On 10/11/09 18:19, Christina Casey wrote:
Hi Klaus,
Yes all the below is possible/easy with the OrderlyStats call centre
management and reporting tool.
It's a free download - please see http://www.orderlyq.com/orderlystats.html
Kind regards,
Christina Casey
Accounts Manager
Orderly
Hi Christina!
From documentation it seems it only supports queues and agents. I do
not have a single queue nor agents. Does it also support real-time
status for normal SIP-SIP calls?
regards
klaus
Christina Casey schrieb:
Hi Klaus,
Yes all the below is possible/easy with the OrderlyStats
On 11 Nov 2009, at 09:06, Alan Lord (News) wrote:
Warning: Your browser may not be able to handle this site! Please
upgrade your browser to the latest version of Internet Explorer,
Firefox, Mozilla or Netscape.
Sorry for any inconvenience.
- The OrderlyQ Team.
I am using Firefox
On 11/11/09 10:00, Steve Howes wrote:
On 11 Nov 2009, at 09:06, Alan Lord (News) wrote:
Warning: Your browser may not be able to handle this site! Please
upgrade your browser to the latest version of Internet Explorer,
Firefox, Mozilla or Netscape.
Sorry for any inconvenience.
-
Ciao,
may be is enough the Free PBX admi web installed with standard Asterisk now
distribution.
Real time status is for sure available.
2009/11/11 Klaus Darilion klaus.mailingli...@pernau.at
Hi Christina!
From documentation it seems it only supports queues and agents. I do
not have a single
Hi,
I am looking for a hosted / virtual IPBX *PLATFORM* for service provider.Such
hosted IPBX platform is aimed to be as a service, so that final customers don't
have to install, maintain, or upgrade any PBX hardware or software.
It should have a control panel for end users to create / edit
Hi,
I am looking for a hosted / virtual IPBX *PLATFORM* for service provider.Such
hosted IPBX platform is aimed to be as a service, so that final customers don't
have to install, maintain, or upgrade any PBX hardware or software.
It should have a control panel for end users to create / edit
Hello. I am trying to execute an fax reception script and i am getting the
following:
[Nov 11 08:40:52] WARNING[12800]: app_system.c:88 system_exec_helper: Unable to
execute '/var/lib/asterisk/scripts/mailfax '
I tried changing the permissions to the mailfax script but Asterisk still can´t
Try aretta.com.
--
Sent from mobile device
On Nov 11, 2009, at 6:41 AM, Paulo Vicentini vizent...@hotmail.com
wrote:
Hi,
I am looking for a hosted / virtual IPBX *PLATFORM* for service
provider.
Such hosted IPBX platform is aimed to be as a service, so that final
customers don't have
Hi people, just a question:
Is it possible to execute Voicemail command in the h extension? (after hangup
the channel).
Because if I put it before it, it works right, but if I put it there, it
doesn't...
The log is:
-- Executing [...@cont-mine:1] NoOp(SIP/3005-096736a8, End of
cont-mine)
'core show application dial' should give you an idea of what to play
around with...
In a similar scenario, once I used the 'm' option, with a special moh
class. The moh class had some soft ticking sound because the remote
system was not correctly indicating ringing, and sometimes delayed the
Hello. I am trying to execute an fax reception script and i am getting the
following:
[Nov 11 08:40:52] WARNING[12800]: app_system.c:88 system_exec_helper: Unable to
execute '/var/lib/asterisk/scripts/mailfax '
I tried changing the permissions to the mailfax script but Asterisk still can´t
Is SELinux enabled on the machine? If it is, you might have a problem with
the asterisk process being able to execute in that directory.
On Wed, Nov 11, 2009 at 4:38 AM, her Garcia herli...@lycos.com wrote:
Hello. I am trying to execute an fax reception script and i am getting the
following:
What are you trying to achieve here? The h extension is for when the
channel hangs up. And if the caller hangs up how will he leave you a
voicemail?
Sent from my iPod
On Nov 11, 2009, at 7:20 AM, Anahi Ludueña a_ludu...@hotmail.com
wrote:
Hi people, just a question:
Is it possible to
Voicemail is by definition an active channel application (you hangup, this
application cant run). Your options would be either an internal callback
to let you leave a voicemail or a callback to the outside caller to ask
him/her to leave a voicemail.
_
From:
Hi all,
I did some call using an asterisk 1.4 PBX and 2 softphone in a private
network;
call is up, but with bad quality.
Someone knows how to debug this problem ?
Thanks in advance for any help.
--
Giancarlo Lombardo
___
-- Bandwidth and Colocation
Darrick Hartman wrote:
Alejandro Recarey wrote:
No I have only tried 1.6.1.6.
