Atis Lezdins schrieb:
GMail webinterface does automatically hides quotations.
It's broken. It doesn't hide the somebody wrote: line which
makes it even worse.
Example:
---cut
Bob wrote:
Are you hungry?
Yes.
Are you thirsty?
No.
Pizza?
OK.
Hi,
Would like to run the software to monitor the quality of the bandwidth.
Suggestions welcome?
Thank you.
Shaun___
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This is because meetme needs zaptel to works:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMe
Please note: A Zaptel timer must be present for conferencing to work! See
Asterisk
timer http://www.voip-info.org/wiki/view/Asterisk+timer
Alessandro R.
On Thu, Aug 23, 2007 at
Well, it seems this opened one large can of worms.
Anyway, just to repeat my previous plea - and to echo David's request - can we
please stop all this 'top post' rubbish and move on with our lives?
Thanks and Merry Christmas
Andy
-Original Message-
From: [EMAIL PROTECTED]
Define quality and bandwidth.
Shaun Wingrin wrote:
Hi,
Would like to run the software to monitor the quality of the bandwidth.
Suggestions welcome?
Thank you.
Shaun
nload will show you current bandwidth usage, but i guess that isn't what
you're looking for?
http://sourceforge.net/projects/nload/
Cheers
Geraint
2008/12/11 Shaun Wingrin [EMAIL PROTECTED]
Hi,
Would like to run the software to monitor the quality of the bandwidth.
Suggestions welcome?
When I call an extension on my Asterisk system, and the extension is
unplugged, I just get silence for the 30 seconds (Dial command ring time)
before it goes to voice mail.
How can I get around this?
Michael
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I've just spotted another interesting CDR 'feature'. Data calls don't
get flagged as such. In other words - if I make an ISDN modem call to
another ISDN modem via. the PSTN, the source and destination channels
are set correctly (as is everything else in the current CDR) but there
is no record if
Hello
Looking for some help with a rather odd problem. We have Asterisk
1.4.10 running on a Linux box, within our Windows domain. Our Domain
Controller is a Windows 2003 server, providing the normal Windows domain
functions, such as DHCP and DNS.
When we lose either our Domain Controller
try the following
http://www.callcentric.com
they are the best i've ever dealt with .. they provide did numbers in Sweden--
AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963
944 618286 USA: +1 347 562 2308 Date: Wed, 10 Dec 2008 15:30:59 + From:
[EMAIL
Michael schrieb:
When I call an extension on my Asterisk system, and the extension is
unplugged, I just get silence for the 30 seconds (Dial command ring time)
before it goes to voice mail.
How can I get around this?
qualify=yes in sip.conf?
Philipp Kempgen
--
Phil Knighton schrieb:
When we lose either our Domain Controller (for a reboot/maintenance) or
external ADSL access, Asterisk drops all SIP registrations - even
internal SIP calls within the building no longer function.
Not sure if it cures your problem but I would suggest running a
caching
just an idea, could it have something to do with DNS being unavailable, but
that wouldn't really explain why it would die when ADSL is down... h.
Cheers
Geraint
2008/12/11 Phil Knighton [EMAIL PROTECTED]
Hello
Looking for some help with a rather odd problem. We have Asterisk 1.4.10
Phil Knighton wrote:
Hello
Looking for some help with a rather odd problem. We have Asterisk
1.4.10 running on a Linux box, within our Windows domain. Our Domain
Controller is a Windows 2003 server, providing the normal Windows
domain functions, such as DHCP and DNS.
When we lose
Thanks Philipp - I'll go ahead and get bind9 installed.
Cheers
Phil
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: 11 December 2008 12:07
To: Asterisk Users
Subject: Re: [asterisk-users] Asterisk dies when external access is lost
Doug,
No, the phones are all Snom phones - a mix of 290s (actually elmeg
IP190), 320s and 360s. Mostly using firmware v6.
Cheers
Phil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: 11 December 2008 12:12
To: Asterisk Users Mailing
hi
how i can prevent asterisk try to make calls using G729 when it don't have
any more licenses?
i want it just reject the call or something like that.
thanks
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
We use an iftop. Very similar to top process monitor.
On Fri, Dec 12, 2008 at 3:49 AM, Shaun Wingrin [EMAIL PROTECTED] wrote:
Hi,
Would like to run the software to monitor the quality of the bandwidth.
