 1.6.x
Obviously 1.6.x is in development. Is it stable enough for production use?
What are the new features being implemented in
Thanks Leif,
That cleared up the versioning.. Is there a list of new features in 1.6.x
versus the 1.4.x version?
On Mon, Aug 3, 2009 at 4:34 PM, Leif Madsen leif.mad...@asteriskdocs.orgwrote:
Michael Cunningham wrote:
Forgive me if this is a FAQ question but I didnt see anything on the
Olivier wrote:
Hello,
Groups in asterisk are summarized here (
http://www.voip-info.org/wiki/view/Channels+and+Groups).
Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
(as I've been advised in another thread, to switch from one notation to the
other and I can't
Olivier wrote:
Hello,
Groups in asterisk are summarized here
(http://www.voip-info.org/wiki/view/Channels+and+Groups).
Is there any difference between DAHDI/G1/0123456789 and
DAHDI/g1/0123456789 (as I've been advised in another thread, to switch
from one notation to the other and I
Thanks for replying !
I've added a link here
http://www.voip-info.org/wiki/view/Channels+and+Groups to
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels.
2008/12/1 Mark Michelson [EMAIL PROTECTED]
Olivier wrote:
Hello,
Groups in asterisk are summarized here
Please read:
http://www.voip-info.org/wiki/view/FreeSwitch
and
http://www.voip-info.org/wiki/index.php?page=Asterisk
Then if you have a specific question about one of them,
come back here to ask about asterisk, and on the freeswitch
mailinglist for more info on that technology.
Or you could
On Tuesday 22 January 2008 11:50:25 love U.all wrote:
what is the difference between FreeSwitch and Asterisk ,
The main difference in functionality is that FreeSwitch is a voip-switch only.
It does not have any method to interface to the PSTN, other than through
using another host which does
what is the difference between FreeSwitch and Asterisk ,
The main difference in functionality is that FreeSwitch is a
voip-switch
only.
Technically, FreeSWITCH is a soft-switch, or a modular media switching
library that can switch more than just voice. Also, technically, FS is
a library,
The TE121 is a PCI Express card (TE122 is standard PCI, 2.2 if I'm not
mistaken).
Gal Barak
Tech support
Atelis PLC
2008/1/16, Guilherme Loch Waltrick Góes [EMAIL PROTECTED]:
What's the difference between the TE121 and TE122. I read the description
on Digium's site and it isn't clear to me.
Hello!
Guilherme Loch Waltrick Góes wrote:
What's the difference between the TE121 and TE122. I read the description on
Digium's site and it isn't clear to me.
Best regards,
The only one difference is interface: one of them have PCI and other
have PCI-Express.
--
Best regards,
Igor A.
The model AP200 that you are giving as example is 2 port only... and
i'm not sure about the price...
I know that codec conversion is one of the most cpu-intensive task
that asterisk has to do, so, you can chose a Digium/Sangoma card with
a powerful server doing the work or you can also use a VoIP
Yehavi,
The release branches (1.2, 1.4) were at one time trunk. When it was decided
to release 1.4, for example, it was branched off from trunk as the
1.4branch. New functionality continued to be added to trunk after
that. Once
the release branches are created, they are feature-frozen and only
Hello Sean,
Does this clear things up?
Yes! Thanks!
__Yehavi:
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'show channels' shows only running calls while 'sip show channels' shows
all running sip sessions including phones trying to register .
On 09/09/2007, ram [EMAIL PROTECTED] wrote:
Hi all
what is the difference between
show channels
sip show channles
i see the difference in both
show
On 9/9/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
'show channels' shows only running calls while 'sip show channels' shows
all running sip sessions including phones trying to register .
thanks
but after my 30 channels of show channels
i see lot of vice break and choppy voice
doing
Michiel van Baak wrote:
On 05:27, Fri 03 Aug 07, bilal ghayyad wrote:
Hi List;
What is the difference between WaitExten function and
TIMEOUT (response)? As I see that both are used to
determine the allowed time to enter the digits, any
one can advise?
WaitExten is waiting for
The difference is in the scope of the command.
Think of it this way:
WaitExten gives the user more time to enter digits before the dialplan
moves on to the next instruction in the dial plan. Timeout is the max
number of seconds to wait at any point in the current context before
deciding the user
Dear Michael;
I understood it in that way (please advise me if I am
correct):
WaitExten is for the time to complete entering the
digits, while timeout is specified wether user
responded by dialing any thing or not.
Please advise.
regards
The difference is in the scope of the command.
Think of
On 05:27, Fri 03 Aug 07, bilal ghayyad wrote:
Hi List;
What is the difference between WaitExten function and
TIMEOUT (response)? As I see that both are used to
determine the allowed time to enter the digits, any
one can advise?
WaitExten is waiting for you to type an extension.
