Displaying 20 results from an estimated 6000 matches similar to: "Please advise: Asterisk on Dell PowerEdge 1750 w/ hyperthreading"
2010 Oct 20
4
Recommendation for a new server
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
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2010 Oct 23
3
Cepstral voice quality not good
Hello list,
I have been using Cepstral's 8KHz voices for my text-to-speech service for
some time now, and have been noticing that the voice quality is really poor,
doesn't matter what phrase I give it to convert. None of the other 8KHz
voices I have ever used were this bad. It doesn't seem good enough system to
be used in a commercial system. Is there any better quality text-to-voice
2008 Oct 17
5
How to add contexts in asterisk realtime?
Hi everybody,
How can we add new contexts in asterisk realtime module? All I could figure
out after googling is that a new context HAS to be declared in
extensions.conf with 'switch => Realtime/@<databasetable>' under the context
name declaration. This works fine as long as we are adding extensions only
to this one context, but doesn't give the freedom to add new contexts for
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2010 Mar 18
6
Asterisk DIES with no trace. PLEASE
Thanks Zeeshan,
SAngoma told me that the asterisk problem is unrelated to wanpipe drivers,
they told me to reinstall asterisk again
But, i still having doubts about the problem :(
Thanks in advance
>
> Message: 10
> Date: Thu, 18 Mar 2010 00:21:06 -0400
> From: Zeeshan Zakaria <zishanov at gmail.com>
> Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE
>
2009 Jul 11
2
Suggestions for web based soft phones
For a while now I've been looking for a good web based soft phone solution,
but so far no luck. A few solutions which I've tried, both Java based and
Flash based, either don't work, or had bad sound quality. I need something
which I could put on my productions server for my clients.
Seems like good web based solutions are all paid ones, nobody is giving it
for free. Any ideas,
2009 Jul 14
3
Why CDR is recording dst value = h?
For a new project, I have written a dialplan and it is pretty straight
forward: The [dialout] context dials out a number, and h extension in this
context writes the CDR. But what is happening is that if the callee hangs up
first, all values in the CDR are fine, but if the caller hangs up first, the
'dst' column is always 'h'. I put a NoOp right in the begining of this macro
to
2011 Jan 21
4
Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
Hi list,
For a client I am setting up a system which will use T1 PRI from Primus, who
offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only
used switchtypes euroISDN and National. Although the documentation says
Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you
have used these protocols on an Asterisk box and if there were any things to
consider. If
2007 Jul 18
5
In Vancouver is it a local to call from 778 to 604, and vice versa?
I've got a 778 DID for vancouver, but don't know if it will be a local call
if dialed 604 and vice versa.
What are the different area codes in Vancouver and why its easier to get 778
DID than 604?
--
Zeeshan A Zakaria
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2010 Oct 30
8
Under heavy attack
My main asterisk server is under unusual heavy attack, and so far Fail2Ban
has blocked about 30 IPs, from various different countries. At this time it
is blocking about 1 IP address every few minutes.
Just wondering if anybody else is also experiencing unusually increased hack
attempts today?
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
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2010 Sep 14
6
How different is implementing Cisco based system than Asterisk based system?
Hello list,
Slightly off the list topic, but I hope I'll get some help here. Somebody
wants me to implement for his project a Cisco based VoIP system. I told him
that I specialize in Asterisk based systems, but he is not even aware of
Asterisk. The requirement of project is such that chances are slim that this
firm will consider Asterisk based system. So I told him that though not
experienced
2010 Oct 16
6
Remote Unix Connection
Hi,
Does anyone know where this is suddenly coming from?
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Thanks
Dan
p.s. sorry about the last post. hit the mouse by mistake and it sent the email.
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2006 Oct 31
3
Asterisk and ARI (Aterisk Recording Interface) integration problem
Anybody knows why ARI gives this error message when I enter extension number
and password.
*Warning*:
file(/var/spool/asterisk/voicemail/default/222/INBOX/msg0000.txt): failed to
open stream: Permission denied in *
/var/www/html/recordings/modules/voicemail.module* on line *525*
It doesn't show the voicemails, although it shows that there is 1 or 2
voicemails in the INBOX.
--
Zeeshan A
2006 Sep 15
3
Hyperthreading ON causes OOPS on boot
I just tried to install CentOS (x86_64 4.4 and i386 4.3, not both at
once, when I found the 64bit version didn't boot, I tried the 32bit
version, same result) on a dual Naconda Xeon system. I get an OOPS just
after the IDE info is printed on the screen during boot (this is booting
off the CD during the install phase).
After mucking about with the bios settings (and many reboots later), I
2009 Jan 19
1
Need help registering Cisco 7960 Phones on Asterisk
Hi everyone,
I googled this followed the instructions, but it hasn't work for me yet.
I have universal setting in SIPDefault.cnf and phone specific settings in
SIPXXXXXXXXXX.cnf. But it doesn't get registered.
I need to register it on two different asterisk boxes. So my
SIPXXXXXXXXXX.cnf looks like this:
phone_label: "Zeeshan A Zakaria"
line1_name: "523"
2006 May 03
3
hyperthreading and zaptel
My Dell 2800 Dual 64bit Proc. machine came in with hyperthreading enabled. (they call it virtual processor??)
I have been intending for a month to disable it.
Tonight, I rebooted, turned it off, and let the system come up.
zaptel loaded and asterisk loaded, but both of my t1s were red. (it is a TE411P)
/proc/interrupts looked OK, zttest gave OK numbers.
I doublechecked all of the files in case
2006 Dec 10
10
Recommendations for QoS, PoE Switches
Hi all,
For a top quality setup, I will need to install high quality VoIP switches
with QoS and PoE. My potential customer should not have any problem with
call quality. Experienced folks, Please advice me what switches to install
and at what price. I may need it for upto 100 phones. What else should I
consider so that phones work without problem along with the computers on the
same network?
2010 Mar 26
7
Asterisk load balancing and failover
Hi List,
I'm finding a solution to provide failover and load balancing features to my IVR system.
Anyone suggest me what is the best solution please?. what the hardware I should use ?.
I heard about RedFone, but someone on the mail list said that it is not good because TDMoE module in asterisk is not so stable and TDMoE is stale. And It seems that RedFone doesn't not support load
2007 May 09
5
Audio going blank for a few seconds and then comes back. What could be the reason?
Hi,
Everything was working fine on this 10 phone office, but for last few weeks
they are complaining that audio goes blank for a few seconds during the
conversation, and then comes back on. It goes blank for both parties.
What are the possible causes for this to happen?
--
Zeeshan A Zakaria
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2010 Jun 15
4
can't seem to register, status unmonitored
Hi everybody,
I am trying to register my softphone(twinkle) on an asterisk server.
Everything seems to be fine.
Here is the output on show registrations in twinkle:
Tue 18:57:51
nikhil: you have the following registrations
<sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013
208 is ip of the asterisk server.
on the server on doing 'sip show peers' , it