Hi,
I am trying to make external sip calls by using asterisk. Please provide
information regarding sip trunk configuration in conf files.
Setup is as below,
* *
*Case A:*
Register two soft phones [X-lite] with 1000 and 10001 numbers to asterisk
PBX [running in 192.168.1.11] and able to
On Wed, Jun 16, 2010 at 3:46 AM, Michael Graves mgra...@mstvp.com wrote:
Some distro's, like Askozia and Astlinux, have been specifically
engineered around running from flash media. This basic form of
operation has been well proven in projects like monowall and pfsense.
I think you hit the
Thanks Danny,
I made it working adapting your dialplan
exten = 12345,1,Answer
exten = 12345,n,System(/bin/ls
/var/spool/asterisk/voicemail/default/${EXTENSION}/temp.wav)
exten = 12345,n,verbose(returned ${SYSTEMSTATUS}
exten = 12345,n,Gotoif($[${SYSTEMSTATUS} = SUCCESS]?play1)
exten =
Hi all,
As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the
ISDN phone does not work.
There is the setting and error.
***Enviroment*
Asterisk-1.6.1.18
dahdi-linux-2.3.0.1
dahdi-tool-2.30.
libpri-1.4.11.2
CentOS-5.5
OpenVox
Hi to all,
I use FreePBX version 2.7.0.2 with dahdi. The first problem is with
dahdi: At the system startup i can't find a way to start correctly
Asterisk with Dahdi.
My boot configuration is the following:
/etc/rc.d/after.local
/usr/sbin/rcdahdi start
sleep 15
/usr/local/sbin/amportal
On Wed, Jun 16, 2010 at 10:28:48AM +0200, Claudio Prono wrote:
Hi to all,
I use FreePBX version 2.7.0.2 with dahdi. The first problem is with
dahdi: At the system startup i can't find a way to start correctly
Asterisk with Dahdi.
My boot configuration is the following:
Tried this... it got connected, but I can't hear any audio now whereas codec
was allowed and both negotiated on alaw.
Deepika
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal Hanif
Sent: 15 June
Sounds like you have a firewall or NAT issue
Deepika Nijhawan wrote:
Tried this... it got connected, but I can't hear any audio now whereas codec
was allowed and both negotiated on alaw.
Deepika
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Jonas Kellens wrote:
Rob,
it's not a macro but a sub. In my previous post I posted more info, I
am not going to post the whole output every time.
I read on the wiki that you set the PICKUPMARK equal to the extension
for that channel, but in my case I'm not using extensions but multiple
-- Forwarded message --
From: nikhil singhania niksingha...@gmail.com
Date: 16 June 2010 12:15
Subject: Re: [asterisk-users] can't seem to register, status unmonitored
To: Zeeshan Zakaria zisha...@gmail.com
Here is my extensions.conf:
[general]
static=yes ; default
It's working now after giving nat=yes, thanks.
Deepika
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Dear all, I have to install an E1 card in my Asterisk 1.4.23 server
and here is my short question:
Is it necessary to install or update any Asterisk/Zaptel/Any extra
module or the default installation is good enough to just plug and run
the E1 card
Thanks a lot
Alejandro
--
Hello List:
I'm working on a funny scenario, where I'm bouncing calls from a Cisco
call center into asterisk. Cisco call center has some logic that if a
customer calls in, an agent is logged into a given extension... if
Cisco sends a customer call to that extension, and there is a ring
with no
Alejandro Cabrera Obed wrote:
Is it necessary to install or update any Asterisk/Zaptel/Any extra
module or the default installation is good enough to just plug and run
the E1 card
You'll need to make sure that libpri and dahdi are installed and configured.
Doug
--
Ben Franklin
Your installation should work, you must configure the card channels and
load the card module on your OS.
Regards.