TIMEOUT is
On 8/3/07, bilal ghayyad wrote:
Hi List;
What is the difference between WaitExten function and
TIMEOUT (response)? As I see that both are used to
determine the allowed time to enter the digits, any
one can advise?
core show function TIMEOUT
for different timeout parameters, I haven't
Call setup/teardown is handled with the SIP protocol while the actual call
audio is handled with RTP I think. Check the config of your NAT devices
relative to RTP.
scd
On 4/16/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I'm wondering about the difference between Cisco Call Manager and
Rodney G. McDuff wrote:
Is the TE411P just a TE410P with hardware echo cancellation?
Yes. Same for a TE405P and TE406P.
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On 3/10/06, AR Tarzi [EMAIL PROTECTED] wrote:
That's because the duration is counted from the time of dialling. billsec is
what you want if it's to calculate the duration the call was active.
To change what shows you need to change call-log.php in
/var/www/html/admin/cdr/
Instead of
What about ResetCDR() just before Dial()?
nik600 wrote:
hi
i've made some test calls, i've notice that a call of the duration of
1:29 minutes is recorded in the cdr database as 1:45 minutes, is it
normal?
i think that 15 seconds are too many... how can i correct this?
thanks
On Fri, 10 Mar 2006 15:04:18 +0100
nik600 [EMAIL PROTECTED] wrote:
hi
i've made some test calls, i've notice that a call of the duration of
1:29 minutes is recorded in the cdr database as 1:45 minutes, is it
normal?
i think that 15 seconds are too many... how can i correct this?
Or maybe, you should try the C flag in your Dial().
[]'s
MM
nik600 wrote:
hi
i've made some test calls, i've notice that a call of the duration of
1:29 minutes is recorded in the cdr database as 1:45 minutes, is it
normal?
i think that 15 seconds are too many... how can i correct this?
ok, thanks for your reply, tomorrow i'll test and let you know
bye
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That's because the duration is counted from the time of dialling. billsec is
what you want if it's to calculate the duration the call was active.
To change what shows you need to change call-log.php in
/var/www/html/admin/cdr/
Instead of duration extract billsec - you can still label it
On Mon, Jul 18, 2005 at 12:03:42AM -0500, Kristian Kielhofner wrote:
trixter http://www.0xdecafbad.com wrote:
On Mon, 2005-07-18 at 07:04 +0300, Tzafrir Cohen wrote:
OT:
Not a Knoppix, actually. You can't do anything useful with it without a
HD install. A while ago I needed badly to test
Tzafrir Cohen wrote:
I should be releasing a much improved Live version of AstLinux
within a week or so. A test version was announced on my mailing list a
while ago, with pretty good results so far. It will be AstLinux 0.2.8, and
available as an ISO (as well as the Windows install package,
On Mon, Jul 18, 2005 at 02:40:40AM -0500, Kristian Kielhofner wrote:
I tried astlinux 0.2.6 as well, however as a live CD it wasn't useful,
because I could not write any modified configuration on the live /etc .
Thus I could not run asterisk with the modified configuration without a
proper
My 2c worth...
For the beginner, AAH is great. The PC that you install on will be
totally reformatted / fdisk-ed (assuming single drive - etc).
With AAH 1.3 - the installation goes to sleep and sort of finishes
when its Syncing with a Time Server. A reboot at this point seems to do
no harm.
As
my $0.02...
I have implemented [EMAIL PROTECTED] 0.9
ina Real Estate/Legal office with 6 phones and 3 incoming lines using
TDM04B card and Polycom/Sipura phones.
I am about to implement 3 more systems ([EMAIL PROTECTED] 1.3) for a construction company.One will
be 6 phones and 4 PSTN lines
That question could start a battle.
[EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the
Knoppix of the Linux distro world. It could work but that is for you
to decide.
On 7/17/05, Michael Felder [EMAIL PROTECTED] wrote:
Hello
What is the difference between these 2 version
I guess I was wondering if it was crippled in some way.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Latham
Sent: Monday, 18 July 2005 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Latham
Sent: Monday, 18 July 2005 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Difference between Asterisk and
[EMAIL PROTECTED
-
[EMAIL PROTECTED] On Behalf Of Michael Felder
Sent: Sunday, 17 July 2005 11:33 PM
To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Difference between Asterisk and
[EMAIL PROTECTED]
I guess I was wondering if it was crippled in some way
I think that AAH would suit you fine for a small business application.
It is not crippled in any way. Far from it in fact. It has more features
installed as default than the latest CVS version.
It is however based on the latest official release of the Asterisk
code (whatever that is at the
Michael Felder wrote:
Hello
What is the difference between these 2 version of Asterisk in terms of
functionality.
For a small office am I going to run into problems if I use the easy
version...
Mike
[EMAIL PROTECTED] is a Linux distribution that makes it easy to install
Asterisk and a few
Mark Phillips wrote:
I think that AAH would suit you fine for a small business application.
It is not crippled in any way. Far from it in fact. It has more features
installed as default than the latest CVS version.
