Hi,
I want to estimate the amount of bandwidth required for Asterisk running on
a T1 in a typical scenario.
Can someone share with me any implementation experience?
Thanks in advance for your input.
Regards,
Mark
___
-- Bandwidth and Colocation
mark morreny wrote:
Hi,
I want to estimate the amount of bandwidth required for Asterisk running
on a T1 in a typical scenario.
Can someone share with me any implementation experience?
What kind of T1? And what codec?
--
Alex Balashov
Evariste Systems
Web:
Hi,
I would like to improve our installation process.
One of my requirement is to enable High Precision Event Timer support.
I'm working with Debian Lenny which is now 2.6.24-based.
Before installating Asterisk, zaptel and so on (and independently of those),
I would like to check HPET is on and
On Fri, 11 Apr 2008 08:40:20 +0200, Olivier [EMAIL PROTECTED] wrote:
Before installating Asterisk, zaptel and so on (and independently of
those), I would like to check HPET is on and working.
$ zgrep HPET /proc/config.gz
CONFIG_HPET_TIMER=y
CONFIG_HPET=y
CONFIG_HPET_RTC_IRQ=y
Hi,
The T1 is 32 x 64Kbps channels ; Codec is GSM.
Thank you for your suggestions.
Regards,
Mark
On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov [EMAIL PROTECTED]
wrote:
mark morreny wrote:
Hi,
I want to estimate the amount of bandwidth required for Asterisk running
on a T1 in a
I contacted the T1 Card manufacturer (Digium), This problem seems similar to
a known issue whose resolution is currently in progress. One of their driver
engineers has some new code in Zaptel that may help in this case. I did
implement it and hopeful that should resolve it. Digium has excellent
Did you provide power to the card? FXS extensions need power.
On Fri, 2008-04-11 at 08:11 +0300, Murithi Martin wrote:
Hi Guys,
I have a TDM400P (2FXS and 2FXO) installed on my asterisk (1.4.18.1)
Zaptel (1.4.9.2) running on Fedora Core 7, below are my configurations
and some diagnosis I did.
Hi,
I'm using Asterisk 1.4.17 and 1.4.19 versions, some time ago I've
noticed that cli command 'core show channels' does not show all data.
It returns only header or one line of data.
After that, auto completition of commands (hitting TAB) freezes cli...
Does anybody has the same problem?
regards,
On Fri, Apr 11, 2008 at 08:47:16AM +0100, Horwich IT Services wrote:
On Fri, 11 Apr 2008 08:40:20 +0200, Olivier [EMAIL PROTECTED] wrote:
Before installating Asterisk, zaptel and so on (and independently of
those), I would like to check HPET is on and working.
$ zgrep HPET
Michael
Check your /etc/asterisk/zapata.conf and if you have
echocancelwhenbridge=yes, remove
Ruben
Michael J. Liberatore escribió:
hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel
1.4.10. They have the hardware echo cancellers. I am having an issue
though, when i
2008/4/11, Godwin Stewart Horwich IT Services [EMAIL PROTECTED]:
On Fri, 11 Apr 2008 08:40:20 +0200, Olivier [EMAIL PROTECTED] wrote:
Before installating Asterisk, zaptel and so on (and independently of
those), I would like to check HPET is on and working.
$ zgrep HPET /proc/config.gz
Flash Operator Panel http://www.asternic.org/
regards,
Drew
Arjan Kroon | Mobillion wrote:
Hi,
I’m not looking for a programma that show the queue logging.
But is there a way to check during a call, which member is connected
to the caller.
Kind Regard,
Arjan Kroon
* From: *
BJ Weschke wrote:
Rilawich Ango wrote:
Thanks. I have checked that the queue.conf. I keep the default
setting as autofill=yes in my tests. That's mean even autofill=yes,
the 1st caller will still stick the whole queue.
asterisk version : 1.4.18
--queue.conf--
; AutoFill Behavior
;
Hi,
I want to estimate the amount of bandwidth required for Asterisk running
on
a T1 in a typical scenario.
