i finished setting my asterisk pbx with 5 phones. thanks for everyones help here, in getting this accomplished. it is greatly appreciated. this is what i set up. Athlon 3500+ cpu (2ghz i think) 1 Gig of RAM Netopia gateway to SBC DSL 6000kbps/600kbps Linksys wire with QOS functionality Trix
Hi all,
Just a little note for the records and archives. We see many small
glitches / troubles in the mailing-list but rarely success stories ...
Here's one :
Asterisk is running perfectly fine in our setup :
Debian 3.0 stable / Athlon 1.8, 256 MB Ram / Digium E-100P / Swisscom
PRI isdn
We hav
Hi Marcel,
Great news. Thanks for posting your success story
john brown
chagres technologies, inc
On Tue, Oct 14, 2003 at 01:02:17PM +0200, Marcel Prisi wrote:
> Hi all,
>
> Just a little note for the records and archives. We see many small
> glitches / troubles in the mailing-list but rar
D]>
Sent: Wednesday, October 15, 2003 12:02 AM
Subject: [Asterisk-Users] Success story
> Hi all,
>
> Just a little note for the records and archives. We see many small
> glitches / troubles in the mailing-list but rarely success stories ...
>
> Here's one :
>
> Asterisk
rs transfer calls to each other? I.e. is
> it announced or blind?
>
> Regards,
> Aaron.
>
> - Original Message -
> From: "Marcel Prisi" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, October 15, 2003 12:02 AM
> Subject: [A
Can you please describe how you have your "el-cheapo" consultative transfers
working?
- Original Message -
From: "Marcel Prisi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, October 21, 2003 1:04 AM
Subject: Re: [Asterisk-Users] Success
Wow.
> [00059002042b]
> context=main
> host=dynamic
> callerid = "John Doe" <123>
> nat=yes
> Line => svip10
That did it. The phone registered with * and a debug msg flys up when I
pickup/put down the reciever.
When I pick up the handset, I can hear a dialtone. But pressing numbers on
the keypad
Hi,
> -Original Message-
> > Line => svip10
>
> That did it. The phone registered with * and a debug msg flys
> up when I pickup/put down the reciever.
>
> When I pick up the handset, I can hear a dialtone. But
> pressing numbers on the keypad doesn't do anything other than
> show up
Dear list,
Wiki say that supported video codecs are H.261 & H.263!
I would like to know if anyone of you have success case with asterisk
and video support ??? With SIP or H323!
Which are the recommended gateway with T1/E1 that support video and work
well with Asterisk ?
I will appreciated an
I've just downloaded the new SIP firmware images and placed them in my tftp
directory. The phone seems to only grab 1 of the 4 files necessary to
upgrade to SIP.
If anyone can shed some light here, would appreciate it.
Thanks,
Matthew
___
Asterisk-User
If anyone has had success in using a Swissvoice IP10S and the SIP firmware
and would like to help me, please contact me offlist. I have successfully
upgraded the phone to SIP and it registers to asterisk, but that's about it.
No dialtone, won't answer a ringing call, etc..
THanks,
Matthew
___
I just wanted to post here and let everyone know that the TE406P (quadspan
T1/E1 with hardware echo can) kicks some serious ass.
We've been running a PRI now for over a year with Asterisk (every single call
in and out is through two Asterisk boxes, including faxes) and while the
software based
-Commercial Discussion
Subject: [Asterisk-Users] success story: TE406P (quadspan with
hardwareechocan)
I just wanted to post here and let everyone know that the TE406P (quadspan
T1/E1 with hardware echo can) kicks some serious ass.
We've been running a PRI now for over a year with Asterisk (
On Sunday 24 July 2005 16:30, Chris Modesitt wrote:
> Man I almost passed from laughing when I read this, that is the best
> description of bad echo I have ever heard:)
:-) Well the bad bad echo I described I am almost positive occurs because the
echo canceller either mistakenly turns off (false
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 SIP
calls. Initially, we could not get past 64 simultaneous digitall
Hello list ,
I´d like to report a success case with a modem based
on chipset : Motorola 62802-51.
It works fine , and zaptel identifies as a X100P
( not clone ) .
Red Alarms can be identified . :) This doesn´t
occurred on MD3200 ambient chipsets.
Best Regards ,
--
- Jefferson Carvalho
Analista de
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 SIP
> ca
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
Patrick,
Thank you for your suggestions.
