...You can declare a variable whose values gets set/used anywhere in the
dialplan.
Regards,
Roland.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Yuan LIU
Gesendet: 25 February 2007 08:41
An: asterisk-users@lists.digium.com
Betreff: RE: AW:
Thanks a lot Joanna ...
Yes _ I made it and it works fine ...
Mohamed Farid ,,
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joanna
Liza Mariazeta
Sent: Wednesday, February 21, 2007 6:19 AM
To: Asterisk Users Mailing List - Non-Commercial
Had the same problem with them. I now use didx.net, and would not say they're
the best but atleast they have a good ticketing/help desk system and someone
does respond. They also have a large selection of numbers.
-- Original message --
From: Brad Templeton [EMAIL
When I listening to messages, VoiceMailMain always goes from the oldest
message to the newest message.
For new messages, this order is ok. But for old/archived messages, I
would like to hear the reverse order. What can I do?
___
--Bandwidth and
Dinesh Nair wrote:
is there a reason why wanpipe stopped working with asterisk ?
My experience from yesterday shows that zaptel.c has been renamed to
zaptel-base.c. This prevents the Sangoma Setup script from patching
zaptel. The fix (Found by Googling) was to rename every instance of
On Saturday 24 February 2007 18:53, shadowym wrote:
Hi there,
Here is my dilema. I have a new small business customer that wants me to
put in a VoIP phone system for them. Based on their requirements, I have
determined that it needs to be a set it and forget it type of thing like
a lot of
What is the mechanism used to SEND TEXT over a Zap channel? Is it FSK?
Roland Ndaka Fru wrote:
Here is how you can send/receive text in the DialPlan using an AGI script:
print STDERR 1. Testing 'sendtext'...;
print SEND TEXT \hello world\\n;
my $result = STDIN;
checkresult($result);
print
Roland Ndaka Fru wrote:
...You can declare a variable whose values gets set/used anywhere in the
dialplan.
Like so:
print SET VARIABLE ret $result\n;
Dialplan:
...,NoOp(the result is: ${ret})
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Agreed. Monitor yes.But why let the system run.. only to find out it is
going to go down after being up for 100 days?Yes, it should be able to
run continually with no issues, but unfortunately asterisk seems to have
memory leaks. 1.2.6 is the only one we've found that will run and run
This page http://www.freecall.com/en/index.html is advertising free
calls to:
Argentina, Australia, Austria, Belgium, Canada, Czech Republic, Denmark,
France, Germany, Hong Kong (+mobile), Hungary, Ireland, Italy,
Luxembourg, Malaysia, Monaco, Netherlands, New Zealand, Norway, Panama,
Poland,
I'm looking for a device that will announce sounds from other devices
to Asterisk to be heard on my Grandstream sip phones throughout the house.
The Grandstream phones are already set up for ammouncements by pressing
a dial code.
I need a device that will detect and play sounds automatically
This worked:
Dial(SIP/[EMAIL PROTECTED],,D(12345678))
however, the problem now exists in the disconnection. Asterisk tries to
bridge the call, play dtmf but never disconnects. What is there a specific
syntax to the D command that specifies a disconnect period.
I am thinking a better solution
I suspect they are the same company to www.voipdiscount.com (same rates,same
windows software same design in yellow). (and even maybe voipbuster.com)
Asterisk setup is easy and works quite well. No quality problem, occasionaly
a fire in the server room ( :) ), or 301s, but they are quite reliable.
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even with
older version 1.2.3
On 2/24/07, Al Bochter [EMAIL PROTECTED] wrote:
Is there anyone sending SMS with Asterisk?
--
Best regards,
Al Bochter
Bochter Services
At 09:10 AM 2/25/2007, you wrote:
I don't have any qualified Windows box to get an account and try it.
Can anybody comment on setup and or call quality?
I've been using it for 6 or 8 months for my calls to New Zeland and
Australia. It's been perfectly acceptable but the people I call know
Dinesh Nair wrote:
On 02/25/07 06:26 Darrick Hartman said the following:
Kristian is working with Sangoma to get wanpipe supported once again
in Asterisk.
is there a reason why wanpipe stopped working with asterisk ?
I meant with AstLinux. Sorry for any confusion.
--
Darrick Hartman
have you tried looking at the CLI to double check on the call flow? do make
sure that you 'set verbose 10' or something like that.
On 2/24/07, Jonathan Solano [EMAIL PROTECTED] wrote:
Hi all, I'm having a problem, with the h extension.
I have an application, when I call it check for the line
On 13:29, Sun 25 Feb 07, Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even with
older version 1.2.3
I can recommend them as well.
Works great and delivery times are good.