Can someone share with me any implementation experience?
Thanks in advance for your input.
Regards,
Mark
Check out http://www.asteriskguru.com/tools/bandwidth_calculator.php it
On Fri, 11 Apr 2008 14:32:36 +0200, Olivier [EMAIL PROTECTED] wrote:
So my question remains :
how can I be certain HPET is included and enabled without messing with
zaptel and subsequent operations ?
HPET is part of the Linux kernel. Messing with zaptel and subsequent
operations is not going
I'm fairly certain the problem is with the phone line. I have all
callerID settings disabled as the Telco is unable to provide it along
with our rollover line setup due to limitations in their antiquated
switch. The CLI and Logs all plainly show the calls as if they were
normal calls
2008/4/11, Godwin Stewart Horwich IT Services [EMAIL PROTECTED]:
On Fri, 11 Apr 2008 14:32:36 +0200, Olivier [EMAIL PROTECTED] wrote:
So my question remains :
how can I be certain HPET is included and enabled without messing with
zaptel and subsequent operations ?
HPET is part of the
Hi,
There are a lot of people who can and will answer questions for
newbies, live on the conference. Just make sure you already Googled
and read The Book ;)
Details on how to hook up with us are here: http://voipusersconference.org
Conference mailing list is here:
Just this minute got the confirm from Terry for today.
We'll be talking about this:
http://www.pikatechnologies.com/english/View.asp?x=539
Note the free port. Try it and let us know how it goes.
r
___
-- Bandwidth and Colocation Provided by
Jerry Geis wrote:
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o
LD [M]
Quoting Ed W [EMAIL PROTECTED]:
one thing I thought about, but never actually did was to install a
damping circuit across the line - a phone plugged in the line never
actually rang or if it did it was so short it was imperceptible. I
figured just a load across the might damp down the test
Is there a way to tell the difference in an agi
between the person actually hanging up the phone
and the manager interface doing a hangup command?
Thanks,
Jerry
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
I'm having problems sending e-mails to the list. Please ignore this
message, I'm just testing. sorry for the inconvenience.
Thiago
Abra sua conta no Yahoo! Mail, o único sem limite de espaço para
armazenamento!
http://br.mail.yahoo.com/
___
Hi everyone,
Sorry if I'm repeating the e-mail, but I'm having problems with the
list.
I'm currently trying to enable call transfer to different domains in
asterisk box (Asterisk 1.2.13 running on Debian etch). I have a
configuration that requires me to transfer call to separate domains
like
That sounds like an E1 to me. Is that 32 DS0 channels or 24?
On Fri, Apr 11, 2008 at 4:18 AM, mark morreny [EMAIL PROTECTED] wrote:
Hi,
The T1 is 32 x 64Kbps channels ; Codec is GSM.
Thank you for your suggestions.
Regards,
Mark
On Thu, Apr 10, 2008 at 11:25 PM, Alex Balashov
I hope these phone / asterisk compatibility questions are not considered
OT for this list. I am currently in Grandstream hell and need a
cost-effective way out :)
Just wanted to know if anybody has experience with the Cisco 7905 / 7911
Running SIP with Asterisk. These seem like a good replacement
Michael J. Liberatore wrote:
hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel
1.4.10. They have the hardware echo cancellers. I am having an issue
though, when i talk, it cuts out the other end. So for example, i
called up another asterisk box and was listening to the
Hi Andrew,
Yes, it is actually a E1.
Your suggestion will be greatly appreciated.
Thanks,
Mark
On Fri, Apr 11, 2008 at 7:50 AM, Andrew Latham [EMAIL PROTECTED] wrote:
That sounds like an E1 to me. Is that 32 DS0 channels or 24?
