Is there anyone is Tasmania (esp Hobart) doing Asterisk work?
PaulH
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Have you looked at AMS?
http://www.intuitivecreations.com/contributions/AMS/
PaulH
David fire wrote:
my budget is 0 rigth now
and i want opensource because i want to customice it... can program in
Java
PHP
C
C++
.NET (i am not proud of it)
so i want to customice it for my clients
Benny Amorsen wrote:
Elliot Murdock [EMAIL PROTECTED] writes:
I am wondering if a queue feature that blocks call-waiting should be
submitted.
Doesn't Queue() already disregard busy phones? I must admit that we
run with callwaiting turned off, so it isn't something I get to test
Jerry Geis wrote:
I upgraded from 1.2 to 1.4.18
After upgrading I get half channel audio on SOME phones.
I have Cisco 7960 that works, I have a wireless polycom 8002 phone that
works.
However, my polycom 501's are getting half channel audio on EXTERNAL calls.
Internal calls are OK.
I
You might have to look at writing a forward macro on the server that
would be dialed by the DND button - that also changed the device status
to busy(via the devstate app?).
My guess is that it would be less than 10 lines of dialplan code, but
maybe 1.6 only
PaulH
cfh wrote:
hi,
I have
.
Is it configurable via asterisk or is it just the re-register settings
on the SNOM phone?
Thanks again Paul.
Veselin K
On Tue, Nov 18, 2008 at 10:21:14AM +1100, Paul Hales wrote:
The process for upgrading would greatly depend on how Asterisk was
installed in the first place.
If Asterisk
for the reply.
Could you please tell me what is the process called so I can
research it further.
Thank you.
Veselin K
On Mon, Nov 17, 2008 at 10:47:47AM +1100, Paul Hales wrote:
This process has been greatly improved in the latest versions of
Asterisk - might be time to upgrade.
PaulH
This process has been greatly improved in the latest versions of
Asterisk - might be time to upgrade.
PaulH
[EMAIL PROTECTED] wrote:
Hello,
I'm running an Asterisk 1.4.14 on a linux machine.
Serving SIP Snom users.
I've noticed that each time Asterisk is restarted, for the first 5-10
Rob Hillis wrote:
Louis-David Mitterrand wrote:
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Your monitoring app is not sending valid IAX2 packets to the server. If
it was sending a true IAX2 POKE, it would be a valid packet and wouldn't
generate this
The linksys phones annoy me because they cannot implement southern
hemisphere DST properly. Grr.
(yes, you can do it with a hack - but why can't the phones just work?)
PaulH
Steve Anness wrote:
Good Day,
I have been tasked with fixing the time on our asterisk server. I am
having a hard
It should ignore the keywords, but you will get lots of errors in the CLI.
My guess is that if you put it all in a DB (and use realtime) you can
probably do whatever you want.
PaulH
Rob Hillis wrote:
Hi guys,
I'm about to embark on a small (undoubtedly to get much larger) project
to
Separate cabling is also useful if the phone system is being deployed by
a separate company - it avoids the 'your computer network is generating
rubbish traffic' arguments. (been there before, sadly)
PaulH
Andrew Latham wrote:
Alex
I see a fair bit of separate physical networks because of
I know a business that tried those phones, and removed them.
They found that Polycom phones were 'more' perfect.
PaulH
Bruno Castelo Branco wrote:
hi
O use around 500 atcom530, they are work perfect
www.atcom.com.cn
Gordon Henderson wrote:
On Wed, 29 Oct 2008, Kev Szaszvari wrote:
There are some decent third part central management systems for the
Linksys phones (the company I work for write one of them) - the same
piece of software will also manage Polycom and Snom phones (As well as
Aastra phones and a few others)
With regards to programmable buttons, the snoms are
I know of someone who was involved in a software project like this -
scanning paper documents and importing files into a massive searchable
database for a large legal company.
Many of the documents were more than 1000 pages long.
The amount of money spend on the project was stunning.
PaulH
Just to check - have you got the right modules plugged into the right
sort of lines?
