I had (and still have) similar problem. Once SPA 2000 registers with
* it all works well for few minutes. After that all incoming calls
are not answered by SPA 2000. Is that what you mean?
If so, I have temporaraly got SPA 2000 to re-register every 3
minutes. This seems to work at the
What SCO code in any kernel, when all's said and done, it probably is the
other way around.
At 23:57 2/03/04 -0500, you wrote:
Does
anyone know if they took out SCO's code in
Linux 2.6 kernel ?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Alex Lopez
Sent: Tuesday, March
Got my test box up and running.
Celeron 2Ghz 1Gb Ram 2x32Gb Hdd, X100P card 2xGrandStream phones.
I've got everything working calls come in get processed, delivered to the
extension...
But what I can't figure out is how to pass the CallerID Name instead of the
CallerID number?
I've been banging
Hi,
But what I can't figure out is how to pass the CallerID Name instead of the
CallerID number?
use this: ${CALLERIDNAME}
Find more info at: /usr/src/asterisk/doc/README.variables
Regards, Girish
_
Need a job? Get head-hunted by
Can anyone advise from experience what size of PC would be needed
to support two TE405P 4xE1 cards to provide conference bridging
for up to 20 concurrent conferences of 10 participants each?
All the participants would be on the E1 trunks, not VoIP.
Thanks in advance,
Tony
--
Tony Mountifield
A large one!!
Sorry I couldn't resist that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tony
Mountifield
Sent: Wednesday, 3 March 2004 8:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Size of PC for conferencing?
Can anyone advise from
Sorry Floks for the noise.. it seems that this is only when using the
sound card. Never found a reference to this is the archives or docs so I
thought this reply might help some other newb.
Regards,
Andrew
On Sat, 28 Feb 2004, Andrew McRory wrote:
Hi Floks,
I am just starting with *
Hi All,
As ztdummy seems to work fine with mp3 playback, etc I would assume it
will also work with IAX trunking. Is anyone using trunking successfully
with ztdummy and are there any problems I should be aware of?
Regards,
Nathan.
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When calling out on a Cisco 7960 there is a short delay before the call
gets setup and the other side can hear your voice.
Anyone know how to compensate for this effect?
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Hi all
Does anyone know where I can get hold of the German 1TR6 ISDN signalling
protocol specification.
Thanks
Jason
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John,
Others have commented already relative to * not routing Notify messages
so I'll pass on that.
Currently, my application can send NOTIFY messages directly to the
phones in order to turn on and off MWI, but I would really like to be
able to send these messages to * and have it handle
On Wednesday 03 March 2004 13:55, James Sizemore wrote:
When calling out on a Cisco 7960 there is a short delay before the call
gets setup and the other side can hear your voice.
Anyone know how to compensate for this effect?
Open Caveats Release 6.2
This section documents possible
If you want a freely usable implementation of SRTP look at
srtp.sourceforge.net.
Regards,
Steve
John Todd wrote:
I have found few VoIP clients that support encryption. The only one
that comes to mind is the Zultys devices (they have a softphone and a
hardphone that support SRTP.) I spoke
Hi
Just one question
do any of the Digium T1/E1 cards do DPNSS signaling?
Robb
--
Robert Boardman
Tel:01617737929
IAXTel:17007737929
FWD:82623
--
Robert Boardman
Tel:01617737929
IAXTel:17007737929
FWD:82623
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-Original Message-
From: [EMAIL PROTECTED] on behalf of Warren H. Prince
Sent: Tue 02-Mar-04 21:33
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Motorola / Vanguard, H.323 and Asterisk
As I write this, I'm trying to imagine what you have in mind. Bring
FXO's and FXS's into the
Reid A. Forrest wrote:
-Original Message-
From: [EMAIL PROTECTED] on behalf of Warren H. Prince
Sent: Tue 02-Mar-04 21:33
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Motorola / Vanguard, H.323 and Asterisk
As I write this, I'm trying to imagine what you have in mind. Bring
Robert Boardman wrote:
Hi
Just one question
do any of the Digium T1/E1 cards do DPNSS signaling?
Robb
Just one answer. No. :-(
Steve
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Jason Penton wrote:
Hi all
Does anyone know where I can get hold of the German 1TR6 ISDN signalling
protocol specification.
Thanks
Jason
Is that still used? I thought they were 100% CTR4 these days.
