What settings are you using for rxgain/txgain? Do you have echotraining=yes on?
On Mon, 28 Jun 2004 00:45:52 -0500, Chris Foster
[EMAIL PROTECTED] wrote:
I Just finished getting HEAD running (had to update to Slackware 9.1
from 9.0 to do it) and much of my X100P's echo is gone.
This is
On 30 Jun 2004, at 05:18, Kevin P. Fleming wrote:
http://www.voip-info.org/tiki-index.php?
page=Asterisk+IAX+authentication
After struggling to understand IAX2 inbound authentication, I figured
I'd document what I learned so others can benefit :-)
Most appreciated. Thanks!
Hi,
-Original Message-
How does one do translation for calls that come in from other
pbx's where the incoming caller ID is an internal extension
number on their pbx? Eg. when I get a call from
Free-World-Dial the CID shows up as 429102 which is
essentially their internal
Todd Lieberman wrote:
Asterisk and dial-up modemsLook at the ZapRAS 'show application ZapRAS'
this only work w/a PRI. TL
And from what I've seen, this will only work ISDN - ISDN. Note the
fact that there is no modem emulation in ZapRAS.
John
___
Ganbaa a écrit :
Hi all,
I would like to call from SIP client to Asterisk then GnuGk, then
Cisco 5300
to PSTN phone. Is this possible? I need simple config asterisk and
gnugk.Can
somebody help me?
Yes. Setup your Cisco as EP in gnuGk, and use the h323 channel from * to
redirect call to GnuGK.
hello,
anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode?
what kernel ?
and second question mISDN driver .. anyone has working solution with
mISDN and maybe fritz card?
what you suggest for DDI - point2point mode (card,kernel,chan_..., ...) ?
thank you,
Tomaz
Hello Tomaz,
Wednesday, June 30, 2004, 10:58:56 AM, you wrote:
T hello,
T anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode?
T what kernel ?
Dunno what DDI is but I'm currently using a HFC card in NT mode
point2point using the package bristuff 0.0.0.2 with fedora core 1 and
We are using the HFC card in point-to-point mode with DDI.
I am using bri-stuff-0.0.2 as well.
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alessio
Focardi
Sent: 30 June 2004 10:12
To: Tomaz
Subject: Re: [Asterisk-Users] zaphfc - hfc pci
Hello Robinson,
Wednesday, June 30, 2004, 11:19:35 AM, you wrote:
RTW We are using the HFC card in point-to-point mode with DDI.
RTW I am using bri-stuff-0.0.2 as well.
Have someone got a list of bristuff compatible ISDN card ?
I have, for example, some DIGI (datafire) cards that have an
There are several cards that use the chipset, and we had to modify the
code to get it to recognise the cards. I have spoken to kapejod and the
next release 0.0.3 of the drivers (due out this week!!) will support
detection of the chipset itself rather than the card ID, so you should
find that the
anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode?
what kernel ?
I have.
I'm using Linux kernel 2.6.7.
what you suggest for DDI - point2point mode (card,kernel,chan_...,
...) ?
In the wiki there's a page where I described how I got Asterisk CVS
working with
Hello Robinson,
Wednesday, June 30, 2004, 11:35:33 AM, you wrote:
RTW There are several cards that use the chipset, and we had to modify the
RTW code to get it to recognise the cards. I have spoken to kapejod and the
RTW next release 0.0.3 of the drivers (due out this week!!) will support
RTW
hi Tim,
what about winbond card ? 6692 or something like that .. this is here
asus isdn pci adapter ..
tnx,
Tomaz
Robinson Tim-W10277 wrote:
We are using the HFC card in point-to-point mode with DDI.
I am using bri-stuff-0.0.2 as well.
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED]
hello Holger,
what card you have? with hfc chipset?
Tomaz
Holger Schurig wrote:
anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode?
what kernel ?
I have.
I'm using Linux kernel 2.6.7.
what you suggest for DDI - point2point mode (card,kernel,chan_...,
...) ?
In the
hi Tim,
what you think this card will work with bri-stuff ? DDI P2P ?