Also - some analog phone interfaces are NOT standard. :(
But the line modules have to be (to work with standard phone lines, of
course)
PaulH
Daniel Johnson wrote:
Hi,
I am trying to interface our old
I would think
1.6 = Windows Vista
:)
PaulH
Steve Totaro wrote:
1.6 = Windows Vista :-P
On Tue, Sep 23, 2008 at 3:05 PM, Joseph [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I need to upgrade my Asterisk, currently I'm using 1.2.27 from
Gentoo portage but I think this
Not at the moment but, just in case, I will try to use tear gas, first ;-)
I have found that with the right diet, teargas is not necessary.
PaulH
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AstriCon 2008 -
I have found that with the right diet, teargas is not necessary.
That interesting to know.
Maybe we should open a new thread on that and let everyone contribute ;-)
I still think it's a valid idea - with the right lunch, you could
guarantee that the office was empty (except for
Gordon Henderson wrote:
Arrange the building to have a master lights off switch. Push it, then
wait for the screams. This was used in a place I worked some years back.
This was exactly what was used in a large company I used to work for.
I have a strong memory of being in the toilets
When does the call actually hang up?
The final line of this (started music on hold, class 'default', on
Zap/6-1) acutally looks successful.
If the next line after this reads something like 'music on hold stopped'
you have a MOH problem.
PaulH
Daniel Johnson wrote:
Here is the rest of the
Femi wrote:
Great!
How stable was the Digium appliance?
Solid
Did it ever lock up or reboot without explanation?
No
Did you have any issues with phones locking up or rebooting?
No
I need to get a feel for how stable the appliances are so I can make a
decision on which to go with
I have worked with both hand-built appliances running Astlinux and the
Digium appliance.
PaulH
Femi wrote:
Thanks for all the pointers
What I need is real world experience though
Femi
*From:* [EMAIL
Guilherme Loch Waltrick Góes wrote:
We tried to setup SIP between Asterisk and the Router, but the SIP
stack in this IOS version is broken and causes the router to reboot.
My biggest problem is, I can't upgrade de IOS version of the router.
Why not?
Surely that would be the easier option.
From memory, this is an issue with Asterisk 1.2 which can be fixed by
moving to 1.4
PaulH
Chris Bagnall wrote:
Greetings list,
We recently installed some Linksys SPA962 + SPA932 sidecars into a client's
offices. The BLF functionality works fine, but call pickup using the BLF is
Nagios?
PaulH
Jacobus van Niekerk wrote:
Dear Asterisk Users
I'm looking for a solution that can be used to monitor Asterisk and the
Telco lines aswell as the network (Servers, WAN LAN links, Router
Switches)
Thanks
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I have used both munin and nagios - both are cool.
PaulH
EdPimentl wrote:
http://www.voip-info.org/wiki/view/Asterisk+monitoring
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ARI really only let people check their voicemail via a web interface -
for CDR's you can install areske cdr interface as that bolts on to vanilla
asterisk with a small amount of work.
Recordings - how complicated an interface do you need? From memory
there's something in the contribs folder
The DISA application should do what you are looking for.
PaulH
Rudolf Ladyzhenskii wrote:
Hi, all
Can someone give me an example on how to do following:
Asterisk receives incoming call from SIP
Asterisk asks for a pin number
Astersisk provides dialtone
Asterisk collects digits from the
I played with the Polycom login/logout function about a year ago, and it
looked brilliant.
I could never get it to work, but at the time I had both Polycom and
Digium agree that it would be worth getting running.
I ran out of time on that project, and have never re-visited it. But it
would
Slightly confused - this isn't to hard to do (I have done it quite a few
times before )
The dialplan to do this should only be several lines long. Can you
provide a copy of your dialplan?
PaulH
Mark Best wrote:
I have a legacy PBX that I want to slowly move off of. Below is a
diagram of
To provide a better example:
(this is untested hack work - as I usually provide to this list)
exten = _2XXX,1,Dial(SIP/${EXTEN})
exten = _3XXX,1,Dial(ZAP/G2/${EXTEN})
exten = _X.,1,Dial(ZAP/G0/{EXTEN})
Clean up and test as appropriate. :)
PaulH
Mark Best wrote:
I have a legacy PBX
Call files can do something like this - you can choose the number to
call and where to connect the call to (within the dialplan)
PaulH
Julien Claassen wrote:
Hello all!
