On Wed, Jun 04, 2008 at 11:02:19PM -0400, John Morey wrote:
Hello,
I've run fxotune
Of which zaptel version, exactly?
at different times but continue to get what seem to be
strange numbers in /etc/fxotune.conf. It ends up with:
5=7,255,251,251,2,255,255,1,255
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to combine multiple accounts and fail back to PSTN
lines
On Wed, Jun 04, 2008 at 05:48:20PM -0500, Bob G wrote:
You can download a FREE browser softphone and or cliick to call that
supports UDP athttp://1ezphone.com/download It works well with Asterisk I
use it everyday
And it's not as if you're affiliated to the company that wrote it,
right?
I
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Why on earth are you running two layers of echo cancellation - hardware
and software? To be honest, I think this is asking for trouble - I've
seen two occasions where having Oslec and hardware echo
I have a local asterisk 1.2 and a remote asterisk 1.4.
Snom 190 can be used with the local asterisk but not with the remote one.
I need some hints where to track down this issue.
Some information:
Snom 190:
Line 1:
Account: 615
Password: OnlyIknowit
Registrar: ast.mydomain.com
I believe Ubuntu is in the process of migrating from sysvinit to
Upstart. Upstart is supposed to be capable of monitoring services to
ensure they don't fail, so I suspect this is likely to be the reason
behind the safe_asterisk script not being used.
Paul Belanger wrote:
I noticed
Hi All,
I have an Asterisk IP-PABX which I need to make the H323 channel up with an
SBC (ACME).
Does anybody have any example configuration guide for this?
I am really really new with Asterisk, well PABX in general. So any help will
be really appreciated.
Thanks in advance.
Kr,
Sema ARCA
On Thu, Jun 05, 2008 at 03:02:28AM +1000, Rob Hillis wrote:
I believe Ubuntu is in the process of migrating from sysvinit to
Upstart. Upstart is supposed to be capable of monitoring services to
ensure they don't fail, so I suspect this is likely to be the reason
behind the safe_asterisk
On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote:
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Why on earth are you running two layers of echo cancellation - hardware
and software? To be honest, I think this is asking for trouble - I've
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote:
If you use a hardware EC (or technically: a span-specific echo
cancellation method) the generic Zaptel echo canceller (software-based,
OSLEC in this case) will not be used.
That's not always been my
On Thu, Jun 05, 2008 at 09:28:52PM +1000, Rob Hillis wrote:
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote:
If you use a hardware EC (or technically: a span-specific echo
cancellation method) the generic Zaptel echo canceller (software-based,
Ronald Wiplinger wrote:
I have a local asterisk 1.2 and a remote asterisk 1.4.
Snom 190 can be used with the local asterisk but not with the remote one.
I need some hints where to track down this issue.
Some information:
Snom 190:
Line 1:
Account: 615
Password: OnlyIknowit
Brent, hope your problems go away soon.
I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are
using asterisk 1.4.2 for a SIP only based configuration. Currently we have
about 200 SIP users which can cause approximately upto 3 simultaneous calls.
We are mainly concerned about
Hi All,
I've trying to force on the ringtone generated for outbound calls with
Dial,r but want the tone to be the UK standard.
I use Zaptel, but don't have any E1/T1 cards at all (am completely IP
based). So I don't think zaptel.conf will come into this (am I right??)
I've tried editing
Hi!
I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We
are using asterisk 1.4.2 for a SIP only based configuration. [...] We
are planning to accomodate about 5,000 users on this server.
Many people on this list will advise you to use a SIP proxy like
OpenSER in front of
Adrian Marsh wrote:
Hi All,
I’ve trying to force on the ringtone generated for outbound calls with
Dial,r but want the tone to be the UK standard.
I use Zaptel, but don’t have any E1/T1 cards at all (am completely IP
based). So I don’t think zaptel.conf will come into this (am I right??)
Tilghman Lesher wrote:
On Wednesday 04 June 2008 22:02:19 John Morey wrote:
Hello,
I've run fxotune at different times but continue to get what seem to be
strange numbers in /etc/fxotune.conf. It ends up with:
5=7,255,251,251,2,255,255,1,255
6=7,255,251,251,2,255,255,1,255
On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote:
On June 4, 2008 06:20:57 pm Joe Carroll wrote:
Interestingly enough, on the syslog messages from the TNT we are seeing
Called = 911, Q850 Cause = 28, SIP Response = 484
That really looks like the switch that the TNT
On Thursday 05 June 2008 09:17:49 Drew Gibson wrote:
Tilghman Lesher wrote:
On Wednesday 04 June 2008 22:02:19 John Morey wrote:
Hello,
I've run fxotune at different times but continue to get what seem to be
strange numbers in /etc/fxotune.conf. It ends up with:
Hmmm..