Our initial runs were recording directly to an NFS mount and they
experienced the same problems as recording to the local disk. In our
final setup, the copy will be done to an NFS mount as long as it
exists, falling back to local disk only when the NF
On 9/20/05, Matt Roth <[EMAIL PROTECTED]> wrote:
Patrick,
Thank you for your suggestions.
Our initial runs were recording directly to an NFS mount and they
experienced the same problems as recording to the local disk. In our
final setup, the copy will be done to an NFS mount as long a
Hey ho,
I suppose you are the person from the digium forum :)
The reason i recommended you to use a ramdisk is because i think the
problem with recording to disk is saving 20ms of stream 1, then 20 ms of
stream 2, then 20ms of stream 3 etc etc meaning you write everytime
very small things.
On Wed, 2005-09-21 at 10:07 +0300, Zoa wrote:
> The reason i recommended you to use a ramdisk is because i think the
> problem with recording to disk is saving 20ms of stream 1, then 20 ms of
> stream 2, then 20ms of stream 3 etc etc meaning you write everytime
> very small things. (with a lot
Also when you do things over the network, disable your onboard network
card, and go for some more expensive network card.
In our tests with small packets, we could increase the throughput with a
factor 2. (related to cpu load).
Zoa.
--
www.asteriskguru.com
trixter http://www.0xdecafbad
On Wed, 2005-09-21 at 11:11 +0300, Zoa wrote:
> Also when you do things over the network, disable your onboard network
> card, and go for some more expensive network card.
> In our tests with small packets, we could increase the throughput with a
> factor 2. (related to cpu load).
I wonder how muc
True... But i tried several brands of cards, and several drivers, the
dual nic gigabit intel card was a lot better than all the other
combinations i tried.
zoa
trixter http://www.0xdecafbad.com wrote:
On Wed, 2005-09-21 at 11:11 +0300, Zoa wrote:
Also when you do things over the network,
All,
This message has generated a lot of responses, so I'm going to address
each of them here in an attempt to consolidate the thread.
Matt,
- I'm very interested in the specifics of your setup.
- How much space is on the RAM disk?
In light of the I/O bottleneck problem I'd have to ask why asterisk
can't just buffer incoming audio and then flush a complete audio file to
disk.. I'm assuming that recordings vary in length.. the problem with
this idea is what happens if 50 recordings all complete at the same
time.. a dump li
I would think memory would be the limiting factor. A 3-4 minute wav
file is what, 30Meg or so? And there is one for each end of the call,
so that's 60Meg. Now let's say it's a 15 minute call and then are 10 of
them at once. That's 30Meg x 5 (5 times the length of my estimate) x 2
(each leg)
The problem is that then it won't work on systems with little memory. 50
streams would eat memory like crazy.
Zoa
Matt Hess wrote:
In light of the I/O bottleneck problem I'd have to ask why asterisk
can't just buffer incoming audio and then flush a complete audio file
to disk.. I'm assuming t
It's true that the average Asterisk implementation doesn't have enough
RAM, but we are replacing a legacy NorTel switch in a call center. If
you look at the cost of traditional PBXs, the cost of additional memory
starts to look a little better. = )
Now for some quick math:
1 minute of PCM a
I think its the best you can do.
Maybe there should be some option to be set for the monitor command to
buffer, with a warning that it will eat memory.
Its also not needed to buffer the complete call at once, just buffering
and writing to disk every 10 seconds would already be a big improvement
On 9/21/05, Matt Roth <[EMAIL PROTECTED]> wrote:
- What format are you recording to?- What codec are the SIP calls being placed over?
We are recording to the PCM format and using the G711 uLaw codec. Highvoice quality is essential to our application (we are a call center) sowe partnered with MCI t
On 9/21/05, Matt Florell <[EMAIL PROTECTED]> wrote:
> We have sevaral call centers as well, and we just restrict a single server
> to 50 recordings at once and then we would pass the next recording as an
> IAX2 channel to another recording server. It's a scalable system for us that
> is relatively
On 9/21/05, izo <[EMAIL PROTECTED]> wrote:
On 9/21/05, Matt Florell <[EMAIL PROTECTED]> wrote:> We have sevaral call centers as well, and we just restrict a single server> to 50 recordings at once and then we would pass the next recording as an
> IAX2 channel to another recording server. It's a sc
Hello Matt,
very interesting setup! are you using asteriak queues for inbound or not
at all?
Bye
l.
In data Thu, 22 Sep 2005 06:25:44 +0200, Matt Florell <[EMAIL PROTECTED]>
ha scritto:
We wrote VICIDIAL(part of the GPL astGUIclient suite
http://astguiclient.sf.net) for our call center ope
On Sep 21, 2005, at 1:46 AM, Zoa wrote:
True... But i tried several brands of cards, and several drivers, the
dual nic gigabit intel card was a lot better than all the other
combinations i tried.
