From: John Novack [mailto:jnov...@stromberg-carlson.org]
Sent: Sunday, July 25, 2010 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Kevin Keane
Subject: Re: [asterisk-users] Using Vertical IP2007 phones with Asterisk?
Kevin Keane wrote:
I recently inherited a Vertical
Hi,
Anybody out there running Adit600s?
I have in my care an Adit600 channel bank connected to an old (version
1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk
(1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail.
I have attempted to add the "slowsequence = yes" line
On Sun, Jul 25, 2010 at 10:20 PM, Janu Mukherjee wrote:
> I installed asterisk server in my linux box. I configured a user 1000 using
> xlite and registered with asterisk server in the same linux box. I
Where on the network is this box?
> configured one more user 1001 in other box and this user
ViciBox actually gives you the option of using the 2.2.1 release or
SVN/trunk versions ViciDial
Also, ViciBox is the officially supported ISO installer of the ViciDial
project.
But, both ViciBox and ViciDialNow are Linux ISO installers that will give
you a functional ViciDial system.
Thanks,
M
Hi,
I installed asterisk server in my linux box. I configured a user 1000 using
xlite and registered with asterisk server in the same linux box. I
configured one more user 1001 in other box and this user also got registered
with asterisk. But i am facing two issues here.
1. When a call is made fr
On 20:59 Fri 23 Jul , Steve Underwood wrote:
> That's just how your images look for me, so I guess your problem is
> described here http://www.soft-switch.org/spandsp_faq/ar01s09.html
>
> Steve
Big thanks for your help, Steve. I tried feh, gqview, gimp and "pages
look an odd shape". Can you s
On 7/25/10 7:54 PM, Nick Brown wrote:
> Hi All,
>
> Facing an issue at the moment with setting the TOS on packets - the
> documentation is a bit light, however is straightforward so unsure if this is
> a configuration issue or a bug.
>
> Following is set in sip.conf;
> tos_sip=CS3
> tos_audio=E
Hi All,
Facing an issue at the moment with setting the TOS on packets - the
documentation is a bit light, however is straightforward so unsure if this is a
configuration issue or a bug.
Following is set in sip.conf;
tos_sip=CS3
tos_audio=EF
And is reflected in the CLI;
IP ToS SIP:
The only big difference I know, is:
VicidialNow - *based on CentOS* - Vicidial 2.0.5.1rc1
ViciBox - *Based on OpenSuse* - Vicidial 2.0.5
The core of the call center for both of them is Vicidial.
Regards.
2010/7/25 Alejandro Cabrera Obed
> Dear all, I need a call center asterisk's based solut
Dear all, I need a call center asterisk's based solution and I see
there are two important solution for 120+ agents:
VicidialNow and ViciBox
Can you tell me the difference between these open source call center
solution please ???
Special thanks
Alejandro
--
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mstults tds.net
Sent: Sunday, July 25, 2010 3:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue
On Sun, J
On Sun, Jul 25, 2010 at 9:52 AM, Kevin Keane wrote:
> I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk
> 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP
> firmware.
>
>
>
> The Asterisk setup relies heavily on queues with dynamic agents. The
> probl
Kevin Keane wrote:
I recently inherited a Vertical Xcelerator IP system with IP2007
phones. I would like to use the phones with an Asterisk system
instead, but there doesn't seem to be much information on it on
Google. Is it even possible? These phones claim that they are SIP phones.
Than
At 12:53 PM 7/25/2010, you wrote:
>A wild stab in the dark, did you Answer() or Progress() before you
>called Dial()? If not, can you add it to your dialplan and retest.
Just added progress with no change.
Ira
--
_
-- Bandwi
I recently inherited a Vertical Xcelerator IP system with IP2007 phones. I
would like to use the phones with an Asterisk system instead, but there doesn't
seem to be much information on it on Google. Is it even possible? These phones
claim that they are SIP phones.
Thanks!
Kevin
--
_
I'm not sure if ALL use the same TFTP address, but I believe so.
My guess is that it is actually the TFTP server that the previous phone vendor
used for the phone's initial configuration before shipping it to us. So in that
sense it would be an old config.
Is there a way to extract the current
It may be pulling a tftp server from dhcp, or it may just have an old
config. Do all the phones (even the ones that work properly) use the
same tftp address?
