Assuming the price of an ADSI screen phone (say, Aastra 390) was the same
as an IP screen phone (say, Cisco 7960) and someone was setting up an *
server for their 20 employees (each of whom would have either an ADSI or IP
phone on their desk), would there be advantages to using the ADSI phones
Tilghman,
[incoming]
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,Dial(SIP/2126,15,t)
exten = s,4,Voicemail(u2120)
exten = s,5,Hangup
exten = s,102,Voicemail(b2120)
exten = s,103,Hangup
I have had reports from users that people at the other end are able to
hear me fine, while to me I hear a
On Sunday, 18 January, 2004 02:04, Ken Alker wrote:
Assuming the price of an ADSI screen phone (say, Aastra 390) was the same
as an IP screen phone (say, Cisco 7960) and someone was setting up an *
server for their 20 employees (each of whom would have either an ADSI or IP
phone on their
On Sunday 18 January 2004 01:06, Brent Franks wrote:
Tilghman,
[incoming]
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,Dial(SIP/2126,15,t)
exten = s,4,Voicemail(u2120)
exten = s,5,Hangup
exten = s,102,Voicemail(b2120)
exten = s,103,Hangup
Try doing the Dial before the Answer or add
John Todd wrote:
The soundfiles I submitted earlier today have been cleaned up, and added
to the Digium CVS server in a more formal manner. Also, some of the
really bad formatting in my .txt description file has been rectified.
All of the sounds on my website are now on the Digium site, and
Usually CLASS stands for Custom Local Area Signalling Services. See
http://www.atomicfrog.com/archives/phreak/ess10.htm
On Sat, 2004-01-17 at 21:35, Samuel Jimenez wrote:
If what u mean by CLASS is Class of Service, ie: the ability to
allow/denny access to users to/from resources like public
this is $318 + taxes. my prices included two ISDN cards and 24% vat.
and dell 'servers' aren't more 'servers' than my home-built servers - at
least not the low-end ones.
roy
Paul Mahler wrote:
I have a Dell 400sc sever on order. It will be shipped on the 27th. It is a
2.4GHz P4 with a 533 MHz
great!!
but when will asterisk use some of these new babies?;))
it would be really great to have app_queue saying you are currently caller
number 7 in the queue (=you-are-curr-call-num.gsm + 7.gsm +
in-the-queue.gsm)
that would be really really great.
when speaking of app_queue. i think it
John Todd said:
...
Ideas welcome for more text; I may have another timeslot with Allison
early next week in which there will be some leftover room for
additional words. Short phrases and meaningful sets of words for
existing applications are desired; please don't give me words for
apps
After installing mpg123 * will no longer start
up. I get the following error.
ERROR[16384]: File asterisk.c, Line 1349 (main): Unable to
connect to remote asterisk
If I remove mpg123, * will run as usual. Any ideas?
~paul
Looking at installing Asterisk and have not been able to find any info on
minimum or recommend system hardware.
I have a box P200MMX 128mb 4.5gb HDD which is running Redhat 8.0 fine.
Noticing on the Digium web site it mentions that the single fxo card
(Wildcard X100P) requires a minimum pIII
At 2:55 PM +0100 1/18/04, [EMAIL PROTECTED] wrote:
Subject: Re: [Asterisk-Users] New sounds also now in CVS
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Sun, 18 Jan 2004 14:55:25 +0100 (CET)
John Todd said:
...
Ideas welcome for more text; I may have another
Hi!
Can multiple FWD accounts be registered?
Yes.
I have the following output in my sip.conf file:
register=74928:[EMAIL PROTECTED]/74928
register=75160:[EMAIL PROTECTED]/75160
register=74573:[EMAIL PROTECTED]/74573
Ok so far. Now consider this:
1. you register in order to be called:
run * in console mode and send the
log.
asterisk -cv
- Original Message -
From:
Paul
To: [EMAIL PROTECTED]
Sent: Sunday, January 18, 2004 11:31
AM
Subject: [Asterisk-Users] No startup
after mpg123 install
After installing mpg123 * will no
John Todd [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Ideas welcome for more text; I may have another timeslot with Allison
early next week in which there will be some leftover room for
additional words. Short phrases and meaningful sets of words for
existing applications are
Is nufone having problems taking gsm calls
today
i had some issues dialing overseas to call my
folks.
here's snip of what the console
displayed
-- Executing
Dial("SIP/2204-a279", "IAX2/[EMAIL PROTECTED]/011351217907000|100|T")
in new stackJan 18 10:30:02 WARNING[1200884528]:
Just dropping a note about hot-desking
I believe hot-desking requires hot phone numbers. These hot phone #
should not be connected to physical phones but rather to a database
entry from where the physical phone(s) is(are) retrieved.
