Hi List,
I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years
on my lab test setup and I appreciate it. Moreover the global quantity of
fax handled by this setup is not very high.
I'll be involved in a more complex system for a customer and I would like
to ask to All of you
Is it the Windows software, or other? I noticed the Nokia E71 mobile
has an option for Cisco IP Communicator (besides the built-in SIP
client)
On Wed, Mar 4, 2009 at 22:32, Dorien K. Takeshi
dorien.take...@webhad.co.nz wrote:
Hi guys,
Has anyone had any luck with getting the Cisco IP
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network) =
[Asterisk with TDM800p] = Fax/Copy Machine
This seems to work fine, a few failed Fax or very slow sending/process
sometimes but no complaining
users, so this must be ok :)
My previous try was:
E1 =
On Sun, 8 Mar 2009, Marco wrote:
Hi List,
I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years
on my lab test setup and I appreciate it. Moreover the global quantity of
fax handled by this setup is not very high.
I'll be involved in a more complex system for a customer
On Sun, 8 Mar 2009, benoit wrote:
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network) =
[Asterisk with TDM800p] = Fax/Copy Machine
The TE220P and the TDM800P are in different Asterisk boxes? Any particular
reason for that? I now have an E1 coming in to
Remco Barendse a écrit :
On Sun, 8 Mar 2009, benoit wrote:
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network) =
[Asterisk with TDM800p] = Fax/Copy Machine
The TE220P and the TDM800P are in different Asterisk boxes? Any particular
reason for
In makemenuconfig at apps you have rxfax and txfax you can disable
them from there . I think the name is app-fax i dont remember exactly
now but it is there
Enviado desde mi iPhone
El 08/03/2009, a las 04:33 a.m., Remco Barendse aster...@barendse.to
escribió:
On Sun, 8 Mar 2009,
Hello Everybody!
I am currently setting up an Asterisk server for medium to high load
(approximately 20-35 concurrent phone lines).
Do you think the following specs will sufficiently satisfy this system?
CPU: XeonQC3220 2.4GHZ 8M
RAM: 2X2GB/800
Harddrive: 1X250GB
I could add harddrives and
benoit wrote:
Remco Barendse a écrit :
On Sun, 8 Mar 2009, benoit wrote:
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network) =
[Asterisk with TDM800p] = Fax/Copy Machine
You'll find that faxing over IAX is problematic at best.
If
Hello,
setting up Meetme was very easy. I jut added the MeetMe Application to
an internal extension to be reachable by SIP and to an external
extension to be reachable by ISDN.
What I don't understand however is how to call somebody and drop him
to the conference?
I'm using Asterisk 1.4 from
Doug Lytle a écrit :
benoit wrote:
Remco Barendse a écrit :
On Sun, 8 Mar 2009, benoit wrote:
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network) =
[Asterisk with TDM800p] = Fax/Copy Machine
Look like good. I have an similar server for 100 ext.
Regards,
Luis Morales
On Mon, Mar 9, 2009 at 8:01 AM, Elliot Murdock murdo...@gmail.com wrote:
Hello Everybody!
I am currently setting up an Asterisk server for medium to high load
(approximately 20-35 concurrent phone lines).
Do you
Elliot Murdock wrote:
Hello Everybody!
I am currently setting up an Asterisk server for medium to high load
(approximately 20-35 concurrent phone lines).
Do you think the following specs will sufficiently satisfy this system?
CPU: XeonQC3220 2.4GHZ 8M
RAM: 2X2GB/800
Harddrive: 1X250GB
Hi., do you think that sbr policy in queue strategy will be useful?
Bye
-- Forwarded message --
From: nik600 nik...@gmail.com
Date: Sat, 7 Mar 2009 15:21:14 +0100
Subject: add a new queue strategy: SBR
To: Asterisk Developers Mailing List asterisk-...@lists.digium.com
Hi to all
My recommendation would be to stay away from VoIP even T.38 whenever possible.
That said yout best option is to use TDM, for that you can either use
1 single span T/E1 from digium and an analog TDM card for FXS.
Or you could uee a dual span T/E1 card and a channel bank with FXS
ports. While the
When I said to stay away from VoIP I meant when it comes to faxing.
On 3/8/09, C F shma...@gmail.com wrote:
My recommendation would be to stay away from VoIP even T.38 whenever
possible.
That said yout best option is to use TDM, for that you can either use
1 single span T/E1 from digium and
the queue already have prioritys.
David
2009/3/8 nik600 nik...@gmail.com
Hi., do you think that sbr policy in queue strategy will be useful?
Bye
-- Forwarded message --
From: nik600 nik...@gmail.com
Date: Sat, 7 Mar 2009 15:21:14 +0100
Subject: add a new queue strategy:
but priority are se to the call, not to the agent!
or am i wrong?
On Sun, Mar 8, 2009 at 5:32 PM, David fire ddf...@gmail.com wrote:
the queue already have prioritys.
David
2009/3/8 nik600 nik...@gmail.com
Hi., do you think that sbr policy in queue strategy will be useful?