Well indications.conf does have:
country=uk
But I've definitly just hearing a long-tone tone, long break, long tone
But the file is set to:
[uk]
description = United Kingdom
ringcadence = 400,200,400,2000
; These are the official tones taken from BT SIN350. The actual tones
; used by
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as screeching,
feedback, static, or other useless terms. If users report static
on a system where there cannot be static (all digital, PRI, SIP, etc),
you might be
Adrian Marsh wrote:
Hmmm..
Well indications.conf does have:
country=uk
But I've definitly just hearing a long-tone tone, long break, long tone
But the file is set to:
[uk]
description = United Kingdom
ringcadence = 400,200,400,2000
; These are the official tones taken from BT
On Thursday 05 June 2008 09:50:05 Eric ManxPower Wieling wrote:
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as screeching,
feedback, static, or other useless terms. If users report static
on a system where there
So I wonder, is it asterisk itself generating the tones in Dial(), or
does it comefom the psedo zaptel driver that generates it ??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: 05 June 2008 16:13
To: Asterisk Users Mailing List -
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent calls.
My problem is how to
Hello list,
I have a problem that looks quite simple but I cannot find a way to fix.
I have a Dial() command and want to detect which party of the call hung up
- if it was the caller or the callee.
In the dialplan, I have the folllowing commands...
exten =
exten =
On Thu, Jun 5, 2008 at 6:57 PM, Lenz [EMAIL PROTECTED] wrote:
Hello list,
I have a problem that looks quite simple but I cannot find a way to fix.
I have a Dial() command and want to detect which party of the call hung up
- if it was the caller or the callee.
In the dialplan, I have the
Gordon Henderson a écrit :
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support at max only two concurrent
Hi,
Was wondering if anyone had any tips or experience in getting a Nortel CS1K
and Asterisk 1.4.19 to talk to each other via NRS? So far I've gotten
asterisk to place calls to the CS1k via the NRS, however calls originated by
the CS1K get rejected by the NRS with a 404 Not Found message. If
I`m curious: did going with numerical IP addresses fix your problem?
Mick
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent: Wednesday, June 04, 2008 13:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Philipp von Klitzing wrote:
Hi!
I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We
are using asterisk 1.4.2 for a SIP only based configuration. [...] We
are planning to accomodate about 5,000 users on this server.
Many people on this list will advise you to use a
I wonder why more vendors haven't adopted IAX yet?
I expect that before major players adopt this protocol it'd need to be
confirmed as a standard by some form of international body. That was
underway, but lacking anyone to push the process along.
I would've thought that Digium would be the most
Tilghman Lesher wrote:
On Thursday 05 June 2008 09:50:05 Eric ManxPower Wieling wrote:
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as screeching,
feedback, static, or other useless terms. If users report
On Thu, 2008-06-05 at 13:45 -0500, Michael Graves wrote:
I would've thought that Digium would be the most likely lead proponent,
but that doesn't seem to be the case.
Actually, Digium has been quite active in helping to try to get the IAX
protocol adopted as a standard. See
5 jun 2008 kl. 20.45 skrev Michael Graves:
I wonder why more vendors haven't adopted IAX yet?
I expect that before major players adopt this protocol it'd need to be
confirmed as a standard by some form of international body. That was
underway, but lacking anyone to push the process along.
Just released by the CPSC on their recalls mailing list; please
forward to any venues where you feel operators or resellers of the
SoundStation might be, with this preface included.
My 2W had a battery with the part code 1520-07804-002; its date code
was GP0806, and therefore predates the recall
Solved -
I thought I would follow up in case anyone else on the list is using
gentoo. Got some guidance from the gentoo forum. There is a difference in
this function between 1.33 and 1.34 (1.34 is current in gentoo portage)
1.33:
BOOST_FILESYSTEM_DECL bool no_check( const std::string
Hello
My Flash Operator Panel keeps resetting timers everytime i open it or refresh
it..
is there a way or config to force it to maintain timers ?
