We have set the do not disturb feature on our polycom phone such that
incoming calls will not be rejected and sent to 'Busy' status.
The user can still toggle dnd on/off, but incoming calls will still get in,
indicated by the blinking light and the screen status.
We were able to do that by setting
Which versions of Zaptel have you tried to build?
PaulH
Jay Ray wrote:
> Any ideas, please they are highly appreciated
>
> --- On *Mon, 8/4/08, Jay Ray /<[EMAIL PROTECTED]>/* wrote:
>
> From: Jay Ray <[EMAIL PROTECTED]>
> Subject: [asterisk-users] FC2 and Zaptel
> To: asterisk-u
Any ideas, please they are highly appreciated
--- On Mon, 8/4/08, Jay Ray <[EMAIL PROTECTED]> wrote:
From: Jay Ray <[EMAIL PROTECTED]>
Subject: [asterisk-users] FC2 and Zaptel
To: asterisk-users@lists.digium.com
Date: Monday, August 4, 2008, 12:02 AM
Hi,
I am using an older Fedora - FC2 and
Dan Peters wrote:
>
> We have had Asterisk up and running for a while now and it works very
> well. Recently we tried to integrate a Linsys SPA962 with the
> associated SPA932 console. We can get the BLF lights to blink when a
> phone is ringing and we can get the BLF lights to go solid when t
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I will
You might want to look at ForkCDR() and ResetCDR() to see if they meet
your needs. Or if you really need the accounting that cisco provides,
bounce your calls through that cisco box first.
Honestly, if your hardware is crashing a lot, I'd worry more about the
root cause of that than the fact that
There is no BLF state for 'off hook' (at least in Asterisk) - so what
you have here is what you are going to get.
regards,
PaulH
Dan Peters wrote:
>
> We have had Asterisk up and running for a while now and it works very
> well. Recently we tried to integrate a Linsys SPA962 with the
> ass
THis machine is well OVER POWERED for the task you define.
The greater issue will be the media bandwidth.
I have operated with 220 simultaneous g711 participants
in 18 ~ 20 different conferences.
At 02:45 PM 8/11/2008, you wrote:
Hi list
I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Alex Balashov wrote:
> Gerard A. Matthew wrote:
>
>> I'm trying understand, if it's possible to run an agi script to obtain a
>> user's account balance and from there asterisk would be able communicate
>> that value back to a sip phone. Is that phon
Hey Ronald what about a http://www.taa.com/products-vdex-40.html - it's
got 4 fxo ports built in and uses the druid embedded os so easy enough
to program by anyone using nothing more than a browser.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On B
We have had Asterisk up and running for a while now and it works very
well. Recently we tried to integrate a Linsys SPA962 with the
associated SPA932 console. We can get the BLF lights to blink when a
phone is ringing and we can get the BLF lights to go solid when that
call is picked up. My ques
Depending upon your needs you may find it cheaper to run it on a thin
client device. I find HP T5700 series for around $100-150 on Ebay.
These genrally have 1 GHz processors and 256 MB RAM. Not dual NICs, but
you could add that if necessary. There's an expension chasis for $30
from HP that allows f
> Hm. $300 in the US and the UK disty is selling them for just short of
> £240, so they can go stuff themselves, low-power or not. (I buy 1GHz
> systems with 1GB of RAM, running at 15W for half that. No drive though)
Gordon,
If you don't mind my asking: What do you get for $150.00 ?
___
>
> Hi,
> I got some errors about not being able to create subdir [already
> existing] on a 'make update' in my zaptel 1.4.
> I removed the directory and did a new svn co of zaptel 1.4
> [ svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel ]
>
> now I get:
>
> /usr/bin/install -c
Hi,
I got some errors about not being able to create subdir [already
existing] on a 'make update' in my zaptel 1.4.
I removed the directory and did a new svn co of zaptel 1.4
[ svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel ]
now I get:
/usr/bin/install -c -D -m 644 tonezone
Michiel van Baak wrote:
> On 01:30, Wed 13 Aug 08, Ronald Wiplinger wrote:
>> I had installed in the office an Asterisk server, but the company is
>> gone and I could keep the server.
