I suspect they are the same company to www.voipdiscount.com (same rates,same
windows software same design in yellow). (and even maybe voipbuster.com)
Asterisk setup is easy and works quite well. No quality problem, occasionaly
a fire in the server room ( :) ), or 301s, but they are quite reliable.
Hello,
I've a monitoring problem with app_meetme,
I'd like to record a zap channel, which goes to a meetme conference
Monitor doesn't record the voice of another members in the conference.
Thanks any help
Tamas
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: 40994
Thanks any help,
Tamas
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Hi,
I'd like to ask for
help if there is a solution. I have already went thru the archives, but nothing
that'd solve my problem came up.
My problem is that
since we changed to asterisk and voip phones at the Co, using a zaptel tormenta2
dual T1 channel card, and an Adtran TA750 channelbank, we
in advance,
Tamas
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Kristian Kielhofner wrote:
Tamas wrote:
Hello,
- a call comes in with protocol SIP, codec g729
- connects to the IVR [playing prompts, collecting digits]
- dialing forward to protocol IAX2, codec g729
I guess, while the call is in IVR, I need one g729 license for the call.
If the call
application] can be
very complex, allowing different skill-based routing systems.
Any help, idea appreciated.
Kind regards,
Tamas
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Hi,
how do you record calls? Monitor app. or MixMonitor or something else?
How does your storage backend looks like?
What kind of channels do you use? Do you record IAX2 channels?
Regards,
Tamas
Wai Wu wrote:
You got to be kidding about 53 calls being recorded at sametime is an issue.
I have
Kevin P. Fleming wrote:
Matt Roth wrote:
These statements seem contradictory. I know of no way (short of a
custom patch) to tell Monitor() to mix the in and out legs prior to
writing them to disk. On the other hand, MixMonitor() does just that
and I believe it also buffers the writes
Wai Wu wrote:
Except that mixmonitor still has a bug in it.
What kind of bug? Issue number?
FYI: yesterday one issue has been fixed :D
http://bugs.digium.com/view.php?id=6457
Did you mean that type of bug? If something else, please let us know...
T.
-Original Message-
From:
force asterisk to CPU0, would it be able to
handle 60 speex channels and monitor application?
Thanks in advance!
Regards,
Tamas
ps: specs:
Supermicro X6DH8-XB, 2x Xeon 2.8GHz, 2GB RAM, 3Ware 9500S-4LP in RAID5,
4xSATA 200GB disk, onboard NIC [Broadcom NetXtreme BCM5721], 2x Digium
TE110P
Linux
Tamas wrote:
Matt Schulte wrote:
Could anyone either recommend a website or howto on optimizing Linux to
run asterisk. Such examples of what I mean are..
Renice of asterisk pid's
Forcing irq smp_affinity (For interupt hogging T1 cards)
.. That kind of stuff, I looked on the wiki
Hello,
I did some tries with efax and it worked pretty well, however I sent
faxes to the same machine, so I don't know how good is this combination
in real life faxes.
Regards,
Tamas
Carlos Chavez wrote:
Has anyone tried to use an Asterisk server with iaxmodem and
efax? I have
!
Kind regards,
Tamas
Tony Mountifield wrote:
In article [EMAIL PROTECTED], Tamas [EMAIL PROTECTED] wrote:
Hello,
I have some ugly numbers given by zttest for ztdummy on an AMD64 box
running linux-2.6.15 compiled for Athlon64.
Don't be misled by the apparent ugliness
Steven Ringwald wrote:
On Mon, 2006-01-16 at 22:30 +0100, Tamas wrote:
/var/log/dmesg:
...
CPU: L1 I Cache: 64K (64 bytes/line), D cache 64K (64 bytes/line)
CPU: L2 Cache: 512K (64 bytes/line)
mtrr: v2.0 (20020519)
CPU: AMD Athlon(tm) 64 Processor 3000+ stepping 02
Using IO-APIC 2
: 99.694824 -- Average: 99.951973
HW:
Tyan Tomcat K8E, Athlon64 3000+, 1GB RAM, 3ware 8006, 2x Maxtor HDD
SW:
Ubuntu 5.10, linux-2.6.15, zaptel from 1.2 branch
Any idea what can be wrong?
Thanks in advance,
Tamas
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Steven Ringwald wrote:
On Mon, 2006-01-16 at 17:43 +0100, Tamas wrote:
Hello,
I have some ugly numbers given by zttest for ztdummy on an AMD64 box
running linux-2.6.15 compiled for Athlon64.
linux-2.6.15, zaptel/branches/1.2 r900, jiffies
./zttest
Opened pseudo zap interface, measuring
of device :00:0e.0 to 64
pcie_portdrv_probe-Dev[005d:10de] has invalid IRQ. Check vendor BIOS
assign_interrupt_mode Found MSI capability
Allocate Port Service[pcie00]
Real Time Clock Driver v1.12
...
Maybe this says somehting...
Regards,
Tamas
Steven Ringwald wrote:
On Mon, 2006-01
up the not used agent? (transferer)
Thank you in advance!
Kind regards,
Tamas
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experiencing this problem?
Regards,
Tamas
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Steve Underwood wrote:
Tamas J wrote:
Hello,
I have seen that SpanDSP supports V.17 faxing, however when I tryed to
send pages, I eneded with very ugly pages (unreadable). Did anybody else
try that?
Yes, I checked frame slips and clocking on PRI, everything has to be OK.
Regards
Hello,
I have seen that SpanDSP supports V.17 faxing, however when I tryed to
send pages, I eneded with very ugly pages (unreadable). Did anybody else
try that?
Yes, I checked frame slips and clocking on PRI, everything has to be OK.