Regards,
Steve
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Hi,
One of my CPU boards is MS-5169(ATX AL9 mainboard). The chipset is
Aladdin M1531/M1543. The USB is 'usb-ohci'. ztdummy.c and ztdummy.h
uses 'usb-uhci', so I changed from 'uhci' to 'ohci' in ztdummy.c and
ztdummy.h. and tried /sbin/modprobe ztdummy, never succeeded.
Is it impossible to use
I think what James is referring to is the delay once the call already been dialed.
It's not specific to Ciscos, as I'm experiencing the same problem on my polycom
phones. Must be SIP related.
The problem is that once a call is dialed, when the remote party picks up the phone,
the first half
Hi
Just one question
do any of the Digium T1/E1 cards do DPNSS signaling?
No.
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Bisker, Scott (7805) wrote:
I think what James is referring to is the delay once the call already
been dialed. It's not specific to Ciscos, as I'm experiencing the
same problem on my polycom phones. Must be SIP related.
The problem is that once a call is dialed, when the remote party
picks up
I think the zaptel compile depends on a softlink 'linux-2.4' pointing to
'linux-2.4.25' (or whatever). Your compiles may be including .h files
from an older source tree.
Mike
Francois wrote:
On Thu, 5 Feb 2004 22:32:34 -0500, Tim Sailer [EMAIL PROTECTED] wrote:
Does anyone have the zaptel
dkwok wrote:
Has anyone tried to hang GS101 phones on a wall?
It has recess holes at the back of the base where you can hang it on a
wall. What it lacks is that the handset is not supported for this
upright position.
Has anyone done any modification on it? I was thinking about velco the
Hi All
I want to configure my * box for my idsn 2 line which I ordered from KPN
(Netherlands).
Does anyone have any configuration for this that can help me?
Thanks
Mark
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Hey Steve
Apparently so :-(. It is used in our legacy PBX with which I would like to
connect my Asterisk box.
Cheers
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Underwood
Sent: 03 March 2004 03:56 PM
To: [EMAIL PROTECTED]
Woops, should have said symbolic link :-)
cd /usr/src
ln -s linux-2.4.25 linux-2.4
Cheers
Michael Welter wrote:
I think the zaptel compile depends on a softlink 'linux-2.4' pointing to
'linux-2.4.25' (or whatever). Your compiles may be including .h files
from an older source tree.
Mike
Tony Mountifield wrote:
Can anyone advise from experience what size of PC would be needed
to support two TE405P 4xE1 cards to provide conference bridging
for up to 20 concurrent conferences of 10 participants each?
From working with 2 TE410P in one PC I can tell you that it will not
work. And
I have had some success at getting ringback tone
working with the X100P card. Unfortunately, it will not meet my
requirements.
The following provides ringback tone for an
incoming caller. Please note that the Zap channels are configured to use the
inbound context.
[inbound]
exten =
Hi,
-Original Message-
I want to configure my * box for my idsn 2 line which I
ordered from KPN
(Netherlands).
Does anyone have any configuration for this that can help me?
Sure, I use this a lot at home. What channel type (ISN card) are you using ?
Mail me off-list if you
htguy wrote:
Got my test box up and running.
Celeron 2Ghz 1Gb Ram 2x32Gb Hdd, X100P card 2xGrandStream phones.
I've got everything working calls come in get processed, delivered to the
extension...
But what I can't figure out is how to pass the CallerID Name instead of the
CallerID number?
I've
Hi Mark,
On Wed, 2004-03-03 at 15:08, Mark wrote:
I want to configure my * box for my idsn 2 line which I ordered from KPN
(Netherlands).
Should be no problem.
Does anyone have any configuration for this that can help me?
Need more input.
The software configuration depends (of course) on
On Wed, 2004-03-03 at 14:57, Zen Kato wrote:
Hi,
One of my CPU boards is MS-5169(ATX AL9 mainboard). The chipset is
Aladdin M1531/M1543. The USB is 'usb-ohci'. ztdummy.c and ztdummy.h
uses 'usb-uhci', so I changed from 'uhci' to 'ohci' in ztdummy.c and
ztdummy.h. and tried /sbin/modprobe
On Wed, 2004-03-03 at 00:34, Hermann Wecke wrote:
On Thu, 5 Feb 2004, Tim Sailer wrote:
Does anyone have the zaptel modules built for Debian 2.4.24 kernel?
Someone here is running * on debian?
I have * running on Debian stable. I back-ported the zaptel and
asterisk packages from testing.
