Hi,
The ami manager call out with an originate through dadhi to a local number (A).
If this call is answered, then the ami manager redirect this call to a dial
command.
This dial command calls through dadhi to another local number (B).
Number B answers this call and number A en B are connected.
Hello,
Is it possible to install Asterisk on Vmware(centos) from source. Is there
any difference or disadvantage for this compared to asterisk running on
physical machine.
--
_
-- Bandwidth and Colocation Provided by
Tino wrote:
Hello,
Is it possible to install Asterisk on Vmware(centos) from source. Is
there any difference or disadvantage for this compared to asterisk
running on physical machine.
What version of vmware?
Generally it works but it could be a problem if you require access to
dahdi
Thanks Gareth for your quick reply.
It is the lateset version and i think i need access to Dahdi interface. Is
there any disadvantages other than this.
On Wed, Aug 11, 2010 at 2:11 PM, Gareth Blades
list-aster...@skycomuk.comwrote:
Tino wrote:
Hello,
Is it possible to install Asterisk on
Hi
I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon.
With some calls, the value in the `billsec` field in the CDR is exceeding the
value in the `duration` field.
I didn't think this was supposed to happen? Our old installation (some
ancient version, sorry not available)
On 08/11/10 18:46, Tino wrote:
Thanks Gareth for your quick reply.
It is the lateset version and i think i need access to Dahdi
interface. Is there any disadvantages other than this.
If you need access to cards installed in the machine, you can forget
running Asterisk under VMware. VMware
'latest version' doesnt really help. There are multiple products.
Tino wrote:
Thanks Gareth for your quick reply.
It is the lateset version and i think i need access to Dahdi interface.
Is there any disadvantages other than this.
On Wed, Aug 11, 2010 at 2:11 PM, Gareth Blades
That was exactly what I was lamenting - that some common distros do not send
every event, so that AMI ends up being less than reliable. If AMi sends all
events, then it's really trivial to track calls :)
l.
2010/8/9 Motiejus Jakštys desired@gmail.com
On Mon, Aug 9, 2010 at 12:08 PM, Lenz
Hi all,
using Asterisk 1.8 beta3 installed from scratch I am not able to
stop/start/restart Asterisk deamon with
/etc/init.d/asterisk stop|start|restart
It just happens nothing, no warnings, errors etc.
I am running Debian Lenny.
Any ideas what is wrong?
Thanks,
Oliver
--
Hi,
The best cards I used are Sangoma's but if you cant afford them go for
OpenVox, they are exactly the same with digiums and their 4 port ones doesnt
need any driver installation even.
Regards.
--
M. Shokuie Nia.
On Wed, Aug 11, 2010 at 8:55 AM, Faisal Hanif fai...@vopium.com wrote:
Hi,
Sorry if this is a stupid question but you are doing
sudo /etc/init.d/asterisk stop|start|restart
Aren't you?
have you made sure /etc/init.d/asterisk is executable?
Ish
On 11/08/10 11:14, unsero...@aol.com wrote:
Hi all,
using Asterisk 1.8 beta3 installed from scratch I am not able to
Yes I do.
And yes, I checked, it is executable.
--
Sorry if this is a stupid question but you are doing
sudo /etc/init.d/asterisk stop|start|restart
Aren't you?
have you made sure /etc/init.d/asterisk is executable?
Ish
On 11/08/10 11:14, unsero...@aol.com wrote:
Hi all,
using Asterisk
Are you talking about VMware Server, ESX/ESXi, or one of their other products?
The only VMWare product that I can even conceive might work is ESX/ESXi.
Others have already pointed out that in VMware, you won't get direct access to
the hardware. VMWare does have some limited capability to
I have an asterisk 1.6.2 server running on Debian Lenny on ESX which
bridges ENUM SIP and Skype Calls with our Cisco PBX (connected via SIP).
