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...@camp
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-
>>
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 ?
>
>
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.
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 i
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
>
using 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
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
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
MSN:[E
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
Email: [EMAIL PROTECTED]
MSN:[EMAIL PROTECTED]
Centos 5 I experienced a complete OS crash when
calling over HFC misdn channels. Didn't really have time to investigate
and dumps as it was a live machine.
I have tried updating all relevant asterisk software, but to no avail.
Anyone have any ideas?
Kind Regards
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 Sci
b410p card on the box. When using HFC based cards I have no problem
with the recording.
Does anyone have any ideas? Is it possible to reopen this bug on the
old 1.2 code?
Garth
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:
Fax receive not successful - result (11) Unexpected message received.
The files are only 8 bytes long???
Garth
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appreciated.
Kind Regards
Garth
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{connid} ${query})
Kind Regards
Garth
Garth van Sittert wrote:
Hi All
Does anyone know how to use the MySQL cmd in Asterisk with LIKE and %
in the query?
I have:
exten => s,5,Set(query=SELECT name from contacts where tel like
%${number})
exten => s,6,MySQL(Connect connid hos
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
%${number})
exten =&g
Hi Jon
No luck - it works with the quotes and no % sign but as soon as I add
the % it doesn't work.
Garth
Jon Farmer wrote:
Try enclosing in single quotes. ie.
SELECT name from contacts where tel like '%${number}'
Jon Farmer
Telford, Shropshire, UK
- Original Me
resultid ${connid} ${query})
But there seems to be a problem with the % sign and I don't know how to
hash it out.
It works without the % sign.
Thanks
Kind Regards
Garth
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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
ither. Idefisk works 100% on the same setup.
Kind Regards
Garth
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so
Anybody has any other ideas / suggestions?
Have you tried turning on debug in logger.conf. You should be able to
see what is wrong from there.
Kind Regards
Garth
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part of the processing for the Digium cards (that's why
they are so cheap) so you cannot test the PRI without getting asterisk
up and running. The zaptel drivers will get Layer 1 and I think Layer 2
up, but Asterisk is needed for more than this.
Kind Regards
Hi Eugeniy
You should set the Asterisk configuration as if you are connecting to
your local Telco provider and set the Samsung to fit Asterisk.
Garth
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To
s. Does anyone have any
idea what could be causing this?
Thanks
Garth
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agent still show as the static extension.
Kind Regards
Garth
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BSc (Physics & Computer Science)
-
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
MSN:[EMAIL PROTECTED]
Web:www.bitco.c
know that the agent is busy on a call and try another free
agent? I have worked around this by using call-limit=1 in the sip.conf
file.
Kind Regards
Garth
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To
e the network switches etc, but Cisco? I
fail to see how a switch could bring down a device.
Kind Regards
Garth
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ideas?
Kind Regards
Garth
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BSc (Physics & Computer Science)
-
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Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
MSN:[EMAIL PROTECTED]
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d the metric units in the TIFF header. Convert them to
imperial units?
Kind Regards
Garth
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group=1
channel => 1-15,17-31,32-46,48-62
I have since changed the switchtype to QSIG and the Samsung is now set
up with QSIG.
Kind Regards
Garth
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> Everyone is raving about the all-new Yahoo! Mail
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ls are set correctly.
Echo problems in the past have been straight forward to remove with the
correct echotraning and volumes set. I am quite certain it is due to
calls going through the Samsung. Any ideas?
Kind Regards
Garth
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Hi
I had stability issues with queues on 1.2.9.1. 1.2.7.1 also has queue
issues, but it is a LOT more stable.
Garth
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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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I recently installed one of these plugged into an ATA with a dialplan that calls a predetermined number when the line is "picked up". The sound guys have only to push one button and we get audio into the pbx. Works flawlessly. We did have to boost the input level more that I think we should ha
dst forwarded number in the CDR. No SIP details
at all. How do we track this?
Billing for standard calls from the SIP user to the PSTN is fine.
Regards
Garth
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To
Thanks for the input guys. Nice to understand what's going on.
