You could try using a Intel Little Falls motherboard for that if you
are not going to be recording calls.
It comes with the processor on board.
Garth
van Sittert
BSC
(Physics Comp Sci)
Technical
Director
Tel: 08600 24826
supp...@bitco.co.za
vese...@campbell
Do you not have to answer the channel before the MOH can happen?
Joseph wrote:
No, the same happens when I use SIP phone, no music on internal call.
--
Joseph
On 11/03/09 13:10, Danny Nicholas wrote:
I suspect that IAX is the culprit...
-Original Message-
From:
We have the 870 working great in our test environment so far.
Garth van Sittert
BSC (Physics Comp Sci)
Technical Director
BitCo
08600 24826
www.bitco.co.za
--[ UxBoD ]-- wrote:
Anybody tried one with Asterisk yet ? Views ?
Best Regards
Depends on what you want to do and what your server platform is like.
Garth van Sittert
Technical Director
BitCo
08600 24826
www.bitco.co.za
abdelkader wrote:
Hello,
What is the maximum number of simultaneous calls supported by asterisk.
thks
I would think that VoIP over VPN is a bad idea as UDP packets need to be
in realtime not corrected by the TCP of the VPN.
Garth van Sittert
Technical Director
BitCo
08600 24826
www.bitco.co.za
Aurimas Skirgaila wrote:
Despite the VPN overhead, running VOIP through VPN is good idea
because
As a quick workaround you could use a goto to send to an invalid extension.
Goto(nowhere,1)
Garth van Sittert
Technical Director
BitCo
08600 24826
www.bitco.co.za
Chris Maciejewski wrote:
Hi,
I am trying to send 404 Not found reply, without any luck with the
following:
exten = 555,1
asterisk 1.4.20 and misdn 1.1.8. This never used to happen
on asterisk 1.2. I have also tried the latest chan_misdn on 1.4 with
the exact same results.
I have found no other useful documentation on this.
Kind Regards
Garth
--
Garth van Sittert
Technical Director
BitCo
08600 24826
Hi Satish
You would want to investigate Local channels on Asterisk for this.
Garth
Garth van Sittert
BSc (Physics Computer Science)
-
Main: 08600 BITCO
Phone: +27 (0)11 875 6900
Fax:+27 (0)11 875 6901
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MSN:[EMAIL
Where would you suggest all the logic goes Brian?
Garth
Garth van Sittert
BSc (Physics Computer Science)
-
Main: 08600 BITCO
Phone: +27 (0)11 875 6900
Fax:+27 (0)11 875 6901
Mobile: +27 (0)83 791 6662
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MSN:[EMAIL PROTECTED]
Web
Hi All
Has anyone experienced a crash specific to asterisk 1.2 and Centos 5
when using the misdn hfcpci module that comes with zaptel?
I have an asterisk pack based on asterisk-1.2.17 that I have been using
on dozens of machines that are rock solid and stable. Today when I
tried moving it to
Hi Remco
I have used the IP600 v3 with SIP support on Asterisk... apparently I
was the 1st person globally to run it at a site. The 1st firmware was a
bit buggy at times, but seems to be much better on the later versions.
Kind Regards
Garth
Garth van Sittert
BSc (Physics Computer Science
Hi All
According to http://bugs.digium.com/view.php?id=6457 this has been
resolved since 04-11-2006 and I have seen mentioned since 1.2.7. I have
tried using mixmonitor on asterisk 1.2.13 and 1.2.17 with the exact same
results. The WAV files are recorded but are cut short. I am using a
Hi All
Has anyone managed to get Asterisk 1.2 faxes working reliably with
spandsp 0.0.3? I am running Asterisk 1.2.17 and spandsp 0.0.3pre28 with
a Digium b410p card. Everything compiled smoothly but only about 70% of
faxes come through ok. Debugging shows nothing more than: app_rxfax.c:
Hi All
I have made the move to the Digium b410p BRI card and keep getting the
following intermittent kernel oops listed here in dmesg. I can make and
receive calls to sip devices fine. It only seems to happen when I call
from the Telco and hangup from the Telco before the audio paths on the
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in
the query?
I have:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten = s,7,MySQL(Query resultid
: Garth van Sittert [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, 4 December, 2006 12:38:07 PM
Subject: [asterisk-users] MySQL cmd % pattern matching
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE
I have it working as your example, Doug, but unfortunately I need the
like phrase as the numbers all contain spaces or sometimes even brackets.
Garth
Doug Lytle wrote:
Garth van Sittert wrote:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL
Garth
Garth van Sittert wrote:
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and %
in the query?
I have:
exten = s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten = s,6,MySQL(Connect connid hostname username password dbname)
exten
Hi Matt
Check that your volumes are not too high.