With some scripting knowledge you
You can even add php scripting to send messages via browser
On 2/25/07, Michiel van Baak [EMAIL PROTECTED] wrote:
On 13:29, Sun 25 Feb 07, Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even
with
older
On 14:19, Sun 25 Feb 07, Supa wrote:
You can even add php scripting to send messages via browser
That's what I do in a couple of apps indeed ;)
And the perl we use in nagios to send sms.
Possibilities are endless ...
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key:
Can anyone else recommend any other providers not listed already
On 2/25/07, Michiel van Baak [EMAIL PROTECTED] wrote:
On 14:19, Sun 25 Feb 07, Supa wrote:
You can even add php scripting to send messages via browser
That's what I do in a couple of apps indeed ;)
And the perl we use in nagios
Hi,
Asterisk Version : 1.2.15
Card : TDM11B (1 x FXO , 1 x FXS)
I have internal dialling working okay SIP-ZAP (analogue phone) and ZAP
(analogue phone) - SIP.
The problem comes when I try and make a outbound call.
Here is my extensions.conf :-
Code:
[incoming]
exten =
I followed Marks SNMP howto on Voip Magazine and ran into a small
problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/)
When asterisk is running as a non-root user (asterisk) SNMP request
for for the Asterisk MIB tree return nothing. If I quit asterisk and
run it as root, all is
this dials, and upon answers plays dtmf tones, but does not auto disconnect:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,D(12345678),S(8))
and this disconnects after 8 secs, but does not play dtmf:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,S(8),D(12345678))
any ideas of what wrong
Supa wrote:
this dials, and upon answers plays dtmf tones, but does not auto
disconnect:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,D(12345678),S(8))
and this disconnects after 8 secs, but does not play dtmf:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,S(8),D(12345678))
any ideas of
Thanks that worked, but it still tries to bridge call after dtmf, then fails
instead of moving on to next number to dial and page
On 2/25/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Supa wrote:
this dials, and upon answers plays dtmf tones, but does not auto
disconnect:
exten =
OK. problem solved. It was something dumb on my part. /var/agentx
didn't have enough permissions to let asterisk access the socket.
On 2/25/07, Forrest Beck [EMAIL PROTECTED] wrote:
I followed Marks SNMP howto on Voip Magazine and ran into a small
problem...
Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even
with older version 1.2.3
Anyone out there running it against 1.4.0?
It built just fine for me, but then it crashes the server when I try to
run it.
On 17:53, Sun 25 Feb 07, Brian Capouch wrote:
Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even
with older version 1.2.3
Anyone out there running it against 1.4.0?
It built just fine for me, but
Hi List,
I wonder if there are issues about timing and synchro of differents
telco provider using te412p ?
And if I can use 3 ports as slave and 1 as master ?
Tristan
Julian Lyndon-Smith a écrit :
Is there a trade-in program in place ? I have a te410p and a te405p
that I am not using
Have you tried SMSSEND? It's open source, available as RPM.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Sunday, February 25, 2007 6:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sending SMSa
On
How can i see if snmp is running ok on mi * box ?
Thanks in advance
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Forrest Beck
Enviado el: Domingo, 25 de Febrero de 2007 06:14 p.m.
Para: Asterisk Users List
Asunto: [asterisk-users] Re: Marks SNMP HowTo
El dom, 25-02-2007 a las 20:12 +, --[ UxBoD ]-- escribió:
[internal]
include = outbound-local
include = uri
...
you're dialing 912345678, which has 9 digits
[outbound-local]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion()
exten =
I have looked around with no luck.
Does anyone know of a way to send an email from the dialplan.
The system that I am working on has none thing to do with VoiceMail.
This is something like the SMS command but using sending email
I am working on a prepaid alarm dispatch program for Asterisk if
Figured out myself, just sharing to help others
I have fixed the tables problem in the postgresql database
the parameter tcpip_socket is no longer used in version 8.0 soforget about
that. it has been replaced by listen_address.
and we only want postgres to listen on the localhost so the setting
Al Bochter wrote:
I have looked around with no luck.
Does anyone know of a way to send an email from the dialplan.
The system that I am working on has none thing to do with VoiceMail.
This is something like the SMS command but using sending email
I am working on a prepaid alarm dispatch
Exten = alarm,1,System(/usr/local/bin/sendalarm.sh|[EMAIL PROTECTED])
Or
Exten = alarm,1,AGI(sendalarm)
/usr/local/bin/sendalarm
#!/bin/sh
Mail -s Alarm condition on PBX $1 /dev/null
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf
Al Bochter wrote:
Does anyone know of a way to send an email from the dialplan.