2010/6/16 Alejandro Cabrera Obed aco1...@gmail.com
Dear all, I have to install an E1 card in my Asterisk 1.4.23 server
and here is my short question:
Is it necessary to install
On Wed, 16 Jun 2010, nikhil singhania wrote:
echo(Hello, world!);
echo(Hello, world!);
echo(created);
echo(Hello, world!);
echo $result;
Each time you echo you are violating the AGI protocol. Remember, your
script's STDOUT is feeding requests back to Asterisk. Try enabling AGI
On Wed, 16 Jun 2010, Randy R wrote:
On Wed, Jun 16, 2010 at 3:46 AM, Michael Graves mgra...@mstvp.com wrote:
Some distro's, like Askozia and Astlinux, have been specifically
engineered around running from flash media. This basic form of
operation has been well proven in projects like
Hi,
Am running 1.4.18 at the moment, and am trying to implement inline blind
transfer.
I have :
[featuremap]
blindxfer = *6 ; Blind transfer
in features.conf
And in extensions .conf under [globals] :
DYNAMIC_FEATURES=automon#blindxfr
So what am I missing ??
On voip-info I found a few dated references to TDD support being in the
alpha stage and buggy.
Can anyone direct me to any newer information on this option?
Thanks
--
Karl Harris
--
_
-- Bandwidth and Colocation Provided by
On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com wrote:
pretty much giving up on Skype for Asterisk (and Skype for SIP) now
that I realize that they'll be charging a monthly fee that is
disproportionately high compared to my need to let Skype users call
us. We'll know the
On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi shinacaly...@gmail.comwrote:
Hi!
I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide
a conference bridge for an existing Avaya PBX. I have no control over the
Avaya system, but I am able to speak with the admin in charge
I'm supposing that it is
1. no better or worse than SMS support
2. dependent on the version you are on
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Harris
Sent: Wednesday, June 16, 2010 10:31 AM
To:
On Wednesday 16 June 2010 08:21:17 David Backeberg wrote:
I know if I do not do an Answer() that the call is not yet picked up.
However, if I do a HangUp(), is that functionally equivalent? Can you
Hangup() a channel you never Answer() ed?
A Hangup just returns -1, which causes the dialplan to
On 06/16/2010 11:31 PM, Karl Harris wrote:
On voip-info I found a few dated references to TDD support being in
the alpha stage and buggy.
Can anyone direct me to any newer information on this option?
There are installations where the TDD support in spandsp has been
integrated with
On Wed, Jun 16, 2010 at 11:50 AM, Tilghman Lesher tles...@digium.com wrote:
On Wednesday 16 June 2010 08:21:17 David Backeberg wrote:
I know if I do not do an Answer() that the call is not yet picked up.
However, if I do a HangUp(), is that functionally equivalent? Can you
Hangup() a channel
I am looking at http://www.voip-info.org/wiki/view/Asterisk+cmd+System
I dont see how to execute a system command and set an asterisk variable
to the string that is
in my file /tmp/my_data.
How is that done?
Jerry
--
_
--
Ok got it up and running. In the case for Qwest with NFAS they reserve what
they call Interface ID 1 for the circuit with the backup d channel. In
our case we only have two circuits with a single d channel. The real key
was realizing the logical span number in the spanmap translated into
On Wed, 16 Jun 2010, Jerry Geis wrote:
I am looking at http://www.voip-info.org/wiki/view/Asterisk+cmd+System
I dont see how to execute a system command and set an asterisk variable
to the string that is in my file /tmp/my_data.
Read up on the application readfile().
--
Thanks in advance,
On Wed, 16 Jun 2010, Randy R wrote:
On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com wrote:
pretty much giving up on Skype for Asterisk (and Skype for SIP) now
that I realize that they'll be charging a monthly fee that is
disproportionately high compared to my need to let
On 06/16/2010 11:44 PM, Danny Nicholas wrote:
I’m supposing that it is
1. no better or worse than SMS support
What relevance does SMS support have to TDD/TTY support?
1. dependent on the version you are on
I don't think the TDD support has been touched for years, so I doubt the
Read up on the application readfile().
Steve,
on my 1.4 system help readfile says no such command.
searching a little more shows readfile as an AGI command.
Is this what your refering to?
http://www.voip-info.org/wiki/view/Long+Distance+or+Local+Python+AGI
jerry
--
Do a 'show application ReadFile'
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-16 12:49 PM, Jerry Geis ge...@pagestation.com wrote:
Read up on the application readfile().