Not to sound rude, but that doesn't make any sense. Or I'm reading it
wrong
On Sun, Jul 17, 2005 at 10:18:26PM -0500, Andrew Latham wrote:
That question could start a battle.
[EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the
Knoppix of the Linux distro world. It could work but that is for you
to decide.
OT:
Not a Knoppix, actually. You can't do
On Mon, 2005-07-18 at 07:04 +0300, Tzafrir Cohen wrote:
On Sun, Jul 17, 2005 at 10:18:26PM -0500, Andrew Latham wrote:
That question could start a battle.
[EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the
Knoppix of the Linux distro world. It could work but that is for
trixter http://www.0xdecafbad.com wrote:
On Mon, 2005-07-18 at 07:04 +0300, Tzafrir Cohen wrote:
On Sun, Jul 17, 2005 at 10:18:26PM -0500, Andrew Latham wrote:
That question could start a battle.
[EMAIL PROTECTED] is a bootable Asterisk system. Think of it as the
Knoppix of the Linux distro
Kib Eki wrote:
Hi,
can one summarize the main differences between Asterisk and
[EMAIL PROTECTED] or
point me to a location where i can find such a list?
See [EMAIL PROTECTED] site - http://asteriskathome.sf.net - and asterisk
site - www.asterisk.org.
Basically, asterisk is a program, and [EMAIL
Matt wrote:
I have a STUN server running on my Asterisk box which seems to work
for most of my SIP clients.. but some of them seem to require NAT=yes
turned on. If I go further and turn QUALIFY=yes to on, is there a
reason I need to keep running a STUN server? If so, what's the
difference?
I
Am Samstag 05 März 2005 07:58 schrieb Remco Barende:
Hi list!
While looking for the Snom 190 I found another phone, the Elmeg IP 290
(www.elmeg.de).
Looking at the pictures the specs they seem to be very similar beasts
but the firmware is supposedly not interchangeable.
Does anyone know
On 5 Mar 2005, at 06:58, Remco Barende wrote:
Hi list!
While looking for the Snom 190 I found another phone, the Elmeg IP 290
(www.elmeg.de).
Looking at the pictures the specs they seem to be very similar
beasts but the firmware is supposedly not interchangeable.
Does anyone know the
Hello Remco,
On Sat, 5 Mar 2005, Remco Barende wrote:
Hi list!
While looking for the Snom 190 I found another phone, the Elmeg IP 290
(www.elmeg.de).
Looking at the pictures the specs they seem to be very similar beasts
but the firmware is supposedly not interchangeable.
Does anyone
: [Asterisk-Users] Difference between Snom 190 Elmeg 290?
Hello Remco,
On Sat, 5 Mar 2005, Remco Barende wrote:
Hi list!
While looking for the Snom 190 I found another phone, the
Elmeg IP 290
(www.elmeg.de).
Looking at the pictures the specs they seem to be very similar
PRI comes in 2versions E1 European and T1 US
E1 30 channels T1 23 channels
On Wed, 2005-02-23 at 14:15, Eric Bishop wrote:
Hi all,
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the
Eric Bishop wrote:
Hi all,
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the same thing or is there some technical
difference. Even Newton's telco dictonary seemed a bit fuzzy on this
Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Difference between E1 and PRI
Eric Bishop wrote:
Hi all,
I have seen the term E1 and PRI used interchangably when
referring to
a voice service with 30B channels and 1 D channel. Are they just
different
On Wed, 23 Feb 2005, Eric Bishop wrote:
I have seen the term E1 and PRI used interchangably when referring to
a voice service with 30B channels and 1 D channel. Are they just
different terms for the same thing or is there some technical
difference. Even Newton's telco dictonary seemed a bit
Eric,
E1 is a physical layer protocol, like ethernet. It defines a 2Mbps pipe,
which can be used for data, or can be split into 32 64Kbps telephone
channels, or a mixture. If used for telephone channels, 30 of these
channels can carry one telephone conversation each, and 2 carry
signalling and
Alistair-
Good writeup! Question regarding Q.SIG: Can it be used to solve the
problem of signaling a remote switch to take a call back and extend it
to another channel instead? This, as you know, is always a challenge
when using IVR in a call centre environment, when one wants to extend an
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Stingel
Sent: Wednesday, February 23, 2005 9:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Difference between E1 and PRI
Alistair-
Good writeup! Question regarding Q.SIG
Scott,
Yes, and this is one of the principal reasons people choose Q.SIG.
I've worked on quite a few large voicemail servers, and these tend to do
a lot of transfers for follow-me and operator features. Q.SIG support
can significantly reduce the number of telephony channels needed, as not
only
On Wed, 23 Feb 2005, Scott Stingel wrote:
Good writeup! Question regarding Q.SIG: Can it be used to solve the
problem of signaling a remote switch to take a call back and extend it
to another channel instead? This, as you know, is always a challenge
when using IVR in a call centre
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