ASUS HN 100 ST D 128K (i think it has winbond chip)
tnx,
tomaz
--
Robinson Tim-W10277 wrote:
There are several cards that use the chipset, and we had to modify the
code to get it to recognise the cards. I have spoken to
http://www.voip-info.org/tiki-index.php?page=Asterisk+IAX+authentication
After struggling to understand IAX2 inbound authentication, I figured
I'd document what I learned so others can benefit :-)
All comments welcome, especially if I misunderstood or improperly
explained anything.
http://www.voip-info.org/tiki-index.php?page=Asterisk+IAX+authentication
After struggling to understand IAX2 inbound authentication, I figured
I'd document what I learned so others can benefit :-)
All comments welcome, especially if I misunderstood or improperly
explained anything.
what card you have? with hfc chipset?
Two D-Link LCS-8051 cards.
Of course with HFC chipset, otherwise I wouldn't use zaphfc ...
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ASUS HN 100 ST D 128K (i think it has winbond chip)
If you think then probably no one can say something for sure.
Use lspci and post the output for this card.
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Is there a way of getting the dialled number from an
AGI? Is it passed in the initial variables, or can it be pulled out or passed
across from the dial plan?
Cheers,
Ben Merrills
Griffin Internet
I get a compile warning when building zaptel (current CVS head) against
2.4.18 kernel (Debian stable dist)
zaptel.c: In function `zt_net_close':
zaptel.c:1238: warning: implicit declaration of function `hdlc_close'
It completes but fails to install with modprobe finding unresolved
references.
Hi Ben,
in my dialplan I use:
exten =
_0.,1,AGI,lcr.agi|${EXTEN:${TRUNKMSD}}
then you can get this argv in your AGI
with:
$number = $ARGV[0];
should give a you a point where to start... If you
have questions please ask!
bye...
Thorsten
- Original Message -
From:
Ben
I'm taking a slight tangent here, but stay with me.
It looks like there are three methods of using HFC-S based ISDN BRI cards
with *. Capi (via capi.conf), zaphfc (via zapata.conf), and isdn4linux (via
modem.conf). Why, and which one is better and for which reasons?
Gary
Notice: Spelling
Anyone have any experience using gr303?
May have a need to interface * to a Siemens Class-5 CO for pstn
trunking (inbound and outbound).
Rich
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I'm taking a slight tangent here, but stay with me.
It looks like there are three methods of using HFC-S based ISDN BRI cards
with *. Capi (via capi.conf), zaphfc (via zapata.conf), and isdn4linux (via
modem.conf). Why, and which one is better and for which reasons?
zaphfc is the best
Am Mi, 2004-06-30 um 12.29 schrieb Holger Schurig:
ASUS HN 100 ST D 128K (i think it has winbond chip)
If you think then probably no one can say something for sure.
Use lspci and post the output for this card.
Just read what it says on the chipset. If it is a Winbond 6692 then
just wait
On the menu General - Language the option for portuguese is wrote in a
wrong way the correct form is portuguese.
Kind regards,
Miguel
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Hi!
It looks like there are three methods of using HFC-S based ISDN BRI
cards with *. Capi (via capi.conf), zaphfc (via zapata.conf), and
isdn4linux (via modem.conf). Why, and which one is better and for which
reasons?
i4l: Trouble with outgoing DTMF (not working), less ISDN features than
Hi!
Is there a way of getting the dialled number from an AGI? Is it passed in
the initial variables,
Yes.
You'll need to do some reading, though, start with Communicating with
Asterisk at http://home.cogeco.ca/~camstuff/agi.html.
Then continue with
Hi,
Anyone know if Asterisk and Digium Hardware supports Centos-3.1 which is clone of
Redhat Enterprise 3.1 server.?
--
Best regards,
Frankie ([EMAIL PROTECTED])
mailto:[EMAIL PROTECTED]
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Asterisk works perfectly fine on RHEL 3 and on whiteboxlinux. So I see
no reason it should not work on centos :)
On Wed, 2004-06-30 at 14:49, Frankie Gravato wrote:
Hi,
Anyone know if Asterisk and Digium Hardware supports Centos-3.1 which is clone of
Redhat Enterprise 3.1 server.?
Anyone know if Asterisk and Digium Hardware supports Centos-3.1 which
is clone of Redhat Enterprise 3.1 server.?
No, it is not **SUPPORTED**, when you don't buy support.
But yes, it is probably running out of the box if you follow the docs.
Linux distros are quite similar from a program's
Thanks for the example. But my question is how does Asterisk know where to
send data if there doesn't seem to be a facility for people to register
their IP to an extension?