I could consider downgrading, but upgrading to a bleeidng edge release
makes me kind of nervous, as the box is in production right now. 1.6.1.0
is recommended?
NO! If you're using a specific 'branch' of
Darrick Hartman wrote:
Alejandro Recarey wrote:
Hello all,
I have a pretty much standard installation of an Asterisk 1.6.1.6 with
no PRI cards of any type (full VoIP).
Occasionally (it happens every 2 weeks or so), it just stops running. I
was using safe_asterisk but it seems that
Dear all,
I'm not sure that mail was correctly delivered.
Best Regards
Giancarlo Lombardo
-- Forwarded message --
From: giancarlo lombardo gianclomba...@gmail.com
Date: 2009/11/11
Subject: Bad quality of call
To: asterisk-users@lists.digium.com
Hi all,
I
Did you use a headset for you softphone? Check the settings of your headset
(mic volume particularly) because they pickup all the noise around (computer
fans, AC, etc..).
reygue
On Wed, Nov 11, 2009 at 9:01 AM, giancarlo lombardo gianclomba...@gmail.com
wrote:
Hi all,
I did some call using
Hi Klaus,
Yes it does also support real-time status for normal SIP-SIP calls.
You just add the phones you want to monitor by hitting the Add Interface
button in the Interfaces section of the Agents page.
Regards,
Christina Casey
Account Manager
Orderly Software
Hello,
I'm running Asterisk 1.4.26.2, DAHDI 2.2.0.2, and libpri 1.4.10.2 on Ubuntu
with a Digium TE122 connected to my PRI.
I have an extension setup so that managers can monitor calls:
# grep 9900 /etc/asterisk/extensions.conf
exten = 9900,1,DAHDIScan()
exten = 9900,2,Hangup
For
Hi,
I would like some advice from you on how to configure a multi line phone
the best way!
So far I have given the phone 4 sip accounts one for each line, this is
a lot of work and gets messy.
Is it a better way to do this?
Thank you!
Best regards
Helge-Bjørn
Hi,
I have a weird issue that I hope someone can help me with. I have 2 test
computers and I've changed each the roles of each one with the same results.
I have one xlite client running across a VPN and another connecting directly
from the WAN via the external IP. The client connecting
I'm sure you'll get a better answer than this, but IMO the Best Practice
is to have the phone register to Asterisk and have Asterisk register to the
providers. By sip accounts, I assume you mean external SIP accounts like
5551...@bandwidth.com. So the phone has 101, 102, 103 and 104 as it's
One of the coolest things about Asterisk, and SIP/VoIP in particular is the
abstraction of the logical (numbers) and the physical (lines). 'Lines' and
'numbers' are no longer one and the same. For example, I have something
like one hundred telephone numbers, 50 phones, and only ONE sip
Karl Fife wrote:
Question about the proper way to update LibPRI:
'Bouncing' asterisk after an installing the new LibPRI version does
indeed reflect the update:
asterisk*CLI pri show version
libpri version: 1.4.10.2
Hmm. What asterisk version are you running?
On 1.6.0.18-rc2:
pbx*CLI
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Monday, October 19, 2009 9:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] digium
Karl Fife wrote:
Question about the proper way to update LibPRI:
'Bouncing' asterisk after an installing the new LibPRI version does
indeed reflect the update:
asterisk*CLI pri show version
libpri version: 1.4.10.2
Hmm. What asterisk version are you running?
On 1.6.0.18-rc2:
pbx*CLI
Hi all,
I'm not sure that this mail was received.
Thanks again
-- Forwarded message --
From: giancarlo lombardo gianclomba...@gmail.com
Date: 2009/11/10
Subject: user extension in asterisk GUI
To: asterisk-users@lists.digium.com
Hi all,
I just configured some user in sip.conf
It was, but it was a FreePBX question not an Asterisk one.
The answer is that it doesn't read custom config. You can use FreePBX
php libraries to add extensions though.
Steve
On 11 Nov 2009, at 17:23, giancarlo lombardo wrote:
Hi all,
I'm not sure that this mail was received.
Thanks
Thanks, that is what I checked, there is nothing in there that would
appear to do it.
I wasn't sure if there were any hidden variables I could set beforehand.
I'll try the MOH class as it might work. The ringback tones are
indicating that an external system is being called, and we are trying to
2009/11/10 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote:
Hello,
1. How can specify in /etc/dahdi/genconf_parameters file that a port from
a
B410P board is to be disabled.
There's currently no way to do that.