Fun fact: short of buying a 10gigE card, your best performance will
probably be from a cheap
Hi Matt,Is your solution 100% Asterisk or are you using other "helpers" such as SER or XXXproxy or whatever?Thanks,WaldoOn Sep 21, 2005, at 12:45 PM, Matt Roth wrote:All, This message has generated a lot of responses, so I'm going to address each of them here in an attempt to consolidate the threa
We are 100% Asterisk on the VOIP side. We use SIP, IAX and Zap(channelbanks) for phones and Zap T1s for telco termination.
MATT---On 9/22/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
Hi Matt,Is your solution 100% Asterisk or are you using other "helpers" such as SER or XXXproxy or whatever?Than
> - The reason i recommended you to use a ramdisk is because i think the
> - problem with recording to disk is saving 20ms of stream 1, then 20 ms of
> - stream 2, then 20ms of stream 3 etc etc meaning you write everytime
> - very small things. (with a lot of seeking).
I was thinking about
Hi everyone,
This is just another attempt to address everybody in one place and
consolidate the thread.
Zoa,
- I think its the best you can do.
If something pops into your head, even if it's off the wall, don't
hesitate to share i
On 9/22/05, Matt Roth <[EMAIL PROTECTED]> wrote:
Matt,- Have you tried recording directly to GSM format? It will help reduce the
- bottleneck on disk IO although it will use more CPU cycles(in your case- on a RAM drive this may not help at all)We don't want to do any transcoding on the Asterisk ser
Matt,Thanks for the information. If you don't mind answering: are you guys developing this solution for your internal needs (meaning serving UAs from within your enterprise) or are you planning on offering services to the public?It's not that I'm really interested in your business or business model
Waldo,
Thanks for the information. If you don't mind answering: are you guys
developing this solution for your internal needs (meaning serving UAs
from within your enterprise) or are you planning on offering services
to the public?
This solution is being developed for our internal needs.
It
Which version of asterisk and zaptel are you using?
Will they work with 1.0.9 ?
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237
Hi Andrew,
I'm using a TE406P too, and I have "echocancel=yes" in zapata.conf.
Is this redundant? Should I take the line out?
Please advice.
Thanks.
AK
On 10/3/05, Rod Bacon <[EMAIL PROTECTED]> wrote:
Which version of asterisk and zaptel are you using?Will they work with 1.0.9 ?=
Andy Kuo wrote:
Hi Andrew,
I'm using a TE406P too, and I have "echocancel=yes" in zapata.conf.
Is this redundant? Should I take the line out?
Please advice.
No, if you don't put 'echocancel=yes' in zapata.conf, Asterisk will not
request echo cancellation on the channels. If it doesn't reques
Hello list ,
I´d like to report a success case with a modem based
on chipset : Motorola 62802-51.
It works fine , and zaptel identifies as a X100P
( not clone ) .
Red Alarms can be identified . :) This doesn´t
occurred on MD3200 ambient chipsets.
can you send us more info?
driver,versions,logs, aud
y 19, 2005 2:41 PM
Subject: [Asterisk-Users] :: Success Case => Motorola 62802-51 as FXO
device::
> Hello list ,
>
> I´d like to report a success case with a modem based
> on chipset : Motorola 62802-51.
> It works fine , and zaptel identifies as a X100P
> ( not clone ) .
&
Hello!
In spite of a number of complaints, I have tried to use a TDM400 on a Via
EPIA-MII motherboard with a 1.2GHz C3 CPU. cpuinfo and interrupts are
included at the end of this e-mail. I have had no problems with it so far
that I can attribute to the computer. I have, though, had continual
iling List - Non-Commercial Discussion"<asterisk-users@lists.digium.com>Sent: Wednesday, January 19, 2005 2:41 PM
Subject: [Asterisk-Users] :: Success Case => Motorola 62802-51 as FXOdevice::> Hello list ,>> I´d like to report a success case with a modem based> on chip
Walter Willis wrote:
not work fine
Actually it is recognized as an x100p device:
Nov 21 19:54:34 asterix kernel: Zapata Telephony Interface Registered on
major 196
Nov 21 19:54:34 asterix kernel: PCI: Found IRQ 5 for device 00:0d.0
Nov 21 19:54:34 asterix kernel: wcfxo: DAA mode is 'FCC'
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