Bye
Hello!
Oh, yes, I will be mirroring the harddrives in case of any failures.
What is your opinion about using (software) RAID? Do you think the
overhead impacts performance too much?
In an ideal situation, I would use hardware RAID, but that is not
feasible right now.
Thanks,
Elliot
On Sun,
On 8 Mar 2009, at 17:04, Elliot Murdock wrote:
What is your opinion about using (software) RAID? Do you think the
overhead impacts performance too much?
There *should* be very little disk access if you get it right. We run
plenty of voice stuff on software raid. Wouldn't worry.
Steve
On Sun, 8 Mar 2009, Elliot Murdock wrote:
Hello!
Oh, yes, I will be mirroring the harddrives in case of any failures.
What is your opinion about using (software) RAID? Do you think the
overhead impacts performance too much?
In an ideal situation, I would use hardware RAID, but that is not
Faxing over IAX locally works fine.
--
Sent from mobile device
On Mar 8, 2009, at 8:46 AM, Doug Lytle supp...@drdos.info wrote:
benoit wrote:
Remco Barendse a écrit :
On Sun, 8 Mar 2009, benoit wrote:
Here is my current setup:
E1 = [Asterisk with TE220p] = IAX Trunk (routed network)
Just transfer them to your meetme extension after you've called them.
Just like you would transfer someone who has called you.
* will then put them into that conference.
Thanks.
On 08/03/2009, Sven Geggus use...@fuchsschwanzdomain.de wrote:
Hello,
setting up Meetme was very easy. I jut added
Hello!
There will be disk writing in these areas:
1. Logs
2. CDRs
3. MYSQL Call logs
4. Faxes and voicemail
Also, there will be a lot of codec encoding/decoding from/to the PRI
devices, which is my main concern with CPU load.
Cheers,
Elliot
On Sun, Mar 8, 2009 at 7:56 PM, Gordon Henderson
On Sun, 8 Mar 2009, Elliot Murdock wrote:
Hello!
There will be disk writing in these areas:
1. Logs
2. CDRs
3. MYSQL Call logs
4. Faxes and voicemail
I'd not consider these to be a heavy load myself...
Also, there will be a lot of codec encoding/decoding from/to the PRI
devices, which is
Hello Gordon,
Aside from alaw and ulaw, we also use G729.
I am not that familiar as to how Asterisk converts PRI signals into
coded format, but why wouldn't any transcoding be necessary for alaw
and ulaw codecs?
Regards,
Elliot
On Sun, Mar 8, 2009 at 8:52 PM, Gordon Henderson
Sorry, yes it is a Windows Application. I'm running on version 7.0. I have
noted and searched and found people with version 2.0 has been able to get it
working. But getting in touch with those guys is a mission and a half.
Thanks,
D
On 8/03/09 9:53 PM, Andrew Joakimsen joakim...@gmail.com
you are wrong.
when you set up an agent in a queue you can put a priority.
David
2009/3/8 nik600 nik...@gmail.com
but priority are se to the call, not to the agent!
or am i wrong?
On Sun, Mar 8, 2009 at 5:32 PM, David fire ddf...@gmail.com wrote:
the queue already have prioritys.
David
On Sun, 8 Mar 2009, Elliot Murdock wrote:
Hello Gordon,
Aside from alaw and ulaw, we also use G729.
I am not that familiar as to how Asterisk converts PRI signals into
coded format, but why wouldn't any transcoding be necessary for alaw
and ulaw codecs?
It shouldn't have to convert them as
Noojeeclick?
http://www.noojee.com.au/Page/NoojeeClick
ADM? (asterisk desktop manager?)
PaulH
Alan Lord (News) wrote:
Dean Collins wrote:
ADA Forums: http://forums.digium.com/index.php?c=8
ADA Download: http://dl1.digium.com/ADA/ADAInstall.exe
ADA Administrators Guide:
On 9/03/2009 8:08 a.m., Elliot Murdock wrote:
Hello Gordon,
Aside from alaw and ulaw, we also use G729.
I am not that familiar as to how Asterisk converts PRI signals into
coded format, but why wouldn't any transcoding be necessary for alaw
and ulaw codecs?
Because a T1 uses mulaw and an
Gavin Henry gavin.he...@gmail.com wrote:
Just transfer them to your meetme extension after you've called them.
Hm, how would I do this? Until now call switching usually ended for me when
the call has been established.
I'm using a SIP phone connected to an asterisk box which is connected to the
David fire wrote:
you are wrong.
when you set up an agent in a queue you can put a priority.
David
The term used in Asterisk for a queue member's priority is the word
penalty. When you set up a member in queues.conf, the penalty is the
third option for a member. Here's an example:
member =
I manged to get something working but It's only working when a
grandstream ip phones is the one tranfering calls. With linksys IP
phones I get a busy yone
I edited the extensions.conf
last lines of [macro-exten-vm]:
; Extensions with no Voicemail box reporting BUSY come here
exten =
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