Thanks,
--Warren Selby
On Jul 25, 2010, at 4:47 PM, Kevin Keane
wrote:
> Stupid question (sorry, I'm pretty much an Asterisk begi
Stupid question (sorry, I'm pretty much an Asterisk beginner) - where do I find
the dialplan.xml? As far as I can tell, there is no TFTP server in this
network. I found the IP address that the phone tries to use for TFTP
(192.168.1.7 in this case) but there is nothing at that device.
Thanks!
-
On Sun, Jul 25, 2010 at 2:53 PM, Ira wrote:
> I assume this should also be in the bug tracker?
>
A wild stab in the dark, did you Answer() or Progress() before you
called Dial()? If not, can you add it to your dialplan and retest.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.bel
Tilghman Lesher wrote:
> On Saturday 24 July 2010 23:52:39 cov...@ccs.covici.com wrote:
> > Hi. I hava a variable and in 1.6 I set the string variable to "" and it
> > got the null string. In 1.8, it gets the quotes, I have to set it to
> > nothing at all to make it get the null value.
>
> Ple
I'm having some mysterious problems installing 1.6.2.10 on Centos 4.8
(totally up to date). I can't see anything on Google or the list regarding
this issue, which I find a bit odd considering 1.6.2.10 was released a few
days ago. I'm therefore assuming there's something weird about my setup,
even t
At 02:58 PM 7/23/2010, you wrote:
>The Asterisk Development Team has announced the release of Asterisk
>1.8.0-beta1.
>This release marks the beginning of the testing process for the
>eventual release
>of Asterisk 1.8.0.
One more problem. Everything seems to work fine but this morning I
decided
At what stage will there be versions of the G.729 codec, res_cepstal,
skypeforasteric, Vestec, etc that'll work with 1.8? It would be good if
people using that software could participate in the Beta.
--
_
-- Bandwidth and Coloca
On Saturday 24 July 2010 23:52:39 cov...@ccs.covici.com wrote:
> Hi. I hava a variable and in 1.6 I set the string variable to "" and it
> got the null string. In 1.8, it gets the quotes, I have to set it to
> nothing at all to make it get the null value.
Please read the 6th item in UPGRADE.txt.
Check your dialplan.xml file that the affected phones are loading.
Thanks,
--Warren Selby
On Jul 25, 2010, at 10:52 AM, Kevin Keane
wrote:
> I recently inherited an Asterisk system (PBX in a Flash, based on
> Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones
> with the
Steve Edwards wrote:
>>> On Sun, 25 Jul 2010, Muro, Sam wrote:
>>>
I am having a problem understanding the way to retrieve some
parameters to asterisk via AGI or what ever method that fits. I have
an executable program that accept one parameter (CALLERID) and return
customer sta
>> On Sun, 25 Jul 2010, Muro, Sam wrote:
>>
>>> I am having a problem understanding the way to retrieve some
>>> parameters to asterisk via AGI or what ever method that fits. I have
>>> an executable program that accept one parameter (CALLERID) and return
>>> customer status from the database se
Kyle Kienapfel wrote:
> On Sun, Jul 25, 2010 at 9:04 AM, Muro, Sam
> wrote:
>> Kyle Kienapfel wrote:
>>> On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam
>>> wrote:
I am having a problem understanding the way to retrieve some
parameters
to
asterisk via AGI or what ever method that f
On Sun, Jul 25, 2010 at 9:04 AM, Muro, Sam wrote:
> Kyle Kienapfel wrote:
>> On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam
>> wrote:
>>> I am having a problem understanding the way to retrieve some parameters
>>> to
>>> asterisk via AGI or what ever method that fits. I have an executable
>>> program
Steve Edwards wrote:
> On Sun, 25 Jul 2010, Muro, Sam wrote:
>
>> I am having a problem understanding the way to retrieve some parameters
>> to asterisk via AGI or what ever method that fits. I have an executable
>> program that accept one parameter (CALLERID) and return customer status
>> from th
Paul Belanger wrote:
> On Sun, Jul 25, 2010 at 11:39 AM, wrote:
> > OK, the line actually is:
> > exten => s,n(auth),Set(password="") which sets to a null value in 1.6.2
> > but does not in 1.8.
