Unless you're in China, you should advertise you telephone number as:
+1 863 248 1195
Is your provider Quest or Qwest? Have you removed the disconnected
numbers from your dial plan?
[EMAIL PROTECTED] wrote:
Sorry for disturbing you, but we have some problems with an installation
with
://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
What is the mechanism used to SEND TEXT over a Zap channel? Is it FSK?
Roland Ndaka Fru wrote:
Here is how you can send/receive text in the DialPlan using an AGI script:
print STDERR 1. Testing 'sendtext'...;
print SEND TEXT \hello world\\n;
my $result = STDIN;
checkresult($result);
print
Go to your book store and get the Fedora/Linux reference.
Get yourself a PC with 20GB drives, a CD burner, and decent ram. The PC
should have either an i386 or x86_64 processor. If you'll be purchasing
a PC, go to the computer store, purchase the piece parts, and assemble
it yourself (I
I can provide Polycom phones, and I have provisioning scripts. Is that
what you need?
Eric Bishop wrote:
Any kind Polycom dealers out there?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
The advertised datarate (8mb/448k) are the speeds at which the circuit
between the customer and the central office is clocked and has no
relationship with *effective* throughput. At the central office are
*shared* facilities than connects each DSL connection with the network,
and over
Three sites are experiencing ~10sec period of one-way audio. This
happens several minutes into the call, and it is very intermittent
(infrequent). It does not happen on inter-office calls but only on
calls to/from the PSTN.
Occasionally, a spurt of white noise precedes the drop-out much
Is anyone having problems with Cisco's 2960/3560 LAN switch? Problems
causing retries exceeded in Asterisk?
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
More info: I've noticed that Asterisk CPU utilization has spiked to
100% for a period of 10-20 seconds.
Michael Welter wrote:
The commonality between all sites is Asterisk/Zaptel 1.4.0. The TDM04B
site started reporting this problem after the upgrade to 1.4.0
Each employee has a Polycom phone at his desk at the real office as well
as a Polycom at his home office.
I'd like a call to the employees extension to ring both phones. I'd
also like one entry in the buddy list for each employee, and the buddy
list to indicate he was on a call no matter
The 501 is 12VDC, and the 601 is 24VDC, as I recall. There was a post a
few months ago that said that plugging the 24VDC into a IP501 will fry
the phone.
Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom IP 500
and IP 600? We've mixed up our
Has anyone tried recording to a ramdisk? To an NFS mount? Was there a
benefit?
[EMAIL PROTECTED] wrote:
Hi,
We want to build an Asterisk system that needs to be able to record,
when in a peak situation, a maximum of twenty calls simultaneously. I
could not find any reference to
Is there a programmatic to to trim the silence from the beginning and
end of a recording? From a .wav file? From a .ulaw file?
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
believe the protocol between the calling TTY and the local TTY is FSK.
In any case, would two TTY devices be able to communicate over an RTP
stream using g711?
Or should the TTY device be attached to a POTS circuit away from the PBX?
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver
Does anyone have experience with this product?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
It happens both when the hold button is pushed and when not pushed.
Blind transfers seem to work properly.
Doug Lytle wrote:
Michael Welter wrote:
The attendant attempts an attended call transfer (all phones are
IP501). The attendant pushes hold, transfer, dials the extension
and announces
for the call
remains busy. Subsequent inbound calls on that channel are rejected.
Asterisk 1.2.12.1, Polycom SIP 1.6.6.
Has anyone seen this? Thanks.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
The docs say that SMDI works only with an FXS interface. What does that
mean? What if Asterisk was a voice mail system on a legacy PBX,
connected via a T1 and SMDI?
Joe Pukepail wrote:
I seen something in the bug tracker and svn about SMDI. Not sure if it
was included it 1.4 though.
replaces Cisco CallManagers, Nortel PBXs
with Linux-based VoIP and messaging servers
http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1
Doug
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk
provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
periodic channel hangs--one account on an Eschelon PRI and the other on
a Nortel PRI. One account is on Asterisk v1.2.9.1 and the other v1.2.7.1 .
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
This seems to happen when an agent makes an attended transfer. Does
anyone have more information?
