Eric ManxPower Wieling wrote:
Leo Ann Boon wrote:
Hi all,
I'm using 'show translation' to help dimension my system, but I
confused by the results I get. My 2 test systems (results below): an
AthlonXP 2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz)
produced similar results (D930 is
I tried it on a intel 3 ghz p4 box and a athlon 3000 768 mb ram running
vista and host for centos 4 ( vmware ) considering the load on athlon
running asterisk ( that too under vista plus vmware ) while intel 3 ghz p4 1
GB ram box was sitting idle with centos , there was hardly a 1 ms difference
I have 3 toll free did's with nufone since 1 month .. Until now i dont have
a problem with them .. their portal was good enough to do proper
configuration and call quality wasnt bad ( even though i havent used them
in really huge traffic yet ) .
On 23/12/06, John Novack [EMAIL PROTECTED]
this really is a great program as far as i have heard even though i am not
able to make it work for me _
On 23/12/06, Matt Florell [EMAIL PROTECTED] wrote:
Hello,
We've released another update to our astGUIclient suite: 2.0.2
http://astguiclient.sf.net/
The client suite runs on most modern
Am Freitag, den 22.12.2006, 09:53 -0500 schrieb Doug Crompton:
Wow what a mess! I can imagine how much easier it would be if the world
adopted a country/area/exchange scheme like in the US with known length.
It must be complicated in Germany just within the country. At least in the
US we know
On 12/23/06 09:51 Leo Ann Boon said the following:
I would love to hear how others are using the results from show
translation in system dimensioning. So far, I feel that dimensioning an
Asterisk box is still mostly guesstimation :). Currently, I'm using the
30MHz per call rule to dimension.
Vicky wrote:
I tried it on a intel 3 ghz p4 box and a athlon 3000 768 mb ram
running vista and host for centos 4 ( vmware ) considering the load
on athlon running asterisk ( that too under vista plus vmware ) while
intel 3 ghz p4 1 GB ram box was sitting idle with centos , there was
hardly
Phil Finkler wrote:
Hi all,
I’m trying to incorporate using the i extension in my callplan to
determine if someone enters an invalid extension. My internal
extensions are all 3 digits (100-104). The problem is, the callplan
doesn’t see that say, extension 600 is invalid, it just goes back
You have to join a conference using the dialplan. If you want the
Manager Client to be able to make an existing call join a conference,
set up an extension in the dialplan that does what you want, and
then use the Manager Redirect command to transfer the channel
to that extension.
Many
I decided to give the whole family IP phones for christmas,
all hooked into my asterisk server, so all the nephews can
have their own lines.
However, one of the phones I got was the SNOM 200. That's worked
fine for me on my own network, but I'm having bad luck getting
it to work behind NAT
On 12/23/06, nik600 [EMAIL PROTECTED] wrote:
You have to join a conference using the dialplan. If you want the
Manager Client to be able to make an existing call join a conference,
set up an extension in the dialplan that does what you want, and
then use the Manager Redirect command to
On Sat, Dec 23, 2006 at 02:02:18AM -0800, Brad Templeton wrote:
I've tried all the various NAT settings on the SNOM 200 (with
the last firmware rev they made) but reports are that's broken.
The SDPs and Contact headers it sends out are always the natted
address, even if I tell it to use STUN
Hi
I'm not familiar enough with Sangoma. I do hope I can slightly help in
isolating the problem.
On Fri, Dec 22, 2006 at 01:41:08AM -0200, Josué Conti wrote:
Hi all, as good?
I try to install asterisk-1.2.14, zaptel-1.2.12,libpri-1.2.4,addons-1.2.5 ,
sounds-1.2.1 and wanpipe-2.3.4-3 and
On Fri, Dec 22, 2006 at 09:58:22AM -0700, Colin Anderson wrote:
LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o
Building modules, stage 2.
MODPOST
*** Warning: register_wanec_iface
[/root/wanpipe/patches/kdrivers/wanec/wanec.ko] undefined!
*** Warning: unregister_wanec_iface
On Sat, Dec 23, 2006 at 09:51:24AM +0800, Leo Ann Boon wrote:
Hi all,
I'm using 'show translation' to help dimension my system, but I confused
by the results I get. My 2 test systems (results below): an AthlonXP
2000+ (1.3GHz) and a Pentium D930 (duo-core, 3.0GHz) produced similar
Doug Crompton wrote:
In order for the external MWI to work you must turn on the message
indicator and for units that have answering machines the machine must be
turned off.