Steven, how do you know that particular diff is the fix for that particular bug (not that I doubt you, just curious how to reference the code to the bug).
Thanks again,
Garth
On 5/8/06, Steven <[EMAIL PROTECTED]>
mail app rather than wait for a release.
Can anyone give me some more info?
Garth
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xecuted by Cron (probably because the above examples are time based)?
Thanks for any input.
Garth
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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
.ELsmp.
I have both the kernel and kernel sources installed:
kernel-2.4.21-37.EL
kernel-pcmcia-cs-3.1.31-13
kernel-utils-2.4-8.37.12
kernel-source-2.4.21-37.EL
kernel-doc-2.4.21-37.EL
kernel-smp-2.4.21-37.EL
Can anyone help with this?
Garth
etimes too long or too short.
Anyone know what could be causing this? I would like to find some more
info on the ISDN layers and protocols, but I haven't found a good source
on this.
Thanks
Garth
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Hi Jean-Marc
I tried removing the call-limit setting. It still doesn't work. I am
using a SNOM 360 to monitor the line status'.
Do I still need to activate the busy lamp on the IP10S' or is this only
if you want the IP10S' to monitor the hints?
Garth
Jean-Marc Sals
,12)
exten => _0X.,1,Macro(dial-external,${EXTEN:1}) ;
External calls
How does the hints work? Do you know anything about the flow?
Thanks
Garth
Mike Pollitt wrote:
Hi Garth --
Other users have also reported problems with the status being set by the
SwissVoice phones - oh wait a minute..
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
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
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'.
Th
Hi
I have a test setup of a sirrix card installed in NT mode connected to a
PBX. I keep getting the following error:
D-Channel receive message aborted, discarding frame (RSTAD=0x1c)
What does this mean? What could be causing it?
Garth
I have that set up, but I cannot get some of the phones to change the
hint State. The SNOM phone show State:InUse, but Swissvoice phones show
State:Idle even when on a call.
I use 'show hints' to see this.
Kind Regards
Garth
Colin Anderson wrote:
Breeze to set up, too. To m
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 wor
The silence suppression is a client setting. Asterisk does not have
silence suppression as far as I know.
Garth
Dan Elder wrote:
Hi all, I'm getting some noise gate like effects on our sip lines & I think I need
to disable silence supression, I'm searching docs & not find
dialog-info+xml
10.0.0.1011 3c267009b71 12 Idle
dialog-info+xml
Regards
Garth
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Could we possibly see your settings to get this right? I am trying to
get it working at the moment.
I can see the phone buttons have subscribed to asterisk, but they just
don't light up. We are using 4.1 firmware and are upgrading to 5.3 to
see if it helps.
Regards
Garth
Darrell
cture around this,
that would be great.
Thanks
Garth
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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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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
AN is used purely for
VoIP traffic.
Garth
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Hi Abdul
You will need to download and install the Intel API which is then used
to compile the patched G723 codec.
Hope this helps.
Kind Regards
Garth
Abdul Lateef wrote:
Hi All,
I have one Carrier which is supporting only G.723.1,
how i can put in my extentions.conf to send calls to
Hi Alex
I tried your exact example below and still the same thing. I am getting
403 Denied after I see the Pickup cmd in the CLI. If you do a show
channel SIP/XXX when the phone is ringing, do you get a value for
Extension:??
Kind Regards
Garth
Alex Barnes wrote:
-Original Message
= 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:
B
N:1})
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 Picku
Hi Dan
Have a look at setting up queues.
Kind Regards
Garth
Dan Journo wrote:
I thought someone was going to say that.
Does anyone know a way to do the following:-
1) Answer incoming call
2) Begin dialing an extension
3) While extension is ringing play a welcome message to the caller
4
= *8
Kind Regards
Garth
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BSc (Physics & Computer Science)
-
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
Phone: 08600 BITCO
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? 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
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
Hi James
I would consider Hylaxfax if you are going to do purely faxing.
Garth
James Harper wrote:
I am trying to set up a linux based faxing solution for a client, and
have found that the modem they have (ancient dataplex external unit)
just isn't up to the job. It talks to some remot
There is a good utility called iaxping to test IAX latency.