Kind Regards
Garth
Garth van Sittert
BSc (Physics Computer Science)
-
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
MSN:[EMAIL PROTECTED]
Web:www.bitco.co.za
Matt wrote:
Hello,
I had had
Hi All
Has anyone used Cubix / Firefly successfully with Asterisk? When
someone calls a Cubix softphone, Cubix never seems to answer the call
correctly. The other person just hears ringing even though it has been
answered. I am using IAX as the SIP support doesn't seem to 100%
either.
Joseph wrote:
On Mon, 2006-10-09 at 01:45 -0300, Hermann Wecke wrote:
On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote:
I have bind-address = 127.0.0.1 in my.cnf
the cdr was working find with asterisk 1.0.1 just after upgrade
something is not connecting.
I don't know if
Hi All
I have a site with 50 Budgetone 102's and about 5 snom phones.
At random intervals during the day about 20 or 30 of the Budgetones lose
their connection to the network all at the same time. It happens about
once a day. The Snom phones are fine and never get disconnected. I
can't
Eugeniy Khvastunov wrote:
What signaling method i should use for connecting Asterisk(Gentoo,
Tormenta 2) + Samsung OffServ 500 by PRI flow?
What parametrs in zaptel.conf, zapata.conf?
---
Какой метод сигнализации нужно использовать при подключении
Asterisk(Gentoo, Tormenta 2) + Samsung OffServ
Andy Chung (Power-All) wrote:
Hi all,
I have connected a T1IDA-P to the Digium TE405P. Checked with the
Telco, and confirmed the T1 is up and connected. However, I have no
idea how to test the T1 is really work, because the Asterisk server
not yet be configure. Anyone has the method on how
the
agent still show as the static extension.
Kind Regards
Garth
--
Garth van Sittert
BSc (Physics Computer Science)
-
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
MSN:[EMAIL PROTECTED]
Web:www.bitco.co.za
Hi All
I have set up queues in Asterisk to use dynamic members using the
addqueuemember application. The application works well, but I have one
issue with it; Asterisk keeps passing a waiting caller to a member even
if the member is on a call from the queue already. Surely Asterisk
should
Garth
--
Garth van Sittert
BSc (Physics Computer Science)
-
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
MSN:[EMAIL PROTECTED]
Web:www.bitco.co.za
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Hi All
Just wondering if anyone knows of a solution to the squashed tiff
problem with spandsp (or rather Windows Image Viewer) other than
converting to a PDF. I find the PDF image quality is not nearly as good
as the original TIFF. Apparently the Windows Image Viewer doesn't
understand the
Hi All
I have an Asterisk box with 2 PRI's going into a Samsung DCS 500. After
a day or so of usage, the PRI channels are all used up and users cannot
make or receive calls over the PRI's. I get many of the following
messages in the logs:
Ring requested on channel 0/1 already in use on
Hi All
Has anyone had experience with rxfax on asterisk 1.2.x with a sirrix
quad BRI card?
Does it work with the Sirrix cards?
Garth
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Hi All
Does anyone know how to track calls that come in from the PSTN, go to
the user SIP phone and then get forwarded from the user SIP phone back
out to PSTN (typically the users mobile number)?
It is a headache trying to bill the users who do this. There is only
the src PSTN number and
Hi
When trying to compile zaptel-1.2.5 I am getting the following errors:
/usr/include/linux/modversions.h:1:2: #error Modules should never use
kernel-headers system headers,
/usr/include/linux/modversions.h:2:2: #error but rather headers from an
appropriate kernel-source package.
Upgraded kernel and sources. Seemed to sort it out.
Garth
Garth van Sittert wrote:
Hi
When trying to compile zaptel-1.2.5 I am getting the following errors:
/usr/include/linux/modversions.h:1:2: #error Modules should never use
kernel-headers system headers,
/usr/include/linux/modversions.h
Hi All
I have an Asterisk box using a Sirrix card sitting between our PSTN and
an ISDN pbx. Calls from the PSTN are forwarded to the PBX ok.
Calls from the PBX are having problems - the digits being passed are
being garbled. The numbers from the PBX are totally incorrect and
sometimes too
-
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On
Behalf Of Garth van Sittert
Sent: 24 February 2006 07:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk hints
Hi Mike
I have build 18 on the IP10's and I
? Also check that you're
running the latest firmware on the Swissvoice (I think it's build 18), since
I know they've been tinkering with the presence features recently.
Cheers,
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garth van
Sittert
Sent
Hi All
Does anyone know how the hints in asterisk works? How does a SIP phone
interact with the hints? I am having a problem with certain phone
models that do not set the hints correctly when I list the hints with a
'show hints'.