Michael Welter wrote:
There was activity in late 2005 concerning PRI channel lockups. The
telco sends a call to channel n, but Asterisk thinks channel n is
busy and rejects the call
bug, but it
has to do with queues.
Can someone shed some light on this situation?
Thanks,
P.S. This system often fails to reboot properly. zaptel doesn't load
correctly, and Asterisk goes into continual restart.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
be doing to narrow down
on this problem.
Thanks for your help.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
From the Eyebeam softphone, is there a way to capture the AVI stream to
a file? I tried PHP record_file, but it seems to want an audio file.
Thanks
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
don't always have control of the DHCP server.
Is there a way to set the phone to find the tftp server on its own?
Thanks
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth
?
Thanks,
Kevin
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver
and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
Is anyone using the HDLC facility in Zaptel to bring a data T1 into an
Asterisk system? I know this was available in kernel 2.4.19--is anyone
using it in kernel 2.6.x?
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is
there a simple way to reduce the gain without having to remix the tracks?
Thanks
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided
IP501 phones
Private network between the Asterisk system and the CLEC.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com
/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE
I was just spammed from exoxshaindia.com. The spoofed from address was
asterisk-users asterisk-users@lists.digium.com, and the subject was
Fw: Real show and it containd attachment 3.92315089702606R02.UUE. I
believe UUE is a compressed executable.
WATCH OUT!
--
Michael Welter
Telecom
My customers are reporting that the contact directory can only hold
about 45+ entries.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
for your help.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit
.
Thanks for your help.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
connection and go out to
NuFone on IAX. Are these calls bridging away from the Asterisk server?
How can I get accurate billing data?
I tried to Google the archives but I'm still getting page not found.
Thanks
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL
choppiness problems with
teliax. So it's not just me.
-Dan
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation
?
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
And, what does traceroute say about your connection with Teliax? Hmm?
[EMAIL PROTECTED] wrote:
On Fri, 31 Mar 2006, Michael Welter wrote:
Having said all that, I see where Teliax have installed the voip-co4
host on Viawest. Are you using that host for your analysis?
I have used every
different Asterisk system using the same
CLEC. One is * v1.2.4 and the other is v1.2.5. The systems have
totally different motherboards.
Has anyone had a similar problem, and what was the cause?
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL
I'm seeing quite a few Not Found pages when I google lists.digium.com.
Is anyone else getting this?
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation
I have a problem with my Polycom phones. In the buddy list, the phone
displays all but three employees.
For those three employees, there is no difference in any of the
configurations.
Is there a secret to getting all employees into the buddy list?
Thanks,
--
Michael Welter
Telecom Matters
on lists.digium.com. Are
the older posts being purged?
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
hoped that I could just plug a CAT-5 patch cable from my
RJ45 wall outlet into the phone.
On Mar 5, 2006, at 5:17 PM, Michael Welter wrote:
As I understand 802.3af, the phones go through a negotiation with the
unit supplying the power. I don't think it's a matter of -48VDC on a
particular pair
? Agents
from different time zones signing on in their morning and signing off in
their evening. The call center itself running 24/7.
Issues?
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
Doug Lytle wrote:
Michael Welter wrote:
I'm having difficulty with an Asterisk system. The external party has
very good call quality, but the internal party hears clipping and drop
outs.
RX Gains too high
IRQ sharing of the of the ZAP device
There is no ZAP device (it is a SIP-only
Doug Lytle wrote:
Michael Welter wrote:
Doug Lytle wrote:
Michael Welter wrote:
The machine is totally idle.
The T1 vendor noticed 2% packet loss during a ping flood originating
from outside. We changed the Cisco IAD, and there is no longer packet
I've noted from employees
.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
click can be
heard, the tone starts, but the tone is clipped and there is silence
until the next click.
I've verified that QoS is enabled in the IAD.
I would appreciate your thoughts.
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED
server in order to
record his movements.
Does anyone have experience in this area?
Thanks,
Mike
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided
Andres wrote:
Did Linksys really use Asterisk for the SPA9000 software?
I certainly hope so. Have you checked what SNOM uses for their phones?
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
they
do this? What happens when I get a second toll-free number for a
different business--will I be able to differentiate the called number?