Is there a way to (mis)use asterisk itself as a softphone? Can I make a
call
from within the CLI? Can asterisk from
One of the Asterisk people down here in Melb set it up for the company
they used to work for, and I played with it once and it seemed to be usable.
PaulH
Lee, John (Sydney) wrote:
Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?
Does
Asterisk?
PaulH
Jeremy Mann wrote:
Does anyone know of a pri splitter device? Something that would take
an incoming PRI, and based on DID send that out one of other multiple
PRI ports?
I’m needing to take a single PRI from the telco, and send it to two
separate phone systems(one
Asterisk.
PaulH
Tom Moore wrote:
Hi guys,
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?
Tom
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Are you looking for a hardware suggestion or a software suggestion?
PaulH
Tom Moore wrote:
No, these are mainly Samsung pbx systems.
I know I can use Asterisk to do this but what be a solid platform to
go with that can go in the phone closet?
tom
Is there a maximum string length for use with the legacy interface
chan_string?
Does it depend on the type of cup used? Does styrofoam give better range
than paper?
regards,
Drew
A lighter material for the cup will give better dynamic range than a
heavier one, at the
Joe Carroll wrote:
I’ve search the world over…. but I haven’t figured out a way to have
valid/invalid options for entry when using the Read command…
I need to set a variable, but only want to allow certain values to be
valid options for that variable…
Any ideas?
Thanks in advance..
This is a known issue with 1.4.21 - see bugs.digium.com -
http://bugs.digium.com/view.php?id=12921
later,
PaulH
Carlos Chavez wrote:
I am having a very strange problem with a new Asterisk installation. I
am using Asterisk 1.4.21.2 on a CentOS 5.2 server. The problem is that when
Joseph wrote:
Does anybody know if the process of upgrading firmware on Linksys
SPA3102-NA in Linux is the same as on Sipura 3K as described on voip-info.org
http://www.voip-info.org/wiki/view/Sipura
I'm pretty sure it works - I used it to upgrade a (god help me) SPA 9000
the other
Most likely Asterisk has been built without zaptel support. (if you
built Asterisk first then zaptel second, this will happen)
Rebuild Asterisk.
PaulH
Shaun Wingrin wrote:
Hi All, Hope someone can help.
Asterisk version 1.4.14 is running and just installed
zaptel-1.4.11with only
Ouch - upgrade.
PaulH
Jay Ray wrote:
Yes, I am running 1.0.5
--- On *Thu, 8/14/08, Tzafrir Cohen /[EMAIL PROTECTED]/* wrote:
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Unable to create ZAP Channel
To: asterisk-users@lists.digium.com
Date:
Output of 'zap show status' (in asterisk)
Run 'ztcfg' from the command line and check that out too...
PaulH
Jay Ray wrote:
Here it is:
[asterisk]# asterisk -rx 'zap show channel 1'
Unable to find given channel 1
Verbosity is at least 20
--- On *Thu, 8/14/08, Tzafrir Cohen /[EMAIL
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
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:
Joseph wrote:
On 08/11/08 14:38, Joseph wrote:
My phone rings once and stops before playing message; how to stop this
behavior.
I think it has something to do with Linksys SPA 3201 with Setting under:
PSTN-To-VoIP Gateway.
PSTN-To-VoIP Gateway Enable: Yes
PSTN Ring Thru Line
Bill Andersen wrote:
I am thinking about a change to our company's phone layout and would like
to get comments from people who have done something similar.
Currently, we have 3 locations - each with their own Asterisk PBX. The
corporate office has a PRI. Each remote location has a SIP
I suppose the bit to check is the features ('show features') and then
try to record a call (*1) and see what the terminal says...
PaulH
Bill Michaelson wrote:
My client needs call recording features and would like to initiate the
process in-call (typically *1). I'm installing Asterisk
Plantronics.
PaulH
Simon wrote:
Hi there,
Is anyone using a headset with one of these phones? If so, can you
recommend any?
Thanks
Simon
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AstriCon 2008 - September
That's a good question - the plantronics are available with
interchangeable ends - which makes them easy to move between different
phones.
PaulH
Simon wrote:
So any 2.5 headset will work with the SPA922?
On Fri, Aug 1, 2008 at 12:23 PM, Paul Hales [EMAIL PROTECTED] wrote:
Plantronics
Dan Austin wrote:
John wrote:
Thanks Steve for your suggestions.