Hi Ariel-
I wonder if you could please expand on that a little? What was your
configuration for conferences when you had the problem (how big were the
conferences, were there errors in the /var/log/asterisk/messages file, etc)
I do have problems in a setup with two TE410P's, although my
Folks,
I have subscribed to galaxyvoice for $20 and so far everything is fine as
long as I use the Grandstream phone I bought from them.
What I want to do is use *. They claim that they support SIP and that I
can use any SIP client with them. However, their tech support sucks and
I'm unable to
RTP stream not passed through the * server in case of SIP(Sipura)
H323(Cisco) traffic (Grand Stream was doing OK so we suspected SIPURA).
After SIPURA firmware upgrade (What ever latest) started working
correctly. No confirmed reason but Firmware upgrade did the trick.
- SamW
-Original
Try
[phone]:[password]:[EMAIL PROTECTED]
this is what I had to use for one of the providers I use (iconnecthere)
Martin
On Wednesday 03 Mar 2004 3:00 pm, Mark Phillips wrote:
Folks,
I have subscribed to galaxyvoice for $20 and so far everything is fine as
long as I use the Grandstream
Earlier on I read that there is an IAX image for the SNOM 200. Is this true? Does it
work? Where might I get this?
Michael
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Does the fax detection only work with an X100P or Zaptel card? Will
asterisk auto detect a fax on an incoming SIP call from a Background
menu?
Thanks,
Kevin
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Email back from ipblue:
Thank you for your inquiry. At the present time we work under SCCP or
H.323 protocols. We are developing a SIP phone which will be available
sometime in Q2.
Regards,
Andrew Schecter
Vice President of Sales
IP blue Software Solutions
15 East 26th Street
New York, NY
Sorry for such a basic question, but Googling and wiki searches haven't
lead me anywhere.
Can a called phone transfer a call to another number?
In detail, I have ISDN/BRI via chan_capi - * - SIP to some xlite
workstations.
All my dial() strings have tT on the end of them, typical is:
exten =
When I upgraded from OSS to the ALSA soundsystem, I could no longer get
noice from the microphone. I changed my modules.conf to;
load = chan_alsa.so
noload = chan_oss.so
And my alsa.conf to;
input_device=hw:0,24
output_device=default
or
input_device=hw:0,0
Anyone else seeing SIP registration requests rejected by FWD? I don't seem
to be able to register any longer - even though my SIP config remains the
same.
Iain
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Hello Oliver,
okay, this was not easy and will make a long e-mail that I will also CC
to the list. I will answer in English because it is my native language.
I lived in Germany for 2.5 years and can speak German okay,
however I will spare you all of the declination failures that I make
on a
Alright, this may seem like something relatively easy to do
but I must be missing something or had a neuron misfire. I am trying to
get
The Status lights on my Snom200 hardphones to display the
status of each one of my PSTN lines in my Asterisk server.
Current Config:
3 X100P cards
On Tuesday 02 March 2004 22:30, Alex Lopez wrote:
I don't believe this!! SCO got some one to pony up 7 figures!!
Please don't post off-topic crap like this. I get enough of this on
other lists.
-Tilghman
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On Wed, 2004-03-03 at 17:44, Iain Stevenson wrote:
Anyone else seeing SIP registration requests rejected by FWD? I don't seem
to be able to register any longer - even though my SIP config remains the
same.
Same here. Error message is:
Mar 3 17:57:30 NOTICE[147466]: chan_iax2.c:3209
On Wed, 2004-03-03 at 17:44, Iain Stevenson wrote:
Anyone else seeing SIP registration requests rejected by FWD? I don't seem
to be able to register any longer - even though my SIP config remains the
same.
Sorry for my previous post... it's my iaxtel registration that fails!
My FWD
Robert Sprockeels wrote:
Mar 3 17:57:30 NOTICE[147466]: chan_iax2.c:3209 authenticate: Asked to
authenticate to 69.73.19.178 with an RSA key, but they don't allow RSA
authentication
I've found FWD to either work or not, it's been on and off for the last
few days...
which is why i turned to enum
From: Iain Stevenson [EMAIL PROTECTED]
Anyone else seeing SIP registration requests rejected by FWD?
I don't seem
to be able to register any longer - even though my SIP config
remains the
same.
Iain
Yes, me too - for about the last week I'd guess. I'm guessing that it is
this
Sorry if this is a daft question but when a PSTN call comes in on my
X100P the console shows the following;
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2
(Ring/Answered)...