Thanks
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided
I'm getting the following messages when a call is answered by a SIP device:
Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP
Transmission error to 192.168.1.254:19262: Operation not permitted
For a Cisco 7940 line, I have the following sip.conf entry:
[desk2]
type=friend
Michael Welter wrote:
I'm getting the following messages when a call is answered by a SIP device:
Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP
Transmission error to 192.168.1.254:19262: Operation not permitted
For a Cisco 7940 line, I have the following sip.conf entry
Michael Welter wrote:
Michael Welter wrote:
I'm getting the following messages when a call is answered by a SIP
device:
Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP
Transmission error to 192.168.1.254:19262: Operation not permitted
For a Cisco 7940 line, I have
to pins 45 on the second plug. Pins 12 on the second
plug to pins 45 on the first plug.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation sponsored
.
Do a traceroute on all three. Look for common failure points.
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk
Kevin P. Fleming wrote:
Michael Welter wrote:
Nov 15 20:09:15 NOTICE[27290]: chan_zap.c:7395 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
I ran top during that time, and there was no significant cpu usage.
Probably interrupt starvation... are there any
, and there was no significant cpu usage.
The system uses Asterisk 1.0.7. The cpu is an Opteron on a Tyan
motherboard.
Can anyone explain this?
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
to the configs.
Can anyone help me with this?
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
?
Is there something about the Westell that needs to be changed? From
what the client says, outbound traffic is unlimited. In sip.conf, I
have nat=yes and qualify=yes.
Thanks
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
I have an installation with four Qwest POTS lines. For some unknown
reason, Qwest drops the first digit in the dial string, and the call
fails. To fix that problem, I put a 'W' in the dial string:
QWEST=Zap/g2
exten = _9303NXX,1,Dial(${QWEST}/W${EXTEN:1})
The client has since
I just installed several 3104s in S. Calif. Didn't have any problems--I
was able to call from one line to another on the same unit and between
lines on different units.
Jerry Jones wrote:
Has anyone been able to get the 3104 to register more than one line
correctly? It seems to work OK
What version of libtiff are you using. Has anyone tried 3.7.x with spandsp?
Doug Lytle wrote:
Alexander Lopez wrote:
I have used the pre20 package, with the latest CVS-head. COmpile goes
cleanly, NO ERRORS.
then I get this when I try to load asterisk -cvv
[app_rxfax.so]Sep
Patrick wrote:
On Tue, 2005-09-20 at 18:37 -0400, Matt Roth wrote:
List users,
Over the last few days we have been working with MCI's development lab
to test our Asterisk setup. We were using a piece of hardware called an
Abacus 5000 that is capable of creating and terminating thousands of
I got five of these doctor's phones as well.
You need your server set-up with dhcp and tftp. In the tftp directory,
you'll need files SIPmac.cnf and SIPDefault.cnf available. You'll
also need the current SIP firmware load.
Then, you have to let the phone power up, then enter the password
I'm proposing to install an Asterisk PBX at a collocation facility for a
remote customer. Each of the customer locations will have an SPA-3000
with the FXO port connecting a POTS circuit and the FXS port connecting
a fax machine or red phone.
In addition to voice traffic, the customer has a
Same problem here.
./asterisk stop;./zaptel restart;./asterisk start
seems to get it working.
Question: will ztcfg -vv alone get it working?
Andy Howell wrote:
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen.
I have several installations running Asterisk over FC3 x86_64 on Tyan
Opteron motherboards. I also have one installation on an Athlon 64.
When compiling zaptel, be sure to 'make linux26' for the 2.6 kernel.
Massimo De Nadal wrote:
Joseph ha scritto:
On Mon, 2005-09-12 at 09:56 +0200,
Have you seen the 3Com LAN-switch-in-a-wall-jack device? It's a four
port device that could also be used for the guest's PC. It can supply
PoE to the phone using a wall wart or PoE on the incoming LAN circuit.
You'll need some logic between Asterisk's management interace and the
property
Michael Welter wrote:
My preferred LD vendor requires g729 and SIP.
Is there a method to test, prior to initiating a call, whether a g729
codec is available? Will ChanIsAvail test g729 availability?
To clarify:
I have n g729 licenses for my system. If I have n g729 calls in
process
My preferred LD vendor requires g729 and SIP.