In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is
much more common.
This is exactly my current problem.
NETCOM in Shanghai just told my local contact it is an E1 and that's it.
I
Lee, John (Sydney) wrote:
I am trying to build a simple queue with several agents using
AgentCallBackLogin.
From what I read on the Internet and tried briefly, it seems to suggest that
I should be coding my own queue system for AgentCallBackLogin using AEL2
instead of using the
Wow - that's nasty.
Almost like a broken card or MB. Ouch.
Should you call the supplier of the card and ask them about warranty?
PaulH
Lee, John (Sydney) wrote:
This time, I am trying to remotely install Asterisk in China.
I was told that an E1 line has been installed and so I plug it into
randulo wrote:
On Sun, Jul 27, 2008 at 12:19 PM, Matt Watson [EMAIL PROTECTED] wrote:
I've seen it before infact there is a website setup where people can
post stuff made with it... kind of super nerdy!
http://www.ratemydialplan.com
Cannot find lib path
too nerdy!
Agreed
Ricardo Melendez wrote:
Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have
the feature to transfer calls (Incoming call - Answer - FLASH -
Dial Number to transfer - Answer - FLASH+4) and the call is
transferred, but I have the need to implement an internal ACD using
Alex Balashov wrote:
Conrad Wood wrote:
Unless I am mistaken and there *is* some way to run 400 simultaneous
calls over 2 PRIs...
Traditionally, there hasn't been. But now that they've got that Large
Hadron Collider going... :-)
Are you thinking that with dark matter we
David Nedved wrote:
Interestingly enough, I've had my Grandstream suffering
from the same
problem since I upgraded to 1.4.20, although my config is
static rather
than realtime. I'd actually written it off to typical
Grand-heap-of-$#!+-stream behaviour. :)
I didn't say because I
David Nedved wrote:
Hi David,
It may be IAX2 bug, do you use IAX? In my case downgrading
back to 1.4.19
did the job.
No IAX for me. I don't recall ever having this issue on 1.4.19 so unless I
hear any other suggestions as to how to troubleshoot this I'll go back to
that version
I remember driving past a building with a magnetic door lock (where a
friend worked) late one night, and he noticed that their door was open
and swinging in the breeze. Turns out that if you lost power, this
particular door would open up as well.
PaulH
Steve Totaro wrote:
One of the last
Lee, John (Sydney) wrote:
Do you do contract work?
Thanks for making my day :-)
I am sure there are lots of much more experienced Asterisk people out
there who will respond to your email.
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Lee, John (Sydney) wrote:
exten=7000,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn)
7000 is the extension of main menu
Where do I put the reference to open hours menu in the statement
above.
exten=7000,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn)
[...code for office
guys.
Thanks anyway.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Wednesday, 16 July 2008 5:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] how to incorporate open hours
Lee, John
Zach Segal wrote:
Paul Chambers wrote:
It's probably total overkill, but nagios is a very sophisticated system
monitoring tool. The learning curve is quite steep, but once there,
it'll be able to monitor just about anything of concern.
Paul
I have to second this while
Ein Bielaczyc wrote:
While I respect and appreciate your biased opinion, I was hoping for
more input from objective users.
Thanks :-)
I object to your objective of finding objective users.
In fact, I find it objective.;)
PaulH
___
Have you upgraded to the latest version?
We found a few bugs went away on our test unit when we did that.
PaulH
Sydney Web Hosting wrote:
HI all,
I am having issues with the gui on my AA50.
under Voice Menus Add new Step Go to Time based rule.
It allows me to select “Go to Time based
From memory, I have seen something similar done with the SIPPEERS
function (curcalls) but it's too fuzzy for me to remember it fully.
Paul Hales
NTS
Carlos Chavez wrote:
I have a customer with a small outgoing call center. Usually only 3 to
5 agents online. We are still using Agent
You should probably avoid giving incoming access to outgoing..
PaulH
Lee, John (Sydney) wrote:
With an ISDN10/20/30/etc, I would just put all the lines into an
'incoming' context - and make sure that incoming context doesn't have
any includes (unless you really need them...)
Can
You should probably look at having another context - maybe even
'sip-phones' for your sip phones.
Then include everything you need there.