>>
>> However, for my family with three members and two phone lines this
>> server is overkill. I am looking for a
When you provide TWTelecom with an LOA (Letter of Authorization) you want to
be sure it's absolutely accurate and signed by someone with authority over
the account that the numbers are coming from. That's why it's desirable to
have a CSR (Customer Service Record) to make sure that your LNP request
A CSR is nothing more than a listing of the numbers by your current
provider on some sort of letterhead to indicate you actually are the
subscriber who these numbers belong to (ie, you pay the bill for
them).
Is it necessary for the actual LNP process - no, not technically but
companies require it
What is the deal with "CSR's"?
TWTelecom is telling me that I can't port a number to their service
without a Customer Service Record. Apparently this is easy with
Verizon, and not so easy with some other companies.
Basically I'm at a brick wall with a couple of ports because TWTelecom
is tell
On Tue, 12 Aug 2008, mail-lists wrote:
I can't see why not. You should easily have enough power for asterisk.
You can probably also run it as your firewall in a home environment
thanks to the dual RJ45's
I don't know whether or not you can use the built in RJ11 to interface
with your POTS line
Tried from branch and had the same problem.
gcc -o cmenuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a
-lncurses
gcc -g -c -D_GNU_SOURCE -Wall -DXTHREADS -D_REENTRANT -DXUSE_MTSAFE_API
-I/usr/include/gtk-2.0 -I/usr/lib/gtk-2.0/include -I/usr/X11R6/include
-I/usr/include/atk-
I can't see why not. You should easily have enough power for asterisk.
You can probably also run it as your firewall in a home environment
thanks to the dual RJ45's
I don't know whether or not you can use the built in RJ11 to interface
with your POTS line though - maybe someone else could speak
Ronald Wiplinger wrote:
> I had installed in the office an Asterisk server, but the company is
> gone and I could keep the server.
>
> However, for my family with three members and two phone lines this
> server is overkill. I am looking for a compact solution, which is more
> suitable for me.
>
> I
Hi everyone,
I really need your help. Just now my queue member status are not being
refreshed correctly, when a call is answered the status is set as UNKNOWN
instead of IN USE. After the call is hanged up the state persists as
UNKNOWN.
I have tried using module reload app_queue.so but the only r
I am successfully using this in my dialplan for a number of Sipuras
(modified to fit your dialplan):
exten => 700,1,SIPAddHeader(Alert-Info: info=)
Not saying there's no other way to get it accomplished, but this is known to
work (1.4.21.2).
___
-
Hi,
I'd like to install Asterisk at home. But don't want to use a full blown PC
to host it. I was thinking of using fitPC www.fit-pc.com to do all the
Asterisk work, interfacing with the local Bell Canada line, and using a SIP
VoIP line as well.
What do you experts think of it?
Thanks,
Steve Totaro schrieb:
> Check this link out. http://www.rowetel.com/ucasterisk/ip04.html
I'd stay away from Blackfin unless you are prepared to run
into various issues when compiling stuff.
Grüße,
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma Gmb
Only on -addons, asterisk compiled fine from trunk.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher
Sent: Tuesday, August 12, 2008 8:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unabl
Check this link out. http://www.rowetel.com/ucasterisk/ip04.html
Thanks,
Steve Totaro
On Tue, Aug 12, 2008 at 1:30 PM, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> I had installed in the office an Asterisk server, but the company is
> gone and I could keep the server.
>
> However, for my family
On 01:30, Wed 13 Aug 08, Ronald Wiplinger wrote:
> I had installed in the office an Asterisk server, but the company is
> gone and I could keep the server.
>
> However, for my family with three members and two phone lines this
> server is overkill. I am looking for a compact solution, which is mor
I had installed in the office an Asterisk server, but the company is
gone and I could keep the server.
However, for my family with three members and two phone lines this
server is overkill. I am looking for a compact solution, which is more
suitable for me.
I want a small & silent box, which can
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I will
On Tue, 2008-08-12 at 16:39 +0200, Klaus Darilion wrote:
> Hi!
>
> I wonder how Asterisk measures the call duration. The CDR files have a
> accuracy of seconds. Thus, what happens if the call duration is 0.3
> seconds. What will Asterisk report? 0 seconds? 1 second?
>
> What logic will be used
On 08/12/08 14:28, John Fawcett wrote:
>-BEGIN PGP SIGNED MESSAGE-
>Joseph wrote:
>| My phone rings "once" and stops before playing message; how to stop
>this behavior.