Regards,
Tamas
to me some timing
problems with asterisk, see my posting with subject: real-time priority
Hello Joseph,
how did you connect asterisk with hylafax? Could you share that?
Regards,
Tamas
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feature, however we can save on IP stack when AGI requested handled locally.
Any idea how can I stabilize the FastAGI running? On the other side is a
python threading socketserver.
Thanks in advance,
Tamas
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Asterisk
] servers with Dual Xeons [64bit] with 3Ware SATA RAID.
We will put 2xTE110P into the box. Hmmm... we can expect problems?
Eric, what problems did you have? Could you share? Which board, what
kind of configuration?
Thanks in advance,
Tamas
Hello!
Thank you for the hit :)
Do you know if it works well with asterisk or not?
Thanks,
Tamas
Andrew Latham wrote:
Just get one of these.
The PCI 921-CDS is a low-cost channelized DS3 WAN adapter that can be
used in ImageStream's Industrial Series routers or OEM products
Hello!
Does anybody know what will be the around price for the announced E3
card from Digium? When is it planned to be ready?
What kind of harware can be used for a system with such card which makes
only IVR stuff?
Thanks in advance!
Kind regards,
Tamas
black holes right now.
Any help appreciated!
Kind regards,
Tamas
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.
Only turning off/on helps. What can I do? It's very annoying because
the box is in hosting and not easy to just restart.
The wct1xxp worked fine in the same box (with restarts also).
Any idea, hint?
Kind regards,
Tamas
modprobe wcte11xp
/lib/modules/2.4.28-magic/misc/wcte11xp.o
Monday, December 20, 2004, 12:20:11 PM, Matt wrote:
MR Tamas J wrote:
Hello!
I discovered, that I'm unable to load ther kernel module under 2.4.28.
Before that I had 2.4.26 and tryed to upgrade the kernel to 2.4.28.
After restart (reboot - soft restart) I can't load the module. When I
go
Monday, December 20, 2004, 12:44:36 PM, Matt wrote:
MR Have you tried doing a modprobe -r first?
Before reboot I did rmmod wcte11xp. If you mean that.
now modprobe -r wcte11xp doesn't do anything, still can't load the
module. :(
Tamas
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channel = 17-31
What could relate to this problem? I don't have any idea why this
problem comes out and why around 7 calls.
Any hint?
Thanks in advance!
Tamas
ps: linux-2.4.28, debian sarge, P4 3.0GHz Prescott, 512M RAM
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with a CRM
(so the goal would be to get the customerID from CRM into CDRs). Also
there are many things which can be done by TEXT messages :)
Please let us/me inform about your findings ;)
Kind regards,
Tamas J.
Friday, December 3, 2004, 12:17:38 PM, reseaux wrote:
r Dear Tamas
r im
functions in iax2 (iax2_sendtext) and in asterisk
(ast_sendtext), but my knowledge is far from that I would be able to
answer previous questions.
Any idea, hint?
Thanks in advance,
Tamas J.
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[EMAIL PROTECTED]
http
Saturday, November 20, 2004, 7:03:53 PM, Steven wrote:
SC On Sat, 2004-11-20 at 18:48 +0100, Tamas J wrote:
Hello!
I would like to know wether it is possible to have end-to-end codec
negotiation in iax2?
What I mean is...
In case the user dials a number available through PSTN, let's force
Hello!
I would like to know wether it is possible to have end-to-end codec
negotiation in iax2?
What I mean is...
In case the user dials a number available through PSTN, let's force to
use alaw (the client is in LAN) to overcome unneeded transcoding:
iaxphone-1st asterisk - PSTN
In case the
How can I use ztmonitor to figure out the caller id sent by the telco?
Because it is not working for me in Chicago.
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 21, 2003 9:03 PM
Subject: Re: [Asterisk-Users] anyone with X100P
Seems like windows messenger is using it for video comm. and file/session
sharing too. And of course for messaging.
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 01, 2003 9:26 AM
Subject: RE: [Asterisk-Users] * Video changes
I thought
Hey,
I've installed 0.59r mpg123 on a redhatbox. I
set the extension up for the mp3player. I called and it was playing the file
back,but it was full of drops. like sound - silence - sound continued. I thought
let's try with the developer version of mpg123 cos other extensions like echo
, 3 Apr 2003, Tamas Levente wrote:
Hey,
I've installed 0.59r mpg123 on a redhat box. I set the extension up for
the mp3player. I called and it was playing the file back,but it was full of
drops. like sound - silence - sound continued. I thought let's try with the
developer version of mpg123 cos
]
To: [EMAIL PROTECTED]
Sent: Thursday, April 03, 2003 7:44 PM
Subject: Re: [Asterisk-Users] MP3player problem
I woudln't write that if it wouldn't support mp3.
On Thu, 3 Apr 2003, Tamas Levente wrote:
And does playback support mp3?
- Original Message -
From: Martin Pycko [EMAIL
same wonder here, cos I wasn't able to compile zapata on a mppe patched
kernel which includes ppp, but a patched version for pptp. I had tu
uncomment this line.
- Original Message -
From: Adam Goryachev [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 04, 2003 4:57 AM
Subject:
: [Asterisk-Users] X100P incoming call handling
quick quess is you are not answering the line.
exten = s,1,Answer
exten = s,2,Dial,SIP/snomphone
On Sun, 2003-03-23 at 17:05, Tamas Levente wrote:
Hi,
I have some problem with a x100p hadling incoming calls. It goes to
the incoming context
I got it working by authentication on snom: realm=asterisk
and sip.conf. I think you don't need the [sip:1114] simly use: [1114]
- Original Message -
From: Benjamin Miller [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 24, 2003 9:05 PM
Subject: RE: [Asterisk-Users] Other IP
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