be affecting system
load when your call load increases. Context switches are simply a symptom
and you still need to find the culprit.
Regards,
Bryce Chidester
Rhino Equipment Corp.
br...@rhinoequipment.com
Tel: +1 (480) 621-4000, +1 (877) RHINO-T1
FAX: +1 (480) 961-1826
On Mon, 2005-08-15 at 13:20 -0500, Chris Wade wrote:
First, FXS = handset / FXO = telco line.
Ditto this.
Maybe something like fax-callback; call-in, hangup, Asterisk dials back
on the other channel using the CID received - a purely physical
solution. Otherwise, have the telco setup a rotary
On Mon, 2005-08-15 at 23:32 +0200, Roland Zagler wrote:
Hello everyone,
I want to build an Asterisk Box where i need 8 FXS interfaces
to connect 8 phones to. The problem is, that there is only one
PCI slot available. What i have is 4 USBs 2.0 interfaces free
(if this helps).
So here's my
http://www.voip-info.org/tiki-index.php?page=Asterisk+CMD+voicemail
On Fri, 2005-08-12 at 11:37 -0400, kurt x wrote:
A user has their unavailable message played and once that message
is over the Comedian
message is played right after. Is there any way to prevent the
Comedian message being
On Fri, 2005-08-12 at 20:09 +0300, Iraklis Zografos wrote:
Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0) '3118' ]
That seems to be fairly clear to me. As I understand it, the Avaya is
rejecting the call. Check that 3118 is an acceptable input (i.e.
On Thu, 2005-08-11 at 18:06 +0100, Kevin Walsh wrote:
Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Justin Selleck wrote:
Is asterisk 2.0 real? Running in c#? I see references to it but cannot
find it anywhere.
r: Generate a ringing tone for the calling party, passing no
On Wed, 2005-08-10 at 15:51 +0100, Joao Pereira wrote:
exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
Just an observation that you have an invalid address there; you have
1193 instead of 193 I believe. Fix this and I see no reason for your
problem to remain.
--
-Bryce
[EMAIL PROTECTED]
On Tue, 2005-08-09 at 10:59 -0700, Edwin Lam wrote:
hi folks.
i'm planning to connect * to 120 POTS line. i've done some research
on FXO cards but unfortunately most manufacturers only make 4 ports/card.
the most i've found is 12 ports. so do i have to get 10 of these cards
and setup 3
On Tue, 2005-08-09 at 12:54 -0700, Edwin Lam wrote:
i guess the way to go is using channel banks to convert those to E1 then
connect Asterisk that way.
further research, how about using these:
http://www.welltech.com.tw/product_e_03.htm
will that work?
Sure, that would work, all 20 of
On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote:
I'm attempting to set up call recording with Asterisk. Using
automon = *1 ; One Touch Record
in features.conf does not appear to be working. I'm using Polycom 501's
but when someone dials *1 while in a call,
On Mon, 2005-08-08 at 16:33 -0500, Eric Wieling aka ManxPower wrote:
Bryce Chidester wrote:
On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote:
I'm attempting to set up call recording with Asterisk. Using
automon = *1 ; One Touch Record
in features.conf does
On Wed, 2005-08-03 at 10:23 +, Andres Tello Abrego wrote:
Assign an extension to the fax at extension.conf
Create a menu.
Why even bother to do that much? Just put the 3rd port/line into its own
extension where s automatically dials the fax machine on 4. You can
still use 1, 2, and 3 for
On Thu, 2005-21-07 at 23:08 -0500, Steve Maroney wrote:
I have the original version of the IAXY. I had it laying around collecting
dust, now Im actually putting it to use. When I call my voicemail
extension (8500), Before I get the voice prompts from the voicemail app,
I hear tones that sound
On Thu, 2005-07-21 at 13:00 +0900, Vic wrote:
Dear all,
I had Tom Rymes and several others suggest how I can implement sending
fax using Asterisk. The idea is to have On-Demand-Fax.
Unfortunately, I wrote down the wrong workflow: the real one is:
1. Person will call our phone
On Wed, 2005-07-20 at 08:26 -0400, Ousmane Doukara wrote:
Hi,
I am unable to get a dialtone from iaxy (the old model). When dial a
mailbox, I can see the mailbox app reacting.
iaxy gets registered. I can make call and the remote phone can hear me. No
sound for iaxy user.
./iaxyprov
On Wed, 2005-07-20 at 16:09 -0600, Scott wrote:
I have tried to get MRTG to graph my Asterisk box but have run into a
problem. When I run the perl script provided at:
http://karlsbakk.net/asterisk/ I get the following error:
[EMAIL PROTECTED] asterisk]# ./asterisk-mrtg -h
to the table.
You're looking for DID service
(http://www.voip-info.org/tiki-index.php?page=DID). Contact your T1
provider to set this up.
--
Bryce Chidester [EMAIL PROTECTED]
Rhino Equipment Corp.
___
Asterisk-Users mailing list
Asterisk-Users
When you restarted Asterisk, did you kill the mpg123 processes?
-Bryce
[EMAIL PROTECTED]
NOTICE: The views expressed in this e-mail do not neccesarily reflect
those of my employer. This is a personal e-mail and as such, the
opinions expressed are my own.
On Jul 12, 2005, at 11:52,
It's a common (and commonly overlooked) problem and whenever there appears to be no logic behind irrational behavior, the RAM is the first place I look. Because the RAM is effectively changing the running program's code at the bit level, any and all actions are unpredictable, along with their
cleanly with an E1 or 2 T1s.
