2009/12/29 Zhang Shukun :
> OK. Thanks
>
> 2009/12/29 ram :
>>
>>
>> On Mon, Dec 28, 2009 at 10:39 PM, Zhang Shukun wrote:
>>>
>>> hi,
>>> i have installed a2billing , when i open /admin web pages. errors as
>>> follow:
>>>
>>> Fatal error: Call to undefined function bindtextdomain() in
>>> /usr
Hi,
How does Asterisk CDR work? How can I have in CDR records calls without BYE
message? I checked my wireshark traces and some calls has no BYE messages,
but they appears in CDR as answered call.
Thanks
Szabolcs Szasz
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>> I can't believe anyone would use RJ-11 any more. You can multi-purpose
>> RJ-45 jacks to work with POTS lines. Run everything down to a central
>> panel and send pots over the jacks that you need to. That way if you
>> decide you need/want to go IP in the future, you're all set.
>>
>> Dar
On Monday 28 December 2009 23:49:13 Tilghman Lesher wrote:
> On Monday 28 December 2009 18:09:15 JR Richardson wrote:
> > I turned on console debug to see the actual mysql queries and to my
> > surprise and concern, I see every query for an extension priority
> > repeated 3 or more times prior to d
OK. Thanks
2009/12/29 ram :
>
>
> On Mon, Dec 28, 2009 at 10:39 PM, Zhang Shukun wrote:
>>
>> hi,
>> i have installed a2billing , when i open /admin web pages. errors as
>> follow:
>>
>> Fatal error: Call to undefined function bindtextdomain() in
>> /usr/local/src/a2billing/common/lib/languageS
On Mon, Dec 28, 2009 at 10:39 PM, Zhang Shukun wrote:
> hi,
> i have installed a2billing , when i open /admin web pages. errors as
> follow:
>
> Fatal error: Call to undefined function bindtextdomain() in
> /usr/local/src/a2billing/common/lib/languageSettings.php on line 130
>
> do you know wha
hi,
i have installed a2billing , when i open /admin web pages. errors as follow:
Fatal error: Call to undefined function bindtextdomain() in
/usr/local/src/a2billing/common/lib/languageSettings.php on line 130
do you know what's wrong?
--
Thanks,
Sucan
__
hi,
Does A2Billing has mial list?
--
Thanks,
Sucan
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Thats one REALLY ugly website. Can you at least change the background
colour from black so we can read it?
BillK
On Tue, 2009-12-29 at 00:37 -0500, Dean Collins wrote:
> UPDATE - This is really really bad - check out the paypal phishing
> example on my blog already using Cyrillic characters
>
When I place an outbound call from asterisk 1.6.1.12 to a FXO port on
my Cisco 1760V 12.4, the channel changes - seemingly incrementing:
e.g., in the first call, below, the channel name is
"SIP/vgw1-0075" -- the second call (on the same FXO port after a
soft hangup on the CLI) is "SIP/vgw1-
On Monday 28 December 2009 18:09:15 JR Richardson wrote:
> I turned on console debug to see the actual mysql queries and to my
> surprise and concern, I see every query for an extension priority
> repeated 3 or more times prior to dialplan execution. For instance my
> first dialplan activity is al
UPDATE - This is really really bad - check out the paypal phishing
example on my blog already using Cyrillic characters
http://blog.collins.net.pr/2009/12/de-latinisation-of-web.html
Please forward to everyone in a position to stop ICANN, i cant believe
they didn't think of this in advance.
At 16:13 12/28/2009, Rick Huebner wrote:
>My brother-in-law is finishing up his McMansion and I've done all of the
>low voltage wiring and am starting the trimout. We are batting around
>what to do for a phone system and I'm torn between a Panasonic
>TAW824/TVA50 and using an Asterisk implemen
> Rick Huebner wrote:
> > My brother-in-law is finishing up his McMansion and I've done all of the
> > low voltage wiring and am starting the trimout. We are batting around
> > what to do for a phone system and I'm torn between a Panasonic
> > TAW824/TVA50 and using an Asterisk implementation. I'
Greetings-
I'm in the process of turning up an Asterisk box for a customer and was
wondering if anyone had any good code they could share for implementing
vertical service codes within Asterisk. I'd really rather not have to
spend hours making new wheels if someone has one or more that will fit
i want to thank all the developers at asterisks .many many thanks to alain
chaipainos at netbricks /enea
work hard an dimplement the machine.