_
It’s easy to add contacts from Facebook and other social sites through Windows
Correct me if I'm wrong, but unless you pass specific options to the
dial command to have it override the ringing then when you dial out, you
hear the audio from whatever channel you're dialing on. So the tones
you are hearing are from the telco. The ring cadences defined in
indications.conf
Hi everybody,
Is it possible to create similar extension numbers for multiple users. I am
looking at a case of virtual PBX with 5 tenants on one server. Any
applicable ideas or suggestions would be highly appreciated.
--
Zeeshan A Zakaria
___
--
I'm considering using a PoE switch like this...
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=3023334CatId=2800
...to power as many as 24 Polycom phones of varied kinds.
The sales lit indicates 190 watts available for PoE devices. But I'm
concerned about a
As long as each tenant has its own context you can use the same
numbering plan. The only thing you need to keep unique are the names
for the SIP devices. If you want your tenants to be able to call each
other then you would need to set up a special prefix for each tenant.
On Thu,
Brent Davidson a écrit :
...I wonder why more vendors haven't adopted IAX yet?
Well, even ZoIPer (ex IdeFisk) team, still recommend using SIP over IAX
as SIP is more mature and reliable in asterisk and zoiper,
--
Benoit
begin:vcard
fn:Benoit Plessis
n:Plessis;Benoit
email;internet:[EMAIL
Hi.
At the company I work for, we use Asterisk to communicate with our
offices all around the world. Recently, I've been asked to implement a
video conference system, asterisk compatible/interoperable as possible.
It's preferred but not required to be an open source solution.
What options do I
Benoit Plessis a écrit :
Gordon Henderson a écrit :
On Thu, 5 Jun 2008, benoit plessis wrote:
Hi,
Now that we have a working asterisk server, i'm looking
toward cost optimization :)
We are actually testing a SIP provider, which has an interessting
limitation: each account support
The zaptel version is SVN-branch-1.4-r4257
On Thu, Jun 5, 2008 at 2:57 AM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Wed, Jun 04, 2008 at 11:02:19PM -0400, John Morey wrote:
Hello,
I've run fxotune
Of which zaptel version, exactly?
at different times but continue to get what seem to
Tilghman,
Thanks for the pointer. I'll check this tomorrow and let you know.
John
On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 04 June 2008 22:02:19 John Morey wrote:
Hello,
I've run fxotune at different times but continue to get what seem to
Drew,
I'm also getting complaints of static. Well actually I've complained about
it myself and have asked them to have ATT check the lines just to make sure
the problem is not on that side.
John
On Thu, Jun 5, 2008 at 10:17 AM, Drew Gibson [EMAIL PROTECTED] wrote:
Tilghman Lesher wrote:
On
Yes, we are using the max tnt to aggregate several PRIs both inbound and
outbound from multiple carriers. This PRI is a normal two way circuit that a
carrier would deliver to an end user...
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Jay R.
Would it be possible to have a context with includes for each tenant and
include that context in the specific tenant contexts that you would have
calling each other.if that makes any sense whatsoever..
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On
Wow, rough groupBut good input thanks.I have my tech looking into the
user info and CDRs pages.I will keep working on it, thanks agin good
input for the most part.I hope some of you downloaded the softphone or
clcik to call and tried them.Maybe you could provide with some usefully
info, but
On Jun 5, 2008, at 5:08 PM, Bill Michaelson wrote:
I'm considering using a PoE switch like this...
http://www.tigerdirect.com/applications/SearchTools/item-
details.asp?EdpNo=3023334CatId=2800
...to power as many as 24 Polycom phones of varied kinds.
The sales lit indicates 190 watts
El vie, 06-06-2008 a las 00:24 +0200, Matias Surdi escribió:
At the company I work for, we use Asterisk to communicate with our
offices all around the world. Recently, I've been asked to implement
a
video conference system, asterisk compatible/interoperable as
possible.
It's preferred but
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tilghman Lesher wrote:
Well, the issue is that some enterprising person needs to track down exactly
which optimization in gcc is causing this problem and point it out to them.
We've filed a bug report with them, but without more specific
Hi All -
I hope somebody can clarify for me what exactly fxotune does, and how
it is related to gain settings. I've been reading what appears to be
conflicting information from various sources.
I've got a box with an AEX800 with 6 lines (from Qwest) running
asterisk and zaptel versions 1.4.20.1
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