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
Disclaimer: I work for a company that makes channel banks. I believe
we make a very fine product
I choose not to acknowledge Sangoma's existence whenever possible.
I've had some very poor experiences with the quality of their cards,
firmware, and drivers, so I tend to /not/ recommend them.
-Bryce
On Jul 6, 2005, at 16:24, TC wrote:
this would be with a couple channel banks and a
Just a thought, but I seem to recall that in the dialplan, inlcude
and other similar statements are not prefixed by the hash character
(#). Try include = .
-Bryce
On Jul 4, 2005, at 00:05, [EMAIL PROTECTED] wrote:
We are running * V1.0.9 on a demo box.
We have set up everything in our
Sell 'em quick, 'cause here comes IPv6 and something tells me the
market's going to be saturated. Hmmm... what to do with four and a
quarter billion (round numbers) addresses... I know, porn sites!
-Bryce
On Jul 3, 2005, at 11:08, Mark Charlton wrote:
On 7/3/05, Jerry Glomph Black [EMAIL
You're leaving the :1 in the dial expression, which cuts off the
first digit so what's really being dialed to the server is only 88.
-Bryce
On Jul 3, 2005, at 19:01, Joseph wrote:
On Mon, 2005-07-04 at 01:09 +0200, Roland Zagler wrote:
Hi Joseph,
here is how i did it:
iax.conf of
I think the easiest way to accomplish that would be through a channel
bank which could rackmount as well, and interface via T1 to the
computer, thus only one PCI card and room to grow to 24 channels, FXO
and even FXS.
That or go with a 2U rackmount computer.
Regards,
Bryce Chidester
Rhino
to off-list e-mails, but definitely check us out.
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
On Jun 30, 2005, at 15:26, Chris Gamble wrote:
Any chance you could drop some product
/number.
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
On Jun 29, 2005, at 08:33, Chee Foong Chiew wrote:
Hello,
I have the following situation:
I have a PRI with 200 DID numbers
The callerid on outside lines is set by your carrier. Talk to them.
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
On Jun 26, 2005, at 14:25, Jeff Glassman wrote:
I have two
if you're looking for a less expensive, cinch to configure channel
bank, I would look at the Rhino Channel Bank (http://
www.channelbanks.com). I must admit that I work for them, but I
guarantee Asterisk compatibility and good, personal technical support
should you need it.
Regards,
Bryce
modules.conf:
noload = chan_alsa.so
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
On Jun 17, 2005, at 13:34, Conrad Beckert wrote:
Hi
... probably one of those RTFM kind
DID number not the main number
Jeff
Talk to the T1 provider - they're the ones that set it.
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
}.')
exten = 789,2,Hangup
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
On Jun 14, 2005, at 06:00, Ronald Wiplinger wrote:
I would like to setup a test number, that speaks back my phone
the
channel, then it won't know it's ringing.
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
___
Asterisk-Users mailing list
maintainer, he's in the process of
upgrading the software. He may have run into some snags that are
taking awhile to resolve.
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
afford it but they're more willing to pay a
higher price.
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
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Asterisk-Users mailing list
(not to mention the
intolerably slow speeds for even one computer) I haven't implemented
it. However, I have met quite a few that spread their connects across
various DSL, cable, T1, and other frame-relays so it certainly is
doable.
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED
Same here. There goes my quick reference... at least Google has cached
copies and a better search!
On Thu, 2005-06-09 at 14:33 -0700, Chris Coulthurst wrote:
Anyone else unable to get to www.voip-info.org? Site is returning
'connection refused' here.
Chris Coulthurst
[EMAIL PROTECTED]
You might try IAXComm. It's a bit immature but works fairly well on Windows, and is cross-platform as well. However, I've found SIP clients to be better generally and better supported. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] +1 (480) 940-1826
help with this, or has anyone seen this? The mp3s play fine on
any computer and haven't changed since they did work.
Those wishing to hear for themselves, feel free to call extension 8800
at the number/addresses below.
Thank you,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP
I wonder if once the new ones come out, I can purchase the old in
large enough quantities such that when they fail, popping in a
replacement would be no sweat and relatively cheap too?
I too, of course, would be interested in anyone's results with the
new model.
Regards,
Bryce Chidester
and haven't changed since they did work.Those wishing to hear for themselves, feel free to call extension 8800 at the number/addresses below.Thank you,Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305 IAX: [EMAIL PROTECTED]/305
and txgain. If you're using externally-powered phones
(as in not your ordinary joe-schmoe analog phone), I have found that
they're usually pretty hot (loud) and Asterisk can't understand
what is said.
Good luck!
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL
On May 15, 2005, at 00:38, Jean-Yves Avenard wrote:HelloOn 15/05/2005, at 4:40 PM, Steve Underwood wrote:span=1,1,0,ccs,hdb3,crc4 The second parameter now says "treat this E1 as the first priority as the clock source". Your box should lock itself to the PSTN's clock. If that makes no sense, the
On May 15, 2005, at 00:55, Jean-Yves Avenard wrote:
Hi
On 15/05/2005, at 5:51 PM, Bryce Chidester wrote:
I have the same trouble with wct1xxp. I'd just chalked it up to a
PCI bug or other low-level hardware problem with the Digium card.
I've simply learned not to, though it would be nice
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