Users should be able to modify (securely) their own database
Hi!
Also, when loading the Asterisk configs as shown below, it displays a
message about Removed default indication country au and at the end
proceeds to set default indication country to au¦ the Removed part has
me thinking its forgotten all about the particular indications for au?
I
Have you experienced a hardware failure yet that you had to come back
from? If you loose a drive, it is a high probability that you will loose
the controller. So unless you have a add on card, or some motherboard
Yes, many times. I have _never_ lost a controller when the drive went; the
On Sun, 18 Jan 2004, Ulexus waxed:
On Sunday, 18 January, 2004 02:04, Ken Alker wrote:
Assuming the price of an ADSI screen phone (say, Aastra 390) was the same
as an IP screen phone (say, Cisco 7960) and someone was setting up an *
server for their 20 employees (each of whom would have
[EMAIL PROTECTED] wrote:
Is nufone having problems taking gsm calls today
i had some issues dialing overseas to call my folks.
here's snip of what the console displayed
-- Executing Dial(SIP/2204-a279,
IAX2/[EMAIL PROTECTED]/011351217907000|100|T
mailto:IAX2/[EMAIL
Had a look at your code and is looking good - need to add it to *
Looked for a conversion tool as well for WAV/GSM G.723.1. (could not
find lbccodec)
Does anybody have a suggestions where to find conversion tools
Cheers
Dan
Andrei Koulik wrote:
I solve it for h323 in follow way:
1. Exclude
My Dell 400sc server was $318 delivered including tax.
They are indeed servers, not a PC. They are engineered, built, configured,
and maintained as severs. They come with on-site maintenance, which is great
if the server is in a different location than I am in.
There is no way you can build a
Paul,
I wholly agree with what you're saying - I too ensure that we have at
the very minimum, a set of full spares.
However, this thread really has the wrong name at this point... We're
now looking at embedded solutions, in the same way Cisco has with it's
ICS 7750 solution. I'm looking to
Hi,
I'm running a simple test from asterisk towards a
public telco switch (AXE10) over E100P.
Here is the test case:
1) 30 calls are setup simultainously, 20 sec ringing
time.
2) no calls answers (just calling a vacant public tax
office :=)
3) Each channel will continue on its own with the same
I have coded chan_sip.c so that you can have
// sip.conf
register = username:[EMAIL PROTECTED]/redirectconfig
[redirectconfig]
redirect=yes
redirecturi=sip:[EMAIL PROTECTED]
redirecturi=sip:[EMAIL PROTECTED]
redirecturi=sip:[EMAIL PROTECTED]
so when you receive a call it will redirect to
On Sunday 18 January 2004 12:01, Adthrawn wrote:
Spell out RAID - Redundant Array of Inexpensive Disks. Bingo.
That's Independent Disks. It's the independence of each spindle that
is valued, not the cost proposition. If one spindle goes, it's not all
of your data which goes with it. Consider
Hi,
I wonder if it is possible to switch ISDN data calls with asterisk.
What I want to do is have a Digium TE400P in an asterisk server, one E1
interface connected to a PRI ISDN line and one in PRI network-side mode
connected to a Cisco PPP dialin router as an extension capable of
handling 30
Alfred R. Nurnberger wrote:
There are a number of paging interfaces available which connect to a regular
phone line on one side
and to a paging amplifier on the other side.
Could you provide a pointer?
The search terms pager and telephone together are giving me a heck
of a lot of noise. . .
Could you please explain what you want to do, why you want asterisk to register but
not take
the calls?
You could take the calls into the dialplan (extensions.conf) and dial out from there
with an agi
script that performed the same thing. If you have canreinvite=yes, asterisk will leave
the
On Sun, 2004-01-18 at 14:11, Jan Baumann wrote:
Hi,
I wonder if it is possible to switch ISDN data calls with asterisk.
yes
What I want to do is have a Digium TE400P in an asterisk server, one E1
interface connected to a PRI ISDN line and one in PRI network-side mode
connected to a
Hello,
I still have problems with compiling Asterisk, and I am still on the first
step at zaptel make clean; make install.
I assume, that the troubles I have stem from a recent kernel-update I made.
I upgraded from k_athlon_2.4.21_99_i586 to k_athlon_2.4.21_166_i586 via YaST
Online Update..
Now
Hi Knut-
This may be related to a call volume problem that I've had a lot of problems
with in a very busy IVR environment. I can easily duplicate this on my own
machine by just looping one span to another.