On Yaum al-Arbi'a 12 Jumaada al-Awal 1425 04:41 am, administrator tootai
wrote:
Brian Wilkins a écrit :
Hi,
I am
Hello Michael,
Wednesday, June 30, 2004, 8:54:10 AM, you wrote:
MB Asterisk works perfectly fine on RHEL 3 and on whiteboxlinux. So I see
MB no reason it should not work on centos :)
MB On Wed, 2004-06-30 at 14:49, Frankie Gravato wrote:
Hi,
Anyone know if Asterisk and Digium Hardware
Jean-Yves Avenard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I've only been watching this list for the past 2 days.
And it seems to be an one way street:
- -Tell about your problems and what you would like to do.
Usually no answer.
I have to admit I'm rather disappointed with Asterisk,
Rich Adamson wrote:
Goog job. Is the ordering of the if... then... text based on analysis
of the code?
Somewhat, but mostly on Markster's comments to the bug I entered in
Mantis before I knew how this stuff was intended to work.
___
Asterisk-Users
Rich Adamson wrote:
Anyone have any experience using gr303?
May have a need to interface * to a Siemens Class-5 CO for pstn
trunking (inbound and outbound).
I looked into this a couple of weeks back (check the list archives).
There is preliminary GR-303 support in Asterisk CVS, but it appears to
I'm using version 1.9.1 build 3908
- next problem is that the text messages won't reach by another firefly
client
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I had rxgain/txgain set to specific values before the latest echo changes.
After the latest echo changes, I get the best results with:
rxgain=0.0
txgain=0.0
echotraining=800
Thanks,
Ed
On Wed, 30 Jun 2004, Brian McSpadden wrote:
What settings are you using for rxgain/txgain? Do you have
On Wed, 30 Jun 2004, Brian McSpadden wrote:
What settings are you using for rxgain/txgain? Do you have echotraining=yes on?
On Mon, 28 Jun 2004 00:45:52 -0500, Chris Foster
[EMAIL PROTECTED] wrote:
I Just finished getting HEAD running (had to update to Slackware 9.1
from 9.0 to do it)
Anyone have any experience using gr303?
May have a need to interface * to a Siemens Class-5 CO for pstn
trunking (inbound and outbound).
Rich
I assume Siemens Class5=EWSD.
EWSD is compatible with GR.303, and AFAIK it works with special national
project.
Which software version (APS) and
I had zaptel and asterisk running on the 2.6.6 kernel of Fedora code 2,
but I had to manually update my kernel source to the 2.6.6 source
before I could get the modules to compile and run. YUM and up2date
didn't automatically update the kernel source on my system because the
rpm name changed
On Tue, 2004-06-29 at 23:35 -0400, Steve Totaro wrote:
I am trying the same thing. Now I get this message in /var/log/messages
after trying modprobe zaptel.
kernel: zaptel: version magic '2.6.5-1.358custom 686 REGPARM 4KSTACKS
gcc-3.3' should be '2.6.5-1.358 686 REGPARM 4KSTACKS gcc-3.3'
Patrick J. Conroy wrote:
I wasn't able to get debugging information the first time around either.
After pulling the latest asterisk from CVS, I was able to build and see
debugging information when I started asterisk to test using
asterisk -vvgc. But I noticed today that I do not get the same
Brian Wilkins a écrit :
Ok sorry, I just re-read your explanation. So what I am getting from this is
that * cannot act as a registration server for h.323, but can route calls
from h.323 to sip. Whenever a call comes into the GK, to a prefix, that it
does not recognize (SIP ext), it forwards it
I tried running those commands, though it didn't do too much. Asterisk
stopped with the same errors as in my original email.
=== start of log ===
[EMAIL PROTECTED] cepcion]# /sbin/modprobe zaptel
[EMAIL PROTECTED] cepcion]# /sbin/modprobe wct1xxp
[EMAIL PROTECTED]
Hello!
I have installed the modified prepaid application and its working god. the
only problem is that when I finish the call it does not update the balance
of the card.
any one has any idea how this could be fixed?
best regards
Hekuran
___
On Tue, 2004-06-29 at 12:55, Paul Concepcion wrote:
[EMAIL PROTECTED] root]# cat /etc/asterisk/zaptel.conf
span=1,0,0,esf,b8zs
fxoks=1-24
This file should be in the /etc directory not the /etc/asterisk
directory.
--
Steven Critchfield [EMAIL PROTECTED]
...my VoicePulse Connect account is timing out on its login requests.
Was working fine a hour ago.