> >
> It's possible something changed, how are you checking if the
> ${password} is null? Post an ex
On Sun, 25 Jul 2010, Muro, Sam wrote:
>>> I am having a problem understanding the way to retrieve some
>>> parameters to asterisk via AGI or what ever method that fits. I have
>>> an executable program that accept one parameter (CALLERID) and return
>>> customer status from the database server
> On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam wrote:
>> I am having a problem understanding the way to retrieve some parameters
>> to asterisk via AGI or what ever method that fits. I have an executable
>> program that accept one parameter (CALLERID) and return customer status
>> from the datab
Kyle Kienapfel wrote:
> On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam
> wrote:
>> I am having a problem understanding the way to retrieve some parameters
>> to
>> asterisk via AGI or what ever method that fits. I have an executable
>> program that accept one parameter (CALLERID) and return customer s
Paul Belanger wrote:
> On Sun, Jul 25, 2010 at 9:34 AM, wrote:
> > I am also getting segmentation fault when doing a reload from CLI.
> >
> I believe this is your issue : https://issues.asterisk.org/view.php?id=17704
>
> If not, create a new issue on the tracker with an unoptimized backtrace.
On Sun, 25 Jul 2010, Muro, Sam wrote:
> I am having a problem understanding the way to retrieve some parameters
> to asterisk via AGI or what ever method that fits. I have an executable
> program that accept one parameter (CALLERID) and return customer status
> from the database server which ca
On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam wrote:
> I am having a problem understanding the way to retrieve some parameters to
> asterisk via AGI or what ever method that fits. I have an executable
> program that accept one parameter (CALLERID) and return customer status
> from the database server
On Sun, Jul 25, 2010 at 11:39 AM, wrote:
> OK, the line actually is:
> exten => s,n(auth),Set(password="") which sets to a null value in 1.6.2
> but does not in 1.8.
>
It's possible something changed, how are you checking if the
${password} is null? Post an example dialplan that works in 1.6.2 an
Le 25/07/2010 02:11, Norbert Zawodsky a écrit :
> Hello again!
>
Hi
> after it being "relatively quiet" her for the last weeks, my Astrerisk
> server was the target of 3 of that nasty REGISTER attacks during the
> last days.
>
[...]
Do like most of us are acting: use fail2ban.
--
Daniel
Paul Belanger wrote:
> On Sun, Jul 25, 2010 at 12:52 AM, wrote:
> > Hi. I hava a variable and in 1.6 I set the string variable to "" and it
> > got the null string. In 1.8, it gets the quotes, I have to set it to
> > nothing at all to make it get the null value.
> >
> Post an example for work
I am having a problem understanding the way to retrieve some parameters to
asterisk via AGI or what ever method that fits. I have an executable
program that accept one parameter (CALLERID) and return customer status
from the database server which can be printed in the console.
#./retrive 011747378
I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4
and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware.
The Asterisk setup relies heavily on queues with dynamic agents. The problem I
am having is that on SOME (but not all) the Cisco phones, the
Hi All
I will do a test call from a soft phone to my mobile. I can speak into my
headset and the audio is heard instantly. But if I speak into my mobile
there is a 1-2 second delay in the Audio. I am using SIP.
I am only finding it in the Zoiper Softphones that we are using. All other
phones don'
> I am also getting segmentation fault when doing a reload from CLI.
>
I believe this is your issue : https://issues.asterisk.org/view.php?id=17704
If not, create a new issue on the tracker with an unoptimized backtrace.
--
Yes, the patch fixed it. Thanks.
--
On Sun, Jul 25, 2010 at 9:34 AM, wrote:
> I am also getting segmentation fault when doing a reload from CLI.
>
I believe this is your issue : https://issues.asterisk.org/view.php?id=17704
If not, create a new issue on the tracker with an unoptimized backtrace.
--
Paul Belanger | dCAP
Polybeaco
I am also getting segmentation fault when doing a reload from CLI.
Asterisk crashes and i see
segfault at 46 ip b752827d sp b2bc38f8 error 4 in libc-2.7.so[b74ca000+155000]
in dmesg.
I use Debian Lenny 32bit.
--
_
-- Bandwidth
On Sun, Jul 25, 2010 at 12:52 AM, wrote:
> Hi. I hava a variable and in 1.6 I set the string variable to "" and it
> got the null string. In 1.8, it gets the quotes, I have to set it to
> nothing at all to make it get the null value.
>
Post an example for working (1.6) and not working (1.8)
--
> So, to make the call bridged, I send it across a dhadi "local" span. My
> DAHDI configuration - system.conf:
>
> fxsks=1-8
>
> dynamic=loc,1:0,31,0
> dynamic=loc,1:1,31,0
>
> bchan=9-23,25-39
> dchan=24
>
> bchan=40-54,56-70
> dchan=55
>
> Channels 1-8 are the analog lines to the Toshiba.
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