On Fri, Apr 11, 2008 at 4:18 AM, mark morreny [EMAIL
Using the online calculator mentioned in this thread will help. There
is a lot to bandwidth and even more to VoIP network traffic than can
be answered with your question. On an E1 that is dedicated to IAX
terminating to a provider that does trunking I would say that you
could get a large number
How about a change in IP from the IP provider?
es
(just a calculated guess, but it was a 286 calculator. :) )
Klaverstyn, David C wrote:
Hi All,
I am having problems with some SIP peers. I
seem to
loose registration. If I reload SIP the registration comes back.
They
On Friday 11 April 2008 09:23:09 Jerry Geis wrote:
Is there a way to tell the difference in an agi
between the person actually hanging up the phone
and the manager interface doing a hangup command?
Not in AGI, no. In the core, there's a bitfield that contains a bit for every
reason, but it's
On Fri, 2008-04-11 at 01:18 -0700, mark morreny wrote:
The T1 is 32 x 64Kbps channels ; Codec is GSM.
That's incorrect... a T1 is 24 channels, and each channel is 64kbps.
There are also a few extra bits for framing, which adds up to 1.544
megabits per second in each direction. The audio
I'll begin working on full cross-version support (Asterisk 1.2, 1.4, and 1.6)
in early May for nvfaxdetect and a handful of other modules.
Justin Newman
Hi,
we are using the app_nvfaxdetect from Newman Telecom with Asterisk 1.4
and tried to build the trunk/next release 1.6 with this
Need SIP KEEPALIVES in Asterisk, but QUALIFY won't presently work for you (due
to it's channel disabling behavior)?
Someone posted on the list that they would like to split keepalives and
qualify into different features. Sounds like a good plan, but until that is
done you can turn qualify=
Did this just start happening with the 1.4 tree?
Have you made any progress on getting it resolved?
Justin Newman
Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tzafrir Cohen wrote:
Let's be more specific here, folks:
What version numbers?
Asterisk, spandsp,
Dave,
Docos for Comedian Mail?
Justin
From: dave cantera [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Need good voicemail documentation
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080208/501668f8/attachment.htm
I'll open the source repository soon for envy and nv suite of tools, including
nvfaxdetect. I have a few handfuls of useful Asterisk add-ons.
Starting on module updates to fully support Asterisk 1.2, 1.4, and 1.6 in May.
Maybe we can get some of these in agx-ast-addons.
Also, I am interested
Hello all,
I have the following problem: if there is a temporary loss of Internet
connectivity, the asterisk server 'hangs' if it has external SIP trunks
configured. By hanging I mean that any calls between the local extensions
and any calls to the voicemail extension stop working. Everything
Marius Muja wrote:
My guess is that the asterisk server tries resolving the names of the
SIP providers when it tries to re-register to them and because there
is no internet connectivity it hangs there for a while. However in
that time all the local calls to the asterisk server stop working.
Hi,
I am using asterisk 1.4.19, my requirement is to find out which agents
were ringed by the queue when a call is abandoned (or connected) in a
call center. While this information is available in parts in
queue_logs and cdr, there is no way to correlate this information. For
example this is the
On Friday 11 April 2008 12:57:17 Rajkumar S wrote:
I am using asterisk 1.4.19, my requirement is to find out which agents
were ringed by the queue when a call is abandoned (or connected) in a
call center. While this information is available in parts in
queue_logs and cdr, there is no way to
It is using a local DNS server.
On Fri, Apr 11, 2008 at 10:43 AM, Steven Kurylo [EMAIL PROTECTED]
wrote:
Marius Muja wrote:
My guess is that the asterisk server tries resolving the names of the
SIP providers when it tries to re-register to them and because there
is no internet
This is a very common issue with Asterisk. There is no good fix, but if
you make sure ALL IP addresses of the server are listed in /etc/hosts on
the server it may help.
Marius Muja wrote:
It is using a local DNS server.