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. [EMAIL
Thank you very much. I just copied over the file and now * has detected
all the channels.
Steven Critchfield wrote:
On Tue, 2004-06-29 at 12:55, Paul Concepcion wrote:
[EMAIL PROTECTED] root]# cat /etc/asterisk/zaptel.conf
span=1,0,0,esf,b8zs
fxoks=1-24
This file should be in the /etc
Hi,
I get this error when trying to dial an outbound extension from a sip
phone:
-- snip --
-- Executing Dial(SIP/2003-02d1, OH323/[EMAIL PROTECTED]|20) in new stack
-- H.323 call to [EMAIL PROTECTED] with codec ALAW
-- Called [EMAIL PROTECTED]
0:33.283 H225
ok,
here is something to add .. correct me if I'm wrong!
tomaz
Philipp von Klitzing wrote:
Hi!
It looks like there are three methods of using HFC-S based ISDN BRI
cards with *. Capi (via capi.conf), zaphfc (via zapata.conf), and
isdn4linux (via modem.conf). Why, and which one is better and for
Thank You! This was just the info I needed...I was
wondering why my kernel source didn't get updated on FC2 by YUM, but I just
haven't had time to track it down.
Thanks,
Ed
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
kbrownSent: Wednesday, June 30, 2004 6:54 AMTo:
Am Mi, 2004-06-30 um 17.21 schrieb Tomaz:
ok,
here is something to add .. correct me if I'm wrong!
ok, i will! ;-)
chan_capi: more features (early dial, call deflection, ISDN hold
retrieve etc), stable, comes with echosquelch, works only with cards that
have CAPI driver support; far
Can anyone tell me how (and for how long) asterisk remembers the IP address
for an IAX2 peer? Voicepulse has been going up and down for me, and it seems
to have something to do with the IP address changing. Is there a way to
force asterisk to run gethostbyname() again for the peer? Or do I just
Brian Wilkins a écrit :
Hi,
I get this error when trying to dial an outbound extension from a sip
phone:
-- snip --
-- Executing Dial(SIP/2003-02d1, OH323/[EMAIL PROTECTED]|20) in new stack
-- H.323 call to [EMAIL PROTECTED] with codec ALAW
-- Called [EMAIL PROTECTED]
0:33.283
On 30 Jun 2004 at 10:17, Michael Graves wrote:
...my VoicePulse Connect account is timing out on its login requests.
Was working fine a hour ago.
Michael
I can confirm that this is the case from here in New Zealand too.
Matt
___
Asterisk-Users
What is particularly weird is that if I connect a call to Echo on my local
server, I don;t usually get this oscillation, however if I connect it to
Echo on another Asterisk server (over IAX), it does happen. The main
difference will be the delay in the echo, as the software versions are
within the
Hi all.
We've been working on this with PRI's from USLEC connecting
to a Cisco AS5300 as the gateway. We are sending the CNAM
information, but they can't read it dynamically. We are
stuck (for now) with manually sending in database update
requests.
Having said all that, we're looking at
Same here in Minnesota, USA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, June 30, 2004 10:46 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 10:10am CST - VoicePulse
appears to be down
On 30 Jun
Ditto here. I can ping but not log in.
--Ernest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, June 30, 2004 8:46 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 10:10am CST - VoicePulse
appears to
When I call into my system I have it set to play a bunch of different
sound files (I'm doing testing right now), but when it connects there is
just nothingness for about the time the sound file should take to play.
A soundcard is installed on my system and working properly. Last night
I
Also when I use '!' to execute a shell command from the * CLI a message
comes up saying sox: Can't open output file '/dev/dsp': Device or
resource busy.
Andrew Elchuk wrote:
When I call into my system I have it set to play a bunch of different
sound files (I'm doing testing right now), but
So, in order to use the parking extension configured in parking.conf,
I have to configure that extension under a [parkedcalls] context in my
extensions.conf? I thought the call parking app was supposed to take
care of that for me?
On Tue, 29 Jun 2004 23:49:54 +0100, Craig Waddington
[EMAIL
Thanks.
I will try.
Jorge
Russ Beaupre, P.E. wrote:
Jorge Mendoza wrote:
Hi,
I'm testing a Polycom IP600.
With firmware version 1.1 the phone reboots at any time.
With firmware version 1.2, the first reboot was an endless reboot.
Then I moved the phone to another lan port, then it worked fine.
Hi
there.