A colleague and I have been looking into this and believe that it may be
related to
Hi,
Can anyone pleaselet me know what is the latest stable version of asterisk.
Thanks,
shailesh
Do you Yahoo!?
Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes
I tried that - no errors reported. I checked one or two of the SQL calls
and none returns an error. I rebuilt and reinstalled mysql and all the
ODBC drivers - still no integers written! The direct MySQL driver logs
calls fine. So it looks like there's a deeper problem with ODBC to sort
I spoke the other day about my preliminary tests with office-wide
paging with Cisco phones using the new SIP 6.1 image which supports
auto-answer. I've got a small and crude recipe for those of you who
want to experiment and hopefully create some better and more complete
examples than the one
John Todd said:
The freenum.org project wants to use your trunks! The freenum.org project
is an ENUM parallel tree, which has as an eventual goal the distribution
of ENUM numbering in nations or areas which due to political or other
issues are not able to get secure, inexpensive, or
Howdy,
Sounds like we are in violent agreement. ;-)
What is the difference between an embedded system and a server?? We are
using 1U rack mount servers that cost more, about $1,200.
One nice thing about the dell server is that it will boot from the USB port.
You can put everything on a USB
There are a number of paging interfaces available which connect to a
regular
phone line on one side
and to a paging amplifier on the other side.
Could you provide a pointer?
The search terms pager and telephone together are giving me a heck
of a lot of noise. . .
Thx.
B.
John Todd said:
The freenum.org project wants to use your trunks! The freenum.org project
is an ENUM parallel tree, which has as an eventual goal the distribution
of ENUM numbering in nations or areas which due to political or other
issues are not able to get secure, inexpensive, or
Tilghman Lesher wrote:
On Sunday 18 January 2004 12:01, Adthrawn wrote:
Spell out RAID - Redundant Array of Inexpensive Disks. Bingo.
That's Independent Disks. It's the independence of each spindle that
is valued, not the cost proposition. If one spindle goes, it's not all
of your data which
Franz,
We tried to use SuSE initially and had no luck compiling zaptel on
either 8.2 or 9.0. We even had Digium take a look. After working on it
for days we finally switched to Red Hat 9. Everything compiled without
problem.
Hope that helps a little.
Dustin
-Original Message-
From:
We are using currently following settings: (melbourne)
Zapata.conf
--
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.5
txgain=0.0
Works fine for us, but you can play of course a bit with the
gains
-Ursprngliche
Nachricht-
Von:
Date: Wed, 14 Jan 2004 15:11:59 +0100 (CET)
Subject: Re: [Asterisk-Users] Re: newbie ISDN question
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Currently i am polishing the driver for the hfc-s pci a chipset,
which i used in numerous el-cheapo
Title: RE: [Asterisk-Users] capacity testing
Jesse,
Thanks
for your feedback.
1. I
am running kernel 2.4.18.3 with linux 7.3, please let me know which version of
Redhat are you running on and which kernel are you running, I wonder if that
could make a difference too. I am surprised that
/O
Attached is the Debug information with the 300 Redirect implementation
with asterisk,
You can get the source code from
http://www.speak2world.com/asterisk/chan_sip.php
and when you compile and run it, you get the following info in the debug
o/p.
pbx*CLI sip debug
SIP Debugging
Tilghman Lesher wrote:
On Sunday 18 January 2004 12:01, Adthrawn wrote:
Spell out RAID - Redundant Array of Inexpensive Disks. Bingo.
That's Independent Disks. It's the independence of each spindle
that
is valued, not the cost proposition. If one spindle goes, it's not
all
of your data
Call forwarding
Call forwarding not reply
call forwarding busy
David Kwok
smime.p7s
Description: S/MIME Cryptographic Signature
Bogen
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 18, 2004 3:35 PM
Subject: Re: [Asterisk-Users] Zone Paging
Alfred R. Nurnberger wrote:
There are a number of paging interfaces available which connect to a
regular
phone
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Albertson
Sent: Friday, 16 January 2004 4:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ultra-cheap asterisk box
snip
What I'm finding is that the PCs are so cheap that the cost
Hi
Has anyone opened up a grandstream phone or handytone ATA to find out what is inside?
What is the CPU? How much RAM?