Suggestions welcome?
Thank you.
Shaun
Hi,
exten = _[0-9]XXX,1,Goto(jump,${EXTEN},1)
seems to allow calls shorter than 10 digits through...
Hope you can help.
Thanks
Shaun___
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Phil Knighton wrote:
Doug,
No, the phones are all Snom phones - a mix of 290s (actually elmeg
IP190), 320s and 360s. Mostly using firmware v6.
I had the same issue a few months back, internet connection went down,
pointed the Polycom's to our internal DHCP/DNS and the phones failed.
Shaun Wingrin wrote:
Hi,
exten = _[0-9]XXX,1,Goto(jump,${EXTEN},1)
The above example is saying:
If the number begins with a 0-9 and is seven digits long.
Which really make no sense, since:
X = matches any digit from 0-9
Doug
--
Ben Franklin quote:
Those who would give up
Thanks to all of you toppers we can now plan on any message with top or
post being treated as spam. Some of us actually read these threads to
learn, not just to hear ourselves talk. If you really have to be top
somewhere, go to FoxSports.
-Original Message-
From: [EMAIL PROTECTED]
I built one in C using AGI. Would you consider licensing the source?
j
On Thu, 11 Dec 2008, Michael wrote:
I want to build my own calling card system on Asterisk.
I looked at this page -
http://www.voipinfo.org/wiki/view/CallingCard+Applications
and it has listed some applications that
How do I customize the digits 0 to 9?
I have tried changing the paths in say.conf and nothing changes.
I would like to do this without over writing the existing files, so I can have
all my custom files in one location.
Michael
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I was running :
asterisk 1.2.24
zaptel 1.2.21
libpri 1.2.6
I remove zaptel and compiled
asterisk 1.4.22
libpri 1.4.7
dahdi 2.1.0
dahdi_cfg -vvv
DAHDI Tools Version - 2.1.0
DAHDI Version: 2.1.0
Echo Canceller(s):
Configuration
==
SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133
On Thu, Dec 11, 2008 at 4:25 PM, Michael [EMAIL PROTECTED] wrote:
How do I customize the digits 0 to 9?
I have tried changing the paths in say.conf and nothing changes.
I would like to do this without over writing the existing files, so I can have
all my custom files in one location.
Hello there,
I am reading Asterisk: The Future of Telephony Chapter four. I am using
a Ubuntu box with Asterisk precompiled at this time so I can learn. I
am finding that I am having a problem when I do asterisk -r from the
command line. It says:
Unable to connect remotely (are you sure
that
Hi,
I would like to tune rx/tx gains using dahdi-monitor for a system which will
be connected to french PSTN.
I'm not aware of any public phone number in France I could call to get a
normalized 1004Hz signal.
My questions are :
1. Does such numbers exist ? Is there a directory somewhere listing
On Thu, 2008-12-11 at 11:37 +, Andrew Thomas wrote:
I've just spotted another interesting CDR 'feature'. Data calls don't
get flagged as such. In other words - if I make an ISDN modem call to
another ISDN modem via. the PSTN, the source and destination channels
are set correctly (as is
I have several branch offices all running Asterisk PBX's that register
to each other via SIP so that calls can be transferred from office to
office. Everything is working great on the office to office transfers,
but I'd like to somehow make the CallerID more useful. Currently if an
extension
Brent Davidson wrote:
I have several branch offices all running Asterisk PBX's that register
to each other via SIP so that calls can be transferred from office to
office. Everything is working great on the office to office transfers,
but I'd like to somehow make the CallerID more useful.
-- -Original Message-
-- From: [EMAIL PROTECTED]
[mailto:asterisk-users-
-- [EMAIL PROTECTED] On Behalf Of Steve Murphy
-- Sent: 11 December 2008 16:26
-- To: Asterisk Users Mailing List - Non-Commercial Discussion
-- Subject: Re: [asterisk-users] CDR Design
--
-- On Thu,
2008/12/11 Dave Fullerton [EMAIL PROTECTED]
Brent Davidson wrote:
I have several branch offices all running Asterisk PBX's that register
to each other via SIP so that calls can be transferred from office to
office. Everything is working great on the office to office transfers,
but I'd
Hi Can you please help me make this into one statement...
It doesn't work if I say _9000[1-9]0[1-8].