happy hanuca to all of you
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as
On Mon, Dec 28, 2009 at 4:13 PM, Rick Huebner wrote:
> My brother-in-law is finishing up his McMansion
[snip]
Asterisk can do all you wish, easily. I've not had any experience
with any Asterisk hardware, never tied a physical line into an
asterisk system, but based on what I've learned in #aste
On Dec 28, 2009, at 8:03 PM, Taylor, Jonn wrote:
> Darrick Hartman wrote:
>>> -Original Message-
>>> On Behalf Of Rick Huebner
>>> Sent: Monday, December 28, 2009 4:13 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: [asterisk-users] Looking at Asterisk for
Darrick Hartman wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Huebner
Sent: Monday, December 28, 2009 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
Hi All,
I'm testing some realtime extension apps with Asterisk 1.4.28 and
addons 1.4.10 using res_mysql. Localhost database is 5.0.32 with
Debian Etch. The apps are working fine all syntax is proper, using
Set with (REALTIME) function, Set with (CUT) function, calling a Macro
with s extensions,
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Huebner
> Sent: Monday, December 28, 2009 4:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Looking at As
Rick Huebner wrote:
> My brother-in-law is finishing up his McMansion and I've done all of the
> low voltage wiring and am starting the trimout. We are batting around
> what to do for a phone system and I'm torn between a Panasonic
> TAW824/TVA50 and using an Asterisk implementation. I'm very s
The layout you describe would be an Asterisk installation using a TDM400P or
TDM410P (possibly TDM800P, since you are describing 5-7 POTS lines) in the
Pentium box using FFA for your Fax service. The handsets would cost you
about $50 a pop or up. All other items (1-5) are less than 2 days work fo
My brother-in-law is finishing up his McMansion and I've done all of the
low voltage wiring and am starting the trimout. We are batting around
what to do for a phone system and I'm torn between a Panasonic
TAW824/TVA50 and using an Asterisk implementation. I'm very strong on
the networking/linux/
- "Tim Nelson" wrote:
> - "Leif Neland" wrote:
> > I want some cheap ip-phones with auto-answer, to work as paging
> system
> >
> > at dinnertime.
> > Options, please.
> >
> > Leif
> >
>
> I've had great luck using the BT201 phones from Grandstream for this
> purpose. In fact, this is
- "Leif Neland" wrote:
> I want some cheap ip-phones with auto-answer, to work as paging system
>
> at dinnertime.
> Options, please.
>
> Leif
>
I've had great luck using the BT201 phones from Grandstream for this purpose.
In fact, this is the only situation where I still use Grandstream
On Mon, Dec 28, 2009 at 03:38:54PM -0600, Danny Nicholas wrote:
> Xlite on your kitchen computer? Check out amazon or ebay for current
> pricing of "real" phones - auto-answer probably starts on the $75 and up
> phones.
And there's still no single existing project to reprogram existing cheap
IP-p
Xlite on your kitchen computer? Check out amazon or ebay for current
pricing of "real" phones - auto-answer probably starts on the $75 and up
phones.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Neland
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Leif
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Ill try another mx of libPRI, it has to be something goofy.
Thanks for the input guys.
On Mon, Dec 28, 2009 at 11:00 AM, Kevin P. Fleming wrote:
> Steve Totaro wrote:
>
> > I would call Digium but last I knew, NFAS only worked across one card.
>
> NFAS is implemented in libpri and Asterisk (cha
What do you mean I should use a global function. I'm kind both well versed and
a newb to *
James Shigley
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez
Sent: Monday, December 28, 2009 12
Hi. Does anyone have a patch or workaround for the 50 BLF limit of
Aastra phones? I have a couple 57i with the 560M console and only the
first 50 BLF lines get registered. I am using the latest firmware from
Aastra but I read that this limit was imposed because of a memory leak.
Obviousl
Is ddwhome defined in global context?? If so, then you should use global
function.
Paste asterisk log to check.