- Original Message -
From: Patrick Lidstone (Personal E-mail) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 03, 2004 5:28 PM
Subject: [Asterisk-Users] FWD registration faillures
From: Iain Stevenson [EMAIL PROTECTED]
Anyone else seeing SIP registration requests
Iain Stevenson wrote:
Anyone else seeing SIP registration requests rejected by FWD? I don't
seem to be able to register any longer - even though my SIP config
remains the same.
I wouldn't worry about it - FWD goes through phases of failed
registrations. It's a very highly used service, and
I will respond off list, to conserve bandwidth.
I feel that this is VERY ON-TOPIC. If this continues, we will be faced
with customers shying away from Linux due to the whole FUD factor (fear,
Uncertainty, and Doubt).
We have all made a commitment be it financial or simply time; we all
have an
Hey all,
Doing a search of the mailing list archives turns up a couple requests for
support of an outgoing SIP proxy, and the following wishlist request:
http://bugs.digium.com/bug_view_page.php?bug_id=359
I'm trying to get a Vonage softphone account working, and it's mostly
working, but
If people on the list have ways to present to my customers ways to help me
sell this product to my customer due to concerns about SCO. I want to hear
it.
I do agree that having general discussions about it is not what this list is
meant for.
Michael J. Mimbach II
[EMAIL PROTECTED]
WNOC /
Hi:
Someone know wich is the best firmware for the ATA 186 with *
Thanks
Erick
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Mee too with the same problem
Patrick Lidstone (Personal E-mail) said:
From: Iain Stevenson [EMAIL PROTECTED]
Anyone else seeing SIP registration requests rejected by FWD?
I don't seem
to be able to register any longer - even though my SIP config
remains the
same.
Iain
Yes, me too
Chances are it's waiting to get the caller ID info (sent between the
first and the second ring)
On Wed, 2004-03-03 at 12:01, WipeOut wrote:
Brian Mulligan wrote:
Sorry if this is a daft question but when a PSTN call comes in on my
X100P the console shows the following;
NOTICE[1217602880]:
I've read and tried
a LOT of sample config's for sip.conf and extensions.conf and no matter what I
do I get registration error's when trying to get SJphone registered to my *
server. I have a XP VMware host with Redhat 9 / * as a guest. The
SJphone is on the host XP trying to register with
On Wed, 2004-03-03 at 06:33, Robert Boardman wrote:
Hi,
With the current CVS as of last night 20:00GMT
I was testing a asterisk with the e100p card using a PRI analyzer to excerise
the 30 channels over and over, just going directly to voice-mail.
Basically, I don't know what is going on
Hit reply but did not change address!!
None the less, I understand your point and respect it.
All I ask is for the same respect.
Alex
Message: 6
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #2989 - 8
msgs
Date:
On Wed, 2004-03-03 at 12:34, Jennings, Mike wrote:
I've read and tried a LOT of sample config's for sip.conf and
extensions.conf and no matter what I do I get registration error's
when trying to get SJphone registered to my * server. I have a XP
VMware host with Redhat 9 / * as a guest. The
Hello all,
Asterisk is segfault dying when I try to park a call from an extension
dialed from an AGI script.
The situation is as follows:
I call from a sip phone (really It doesn't matter if It's SIP or not) to
extension 181 (corresponding to a mgcp DG-104S phone).
.
exten =
v2.16.2 ata18x
Works Fine for me.
- Original Message -
From: Erick Weber V. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 03, 2004 1:18 PM
Subject: [Asterisk-Users] Best ATA 186 Firmware
Hi:
Someone know wich is the best firmware for the ATA 186 with *
Yup, that was it. Set usecallerid=no and it rings right out.
Thanks
On Wed, 2004-03-03 at 18:32, Eric Wieling wrote:
Chances are it's waiting to get the caller ID info (sent between the
first and the second ring)
On Wed, 2004-03-03 at 12:01, WipeOut wrote:
Brian Mulligan wrote:
Sorry
I'd like to get Music on Hold working (so when I hit the 'Hold' button on
my pingtel phone the caller gets music until I pick the phone back up); I
can't seem to find any sample config files on how to do this. I've tested
the music on hold subsystem using:
exten = 3000,1,Answer
exten =
Title: Professional Text to Speech
Hi everyone,
I was wondering about peoples experiences with professional text to speech packages, and their integration with Asterisk? Festival at least in the default install sounds... less than perfect. Any suggestions?