I have not heard of any problems so far regarding timing. Voice is
always clear and fine.
hth
Philipp
On 2010-08-11 13:23, Kevin Keane wrote:
Are
On Wed, Aug 11, 2010 at 06:14:52AM -0400, unsero...@aol.com wrote:
Hi all,
using Asterisk 1.8 beta3 installed from scratch I am not able to
stop/start/restart Asterisk deamon with
/etc/init.d/asterisk stop|start|restart
It just happens nothing, no warnings, errors etc.
Next step:
On 10-08-11 04:36 AM, Tino wrote:
Is it possible to install Asterisk on Vmware(centos) from source. Is
there any difference or disadvantage for this compared to asterisk
running on physical machine.
As mentioned you won't have access to DAHDI hardware as VMware won't permit
access to it.
Make sure to set the voicemail_odbc.so ans voicemail_imap.so modules to
noload in module.conf
On Wed, Aug 11, 2010 at 4:52 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Aug 11, 2010 at 06:14:52AM -0400, unsero...@aol.com wrote:
Hi all,
using Asterisk 1.8 beta3 installed from
On 10-08-10 04:11 AM, Olle E. Johansson wrote:
26 jul 2010 kl. 18.13 skrev Leif Madsen:
On Asterisk 1.6.2, your only option for distributing device state is with
res_ais. I've used it in a labbing system and it works well -- the caveat is
that your machines need to be on a low latency
Hi,
I'm trying to set up an old PBX (that supports SIP) to go through our
new Asterisk server, so that our old phones can be used still for some
time.
How can I set up Asterisk to deliver a trunk sip connection that our old
PBX can connect to? Is it just to sett up a normal sip device in
On Wed, Aug 11, 2010 at 10:12 AM, Kent Varmedal k...@domeneshop.no wrote:
I'm trying to set up an old PBX (that supports SIP) to go through our
new Asterisk server, so that our old phones can be used still for some
time.
How can I set up Asterisk to deliver a trunk sip connection that our old
On Wed, Aug 11, 2010 at 4:36 AM, Tino t...@sparksupport.com wrote:
Is it possible to install Asterisk on Vmware(centos) from source. Is there
any difference or disadvantage for this compared to asterisk running on
physical machine.
This has come up repeatedly on the list.
Basically, the less
Hi all,
using Asterisk 1.8 beta3 installed from scratch I am not able to
top/start/restart Asterisk deamon with
/etc/init.d/asterisk stop|start|restart
It just happens nothing, no warnings, errors etc.
Next step: start tracing.
sh -x /etc/init.d/asterisk start
--
h -x
On Wed, 2010-08-11 at 10:29 -0400, David Backeberg wrote:
On Wed, Aug 11, 2010 at 10:12 AM, Kent Varmedal k...@domeneshop.no wrote:
I'm trying to set up an old PBX (that supports SIP) to go through our
new Asterisk server, so that our old phones can be used still for some
time.
How can
On Wed, Aug 11, 2010 at 11:24 AM, Kent Varmedal k...@domeneshop.no wrote:
We need to upgrade this PBX for it to work with SIP, it is at the moment
using ISDN. And those who delivered it and do the
support/reconfiguration is paid by the hour. We don't have any control
over it our self, so when
Hello,
How to take the values of channel variables like 'agi_uniqueid' and
'agi_callerid' in agi script.
For example
#!/bin/bash -x
T=$agi_uniqueid
I want to save value of 'agi_uniqueid' channel variable into a variable
called 'T' in my script
--
BTW, I believe after all said everyone agrees that non of the
providers send onto the PSTN more than 10 digits the only question is
what they send to their customers.
On Wed, Aug 11, 2010 at 1:35 AM, John Novack
jnov...@stromberg-carlson.org wrote:
C F wrote:
The correct way in the US is to
All,
I have multiple Asterisk servers in various locations running various
1.4 and 1.6 versions (lab and production) and am having trouble with a
new Aastra 6739i (3.0.1.2015) registering. Below is my request to
support and they have looked it over and don't see anything wrong:
Support, Can
On Wednesday 11 August 2010 03:59:28 A J Stiles wrote:
I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon.
With some calls, the value in the `billsec` field in the CDR is exceeding
the value in the `duration` field.