I am trying to
connect Asterisk to a local danish ip-telephony provider. But is having some
difficulties. First I thougt they were related to the provider. But then i
started debugging on the Asterisk (aix2 debug)
When I make a call
using AIX to the provider everything seems to
Tor,
Yes, it sounds strange to me too.
The phone is registered with *.
After 24 hrs of inactivity the phone is still working fine (yesterday
was holiday). The problem occurred just 1 time. We are suspecting the
power adapter. If the problem arise again I will post the cfg files.
Thank You for
Hi,
Is anybody in the UK using Telewest as
a PRI Telco provider?
Are you sending them caller ID?
I've been told by Telewest that:-
Oftel doesn't allow them to accept caller ID
(this is rubbish, and I replied pointing out where in the link to Oftel
that they sent me
Martin Kiefer a écrit :
Hi there.
I am trying to connect Asterisk to a local danish ip-telephony
provider. But is having some difficulties. First I thougt they were
related to the provider. But then i started debugging on the Asterisk
(aix2 debug)
When I make a call using AIX to the
The H323 asterisk channel provide a bridge between SIP/IAX EP
and H323 EP and doesn't act as an h323 gatekeeper. See GnuGK for this.
But isn't that exactly what I am trying to do?
I am calling from OpenPhone to an AIX number. I thought that Asterisk
would make a bridge betveen the AIX and
Hiya,
I sent this bugfix to the asterisk-dev mailing list, and modified it as I
noticed side effects, but now it appears to be finished. Nobody seemed to
notice it there, so I thought I'd post here, as it seems to be something
that will be needed as people update to the latest CVS version.
Title: Message
Hello
all,
I am building a
software based on asterisk to handle incoming answering service
calls.
I have one problem
that I have not been able to figure out a reasonably pricedsolution
to:
The goal of this
software is to allow the agent to be able to do their entire job
administrator tootai wrote:
You will never make it work, openphone is h323 and can't be an asterisk
EP. You should setup a gatekeeper instead asterisk (or parallel to).
Say what? OpenPhone MOST CERTAINLY can make/receive calls to/from
Asterisk with either asterisk-oh323 or chan_h323.
The H323
I am encountering the same issues EXACTLY with a GrandStream BT101. I had to
downgrade to the CVS as of 6/7 in order to get the grandstreams to work. It
seems that a change thereafter is causing this issue...
it only happens when called from PSTN - Asterisk. Calls through asterisk
between phones
Hello,
I know that there was a previous posting regarding this in which the
addition of:
CFLAGS+=-I../asterisk/include
fixed the problem, but I have tried that and am still getting the following
when trying to run make:
./mkdep -fPIC -I../asterisk/include/asterisk -I../asterisk -D_GNU_SOURCE
With these phones, power surges will cause reboots. I've also had
problems with bad ethernet cables and the like causing reboots. On one
phone, I resolved (most) of the reboot problems by putting a hub between
the phone and the rest of the net. On another, I put the ethernet
through the surge
Title: Message
Look
at the 7905G phone from Cisco.
TL
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Michael
Blood, Matraex, Inc.Sent: Wednesday, June 30, 2004 2:50
PMTo: [EMAIL PROTECTED]Subject:
[Asterisk-Users] Answering Service
Title: Message
Does anyone know of a solution where I would be able to
setup some sort of permanent connection to the asterisk server via IP?
I
can't have a dial tone in their ears constantly and I need to find a phone or
solution which is $150 or less (preferably under $100) per
Andres wrote:
Ernest W. Lessenger wrote:
Can anyone tell me how (and for how long) asterisk remembers the IP
address
for an IAX2 peer? Voicepulse has been going up and down for me, and it
seems
to have something to do with the IP address changing. Is there a way to
force asterisk to run
At 12:02 PM 6/30/2004, you wrote:
I am encountering the same issues EXACTLY with a GrandStream BT101. I had to
downgrade to the CVS as of 6/7 in order to get the grandstreams to work. It
seems that a change thereafter is causing this issue...
it only happens when called from PSTN - Asterisk. Calls
try
cvs checkout -D 04/01/04 asterisk-addons
with the latest cvs of asterisk
harold
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Darrin
Johnson
Sent: Wednesday, June 30, 2004 2:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
So then don't tell the EP to register with a GK, just send
calls to an
H.323 Gateway, Asterisk.
And that is exactly what I am trying to do. But with no luck.