Also would like to be able to achieve _9000[1-9]0[1-8],
Asterisk 1.4
exten = _900010[0-8].,1,Goto(route1,${EXTEN:5},1)
exten = _900010[0-8].,2,Hangup
exten =
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my sip peer entries, and
1. Ask your telco, they probably have them, but you may have some difficulty
in finding somebody at your telco that has a clue about what you are talking
about. You can find some lists doing some google searches for the numbers
and hope to get lucky... but as far as I know, there is no official
Hi,
2008/12/11 Matt Watson m...@mattgwatson.ca
1. Ask your telco, they probably have them, but you may have some
difficulty in finding somebody at your telco that has a clue about what you
are talking about.
I can testify it's not easy ...
Wait and see ...
You can find some lists doing
On Thu, Dec 11, 2008 at 11:01 AM, Olivier oza-4...@myamail.com wrote:
I would like to tune rx/tx gains using dahdi-monitor for a system which
will be connected to french PSTN.
I'm not aware of any public phone number in France I could call to get a
normalized 1004Hz signal.
1. Does such
When I do a service dahdi stop I get an error message:
Unloading DAHDI hardware modules: execvp: No such file or directory
the modules remain loaded.
I dont know what to do with this??? Anyone else? I am running centos 4.4
2.6.9-42
This box ran 1.2 with zaptel fine.
Jerry
Jerry,
Jerry Geis wrote:
When I do a service dahdi stop I get an error message:
Unloading DAHDI hardware modules: execvp: No such file or directory
the modules remain loaded.
I dont know what to do with this??? Anyone else? I am running centos 4.4
2.6.9-42
This box ran 1.2 with
Hello
asterisk -vvvgc
Regards
On Wed, Dec 10, 2008 at 7:45 PM, Scott Berry n7...@northlc.com wrote:
Hello there,
I am reading Asterisk: The Future of Telephony Chapter four. I am using a
Ubuntu box with Asterisk precompiled at this time so I can learn. I am
finding that I am
On Thu, Dec 11, 2008 at 01:47:22PM -0500, Jerry Geis wrote:
When I do a service dahdi stop I get an error message:
Unloading DAHDI hardware modules: execvp: No such file or directory
What is the output of:
sh -x /etc/init.d/dahdi stop
the modules remain loaded.
I dont know what to
Jerry,
Jerry Geis wrote:
/ When I do a service dahdi stop I get an error message:
//
// Unloading DAHDI hardware modules: execvp: No such file or directory
//
// the modules remain loaded.
//
// I dont know what to do with this??? Anyone else? I am running centos 4.4
// 2.6.9-42
On Thu, Dec 11, 2008 at 02:38:55PM -0500, Jerry Geis wrote:
the /etc/init.d/dahdi stop is the same command as service dahdi stop (I
think).
sh -x /etc/init.d/dahdi stop
runs the same script, but traced.
--
Tzafrir Cohen
icq#16849755
Hello,
I have BLF working on Snom phones. Ringing state (blinking) or on the
phone state (solid) are working well. So the buttons are configured as
BLF in the Snom webinterface.
Now I would like to add another state for unavailable or dnd. In fact I
would like to turn the LED red in the case the
Does anybody have contact info for Dan Toma, the author of Diax?
I've tried da...@clicknet.ro and da...@rdslink.ro without success.
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Hi Asterisk Users,
we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323
1.18.
We are using MeetMe for conference calls and with two participants there is
no echo problems, but with more than two participants there is a lot of echo
that sometimes disappear for a short time
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't any callerid= entries in any of my
This is the result of sh -x /etc/init.d/dahdi stop
Unloading DAHDI hardware modules: execvp: No such file or directory
[FAILED]
[r...@ebox3850 ~]# sh -x /etc/init.d/dahdi stop
+ initdir=/etc/init.d
+ DAHDI_CFG=/usr/sbin/dahdi_cfg
+
If callers need to just listen, you could run meetme with the -l mode.
Otherwise, you might try the -o mode (optimize, mute non-talker) or -m (set
initially muted).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessandro
Dear All,
I would like to ask please if there is a way to reduce latency on asterisk
or to check what is causing this latency
Regards
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Dave Fullerton wrote:
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If they
have a callerid= entry comment it out and do a SIP reload. When it is
set asterisk overrides the caller ID sent to it.