I didn't think this was supposed to happen? Our old
- Tilghman Lesher tles...@digium.com wrote:
On Wednesday 11 August 2010 03:59:28 A J Stiles wrote:
I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr
addon.
With some calls, the value in the `billsec` field in the CDR is
exceeding
the value in the `duration` field.
I
On Wed, 11 Aug 2010, Tino wrote:
How to take the values of channel variables like 'agi_uniqueid' and
'agi_callerid' in agi script. For example
#!/bin/bash -x
T=$agi_uniqueid
AGI is a protocol.
Basically, Asterisk creates a process to run your AGI compliant program
(the above is not).
On Wednesday 11 August 2010 11:51:10 Tim Nelson wrote:
- Tilghman Lesher tles...@digium.com wrote:
On Wednesday 11 August 2010 03:59:28 A J Stiles wrote:
I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr
addon.
With some calls, the value in the `billsec` field in the
Can anyone help please on this?
I tried same configuration on CentOS as well and got same result i.e. No
sound and hangup.
On 04/08/10 5:58 PM, Davinder Kumar Meen davin...@impingeonline.com
wrote:
Hello,
I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk
1.6.2.9
Hello list,
what does the following mean ?
asterisk*CLI sip show peer test13
Mailbox : 1...@default
VM Extension : asterisk
How can this VM Extension be set to the extension of the mailbox ?!
Kind regards,
Jonas.
--
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Davinder Kumar
Meen
Subject: Re: [asterisk-users] Asterisk not working with Festival
Can anyone help please on this?
snip
[connect-to-me]
exten = s,1,Answer
Exten =
I have DAHDI 2.2.1
my system.conf file is :
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
echocanceller=mg2,1-23
loadzone= us
defaultzone = us
Its telling me:
Running dahdi_cfg: DAHDI_SPANCONFIG failed on span 1: Invalid argument (22)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Subject: [asterisk-users] VM Extension : asterisk
Hello list,
what does the following mean ?
asterisk*CLI sip show peer test13
Mailbox : 1...@default
VM Extension :
On 08/11/2010 08:39 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Subject:* [asterisk-users] VM Extension : asterisk
Hello list,
what does the following mean ?
asterisk*CLI sip show
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Subject: Re: [asterisk-users] VM Extension : asterisk
snip
Hello,
I'm using sip realtime peers in Asterisk 1.4.30. I do not see a field
vmexten defined in the mysql-DB on
Sorry, I am not following:
***read the value of var ${HANGUPCAUSE} next line to dial command.*
*
*
*Where is that value? Next to dial you mean right when the call was placed?
or check next few lines to find HANGUP cause?*
*
*
*Note: This is using ZAP (analogue) and not PRI.*
*
*
*Thanks,*
*Bruce
[Aug 11 17:37:03] WARNING[7503]: chan_dahdi.c:2790 pri_find_dchan: No
D-channels available! Using Primary channel 24 as D-channel anyway!
I am getting the above error with dahdi 2.2.1 libpri 1.4.11.2 and
astersi 1.4.32
system.conf
loadzone=us
defaultzone=us
span=1,0,0,esf,b8zs
bchan=1-23
Does anyone know the mechanism by which companies like YouMail (and MNO's
using their own voicemail system) are able to redirect ALL calls from a ALL
subscribers to *just one* voicemail DID, yet determine WHICH subscriber did
the redirection?
I had always assumed this was simply done using
Jerry Geis wrote:
signalling=pri_net
It needs to be pri_cpe
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
_
--
Depends what its connected to
Nick.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Thursday, 12 August 2010 9:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi,
Your dial-plan could be like this,
here you will dial
EXTEN = _X,1,NoOp
EXTEN = _X,n,Set(WHOHAVEHANGED=CALLER)
EXTEN = _X,n,Dial(ZAP/xyz)
if caller hanged below line will never be executed because control will
go to h extension.
EXTEN = _X,n,Set(WHOHAVEHANGED=CALEE)
EXTEN =
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