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Voipstreet.com seems to have very good service. I have been testing for
several weeks now, without issue.
--
A.G. (Tony) Nichols
. It
seems to add a pause of some 4 sec. when dialing.
This would give you the busy error.
--
A.G. (Tony) Nichols
I.S. Manager
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I talked to tech support today... no 64bit yet.
--
A.G. (Tony) Nichols
I.S
Got my latest Linux magazine (www.linux-magazine.com) and fetured is asterisk in home network.
I've also been in contact with Novel/SUSE about their asterisk pakages. Reinhard Max the maintainer.
He has hinted at new packages for SUSE 10. The current ones work well (in production) however he is
I have been told the next version of SUSE will contain the 1.2.1 build.
I am unsure if the zaptel module will be ready -- but I have hight
hopes!
Per my last post... 10.0 is working very well in production -- including the auto updates.-- A.G. (Tony) NicholsI.S. Manager
I have had problems between the sip/FXO lies and was able to kill the
echo by trying different combinations of the echocancel line to 64 (I
think it has settings in 32 bit increments)
Just kept trying different ones till it went away. Here is my config:
group=1
context=line1
signalling=fxs_ks
line I have to play with at the
moment is our main line.
Paul
I have it in production on 8 servers (2 have pri's) no problems.
One remote site keeps crashing when I do a compile so I left it at 1.0.7
The other 6 are hp d220's with fxo's installed.
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On 6/19/05, Bob Goddard [EMAIL PROTECTED] wrote:
On Friday 17 Jun 2005 18:05, Manuel Casal wrote:
Marco Parmeggiani escribió:
Manuel Casal ha scritto:
I made the make menuconfig and make dep in the kernel sources.
i do not remember well how i solved that problem but i'm sure that
On 6/17/05, David Hajek [EMAIL PROTECTED] wrote:
Do you have analog TDM in it?
-David
Oswaldo Arratia wrote:
I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to
configure it using [EMAIL PROTECTED] scripts and did not work, so I went the
long way and
___
I am running on SUSE 9.3x64 with very good results. zttest shows
99.9877 (lowest)
However the distro supplied the 64bit rpms
A.G. (Tony) Nichols
I.S. Manager
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http
VPN's and
Asterisk so any advice will be appreciated.
Thanks,
Mike
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I have 2 clients using the 3com secure gateways... never had a problem yet!
My cisco 2610 and pix hate it... but some people have luck with them.
--
A.G. (Tony
On Tue, 1 Feb 2005 08:08:50 -0600 (CST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I'd like to open up my firwall so that I can connect my SIP phones to a
test server behind or firewall. I can configure an outside addtess to pass
traffic to the internal address of the Asterisk server. I'm not
On Fri, 24 Dec 2004 01:22:47 -0500, Jon Radon [EMAIL PROTECTED] wrote:
Can I ask why? This is clearly the easiest/best way to go about it.
On Thu, 23 Dec 2004 16:21:12 -0500, Tony Nichols [EMAIL PROTECTED] wrote:
The wikki has an example that uses a db
;Login with *801, log out
,2,GotoIf($${autoattendant} = 1?auto|1)
exten = s,3,Dial(Zap/23,30,t)
exten = s,4,Goto(auto|1)
Is there a way to do it without the dbput/dbget?
Thanks,
--
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I.S. Manager
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On Wed, 08 Dec 2004 22:32:16 -0600, Andrew Aken [EMAIL PROTECTED] wrote:
Does anyone have any experience with running asterisk on multi-processor
computers (dual or quad)? Does asterisk on the latest Linux distros take
advantage of the extra processors, or does it predominately utilize a
are plugged in before the device is powered.
Thanks,
Erik
I remember a note on the list about issues with a cisco switch, and
conecting an iaxy. Mine wouldn't grab an ip either (win2k server),
and a cisco 3500. I haven't had time to try a different switch yet.
--
Tony Nichols [EMAIL PROTECTED
On Thu, 2004-11-11 at 20:46, steve szmidt wrote:
On Thursday 11 November 2004 04:39 pm, Geoff Nordli wrote:
[EMAIL PROTECTED] scribbled on :
ok, mathew and other friends
I have this package only and I don't now what a have to do with it I
repeat im a new linux user I don't now how
On Fri, 2004-11-12 at 14:48, JAMES BOTHAM wrote:
Hi there,
I agree with Greg and also with the documentation
group, we are all great at bitching about * (I know I
have done a lot of it but thats because UK and support
for us is minimal or so it feels) we need to unite,
the only reason
Another new article with asterisk/Digium in mind
http://www.onlamp.com/pub/wlg/5909
t o n y
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I would be interested in the script. Did you do zaptel drivers too?
On Wed, 2004-09-08 at 10:41, Martin Mielke wrote:
Hi all,
I just modified one of the startup scripts provided on the tarball to
fit on my SuSE 9.x system to start/stop Asterisk when the system boots
or goes down.