Is there a method to test, prior to initiating a call, whether a g729
codec is available? Will ChanIsAvail test g729 availability?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Maybe you should read tritechcoa's return policies before you start
recommending them. After suffering through their RMA juggernaut, I'll
never again do business with them.
Tarpo, Louie wrote:
We use both voipsupply and tritechcoa. We've had no problems with either one.
I've received
I've just loaded zaptel 1.0.9 on a new 2.6.12 system (FC4 with updates).
The system has a TE110P card, and zaptel.conf is configured for an E1.
When I do a 'zaptel stop' I get a kernel panic.
Has anyone else seen this?
Thanks,
___
--Bandwidth and
I'm going to build an Asterisk system on a 1G Compact Flash card with
NFS mounts for /var/spool, /var/log, etc. Does anyone have information
on which packages are required for the CF card?
Also, I would like to set the CF card to read only. Does anyone have
information on which directories are
I have a new client who is about to order an Integrated PRI T1--eight
voice channels and the remainder data. He's not sure whether Qwest will
provide a router to split the voice and data.
I remember we had a lot of trouble with HDLC in Linux kernels past
2.4.19. Have those problems been
Paul wrote:
jennyw wrote:
Someone mentioned earlier (I can't find the message now) that they had
a motherboard that allowed you to change IRQ assignments in BIOS. Does
anyone happen to know how to identify motherboards that can do this?
I'm going to put together a new machine now and I'm
John Gruber wrote:
Earlier this afternoon I had this working
exten = 2890,1,Answer
exten = 2890,2,GoTo(12)
exten = 2890,12,Wait(1)
exten = 2890,13,Festival('I can say numbers like')
exten = 2890,14,SayNumber(1230001,f)
exten = 2890,15,Wait(1)
exten = 2890,16,HangUp
I was so very proud of
festival_server.scm -- should be in /usr/share/festival
grep for '(localhost) and replace with nil (don't use double quotes).
Innocent Evil wrote:
Hello,
I have installed festival (the rpm package came with fc4).
But getting this:
client(1) : rejected from myserver not in access list
I'm attempting to use Festival with Asterisk on an x86_64 system. This
IVR application works ok on a P4 system.
I'm using the FC3 x86_64 distro on a single processor Opteron system.
Festival by itself (using the command line and speakers) seems to work
ok, and Asterisk without Festival works
(Nortel/Asterisk). Contact me off list if you would like more
information.
Michael Welter
+1-303-718-2804
Paul Belanger wrote:
See inline for clocking sources:
Alvaro Parres wrote:
Well the Actual Digramm of conecctions is:
E1
I need to place a SIP FXO gateway in Central America. I've been looking
at Quintum products, but the prices are about $150/FXO port.
I have a Dell SC400 on the shelf, and I'm considering just installing
Asterisk and two TDM04B cards and shipping it down. Does anyone have
multiple TDM cards
[EMAIL PROTECTED] wrote:
Mmhh nice !! So, why did Digium forbid it :)?
If Dell is so bad... why is a Dell 2850 server one of the two listed on
the compatibility list for ABE?
http://www.digium.com/index.php?menu=product_detailcategory=softwareproduct=ABEtab=compatibility
Does the Dell
Chris Mason (Lists) wrote:
We have a server running Asterisk and shorewall, three network
interfaces and a T1 card, it functions as our firewall, pbx and connects
to an Adtran 600 for FXS/FXO. We currently have two internet feeds,
hence the three NICs.
Our 10 PSTN lines are currently delivered
Russell Horn wrote:
Since May 05 I have been unable to call any non-geographic number in
the UK via Broadvoice. Thse are numbers such as the 0800 range (free
to call) 087xx (local / national rate calls). Broadvoice support have
been unhelpful, and can't say if there's any intention to fix this.
I will be installing an Asterisk system in Honduras, and I need to
convert many 2-wire analog circuits to E1 PRI. I have no idea if there
is answer or disconnect supervision on the POTS circuits.
An E1 from Hondutel seems to be out of the question because they will
only provide SS7 signaling
Scott Nelson wrote:
In the subdivision where I live, we have a well that time to time has
problems.
How about just fix the well :-)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
1 - 100 of 374 matches
Mail list logo