Saludos,
Juan E. Rodríguez
-Original Message-
From: "James A. Shigley"
Date: Mon, 28 Dec 2009 12:11:35
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject:
On Mon, 2009-12-28 at 12:11 -0600, James A. Shigley wrote:
> Alright I have a SIP phone located off premises with a very annoying
> issue.
>
>
>
> Well I say a sip phone it is actually two phones hooked to a Cisco Spa
> 2102
>
> Link: http://www.cisco.com/en/US/products/ps10026/index.html
>
Thanks Kevin.
with changing the version as u suggested: looks it is trying to connect to the
Asterik server.
and timing out. that is a network issue that I could not use the solution you
suggested.
--
I was able to resolve the issue with
sudo make -c sounds install. in Make file./
I was thinki
Alright I have a SIP phone located off premises with a very annoying
issue.
Well I say a sip phone it is actually two phones hooked to a Cisco Spa
2102
Link: http://www.cisco.com/en/US/products/ps10026/index.html
Each phone being a different line/extension.
Alright either line can ALW
Hi,
Can I use Asterik with out registration?
I want to use Asterik as a B2B.
I dont want to do regisration of UAC or UAS.
All I will do is the dail plan update with the routing information.
is it possible to do.
1) Asterk used in between 2 different SIP Proxy ( so all I know is the proxy
IP
Steve Totaro wrote:
> I would call Digium but last I knew, NFAS only worked across one card.
NFAS is implemented in libpri and Asterisk (chan_zap or chan_dahdi),
which means it has no concept of 'cards' at all. Cards are handled at
the Zaptel/DAHDI layer, and are presented as spans full of channe
On Mon, Dec 28, 2009 at 12:45 PM, Ron McCarthy wrote:
> Hi list,
>
> Ive got a server with 6 ports on it (4+2 port card) we have a DS3
> delivering all voice DS1's to us. Carrier has a trunkgroup for the first 8
> span (we only have the first 6 plugged in right now). Everything works fine
> until
Hi list,
Ive got a server with 6 ports on it (4+2 port card) we have a DS3 delivering
all voice DS1's to us. Carrier has a trunkgroup for the first 8 span (we
only have the first 6 plugged in right now). Everything works fine until we
fail the primary D channel (D's are on 24,48) the secondary the
Hi!
Any familiar with Avaya handsets? Did convert a 9650 handset to SIP. Cant
get the name just the number on the Avaya display.
Did put: SET DISPLAY_NAME_NUMBER 1 in 46xxsettings.txt
When I call from 0317998985 (Siemens DECT) to 0317998975 (Avaya 9650) i
just se 0317998985 in the Avaya displ
Hi, Bruce ,
would you remove Async from your php script,
and give it a try
regards
Dhaval
On Thu, Dec 24, 2009 at 5:45 AM, Bruce Nik wrote:
> Jarrod,
>
> Thanks for the input. Can you please include a sample of your work? It will
> really save me days of headache and tests if I can start with s
To achieve closely to a solution for your scenario (and should make you
happy too :-] ), configure it as follows:-
1. Polycom
1. Configure your polycom to enable registration
2. sip.conf
1. host=dynamic
2. Configure the permit and deny to restrict registration from a
spe
Hi C F,
I forgot to say it does not work with all telcossome telcos want
*67#X in the DIAL string and some others want the "key pad element"
method.
Giorgio Incantalupo
C F wrote:
> Huge thanks for mentioning what type of channel you are using.
>
> On Tue, Dec 22, 2009 at 5:11 AM, Gior
On 12/28/09 01:14, Joseph wrote:
>I solved the problem with calls out via FXO but internal call to to phone
>connected to FXS on AudioCodes is not working:
>
>app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause
>20 - Unknown)
> == Everyone is busy/congested at this tim
I've been using a couple of Polycom 501 phones in my home Asterisk setup. I set
up each phone in sip.conf to be static, i.e. host= so that
registration wasn't required. This has worked fine for me for a couple of years.
Now I just bought a Polycom 335. Since the 501's are now obsolete, I had to
I solved the problem with calls out via FXO but internal call to to phone
connected to FXS on AudioCodes is not working:
app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause
20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [...@inte
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