>|
>| Could it have something to do with this error:
>|
>| channel_find_locked: Avoided initial deadlock for '0x81c04d0'
Hi!
Is there a way in Asterisk to get intermediate accounting records? E.g
Cisco gateways send start, stop and and regular interval intermediate
accounting records.
For example if there is a call and Asterisk (or the hardware) crashes
during the call we do not have a CDR for this call, not eve
Hi All,
I'm trying to move some POTS phones from Zap to sipura. I've searched and read
a several articles which suggest that something like this should work:
exten => 600,1,Dial(SIP/cordless)
exten => 600,n,Hangup()
exten => 700,1,Set(ALERT_INFO=Bellcore-r2)
exten => 700,n,Dial(SIP/cordless)
e
Hi!
I wonder how Asterisk measures the call duration. The CDR files have a
accuracy of seconds. Thus, what happens if the call duration is 0.3
seconds. What will Asterisk report? 0 seconds? 1 second?
What logic will be used by Asterisk: floor? ceil? round?
thanks
klaus
___
Did you set up OpenSER to properly statefully relay REGISTER and its replies?
On Tue, August 12, 2008 9:36 am, Steve Totaro wrote:
> On Tue, Aug 12, 2008 at 9:29 AM, michel freiha <[EMAIL PROTECTED]> wrote:
>> Dear All,
>> I have the below issue:
>>
>> I created an extension(5678) under extension
On Tuesday 12 August 2008 07:28:05 Jonn R Taylor wrote:
> I was able to compile from asterisk-addons-beta4 but not from trunk. I am
> running CentOS 4.6.
If you're compiling against 1.6.0, then you should be getting the 1.6.0 branch
of -addons, not trunk.
http://svn.digium.com/svn/asterisk-addons
On Tue, Aug 12, 2008 at 9:29 AM, michel freiha <[EMAIL PROTECTED]> wrote:
> Dear All,
> I have the below issue:
>
> I created an extension(5678) under extensions_custom.conf to record voice
> messages and playback the voice as you can see below:
> [custom-recordme]
>
> exten => 5678,1,Wait(2)
> ext
Dear All,
I have the below issue:
I created an extension(5678) under extensions_custom.conf to record voice
messages and playback the voice as you can see below:
[custom-recordme]
exten => 5678,1,Wait(2)
exten => 5678,2,Record(/tmp/asterisk-recording:g729)
exten => 5678,3,Wait(2)
exten => 5678,4,
Russell Bryant wrote:
> On Aug 11, 2008, at 12:04 PM, SIP wrote:
>
>
>> SIP wrote:
>>
>>> When calling from our SIP proxy through Asterisk to the PSTN
>>> provider,
>>> we support reINVITES which tend to work seamlessly.
>>>
>>> However, when creating a call file which essentially connect
Hi everybody,
I have linksys phone at my location,
i am using asterisk version 1.4.19,
I have a issue regarding dtmf mode, i have set the Asterisk DTMFmode to Auto
in order to eliminate Asterisk effect on the DTMF transmission. Both Inband
and AVT from Linksys worked with PSTN IVR.
But, We have the
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Hash: SHA1
Joseph wrote:
| My phone rings "once" and stops before playing message; how to stop
this behavior.
|
| Could it have something to do with this error:
|
| channel_find_locked: Avoided initial deadlock for '0x81c04d0', 9
retries!
|
|
| Here is the dial
I was able to compile from asterisk-addons-beta4 but not from trunk. I am
running CentOS 4.6.
Jonn
CC="gcc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect
CONFIGURE_SILENT="--silent"
make[1]: Entering directory `/usr/src/asterisk-addons-trunk/menuselect'
gcc -g -c -D_GNU_SOURCE -W
Russell Bryant <[EMAIL PROTECTED]> writes:
> That is actually more of a debug message, and is not necessarily an
> indication of a problem.
It has in >95% of cases (and believe me, we have hit it a lot)
indicated that some SIP conversation has deadlocked (it seems that
it's often a registration
Hi All
I am trying to implement Inbound call Barging using ChanSpy &
ExtenSpy.Actual requirement is i want to spy or barge Inbound calls received
by specfic group or queue.
For that i use Set(SPYGROUP) concept but as its inbound calls it playing
Inbound channel(whcih is 1st 3 digit of inbo
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