Steve,
on my 1.4 system help readfile says no such command.
searching a little more shows readfile as an AGI
core show application readfile is the new command to not get the
deprecated message. To answer the OP's query, here's the dialplan line
Exten = 1234,1,readfile(foo,/tmp/my_data)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Tue, Jun 15, 2010 at 08:46:17PM -0500, Michael Graves wrote:
On Tue, 15 Jun 2010 07:58:34 -0400, SIP wrote:
Danny Nicholas wrote:
Also cheaper to replace flash card than hard drive.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Am running 1.4.18 at the moment, and am trying to implement inline blind
transfer.
I have :
[featuremap]
blindxfer = *6 ; Blind transfer
Do remember that asterisk needs to be in the media stream for this to work,
so you'll want to make sure (in the case of SIP devices) you've set
Hello.
This is what I have :
suppose ${SIPaccounts}=SIP/testcorp5SIP/testcorp6
exten = group,1,Set(_PICKUPMARK=${SIPaccounts})
exten = group,n,Dial(${SIPaccounts})
This is what happens when I try to pickup an extension :
[Jun 16 20:39:33] -- Executing [...@sub-routing:13]
I'm unable to place any calls through a2billing. I followed instructions here:
http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to DISABLE PIN
number request Prompt for some users but, I'm not able to place any calls.
I created a trunk with the same name as in my sip.conf and I'm
Hi!
suppose ${SIPaccounts}=SIP/testcorp5SIP/testcorp6
exten = group,1,Set(_PICKUPMARK=${SIPaccounts})
If I was doing this I'd rather do
Set(_PICKUPMARK=group)
or
Set(_PICKUPMARK=${EXTEN})
but that is probably just me. But let's look at two of your lines:
Set(SIP/testcorp4-
Hello list,
using Asterisk 1.4.30.
[Jun 16 21:35:12] -- Executing [...@sub-routing:12]
Dial(SIP/user110-005a, SIP/user2|999) in new stack
[Jun 16 21:35:12] -- Called user2
[Jun 16 21:35:12] -- SIP/user2-005c is ringing
[Jun 16 21:36:12] WARNING[1991]: chan_sip.c:13073
On Wed, 16 Jun 2010, Landy Landy wrote:
I'm unable to place any calls through a2billing. I followed instructions
here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to
DISABLE PIN number request Prompt for some users but, I'm not able to
place any calls.
I created a trunk
Yes, I'm using XLite...
Anahi Ludueña
From: l...@virtutel.ca
To: asterisk-users@lists.digium.com
Date: Fri, 11 Jun 2010 20:05:39 -0400
Subject: Re: [asterisk-users] Call ended after 31 seconds
You`re using Xlite/eyeBeam by any chance?
Mike
From:
Yes, so I've noticed that I can name _PICKUPMARK anything I want... OK
so the name does not mather and has nothing to do with the different
SIPaccount that it holds...
Another problem is that when another call come in, the _PICKUPMARK
variable is overwritten and I can no longer pick up the
Hello-
After configuring DAHDI and starting asterisk, I get the following
message continuously on the Asterisk console:
WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels
available! Using Primary channel 16 as D-channel anyway!
My card is a D410P configured for E1, only the
Hi all,
As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the
ISDN phone does not work.
There is the setting and error.
***Enviroment**
Asterisk-1.6.1.18
dahdi-linux-2.3.0.1
dahdi-tool-2.30.
libpri-1.4.11.2
CentOS-5.5
On Thu, Jun 17, 2010 at 09:38:43AM +0800, liuxin wrote:
Hi all,
As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the
ISDN phone does not work.
There is the setting and error.
***Enviroment**
Asterisk-1.6.1.18
Note
On 6/17/10 12:49 AM, Steve Edwards wrote:
On Wed, 16 Jun 2010, Landy Landy wrote:
I'm unable to place any calls through a2billing. I followed instructions
here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to
DISABLE PIN number request Prompt for some users but, I'm not
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