It should be trivial to
I don't know why. I am a registered user.
reygue
On Wed, Nov 11, 2009 at 12:23 PM, giancarlo lombardo
gianclomba...@gmail.com wrote:
Hi all,
I'm not sure that this mail was received.
Thanks again
-- Forwarded message --
From: giancarlo lombardo gianclomba...@gmail.com
Hi all,
My Asterisk problem today involves getting a SIP client on a private
net to register with a server somewhere else on the Internet. This
worked for me about a year ago no problem, but now I see an error
message on the remote server every time the client attempts to connect
(the
Hi Asterisk Users,
We've been experiencing some tough time regarding a new Asterisk installation
connected to the PSTN via an ISDN PRI with a Digium TE121 with the optional
VPMADT032 echo cancellation module.
For now, I'll focus on something very specific which is summarized on this
Hello,
Using 1.6.2-rc5, my settings include:
[local-phone](!)
context=mylocal
type=friend
nat=no
canreinvite=no
host=dynamic
qualify=yes
dtmf=info
language=fr
call-limit=5
subscribecontext=subs
disallow=all
allow=alaw
t38pt_udptl=no
On Wednesday 11 November 2009 14:23:31 Olivier wrote:
Hello,
Using 1.6.2-rc5, my settings include:
[local-phone](!)
context=mylocal
type=friend
nat=no
canreinvite=no
host=dynamic
qualify=yes
dtmf=info
language=fr
call-limit=5
Hi,
I am using zap channels, and by using sendDTFM application, I can control
the duration between two DTMF digits, but I can't find a way to control the
duration of the digits themself. Did search on the Internet and found out
that I can change it in the asterisk source files and recompile
I am about out of ideas
I am not able to keep this gateway stable. I am crashing about 2 times a
day
Is there a way to capture the crash data? I have kdump configured on the
server but it seems to be a hard lockup and not a kernel panic
I have tried the following:
Asterisk
The Zapata.conf tonedur DOES work, you just have to restart Asterisk EVERY
time you change the value.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Wednesday, November 11, 2009 3:18 PM
To: Asterisk
On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote:
2009/11/10 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote:
Hello,
1. How can specify in /etc/dahdi/genconf_parameters file that a port from
a
B410P board is to be disabled.
On 12/11/09 9:04 AM, Jaap Winius wrote:
Hi all,
My Asterisk problem today involves getting a SIP client on a private
net to register with a server somewhere else on the Internet. This
worked for me about a year ago no problem, but now I see an error
message on the remote server every time
This year we recorded quite a few of the AstriCon sessions - 3 out of
the 4 tracks were video taped. The folks at TMC then went through a
fairly painstaking process of synchronizing the video presentations to
the slide decks that each presenter provided, so we have an index-able
and
The reasons for poor call quality are many and varied.
As another poster suggested, the headset you are using might be poorly
configured, or just a poor example.
An under-spec server could also do it - I use two simple, low-spec Virtual
Machines in my dev lab that I bring up when I want to
Hi Danny,
Then I must be doing something wrong here, because I did restart Asterisk
every time I made changes. I have something like this here:
group=25
toneduration=500
context=mapping-custom
signalling=em
channel = 25
Group 25 has only one channel for testing purposes.
Zeeshan
On Wed, Nov
Scott L. Lykens wrote:
Any progress on new Fax for Asterisk modules? Last update here was
October 19 as quoted above; Original discussion is now over six weeks
old. FAA Download Selector still shows modules for 1.6.1.4 as the latest
available.
Yes, there has been progress. The new modules
Toneduration=500 would make any call use .5 seconds for each DTMF digit
pressed/sent. Therefore if you dialed 18005551212, the call would take
about 6-8 seconds to connect.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sorry to reply so late, I am months behind and catching up.
I have been inspecting this on my own systems, and the results are inconsistent
to say the least. I’ve been dumping these to the verbose logs for some time and
monitoring them, but I have not been able to determine why the numbers
Dear all,
thanks for the suggestion;
it seems to be a problem of my speaker;
in fact I have same problem using skype
2009/11/11 Michael Wyres mwy...@cdm.com.au
The reasons for poor call quality are many and varied.
As another poster suggested, the headset you are using might be poorly
DNS doesn't seem to resolve, looks like one of those unfortunate Domain name
registration decisions where the DNS servers and all contact email addresses
for the domain are from the domain itself:
NETXUSA.COM
Administrative Contact:
x...@netxusa.com
Technical Contact:
x...@netxusa.com
Domain
Anyone know what happened to netxusa?
Seemed like they dropped off the web overnight.
-Matt
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Wed, Nov 11, 2009 at 1:11 PM, Matt Florell astma...@gmail.com wrote:
They had a nice booth at Astricon and everything. Haven't heard anything
about them going down, this might just be an unfortunate IT management
incident.