PaulH
Lee, John (Sydney) wrote:
You should probably avoid giving incoming access to outgoing..
Thanks Paul.
[incoming]
...
include =
You really want to avoid people making incoming calls being able to make
outgoing calls.
Especially international ones.
PaulH
Lee, John (Sydney) wrote:
You should probably avoid giving incoming access to outgoing..
Thanks Paul.
[incoming]
...
include = internal
include =
H...you could build a site that used a voip service provided to
hook into people's mobiles or home phones
That way there's very little software development needed, and it 'just
works'.
You could deliver the full 'net phone' stuff as phase 2.
later,
PaulH
Dean Collins wrote:
The Asterisk appliance is a but naughty - is really like the idea of
handling DHCP on it's lan ports (and having the phones on them) and
hooking up the WAN port to your local network.
I have heard you can get around this, but it's a bit of work. (no
personal experience)
Paul Hales
NTS
Dean
Basically, you run the phone lines into the asterisk box, then out of
the Asterisk system into the PABX.
This works reasonably well, and gives you the option to migrate to a
full asterisk setup in the future.
PaulH
Syed Nasruddin wrote:
Thanks Steve,
How I can use it Asterisk as Man In
You really should discuss this at the Asterfax forums:
http://forums.asteriskit.com.au/
later,
PaulH
On Wed, 2008-06-04 at 14:17 +0530, Sukhbir Singh wrote:
Hi All,
I am getting following error when i start AsterFax:
Please help me to solve this issue:
[EMAIL
The plantronics dect units are great - bluetooth units are cheaper, but
worser.
PaulH
On Mon, 2008-05-19 at 12:47 -0400, Mike Clark wrote:
Anyone have recommendations for wireless headsets that work well with
Polycom phones and Asterisk?
Thanks,
Mike Clark
With an ISDN10/20/30/etc, I would just put all the lines into an
'incoming' context - and make sure that incoming context doesn't have
any includes (unless you really need them...)
PaulH
On Tue, 2008-05-13 at 09:43 +1000, Lee, John (Sydney) wrote:
In The future of Telephony, it says ... We
dmesg | grep -i zap
Should give you a version, and an echo cancellation technology.
PaulH
On Thu, 2008-05-08 at 17:05 +1000, Lee, John (Sydney) wrote:
The only things I set in relation to echo cancellation is in zapata.conf
where I put echocancel=yes
Ouch...any idea what echo
Which is reasonably new, but an upgrade to the latest version (1.4.10.1)
will only take 5 minutes and is worth a shot.
PaulH
On Fri, 2008-05-09 at 15:24 +1000, Lee, John (Sydney) wrote:
dmesg | grep -i zap
Should give you a version, and an echo cancellation technology.
Thanks Paul.
Ouch...any idea what echo cancellation your system is using?
PaulH
On Thu, 2008-05-08 at 14:55 +1000, Lee, John (Sydney) wrote:
the relaxdmtf (or similar) option in zaptel can make this work a bit
better...but it's a try at your own risk option!
PaulH
Thanks Paul.
I have further
Where are you located?
That will help with getting contacted by someone local
PaulH
On Wed, 2008-05-07 at 19:31 -1000, Thermal Wetland wrote:
I would like to hire someone to help us tweak our asterisk system for
Snom phones.
We would like to enable things like:
One touch
Tomorrow night is the monthly Asterisk night...in melbourne
(australia)...
The usual stuff - get together, eat, show off tech toys.
At the Pint on Punt, from 7pm.
later,
PaulH
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Snom make a headset box, which does the same thing.
Far, far better than a lifter.
PaulH
On Mon, 2008-05-05 at 19:28 -0400, Mike wrote:
If you want to avoid a mecanical lifter, the only option I know of is a
Jabra GN9350 with a Polycom EHS (electronic hookswitch) cable.
It came out only
I did some dialplan work with numbers starting with + (outlook) and from
memory things like
exten = +X.,1,Answer
Seemed to work fine...
PaulH
Melb, Australia
On Fri, 2008-05-02 at 10:25 +1000, Rod Bacon wrote:
This is probably a very simple question, but I can’t for the life of
me work it
)
Paul Hales wrote:
I did some dialplan work with numbers starting with + (outlook) and from
memory things like
exten = +X.,1,Answer
Seemed to work fine...