Thanks
Garth
Have you checked the permissions on the file? Is it executable?
Garth
Dirgan Putra wrote:
hi All
need help, iam installing areskiCC and have a problem
after that create extension for calling card and after dial
exten = 17000,3,DeadAgi,a2billing.php
i see messages : a2billing.php no
I am using Swissvoice IP10S phones.
Garth
Mike Pollitt wrote:
Garth --
What kind of phones are you using?
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garth van
Sittert
Sent: Wednesday, 22 February 2006 7:29 PM
To: Asterisk Users Mailing
Yes, you need to remove the 'System' part.
You should only have:
exten =
s,n,MixMonitor(${CALLDIR}${CALLFILENAME}.wav||touch/tmp/test${UNIQUEID})
Garth
Alex Barnes wrote:
Has anyone had any success using the MixMonitor() plus command as
nothing I have tried works.
I am using 1.2.1 I
extensions to cover up to 54 extensions.
Only the Polycom 601 comes close.
Regards
Rob
On 2/8/06, *Garth van Sittert* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi All
Has anyone come across a handset that can somehow replace
FOP? Some
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BSc (Physics Computer Science)
-
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Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
Web:www.bitco.co.za
Hi All
Has anyone managed to get the hint priority with Swissvoice IP10S phones
working?
I have 2 phones: a Snom 360, setup as the reception phone on extension
11, and a Swissvoice IP10S on extension 12.
When calling each other (tested both ways) I can only ever see the Snom
360 in the Active
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Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
Hi All
Does anyone have any ideas around what processing power is needed when
bridging PRI channels and recording?
I am not sure how the bridging takes place with and without recording?
I basically have a situation like this:
Telko Asterisk - Legacy PBX
Where the lines are
Hi All
Is there any special configuration needed to send and receive faxes on
an ATA device?
I am using G711.a with a Grandstream Handytone 486. I can send faxes
from a fax machine on the ATA, but receiving doesn't work. I get the
fax signal, but it just doesn't continue. The LAN is used
I am using alaw and I have already enabled the pass through. Does alaw
and ulaw work?
I can fax out, but not receive faxes.
Garth
Johann Steinwendtner wrote:
Enable pass thru fax mode on the HT486, or enable ulaw in your SIP
config.
Hans
Garth van Sittert schrieb:
Hi All
Is there any
Hi All
Has anyone come across a handset that can somehow replace FOP? Some
users don't like FOP unless it is on a dedicated PC.
Thanks
Garth
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Email: [EMAIL PROTECTED]
Phone
]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=alaw
context=internal
[200]
callerid=Reception 200
type=friend
host=dynamic
dtmfmode=rfc2833
username=200
secret=pbx
Kind Regards
Garth
Garth van Sittert wrote:
Show Features produces:
Builtin
-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Garth van Sittert
Sent: 02 February 2006 16:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Directed Call Pickup
Hi All
I am having problems with Directed Call Pickup
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Garth van Sittert
BSc (Physics Computer Science)
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Hi All
I am having a problem setting the outbound callerid number on a PRI E1
in South Africa. The outbound number keeps on appearing as the main PRI
number. How does it work between Asterisk and the Telko? More
importantly how do I get it working?
Kind Regards
Garth
--
Garth van
? Do I need to send the complete number, 3 digit
area code + 4 digit extension to the Telko? Does the zapata.conf add
the prefix? How can I check what callerid number is being passed to the
Telko?
Garth
Steve Underwood wrote:
Garth van Sittert wrote:
Hi All
I am having a problem
Kind Regards
Garth
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BSc (Physics Computer Science)
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mailing list
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Garth van Sittert
BSc (Physics Computer Science)
-
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Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
Web:www.bitco.co.za
})
When I dial 812, in the CLI I can see:
Executing Pickup(SIP/29-707f, 12) in new stack
Any thoughts?
Kind Regards
Garth
Bob Goddard wrote:
On Thursday 02 Feb 2006 16:46, Garth van Sittert wrote:
Hi All
I am having problems with Directed Call Pickup in Asterisk 1.2.1
If extension
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Mobile
Hi All
Does anyone know if multiple Digium cards on a single machine will be a
problem.
Machine specs: Dual Zeon 3.0GHz on Intel server board.
Cards: TE411P, TDM400P, TDM400P
I will turn off all unnecessary PCI devices; USB, parallel, serial, etc...
Thanks
I have had an idea of using two identical servers: Server A with IP
x.x.x.a and server B with IP x.x.x.b. Server A is live while server B
sits in the background monitoring server A. Server B rsync's asterisk
config files daily with server A.
In the event of server A going down, server B
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