Kind Regards
Garth
BitCo Data Communications
http://www.bitco.co.za
Cosmin Prund wrote:
As the subject line says: Is PING a good indicator of network latency? If
not, how can I measure latency?
Thanks,
Cosmin Prund
Hi Cosmin
You should be able to get about 12 simultaneous calls on a 128k line and
about 28 on a 256k line according to asteriskguru's bandwidth calculator
http://www.asteriskguru.com/tools/bandwidth_calculator.php.
Kind Regards
Garth
BitCo Data Communications
http://www.bitco.co.za
C
Don't think there is anything wrong with your setup. We get the same
thing... Maybe they're down, but I would like a third opinion...
G
Michaël Gaudette wrote:
Hi,
I`ve just tried the Voipjet 0.25$ test account, following everything the web
site told me to do (see below).
When I dial a loc
Not an answer to your questions, but just in case you don't know there
is a lot of info on the wiki:
http://www.voip-info.org/wiki/view/AreskiCC+CallingCard+Application
We use Areskicc here, and it works great. However we do not use sip/iax
friends, perhaps both of your problems lie there?
B
If you haven't seen it already, this will be a lot of help to you.
http://www.voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Application+The+idiots+guideV2
You should now be on step 12. :)
G
Omar McKenzie wrote:
Hi
I have gone thru the steps of installing AreskiCC, I wo
This one drove me crazy for a while too. I found out that some
companies don't exactly play fair and don't pass answer supervision on a
call until you are actually speaking with a live person. The person I
spoke to about this wasn't sure if that was even legal, but he said it
happens quite a
I've also only heard of the Clipcomm
Along the same lines...
Why doesn't anyone make a wireless ATA? Am I the only one with a need
for such a thing? By the time I plug in a wireless bridge, an ata and a
cordless phone, I need a five outlet powerstrip and shoebox to hide all
the components.
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 chang
We use the Areskicc calling card system as an authentication system. It
does everything you are asking and can generate great reports and
graphs. I like it very much.
That being said, areskicc is tough to get going, but there is plenty of
info here:
http://www.voip-info.org/tiki-index.php?p
Jet is following the rules, so it's not their problem, but at
the same time I would think they want to be as functional as possible.
Thanks for the help everyone,
G
Garth Summey wrote:
Hi List,
I'm wondering if someone who uses VoipJet as their termination service
would do
Hi List,
I'm wondering if someone who uses VoipJet as their termination service
would do me a favor.
If I call the American Airlines reservation number (1-800-433-7300), the
call gets connected, but after 30 seconds asterisk drops the call
responding that no one answered.
I'm using areskic
Hi,
Can anyone give me any information at all to get app_intercept working?
I've found these pages, but there is just not enough for me to get it going.
http://www.pbxfreeware.org/archives/2005/06/new_download_--.html
and
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002692
Thank
Hi,
Has anyone come up with a clever way of indicating DND is activated?
I've thought of stutter dial tone and using the mwi, but have no idea
how to implement these. I'm using Budgetones. My concern is that users
will activate the DND, then forget about it not realizing that they are
not recei
Has anyone come up with a clever way of indicating DND is activated?
I've thought of stutter dial tone and using the mwi, but have no idea
how to implement these. I'm using Budgetones. My concern is that users
will activate the DND, then forget about it not realizing that they are
not receivi
Let's start basic, we know that both PCs that are running the soft
phones can see the aah server, but can both PCs see each other? Can
they ping each other? (ie, they are not across a NAT router or
something like that?)
G
Mark Anthony C. Delfin wrote:
hi list,
I'm running a newly installe
Ok, it now seems to be a firewall issue. When I turn the FW
completely off, the calls work just fine. I only had one opening in the FW
(5060:udp). Is there another port/protocol that should also be open?
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I have my Asterisk server all setup. But have an odd
problem and hope someone here can help.
I have a Polycom IP 300, a Grandstream GXP-2000, and an
installation of X-Lite. They can each call each other just fine
(extension-to-extension). I can also dial-in from the outside (via Broa
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