Matt
Hello again
After a few minutes of thinking (usefull sometimes :) I solved the
problem of using the AGI to make the dialing decision while avoid doing
the dial from inside the agi application without changing context (to
keep access to other extensions using transfer).
Very simple, using SET
It would be nice to have an option to delete voicemail messages that reached
a certain age, or to delete those that are delivered by e-mail. I gather
from searching previous posts that this has not yet been done, and the
solution is to set up a chron job.
Steven Critchfield wrote:
On Wed, 2004-03-03 at 06:33, Robert Boardman wrote:
Hi,
With the current CVS as of last night 20:00GMT
I was testing a asterisk with the e100p card using a PRI analyzer to excerise
the 30 channels over and over, just going directly to voice-mail.
Basically, I don't
Hi,
I thought it would be neat to put my SIP/IAX reachable systems into
the ENUM system.
But reading about it I see that its rather centrally controlled within
the ITU.
My country code (+27) is not delegated. My country has a monopoly
telco whose only interest in VOIP is to keep it all to
On Wed, 2004-03-03 at 14:36, Robert Boardman wrote:
Steven Critchfield wrote:
On Wed, 2004-03-03 at 06:33, Robert Boardman wrote:
Hi,
With the current CVS as of last night 20:00GMT
I was testing a asterisk with the e100p card using a PRI analyzer to excerise
the 30 channels over
On Wednesday 03 March 2004 14:34, Jim Sneeringer wrote:
It would be nice to have an option to delete voicemail messages
that reached a certain age, or to delete those that are delivered
by e-mail. I gather from searching previous posts that this has not
yet been done, and the solution is to
Hi,
We have just upgraded from a Sdx Index V200 PBX to Asterisk and are having a
few problems.
We have 1 ISDN PRI (E1) provided by BT (UK) for incoming and outgoing calls.
Incoming calls work fine and the are no alarms on back of card or in
/proc/zaptel/1, but with outgoing calls,
all numbers
On Wed, Mar 03, 2004 at 10:40:09PM +0200, Stephen Davies wrote:
So - what to do? If I approach the administrators for e164.arpa
([EMAIL PROTECTED], apparently) will they delegate 7.2.e164.arpa
to me?
I guess that they won't. (It would be fun if they
would, for some definition of fun (I
check with bt what kind of dialplan they use and set it with pridialplan= in
zapata.conf
On Wednesday 03 of March 2004 22:08, Michael East wrote:
fine and the are no alarms on back of card or in
/proc/zaptel/1, but with outgoing calls,
all numbers are rejected with the BT error The number you
v.3.0 works fine too
James Coberly wrote:
v2.16.2 ata18x
Works Fine for me.
- Original Message -
From: Erick Weber V. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 03, 2004 1:18 PM
Subject: [Asterisk-Users] Best ATA 186 Firmware
Hi:
Someone know wich is the best
I seem to be having problems doing an update from Digium's CVS. Has anybody
heard anything? I got no response from anybody on the IRC channel.
Cheers,
Steven
Steven Sokol
Owner/Manager
Sokol Associates, LLC
Phone: 816.822.1807
IaxTel: 700.613.9004
Web:http://www.sokol-associates.com
We have 1 ISDN PRI (E1) provided by BT (UK) for incoming and outgoing
calls.
Incoming calls work fine and the are no alarms on back of card or in
/proc/zaptel/1, but with outgoing calls,
all numbers are rejected with the BT error The number you have dialed has
not been recognized, please
On Wed, 2004-03-03 at 16:06, Steven Sokol wrote:
I seem to be having problems doing an update from Digium's CVS. Has anybody
heard anything? I got no response from anybody on the IRC channel.
Check your DNS. Verify you are using DNS. Earlier this week(I think)
there was 2 servers in the
I've been having trouble getting updates recently, but it does eventually
go through.
Thanks. It finally went through for me.
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Hi,
I have a client that wants to set up an Asterisk-based VOIP solution. While
we can easily handle most of their IT needs, we've never really used
Asterisk.
We're looking for an experienced Asterisk tech to give us some help getting
the box up and running, configured appropriately.
Once up
On Wed, 3 Mar 2004, Nate Carlson wrote:
I'd like to get Music on Hold working (so when I hit the 'Hold' button
on my pingtel phone the caller gets music until I pick the phone back
up); I can't seem to find any sample config files on how to do this.