On Fri, Apr 11, 2008 at 10:43 AM, Steven Kurylo [EMAIL PROTECTED]
Marius Muja wrote:
Hello all,
I have the following problem: if there is a temporary loss of Internet
connectivity, the asterisk server 'hangs' if it has external SIP
trunks configured. By hanging I mean that any calls between the local
extensions and any calls to the voicemail extension
Just came across a ZDnet article on the cogoblue appliance that was
launched last week.
http://blogs.zdnet.com/Greenfield/?p=215
Not commenting on the article or the appliance.
But just wanted to highlight that it's good to see asterisk vendors
reaching out beyond the usual geek marketing
Matthew, I have just emailed support. Do you know what the latest
revision is?
Also, is it ok for mg2 to be in zconfig.h and echocancel=yes ? It will
know automatically to use the hw ec rather than the software one?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Ok I will remove it, may I ask what that will do or how that will help?
Mike
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ruben Zamora
Sent: Friday, April 11, 2008 7:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Michael J. Liberatore wrote:
Matthew, I have just emailed support. Do you know what the latest
revision is?
Also, is it ok for mg2 to be in zconfig.h and echocancel=yes ? It will
Yes. Chan_zap and zaptel know how to automatically use the hardware
echo canceller. The configuration
On 4/11/08, Dean Collins [EMAIL PROTECTED] wrote:
Lol – though nothing is going to top the Forbes article about Mark Spencer
this week http://www.forbes.com/forbes/2006/0410/063.html
That article is over two years old...
--
Kristian Kielhofner
You can read these information in the zapata.conf. Most of the time
when you use hardware cancelation echo these paramater make worse echo.
Its better when you use HPEC that is a software no hardware for that
parameter.
Michael J. Liberatore escribió:
Ok I will remove it, may I ask what
On Thursday 10 April 2008 12:30:24 Michiel van Baak wrote:
On 10:00, Thu 10 Apr 08, Pete Kay wrote:
I would like to be able to log call details in Asterisk. The kind of
logs that I like to generate is like this:
From
To Forward
Hmm weird for some reason it showed up in my google alerts box on
asterisk this week -I saved the url and didn't even notice the date so
thought it was this week.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original
On Fri, Apr 11, 2008 at 09:58:08AM -0400, Jerry Geis wrote:
Jerry Geis wrote:
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o
CC [M]
/home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o
CC [M]
What type of Nortel? How are you connected to the Nortel?
CP
Eugen Soare wrote:
Well I am entering into a realm that I don't know.
3 sites with Asterisk
1 site with Nortel
Asterisk/Sip calls working fine between the 3 sites.
Asterisk to Nortel set calls working fine. (call comes
Succession 1000SG running 4.0. Using SIP trunks.
es
CunningPike wrote:
What type of Nortel? How are you connected to the Nortel?
CP
Eugen Soare wrote:
Well I am entering into a realm that I don't know.
3 sites with Asterisk
1 site with Nortel
Asterisk/Sip calls working
Hi. One of my clients has an old Mitel SX 200 with a separate
computer doing the voicemail and auto attendant and integrated via a
COV card which is in his case an ISA card! We would all like to
migrate to asterisk, but as a first step, can asterisk integrate into
the Mitel, so it can serve as
At 17:32 4/11/2008, John covici wrote:
Hi. One of my clients has an old Mitel SX 200 with a separate
computer doing the voicemail and auto attendant and integrated via a
COV card which is in his case an ISA card!
Is it an ActiveVoice system?
We would all like to
migrate to asterisk, but as
Yep, you guessed it, an activvoice system. Anyway to make Asterisk
act like that for a while?
Thanks.
on Friday 04/11/2008 Doug([EMAIL PROTECTED]) wrote
At 17:32 4/11/2008, John covici wrote:
Hi. One of my clients has an old Mitel SX 200 with a separate
computer doing the voicemail and
I have a customer with a Fortinet Firewall that is having stability
issues with Asterisk and SIP endpoints (PAP2T) outside his network.