PaulH
Melb, Australia
On Fri, 2008-05-02 at 10:25 +1000, Rod Bacon wrote:
This is probably a very simple question
Any suggestions as to what improvements can be made with regards to echo
cancellation on a B410p card...
Ideas?
PaulH
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To UNSUBSCRIBE or
parties at that
point?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Monday, March 31, 2008 9:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before
It can be done via the 'visit a macro' part of the dial command...
If anyone would like, i can post a code sample.
PaulH
On Mon, 2008-03-31 at 15:33 -0400, Dean Collins wrote:
Yes it is.
I'm remote at the moment so I can't send you the code but google for mobile
remote receiver and you'll
, Jeremy Mann wrote:
Please do!
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL
PROTECTED]
Sent: Monday, March 31, 2008 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give
Can't you just use the same bootrom for all your polycom phones?
PaulH
On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote:
I have a question about DHCP and boot server supporting more than 1
model of Polycom phones.
According to Polycom standards, Polycom phone boots up to get a
I thought you weren't going to let anyone know which Asterisk gui you
were using?
PaulH
On Wed, 2008-03-26 at 15:25 -0500, Bill Andersen wrote:
Is anyone on the list reselling (or just using) EvolutionPBX
from Intuitive Voice Technologies?? If so, please contact me
off list. Thanks.
I have found 'make menuselect' useful to find out what is/isn't built
and sometimes a hint as to why.
PaulH
On Tue, 2008-03-25 at 14:37 +1100, Rob Hillis wrote:
Only if you're trying to compile Asterisk 1.2. Asterisk 1.4 also has
the menuselect configuration, though for most applications
Have you tried setting the card as being T1 instead of E1 for the port
connected to the channel bank?
PaulH
On Tue, 2008-03-25 at 15:33 +1100, Lee, John (Sydney) wrote:
Any luck with the channel bank?
Thanks for the reminder Paul but so far no luck.
I have been getting:
1) ***
I think you can set callerid's in zaptel.conf for each analog port - I
did that for a client a while ago. (from memory)
PaulH
On Wed, 2008-03-26 at 15:53 +1100, Lee, John (Sydney) wrote:
Good to here,
I know the time off set US - AU is terrible when you need support.
I have continued to
Any luck with the channel bank?
PaulH
On Wed, 2008-03-19 at 18:09 +1100, Lee, John (Sydney) wrote:
What kind of information are you looking for? configuration or? If you
look in our manuals our cards and the Digium cards configure the same
in zaptel and zapata.
Hi James, I have
Which version of Asterisk are you compiling?
PaulH
On Mon, 2008-03-24 at 21:47 -0400, Kyle Gibbons wrote:
Hi,
I hope this is not too much of a noob question. I am trying to compile
Asterisk and I cannot figure out how to get into the Menuselect menu.
I do #make clean #./configure #make
I believe it is being merged into Asterisk 1.6, as it is a new feature.
PaulH
On Mon, 2008-03-24 at 15:28 -0400, Drew Gibson wrote:
Does it work with Asterisk 1.2?
BerkHolz, Steven wrote:
Asterisk work does not pay all of my bills, so I have joined up with a
company that has a very
Have you looked at the privacymanager function in Asterisk?
PaulH
On Wed, 2008-03-19 at 10:31 +0530, Janu Mukherjee wrote:
Hi,
I have our software with SIP running on it.I configured asterisk
server as proxy. How do I implement the call screening
features(incoming and outgoing) using
Paging works fine on the snom handsets - call all can be a problem,
especially if you don't want to key all the phones in manually. (but I
do have a rough memory of a script that did that...)
What are you looking at doing?
PaulH
On Thu, 2008-03-20 at 09:32 +1100, Rupert Utteridge -
Our office PABX is a via low heat pc, with an ISDN10 and 15 IP handsets.
It gets regularly used and abused by us linux idiots in the office, and
runs like a charm. We test software on it, write silly dialplans and
generally treat it badly.
It could not be described as 'server grade' by any
I think some people here (like myself) started off as Asterisk users,
then moved on to helping other people with their Asterisk systems.
Which makes sense - once your Asterisk box is running well, why not
share how nice your work is/was?
PaulH
On Wed, 2008-03-19 at 16:38 -0500, Bill
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