Use System()/TrySystem() or AGI() to execute a script
in whatever language you like.
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Take a look at the last line of your uri context. It looks like that is
matching before the outbound-local ones are. Try changing the extension
from _X. to _[0-8]X.
Looks like you're running into this issue:
Supa wrote:
Thanks that worked, but it still tries to bridge call after dtmf, then
fails instead of moving on to next number to dial and page
So tack on a g to the end of your dial strong, to continue along the
dial plan upon disconnect.
___
Sergio,
The resource has to be compiled at install. If net-snmp is installed
along with a couple other packages, then it will be installed.
To see if it is there now type module show like snmp in the cli. (1.4.0)
Here is a how to. http://www.voip-magazine.com/content/view/2877/
On
- Original Message -
From: Al Bochter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, February 26, 2007 4:20 AM
Subject: [asterisk-users] Sending Email From the dialplan
I have looked around with no luck.
Does anyone know of a way to send an email from the
- Original Message -
From: Al Bochter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, February 26, 2007 4:20 AM
Subject: [asterisk-users] Sending Email From the dialplan
I have looked around with no luck.
Does anyone know of a way to send an email from the
Michiel van Baak wrote:
On 17:53, Sun 25 Feb 07, Brian Capouch wrote:
Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even
with older version 1.2.3
Anyone out there running it against 1.4.0?
It built
From: Philipp Kempgen [EMAIL PROTECTED]
Date: Sun, 25 Feb 2007 17:34:19 +0100
Roland Ndaka Fru wrote:
...You can declare a variable whose values gets set/used anywhere in the
dialplan.
Like so:
print SET VARIABLE ret $result\n;
This is one technique missed in a previous thread about
From: Michael Welter [EMAIL PROTECTED]
Date: Sun, 25 Feb 2007 08:53:09 -0700
What is the mechanism used to SEND TEXT over a Zap channel? Is it FSK?
show AGI send text indicates that it does so in channels that supports
text. show agi receive text indicates that most channels do not support
From: Supa [EMAIL PROTECTED]
Date: Sun, 25 Feb 2007 15:45:08 -0500
this dials, and upon answers plays dtmf tones, but does not auto
disconnect:
exten = s,2,Dial(SIP/TelaSip-gw4/5198881212,15,D(12345678),S(8))
and this disconnects after 8 secs, but does not play dtmf:
exten =
Yuan LIU wrote:
From: Michael Welter [EMAIL PROTECTED]
Date: Sun, 25 Feb 2007 08:53:09 -0700
What is the mechanism used to SEND TEXT over a Zap channel? Is it FSK?
show AGI send text indicates that it does so in channels that supports
text. show agi receive text indicates that most
Do you think it could have been done with another T1/E1Asterisk box
between
the Nortel PBX and the other Asterisk server ?
Sorry, I do not understand exactly what you are asking. Do you mean using
an
Asterisk with PRI card instead of Cisco? If so, I have no experience with
this.
That was
From: Philipp Kempgen [EMAIL PROTECTED]
Date: Mon, 26 Feb 2007 06:59:46 +0100
Yuan LIU wrote:
From: Michael Welter [EMAIL PROTECTED]
Date: Sun, 25 Feb 2007 08:53:09 -0700
What is the mechanism used to SEND TEXT over a Zap channel? Is it FSK?
show AGI send text indicates that it does so in
I use mime-construct along with the System command - works great.
On 2/26/07, Dovid B [EMAIL PROTECTED] wrote:
- Original Message -
From: Al Bochter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, February 26, 2007 4:20 AM
Subject: [asterisk-users] Sending Email
Yuan LIU wrote:
Mmm. In 1.2.x and 1.4.0,
CLI show agi send text
Usage: SEND TEXT text to send
I posted the CLI help text of the SendText() application which
should basically do the same thing.
Sorry for not making that clear.
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566
On 02/25/07 22:16 Doug Lytle said the following:
zaptel-base.c. This prevents the Sangoma Setup script from patching
zaptel. The fix (Found by Googling) was to rename every instance of
ok, the sangoma scripts on freebsd do not patch the zaptel-bsd source in
any way, so this shouldn't
On 02/25/07 22:16 Doug Lytle said the following:
My experience from yesterday shows that zaptel.c has been renamed to
zaptel-base.c. This prevents the Sangoma Setup script from patching
zaptel. The fix (Found by Googling) was to rename every instance of
ok, the sangoma scripts on freebsd
My wife and daughter, and to lesser extent myself and my daughters
boyfriend would like a communications system which allowed us to talk
to each other, both on a one on one basis, but also occassionally in
conference. My wife and I live in a house with an internal LAN with
each of us with a
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