I've tested the music on hold subsystem
Hi
Just got a brand new Box Cisco VIP30 off ebay, the standard phone
functions work fine, just a couple of questions,
1) how do I program the other buttons not on the standard keypad part..
2) When I hang up the display doesn't clear and keeps the numbers just
dialed on screen, can this be
Hi,
I am trying to confirm the command 'canreinvite=yes' in sip.conf
using grandstream BT101/2s and snom100s. In either case, no description
nor 'canreinvite=yes', media stream always go through *.
Do I need another settings for confirming sip clients directly
communicate each other?
--
Zen
I'm having trouble turning up a PRI to a T100P. I've read on the Digium FAQ's that once the wct1xxp module is loaded correctly, the LED on the T100P will flash red. I believe I've loaded the module correctly because both wct1xxp and zaptel are listed when I do the lsmod command. The LED on the
On Wed, 2004-03-03 at 17:01, [EMAIL PROTECTED] wrote:
I'm having trouble turning up a PRI to a T100P. I've read on the
Digium FAQ's that once the wct1xxp module is loaded correctly, the LED
on the T100P will flash red. I believe I've loaded the module
correctly because both wct1xxp and
On Wed, 3 Mar 2004 [EMAIL PROTECTED] wrote:
I'm having trouble turning up a PRI to a T100P. I've read on the Digium
FAQ's that once the wct1xxp module is loaded correctly, the LED on the
T100P will flash red. I believe I've loaded the module correctly because
both wct1xxp and zaptel are
Steven Critchfield [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
03/03/2004 04:06 PM
Please respond to asterisk-users
To:[EMAIL PROTECTED]
cc:
Subject:Re: [Asterisk-Users] wct1xxp module and the T100P
On Wed, 2004-03-03 at 17:01, [EMAIL PROTECTED] wrote:
I'm
Andrew McRory [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
03/03/2004 04:11 PM
Please respond to asterisk-users
To:[EMAIL PROTECTED]
cc:
Subject:Re: [Asterisk-Users] wct1xxp module and the T100P
On Wed, 3 Mar 2004 [EMAIL PROTECTED] wrote:
I'm having trouble
So, sorry I have general question , h.323 dont work on FreeBSD + asterisk
???,,, I need converter h.323 sip and codec converter for h.323.
I use FreeBSD 5.2.
Thanks all,
Serge.
- Original Message -
From: NetOne Administrator [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
Man your email client is borked!
zaptel looks good. try removing pridialplan=unknown and add group=1.
--
Andrew McRory - President/CTO
Linux Systems Engineers, Inc.
PO BOX 3791
Tallahassee, FL 32315
(850)224-5737
(850)294-7567
___
Asterisk-Users
On Wed, 2004-03-03 at 17:01, [EMAIL PROTECTED] wrote:
I'm having trouble turning up a PRI to a T100P. I've read on the
Digium FAQ's that once the wct1xxp module is loaded correctly, the
LED
on the T100P will flash red. I believe I've loaded the module
correctly because both wct1xxp
Steven,
Perhaps I should have posted my question differently to the list:
After installing the CVS version of Asterisk, I type, modprobe xct1xxp.
The machine accepts the command but the LED on the T100P does not flash.
How do I know that the T100P module has loaded correctly?
Do you see
Stephen Davies wrote:
Considering that they probably won't delegate, how about Asterisk
supporting a second parallel ENUM tree under a domain that we can
control ourselves?
http://e164.freenetworks.org
See my previous posts about this...
--
Best regards,
Duane
http://www.cacert.org - Free
Use default configuration
as below should work if you have PRIline.
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23 dchan=24 loadzone = us defaultzone=us
zapata.conf: [channels] context=default switchtype=national signalling=pri_cpe
channel=1-23
group = 1
If it's not green, make sure you
Zen,
I am trying to confirm the command 'canreinvite=yes' in sip.conf
using grandstream BT101/2s and snom100s. In either case, no description
nor 'canreinvite=yes', media stream always go through *.
Do I need another settings for confirming sip clients directly
communicate each other?
Do you have
To recap:
1. Security vulnerabilities have been found in the ASN.1 parsing of *many*
H.323 implementations. Some security experts consider them quite serious,
others don't.
2. OpenH323 *was* vulnerable when the announcement was made. (About a month
and a half ago, or so.)
3. The OpenH323
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