Maybe
On Thu, 2004-08-26 at 09:38, Andrew Kohlsmith wrote:
On Wednesday 25 August 2004 23:50, Steve Underwood wrote:
Several people have reported problems sending faxes from spandsp-0.0.1k
to Canon FAX machines. A spandsp user had the same problem with another
make of FAX machine, and traced the
On Wed, 2004-08-18 at 17:01, Kris Boutilier wrote:
For the inbound digit problem try adding :
debounce=50 ; Needed to reduce the initial off hook
debounce
in the relevant context for those trunks in /etc/asterisk/zapata.conf
Also, are you using 'immediate=yes'?
I'm running 1 9.1 32bit and 1 9.1 64bit. Both are the shipped kernels
and are working very well.
t o n y
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On Tue, 2004-08-17 at 14:08, Jason Kawakami wrote:
Hello all-
So I have * up and running and connected to a legacy system via em_w lines
and have no trouble dialing from * through the tie line but from the PBX
across the tie line I am having intermittant receipt of the DTMF. T-Berd
testing
On Wed, 2004-08-04 at 10:32, Simon Ward wrote:
Hi everyone,
I'm having some problem trying to set up an IAX connection between two *
servers.
The scenario is :
serverA has an X100p card and will direct all calls from the X100p over
IAX to a specific extension on serverB which is at the
On Tue, 2004-08-03 at 08:21, Brian Hudson wrote:
What an ACTIVE newsgroup!
I'm in the early stages of researching Asterisk. My current environment
is a small college (~1000 sets/~400 student sets), Avaya Definity
G3si/Seimens Rolm Phonemail. As you can imagine, the maintenance,
On Mon, 2004-07-26 at 14:47, [EMAIL PROTECTED] wrote:
Hi All
I'm trying to get two Asterisk servers to talk to each other using IAX(2).
I've read the WiKi and the docs and tried the examples.
I can't get it to work (I have 2 x 7960's registering on one server and 1 x 7960
On Mon, 2004-07-26 at 16:03, albyfromg wrote:
Hi all.
We have just setup and asterisk with a 4 line zaptel board with Cisco
7960 and BudgeTone-100 IP phones. All works fine except for this
nagging echo. Whenever I talk, I hear my voice echo back.. This only
happens whenever I talk on an
On Fri, 2004-07-23 at 11:51, Christopher L. Wade wrote:
Tony Nichols wrote:
On Mon, 2004-07-19 at 13:36, Christopher L. Wade wrote:
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20,
I don't think I'll need the U30, but I'm not entirely sure.
Thanks,
Chris
On Mon, 2004-07-19 at 13:36, Christopher L. Wade wrote:
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20,
I don't think I'll need the U30, but I'm not entirely sure.
Thanks,
Chris
I used the DTI-U10 (DTI-24-U10). Got it from GTS Telephone
Inc.(732-323-8620) for
On Fri, 2004-07-16 at 18:45, Andrew Yager wrote:
On 17/07/2004, at 3:24 AM, Eric Wieling wrote:
Tony Nichols wrote:
After calling a bank, or cc processing center; you have to enter
your
social security number, or the cc number - followed by the # key. The
lovely * voice responds
On Fri, 2004-07-16 at 16:34, Christopher L. Wade wrote:
Hi,
I'm am currently in the process of trying to integrate an * box with an
NEC Electra Elite IPK.
Currently, we have 7 POTS lines coming into our building. These lines
are plugged into our NEC using the appropriate analog line
I'm getting down to the last of my * issues ...
After calling a bank, or cc processing center; you have to enter your
social security number, or the cc number - followed by the # key.
The lovely * voice responds transfering I'm sorry that was an invalade
selection. Sometimes the IVR on the
On Thu, 2004-07-01 at 18:27, chouck wrote:
Thanks robert, But im having a problem trying to add a user that can login,
im using a sipura voip box trying to connect to the server and it always
gives me SIP/2.0 403 Forbidden. Under what config can I allow users and
hows it work exactly? Thanks
On Wed, 2004-06-23 at 14:32, asterisk wrote:
Have some errors with the above.
I have tried make and make linux26
Anyone got any clues ? I've googled but only got the make linux26 help
Asterisk compiles and runs great, libpri compiles with no problems.
TIA
Julian.
pbx:~ # cd
Still no go I have asked Digium tech support to look into it. I need
the later cvs to get around a bug with the latest tdm400 card (load
driver - unload driver - load driver again to make it work.
t o n y
On Thu, 2004-06-24 at 08:15, Tony Nichols wrote:
On Wed, 2004-06-23 at 14:32, asterisk
On Thu, 2004-06-24 at 13:01, Joseph wrote:
Just did a new cvs download and then tried to compile.
I get this error message:
chan_zap.c:59:2: #error You need newer libpri
Then there are some more chan_zap.c errors.
Here is the cvs command:
export CVSROOT=:pserver:[EMAIL
I'm trying to get two * boxes to talk no matter what variation I try
I get No Authority Found and connection refused from 192.168.1.5
I've googled, I've site searched to no avail.
Here is the server a configs (192.168.1.5):
iax.conf
[general]
port=5036
bandwidth=low
disallow=all
On Tue, 2004-06-22 at 10:20, David Cook wrote:
So you're saying that the following would be the same?
iax.conf
[YOUR_REC_SERVER]
secret=mysecret
host=my.receiving.server.ca
context=local
extensions.conf
exten = _5XXX,1,Dial(IAX2/YOUR_REC_SERVER/${EXTEN})
If so, what about the
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