Cheers
Rob
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi,
Has anyone experienced * hang/exit when issuing -
asterisk -r -x reload
Peter
- Original Message -
From: Adthrawn [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 18, 2004 4:54 PM
Subject: [Asterisk-Users] Now: Small Biz Robust Asterisk Solution - SBRAS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ian Pilkington
Sent: Monday, 19 January 2004 1:51
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] minimum system hardware for Asterisk install
Looking at installing Asterisk and have not been able
Title: RE: [Asterisk-Users] capacity testing
Dear All,
I have been using Asterisk "10
days ago" version loaded onto my Redhat 7.3 with kernel 2.4.18-3 running
Jeremy's h323 driver. It has been running okay with a bit of problems, like
system crashing after certain period of time with 15
On Mon, Jan 19, 2004 at 12:25:15PM +1100, [EMAIL PROTECTED] said:
snip
What I'm finding is that the PCs are so cheap that the cost of
electric power to run them is now a large part of the cost.
(assume 0.20/kwh times 200W times 365 days = $350. So you
pay for the PC again every year in
On Sun, 2004-01-18 at 19:25, [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Albertson
Sent: Friday, 16 January 2004 4:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ultra-cheap asterisk box
snip
Smarthome has an inexpensive one at: http://www.smarthome.com/77965.html
for $60.00
This one needs a FXO port though.
Alfred.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian Capouch
Sent: Sunday, January 18, 2004 12:36 PM
To: [EMAIL PROTECTED]
I'm not sure, I just opened mine up to see. Looks like they epoxied over
three of the chips. One is a 73pin, another is a 44 pin, and the last looks
to be a 44 pin.
Ethernet is a RTL8019AS (10mbit) and it's using a Tamarack TC3097-8 repeater
(HUB) which actually supports 9 ports (8 ethernet, 1
Title: RE: [Asterisk-Users] capacity testing
I tried to use it to create a trunk
to Ciscos call manager. The 0.7.1 code worked up to a point.
The call would be established, but audio
was one-way from the Call Manager. Asterisk with
Chan_h323 would not setup the sending rtp
stream. The
On Sun, 18 Jan 2004 [EMAIL PROTECTED] wrote:
On another note, what's the deal with Hold, Mute, Caller ID Review, and
Called Number Review on these things? Do they just not work, or am I missing
something?
Caller ID Review/Called Number Review only works when the phone is off the
hook. I
You can't use chan_h323 with call manager.
bkw
On Sun, 18 Jan 2004, Dan Austin wrote:
I tried to use it to create a 'trunk' to Cisco's call manager. The
0.7.1 code worked up to a point.
The call would be established, but audio was one-way from the Call
Manager. Asterisk with
Chan_h323
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shailesh Alluri
Sent: Monday, 19 January 2004 8:15
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Latest version of asterisk
Hi,
Hi,
I am looking for a DID access provider in the west cost (not Iconnect,
voiceglo, packet8 etc). What I need someone like NuFone, IAX and
Toll/Tolfree multiple presentations possible. Any references appreciated.
Thanks
SW
___
Asterisk-Users
Will the other chan_oh323 work?
Quoting Brian West [EMAIL PROTECTED]:
You can't use chan_h323 with call manager.
bkw
On Sun, 18 Jan 2004, Dan Austin wrote:
I tried to use it to create a 'trunk' to Cisco's call manager. The
0.7.1 code worked up to a point.
The call would be
Will either of the h323 channels work with gatekeepers properly? Ie,
reregister through * reloads and such? I've had problems with chan_h323 in
this regard. Chan_h323 will register fine with a gnugk first time around. But
after a reload, it loses its connection. Or should I try downloading
...
Quoting John Todd [EMAIL PROTECTED]:
As to the specifications of directories for sounds in certain
groups: yes, I think that is a good idea, but I am unsure how to
implement it. Mark and I touched on that last night while adding the
sounds to the CVS server, but I told him
You can also get the same files from ftp.digium.com
I did a mirror to help people get it faster in case cvs was hammered
again.
bkw
On Mon, 19 Jan 2004 [EMAIL PROTECTED] wrote:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shailesh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, January 18, 2004 11:22 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New sounds also now in CVS
[...]
The index for each topic could be a text file with a
Hello all!
I have an FXO port on a cisco router that is directly connected to our PBX.
Our ATA-186 (firmware version 3) registers with asterisk, which connects to our cisco
router's fxo port to give me a dialtone on our PBX from the ATA.
How do I pass the flash button to the PBX? It seems the
From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM
We tried to use SuSE initially and had no luck compiling zaptel on
either 8.2 or 9.0. We even had Digium take a look. After working on it
for days we finally switched to Red Hat 9.
Is there anyone who succeeded in compiling Asterisk
Dear All,
Based on your experience and knowledge, which Redhat (7.3, 8 or 9) and which
kernel is most stable and reliable running the 0.7.1 version of Asterisk?
Thanks
Tom
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Version: 6.0.563 /
Dear All
Should one enable HT in the chip when running Asterisk or if we don't, would
that offer alot less processing power?
Tom
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