-Dave
There aren't
On Thu, Dec 11, 2008 at 03:45:15PM -0500, Jerry Geis wrote:
This is the result of sh -x /etc/init.d/dahdi stop
Unloading DAHDI hardware modules: execvp: No such file or directory
[FAILED]
[r...@ebox3850 ~]# sh -x /etc/init.d/dahdi
I will say, most likely the latency is introduced by the network, not the
server
On Thu, Dec 11, 2008 at 3:42 PM, michel freiha mich...@gmail.com wrote:
Dear All,
I would like to ask please if there is a way to reduce latency on asterisk
or to check what is causing this latency
Regards
Has anyone successfully gotten a HiPath system to route calls over to a * box?
If so, I'd appreciate a quick consult. I've configured the HG card to look for
the * server but it doesn't seem to actually be connecting.
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
/ + initlog -q -c 'unload_module dahdi'
// execvp: No such file or directory
/
Here is your problem. It has failed to execute 'initlog' .
I'm not sure how this is directly related to the dahdi init.d scripts.
I ran initlog -q -c ls and this works. so initlog doesnt appear to be
the
Depends on how much latency. The packetization of voice data (and associated
digitizing, transcoding, etc) introduces some latency. Smaller packet size can
reduce this, but at the expense of needing more packets which eats up more CPU
time, etc. Also the jitter buffer size makes a
Brent Davidson wrote:
Dave Fullerton wrote:
Brent Davidson wrote:
Dave Fullerton wrote:
Check the entries for office1 and office2 servers in sip.conf. If
they have a callerid= entry comment it out and do a SIP reload. When
it is set asterisk overrides the caller ID sent to it.
Jerry Geis wrote:
/ + initlog -q -c 'unload_module dahdi'
// execvp: No such file or directory
/
Here is your problem. It has failed to execute 'initlog' .
I'm not sure how this is directly related to the dahdi init.d scripts.
I ran initlog -q -c ls and this works. so initlog doesnt
Hi guys,
Sorry if I'll be very very stupid but really I write to this conference first.
I have problems with configuration of app_meetme in realtime environment.
I use last stable release of asterisk 1.6.0.3
Now situation is following. I create database and table in it. Th table is
CREATE TABLE
If you run as root, can you run initlog -q -c 'rmmod dahdi' ?
Yes this work without error.
Jerry
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Hi,
I am having a very weird problem with call parking. I have defined call
parking correctly, as it work well when parking calls and picking them up.
The problem is what happens after the the 45 seconds have expired.
The behavior wanted is that the person who put the call on park is
On Thu, Dec 11, 2008 at 04:29:36PM -0500, Jerry Geis wrote:
/ + initlog -q -c 'unload_module dahdi'
// execvp: No such file or directory
/
Here is your problem. It has failed to execute 'initlog' .
I'm not sure how this is directly related to the dahdi init.d scripts.
I ran
Just to add to the previous post, here is a bigger snip from my CLI output:
- Executing [...@internal-local-only:1] Park(SIP/0004f21dd2d8-09e6feb8,
) in new stack
-- Stopped music on hold on SIP/0004f215aabb-0a0271e8
== Spawn extension (park-dial, SIP/0004f21dd2d8, 1) exited non-zero
Carlos Chavez wrote:
Use the h extension and execute DeadAGI.
Seems to be working. I have access to variables too.
David fire wrote:
you can try whit the g option to dial.
David
This works only when the called side hungs up, but not the when caller
On Dec 11, 2008, at 12:19 AM, Shaun Wingrin wrote:
Hi,
Would like to run the software to monitor the quality of the
bandwidth.
Suggestions welcome?
Thank you.
Shaun
I can't tell you how to monitor quality of bandwidth - that sentence
doesn't quite make sense, but I'll make some
Jerry Geis wrote:
If you run as root, can you run initlog -q -c 'rmmod dahdi' ?
Yes this work without error.
Tzafrir committed a change to the trunk of dahdi-tools. Could you give that a
try?
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dahdi is from 1.4.21 and up.
1.2.x dont support it.
2008/12/11 Jerry Geis ge...@pagestation.com
I was running :
asterisk 1.2.24
zaptel 1.2.21
libpri 1.2.6
I remove zaptel and compiled
asterisk 1.4.22
libpri 1.4.7
dahdi 2.1.0
dahdi_cfg -vvv
DAHDI Tools Version - 2.1.0
DAHDI
Hi all,
Has anyone any good recomendation of some Virtual PBX that is based on Asterisk?