Both their toll free and fax numbers go to a re-order
Hi there
I am wondering if anybody can help me illuminate a problem I am having with
my asterisk installation. I am using:
- IP phone (Siemens gigaset S685IP) behind a modem/router that has ports udp
5060 and 1:10100 forwarded to the static ip of the IP phone
(192.168.0.3). This has to go
Asterisk is not receiving replies to the INVITE - probably due to NAT
issues.
marcus wells wrote:
Hi there
I am wondering if anybody can help me illuminate a problem I am having
with my asterisk installation. I am using:
- IP phone (Siemens gigaset S685IP) behind a modem/router that
This is possible as I was just able to get the latest SIP firmware
loaded on my 7942. Make sure to follow the guide using the 7941 as the
SIP firmware differs from the 79x0 versions. Here's two links to help:
No problems resolving anything here. Website appears to be up. Maybe
they had a temporary equipment issue.
Matt Darnell wrote:
On Wed, Nov 11, 2009 at 1:11 PM, Matt Florell astma...@gmail.com wrote:
They had a nice booth at Astricon and everything. Haven't heard anything
about them going
Matt Darnell wrote:
Both their toll free and fax numbers go to a re-order message...seems
like the worst.
These days if you have your PBX, DNS, web, and email all hosted on the
same server, it doesn't take much to have your entire business appear to
be gone.
On Wed, Nov 11, 2009 at 2:04 PM, Kevin P. Fleming kpflem...@digium.com wrote:
Scott L. Lykens wrote:
Any progress on new Fax for Asterisk modules? Last update here was
October 19 as quoted above; Original discussion is now over six weeks
old. FAA Download Selector still shows modules for
Jonathan Thurman wrote:
Any chance that 64 bit Linux will be supported?
There is a small chance; I've done some work in the past week while
traveling to attempt solve the 64-bit problems, and I fixed some of them
but not all of them, so it still won't successfully FAX on a 64-bit
machine. I'll
Hello.
I'm trying to test an Asterisk server by using a VOIP provider for
international calls but, I'm having problems trying to get my server
communicate with theirs. I don't know if I'm having all these issues becuase
I'm behind NAT or what. I have the following in my server's sip.conf:
The 7960 and 79x2 use different sip firmwares and as far a I have seen
the 7960 does not have the same port issue the 7941/2 seems to have
(which technically is not a problem, just an implementation of the sip
protocol that you don't typically see).
As to your issue, are you still seeing
Have you tried nat=yes in the definition in sip.conf?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Thursday, 12 November 2009 13:30
To: asterisk-users@lists.digium.com
Subject:
hi all, i had installed asterisk on Centos 5.3, sip.conf and
extentions.conf are vi /etc/asterisk/sip.conf [general]
port =
5060
bindaddr = 192.168.1.2 (asterisk server ip addr)
context = others
[2000]
type=friend
context=my-phones
secret=1234
host=dynamic
[2001]
type=friend
context=my-phones
Hello.
I see this post many times. I have written this for you to get a start.
This is sip.conf
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060; UDP Port to bind to (SIP standard port is 5060)
You may be doing some thing wrong with Configuration of Softphone. Please
take a tutorial .. Google is a good friend. I suggest you to use X-lite
softphone.
On Thu, Nov 12, 2009 at 11:25 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com
wrote:
Hello.
I see this post many times. I have written
hi all, i had installed asterisk on Centos 5.3, sip.conf and
extentions.conf are VI /ETC/ASTERISK/SIP.CONF [general]
port =
5060
bindaddr = 192.168.1.2 (asterisk server ip addr)
context = others
[2000]
type=friend
context=my-phones
secret=1234
host=dynamic
[2001]
type=friend
context=my-phones
I have replied you already . Please look into it
On Thu, Nov 12, 2009 at 11:31 AM, aster...@opensourcesolution.in wrote:
hi all,
i had installed asterisk on Centos 5.3, sip.conf and extentions.conf are
*vi /etc/asterisk/sip.conf*
[general]
port = 5060
bindaddr = 192.168.1.2 (asterisk
Please stay on list because if some one other face similar problem he can
get help by googling list.
IN domain name u have to specify ASterisk server IP in XLITE.
On Thu, Nov 12, 2009 at 11:36 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com
wrote:
I have replied you already . Please look
Hi,
I hooked up a AASTRA 480i to the external IP from my home and I get the
exact same scene. I can call from it to any internal extension but it won't
receive a call. I also noticed that the Panel in FreePBX GUI doesn't show it
connected eventhough I can still call it from an internal
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