Best regards
Martin Kiefer
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Michael Blood, Matraex, Inc. wrote:
Hello all,
I am building a software based on asterisk to handle incoming
answering service calls.
I have one problem that I have not been able to figure out a
reasonably priced solution to:
The goal of this software is to allow the agent to be able to do
I am forwarding my calls from my packet8 phone number to my Free World
Dialup account using the Packet8 FWD interconnect codes. I have asterisk
registered with my FWD account via IAX2 and have also tried with SIP. When
a call comes in Asterisk interprets any DTMF tones twice. IE: someone
types 1
Hi,
I am using asterisk CVS 2004-06-16 with oh323-0.6.3a
I have a strange problem if I start asterisk with oh323 loaded
/usr/sbin/asterisk -vc
once I am in the console and issue restart now or reload asterisk hangs
and it not stoping or restarting at all, below is the console logging when
Folks,
My question concerns the SIP Notify that is being sent to ...
device. You can see it in the following line:
Voicemail: 0/0
Shows no Voice mail but I did leave a voice mail at the extension.
Any suggestion on what I should look for in my * setup. I am not
worried about the 481
The Ipeya iGate-4-FXO-SIP works with asterisk ?
Are there anyone working with it ?
Kind regards,
Miguel
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Hi,
We are having an issue here. It seems that whenever we initialize Asterisk
on our network, the router that the Asterisk server is connected to (Cisco
7200) crashes and loses it configuration. This has happended five times and
each time we have tested it, it is always when Asterisk
On Jun 30, 2004, at 2:05 PM, Brian Wilkins wrote:
Hi,
We are having an issue here. It seems that whenever we initialize
Asterisk
on our network, the router that the Asterisk server is connected to
(Cisco
7200) crashes and loses it configuration. This has happended five
times and
each time we
[EMAIL PROTECTED] wrote:
Hi,
We are having an issue here. It seems that whenever we initialize
Asterisk on our network, the router that the Asterisk server is
connected to (Cisco 7200) crashes and loses it configuration. This
has happended five times and each time we have tested it, it is
I am forwarding my calls from my packet8 phone number to my Free World
Dialup account using the Packet8 FWD interconnect codes. I have asterisk
registered with my FWD account via IAX2 and have also tried with SIP. When
a call comes in Asterisk interprets any DTMF tones twice. IE: someone
types 1
IOS version 12.xx
As far as a traceback, that's going to be difficult now since we've removed it
from our switch and brought it back here to the office for testing. When we
test it tomorrow or later in the week, I'll see if it crashes again in a test
setting and try to get a traceback to the
I've seen a similar problem caused by the Ethernet card in the server.
Everytime there was any load, it would crash the Cisco.
Changing to a different brand Ethernet card resolved the problem.
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Brian Wilkins [EMAIL
Jeremy McNamara a écrit :
administrator tootai wrote:
You will never make it work, openphone is h323 and can't be an
asterisk EP. You should setup a gatekeeper instead asterisk (or
parallel to).
Say what? OpenPhone MOST CERTAINLY can make/receive calls to/from
Asterisk with either
On Jun 30, 2004, at 2:31 PM, Brian Wilkins wrote:
IOS version 12.xx
As far as a traceback, that's going to be difficult now since we've
removed it
from our switch and brought it back here to the office for testing.
When we
test it tomorrow or later in the week, I'll see if it crashes again in
a
We are having an issue here. It seems that whenever we initialize Asterisk
on our network, the router that the Asterisk server is connected to (Cisco
7200) crashes and loses it configuration. This has happended five times and
each time we have tested it, it is always when Asterisk starts
how could any prepaid application be good if it does not update the balance
:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hekuran Doli
Sent: Wednesday, June 30, 2004 9:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] prepaid application
Folks!
I dont know whether anyone has done this exercise before of putting together a
Wish-List of things that you want to do, if you have all the gadgets you need and have
a client base that needs Asterisk's Features and more. Here are some of the scenarios
I am playing out that I will do,
As far as loosing the configuration...the only reason I could see that
happening is if you either are doing one of the two... not saving the
configuration...or you have the configuration register set to something like
0x2142. look on show version for the configuration register. it should be
hello !
My problem is:
Astriks should create a connection to other members using a german Sip
provider (www.sipgate.de).
there are no problems with connections to:
o Sip- Accounts
o national phone numbers
o mobile phone numbers
but connections to international phone numbers DO NOT WORK (see
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