Many thanks,
Christian
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we offer vps servers running trixbox:
http://gwhosting.net/whmcs/cart.php
if you want to look at them. Just an idea.
michael
On Dec 11, 2008, at 2:29 PM, Christian wrote:
Hi all,
Has anyone any good recomendation of some Virtual PBX that is based
on Asterisk?
Many thanks,
Christian
Hello there,
I am trying to get Asterisk set up by using the book Asterisk: The
Future of Telephony. I am on Chapter 4. I have have set up Zaptel and
zapata.conf and also set up extensions.conf and when I run asterisk -r
at the Gnome-terminal to connect with Asterisk I get the following
Try first just asterisk and after asterisk -r
If still doesn't start try asterisk -c to verbose...
Best regards,
Chris Hariga
--Original Message--
From: Scott Berry
Sender: asterisk-users-boun...@lists.digium.com
To: Asterisk Users
ReplyTo: n7...@northlc.com
ReplyTo: Asterisk Users
I'm looking to send calls to a phone with no callerid data whatsoever shown
on the Polycom as far as missed call.
The specific application for this is that I have a 50 phone install with
some being used for paging. Paging works perfectly, but the problem is that
for every page there is a
Hi,
If Asterisk is running as the root user, I had to do:
sudo asterisk -r
On 2008-12-11 at 23:54 he...@henrythebig.com wrote:
Try first just asterisk and after asterisk -r
If still doesn't start try asterisk -c to verbose...
Best regards,
Chris Hariga
--Original Message--
From:
On Thu, Dec 11, 2008 at 3:19 AM, Shaun Wingrin voi...@gmail.com wrote:
Hi,
Would like to run the software to monitor the quality of the bandwidth.
Suggestions welcome?
Thank you.
Shaun
nprobe and PF_RING are by far the most comprehensive tools I've seen
to do this under Linux:
Hi all
When I call to any mobile and the device is power off the asterisk keep
ringing and I not able to hear the tradicional message saying this
mobile is power off.
When I call from a normal analogic line I got the message.
Somebody have some suggestion to enable asterisk to identify
yes, make sure context line in general area has a dummy context, something
with one line to hangup.
On Fri, Nov 28, 2008 at 12:56 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Fri, Nov 28, 2008 at 11:00 AM, Mike l...@virtutel.ca wrote:
I was looking at my CLI the other day, and
I`m having (a lot of) trouble changing the call parking timeout behavior.
This is my SIP contextÂ…
[internal-local-only-hamel]
exten = s,1,Hangup
include = parkedcalls
What I am trying to accomppish is a quick test where I park a call, wait 45
seconds, and it hangs up.
Here is my
hi;
threw the end of the year we are running a promo, when ordering any
package on
http://gwhosting.net
including our vps servers and trixbox servers, you can get your first
month off. Yes, that's right, enter 30free with out the quote signs
into the coupon code field during checkout to
hi;
I stand to be corrected. In order for you to get this coupon code to
get you a free month, you must sign up for the monthly plans. We did
not activate this coupon for our quarterly payment options, ore our
semi annually or annually payment options. You can only get the 30
Days free if
Caution-top posting. It works for me--ignore it if you like.
Lots of us would be happy to provide a month's free service to demonstrate a
valuable product to a potential client, but we wouldn't choose to do it on a
Non-Commercial Discussion list.
And (flame follows) we would do it using
If the page was 'answered' on the Polycom then it would NOT show up as a
missed call, a received call yes but not a missed call. If you are getting
missed calls from the page application, the users are probably ON the phone
when you page, if so you should put something in your dialplan that checks
Remove the r option to Dial.
Bruno Castelo Branco wrote:
Hi all
When I call to any mobile and the device is power off the asterisk keep
ringing and I not able to hear the tradicional message saying this
mobile is power off.
When I call from a normal analogic line I got the
I put context = xyz in the sip.conf upline supplier configuration and it
ignores this and seems to place it in to default, as the incoming call rule
in extensions.conf only works when placed in [default] ruleset.
Michael
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I put context = xyz in the sip.conf upline supplier configuration and it
ignores this and seems to place it in to default, as the incoming call rule
in extensions.conf only works when placed in [default] ruleset.
Michael
___
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