Perhaps we could put together a list of analog phones that have this
feature. I have been told that both Uniden and ATT
On Sat, 23 Dec 2006, Leo Ann Boon wrote:
At this point, I don't feel that 'show translation' is a useful indicator of
actual transcoding performance. It's OK for relative comparisons but utterly
useless if you need the figures for sizing purposes.
Just to give you another (relative)
Hi
can i set up some conditions in my dialplan?
For example:
exten = 99,1,Answer
exten = 99,2, ... if {RECORD}=yes
then:
monitor...
Dial
else:
Dial.
Or something similar... ?
Many thanks in advance
nik
___
--Bandwidth and Colocation
I used _XX. Since it was used in the examples I got from voicepulse.
Maybe I can modify it so it's standardized by using 's'. Any idea why
they'd use something like that for incoming calls? Are you sure 600
would match _XX.? I thought _XX. Was just two digits.
Thanks for the help,
Phil
Phil
Are you sure you want to fire up a JVM each and every time you run this
command? that's a resource hog and will anyway cause a delay for system
class loading, etc. Maybe attaching to a resident process would be lighter.
k,
On Fri, 22 Dec 2006 13:48:27 +0100, Andre Gustavo Lomonaco
nik600 wrote:
Hi
can i set up some conditions in my dialplan?
Yes, look at the following command:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+GotoIf
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither
In article [EMAIL PROTECTED],
nik600 [EMAIL PROTECTED] wrote:
You have to join a conference using the dialplan. If you want the
Manager Client to be able to make an existing call join a conference,
set up an extension in the dialplan that does what you want, and
then use the Manager
Vicky [EMAIL PROTECTED] wrote:
I have 3 toll free did's with nufone since 1 month .. Until now i dont have
a problem with them .. their portal was good enough to do proper
configuration and call quality wasnt bad ( even though i havent used them
in really huge traffic yet ) .
The reason
Leo Ann Boon wrote:
Phil Finkler wrote:
Hi all,
I’m trying to incorporate using the i extension in my callplan to
determine if someone enters an invalid extension. My internal
extensions are all 3 digits (100-104). The problem is, the callplan
doesn’t see that say, extension 600 is
Doug,
Thanks for the info. I'm glad it works.
One question: Is there some sort of one-button way to dial in to your
voicemail? It seems I read something about it, when I was doing similar
research? I think it was the Uniden CLX-465, which claims support of
Phone Company voicemail. I
The nearest I can do is I have a Linksys 3102 and its also set to
inband and when I call that extension fromm outside using asterisk 1.4
I can hear the dtmf just fine in my ear -- works about the same as
using 1.2. The only app so far which is not working is meetme not
detecting the * in asterisk
Hi all,
I am getting the following popping up in my asterisk CLI. Everything
seems to working ok, but I'm curious as to what exactly these messages mean:
Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from
Lee Jenkins wrote:
Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123
handle_request_subscribe: Got SUBSCRIBE for extension
[EMAIL PROTECTED] from 192.168.1.104, but there is no hint
for that
If I'm remembering correctly, it's a message you'd get if you had a
Polycom phone (Other as
Doug Lytle wrote:
Lee Jenkins wrote:
Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123
handle_request_subscribe: Got SUBSCRIBE for extension
[EMAIL PROTECTED] from 192.168.1.104, but there is no hint
for that
If I'm remembering correctly, it's a message you'd get if you had a
Polycom
Folks, with all due respect: this thread is now wy off topic, as it
has nothing to do with Asterisk whatsoever.
Please take it offline, or to ~biz.
thx.
B.
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
Not that I know of. I guess you could speed dial but then my Asterisk
voicemail is 80 so how hard is it to pick up the phone and dial that. I
never had phone company voicemail on a wired line so I don't know how that
works but I suspect you have to dial your own 7 digit or 10 digit
number???
Doug
On Sat, Dec 23, 2006 at 04:05:08PM -0500, Doug Crompton wrote:
Not that I know of. I guess you could speed dial but then my Asterisk
voicemail is 80 so how hard is it to pick up the phone and dial that. I
never had phone company voicemail on a wired line so I don't know how that
works but I
Brian Capouch
I changed the subject I don't think it was right for this message!!
// Re: [asterisk-users] Need quality toll free 800 number over IAX?
Well I don't agree with you about this thread they are talking about the
good and the bad VoIP providers
This is information that Asterisk
On Sat, Dec 23, 2006 at 05:30:54PM -0500, Al Bochter wrote:
Keep in mind list THE VOIP PROVIDERS and VOIP SUPPLIERS NEED US.
SO the providers and suppliers need to get there acts together.