The first issue I see is that Asterisk sees all phones as the IP
address of the Fortinet. Since the parameter localnet defines the
local
On Thu, 10 Apr 2008 11:46:48 -0700, Eugen Soare
[EMAIL PROTECTED] wrote:
So this is just a general question, Is Asterisk really good?
Yes, but you should also look at an alternative that used Asterisk as
a reference (www.freeswitch.org), and make an informed decision.
Fortinets have a SIP session-helper. Sometime this causes issues,
try turning it off. To do this you need to enable telnet on the
forinet management interface. Telnet into the cli and type the following
config system session-helper
edit 12
set port 5066
end
Instead of turning this off or taking
I am looking at that. hmm... what to do... don't want any regrets you
know! :)
thanks,
es
Vincent wrote:
On Thu, 10 Apr 2008 11:46:48 -0700, Eugen Soare
[EMAIL PROTECTED] wrote:
So this is just a general question, Is Asterisk really good?
Yes, but you should also look
I'm running asterisk 1.2 with Sipura adapters.
I've tried to experiment with different codes but I'm either getting No
compatible codecs if I use gsm or
static noise if I use g726
I was under impression that asterisk would translate between codecs according
to show translation table.
2.) Does
I second Doug advice. Migrate to Asterisk asap.
We have several Asterisk auto attendant integrated with Mitel, even
billing using the Mitel's smdr. But voicemail is different. The COV card
emulate a SS4 phone and receive information needed for a voice mail
system. With FXO/FXS ports is not
On 4/10/08, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, Apr 10, 2008 at 05:49:26PM -0400, Al Baker wrote:
Please share more about this.
What/How are you clustering the boxes ?
Is this all VOIP or TDMF front and VOIP for agents in back ?
What kind of Boxes ? What O/S
Additionally Mark, a Channelized (also called Integrated) T1 offers 24
channels for voice/data, but after bit robbing (for signalling, etc) you
only get around 56kbps per channel. ISDN PRI over T1 has 23 b-channels
of voice/data and one d-channel for signalling, etc. PRI is preferred
and most
wow!
That was cool!
thanks for the pdf.
es
Matt Florell wrote:
On 4/10/08, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, Apr 10, 2008 at 05:49:26PM -0400, Al Baker wrote:
Please share more about this.
What/How are you "clustering" the boxes ?
Is this all VOIP
I just wanted to check one more thing,
system is connected to PSTN via SIP trunk ( No echo) , and terminates to
customer analog phone's via Adit 600 fxs.
I do not see any need for echo cancellation in this setup.
There is no far end hybrid source,
Any other thoughts?
On Thu, Apr 3, 2008 at 8:18
Do you mean autofill works in 1.4.x? But it doesn't work even I set it.
On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote:
Rilawich Ango wrote:
Thanks. I have checked that the queue.conf. I keep the default
setting as autofill=yes in my tests. That's mean even
Jorge is correct you will not get the information need via FXO/FXS
unless you program the Mitel to do DTMF inband. It is possible but a
cludge of a fix at best. We have successfully integrated several Mitel
SX200 and SX2000 switches via the PRI (preferred) or T1 using EM_Wink
(works but you have
Hi guys,
I've been experiencing a very strange issue with my Digium Card TDM400
as of this week. It has two FXS and two FXO.
The FXO modules (both of them) never goes on-hook after hanging up in
Asterisk. It had worked perfectly well for over four years.
I put an ammeter in series with the line
OK, this is exactly what I would like to do, can you either write me
on or off list for further details. This would be the first baby step
toward the 20th Century!!
on Friday 04/11/2008 Alexander Lopez([EMAIL PROTECTED]) wrote
Jorge is correct you will not get the information need via FXO/FXS
FYI, I have probably 10 Fortinet units with multiple SIP phones behind
each and all of the phones work flawlessly. As long as the Fortinet is
ver 3.0 or newer, it does NAT so that you don't need to have nat=yes on
*. No pinholes or static nat or anything, it just works.
As a side note, I
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