The BAD and the SCAMMERS are giving VoIP a bad name and pushing OUR NEW
PBX CUSTOMERS TO VONAGE.
Hello Tzafrir, all good?
It was really a strange and curious problem, but the fact is that I entered
in contact with the support technician of the Sangoma and Mr. Alex and Mr.
Yuan of the Sangoma, had passed me a new package of the Wanpipe (
wanpipe-2.3.4-2.1) that it compiled normally in SUSE
Hello Colin:
Please try ftp://ftp.sangoma.com/linux/custom/Yuan/wanpipe-2.3.4-2.1.gz
Best Regards
Josue
2006/12/23, Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, Dec 22, 2006 at 09:58:22AM -0700, Colin Anderson wrote:
LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o
Building modules,
I seriously doubt trxtel.com scams anyone. I may be wrong but the
person behind it has been with this community for a long time and has
done nothing post insightful and meaningful things to this list and give
back to the community in many other ways as well. It is a unique idea
but that is
Tzafrir Cohen,
Well if you would have asked I don't aim to sell service to VoIP users.
I BUY VOIP TRUNK SERVICE from VoIP Providers.
I BUY VOIP DEVICES from suppliers
I install Asterisk PBX Servers and point the my customers to VoIP Providers and
Suppliers
So the fact is I don't offer a
Steve Totaro,
I will contract you off the list about trxtel that is not my base point
of this.
// Bottom line, you get what you pay for.
I agree.
// Check out a provider, try their customer service, see if there is a
toll free number, call it and see if someone picks up.
You forgot word of
Hi,
If Iam doing UPDATE SQL statements I got an overload for connection.
am doing everytime an Disconnect ${connid}) but this is ignored.
any idea?
best regards
Thomas
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Phil Finkler wrote:
I used _XX. Since it was used in the examples I got from voicepulse.
Maybe I can modify it so it's standardized by using 's'. Any idea why
they'd use something like that for incoming calls? Are you sure 600
would match _XX.?
On Sat, Dec 23, 2006 at 02:14:14PM +0200, Tzafrir Cohen wrote:
On Fri, Dec 22, 2006 at 09:58:22AM -0700, Colin Anderson wrote:
LD [M] /root/wanpipe/patches/kdrivers/wanec/wanec.o
Building modules, stage 2.
MODPOST
*** Warning: register_wanec_iface
Geez Al, let it go. We've heard your rants for what seems like years now
(even though it's only been weeks). No one cares anymore.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
You guys are missing the point of the message I sent!
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VoIP PBX) 1-563-773-6610 EXT: 250
Bill Hackensack wrote:
Geez Al, let it go. We've heard your rants for what seems like years
now (even though it's
Tzafrir Cohen wrote:
If you had just one call, then adding extra CPUs wouldn't have helped.
'show translations' mainly helps you compare different codecs. It is
also handy as a benchmark because it's there. However
I agree with you that with 1 call, more CPU won't help. I'm just
surprised
You have to ask what they have open on the box on thier firewall. A good way to
learn asterisk is to get a p3 and play at home.
- Original Message -
From: blackwater dev
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, December 20, 2006 11:25 AM
Never got the financial backing for it. I looked in to mass producing it in
China and I was in way over my head. I am currently working with an
investor. I will let you guys know if anything comes up.
Dovid
- Original Message -
From: Jerry [EMAIL PROTECTED]
To: Asterisk Users Mailing
Just to give you another (relative) comparison... This is from a VIA
processor running at 533MHz, (64KB Cache) asterisk compiled as i586 as
it's missing some of the nicer MMX instructions:
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex
ilbc
g723 - -
Hello everybody
HAPPY and Merry Christmas to all.
___
--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
Ditto, Happy Holidays everyone!
On 12/23/06, Carlos Rojas [EMAIL PROTECTED] wrote:
Hello everybody
HAPPY and Merry Christmas to all.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Anyone seen this and know how to fix it? (note the Assembler messages at
the end). Thanks in advance:
server# make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA
-DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o
gendigits.o gendigits.c
cc -o gendigits
Thers a box we can get here in Australia that has an RJ45 plug, a built
in web server that has a config page and URL's to close/open one of 4
included relays. I use phpagi to hit the url and open/close doors that
way. If anyone is interested, ill let you know URL's but its being sold
from our
When you built Asterisk, it must have refused to build the ilbc codec -
I have never seen an Asterisk box that could not transcode ilbc, in over
3 years of working with Asterisk.
PaulH
On Sun, 2006-12-24 at 14:12 +1100, James Harper wrote:
Just to give you another (relative) comparison...
53 matches
Mail list logo
