I have been running FC5 for 3 years with no issues, I also started using for
FC7 the last 2 months. I have not had any issues.
On 2/3/08, Goke Aruna [EMAIL PROTECTED] wrote:
Tzafrir Cohen wrote:
On Sat, Feb 02, 2008 at 05:13:39PM -0600, [EMAIL PROTECTED] wrote:
Ubuntu server for me please
My main problem is when I have two phones at the home office, the second
phone cant register, and/or, you cant here the voicemail greeting when you
try to check messages.
I have seen this before on badly behaved home routers that have a hidden SIP
Proxy, notably Zyxel wireless units. I've not
Hello,
On Feb/01/2008, Doug wrote:
The client is travelling much of the time.
Is there some way that he can use Port 80 so
that the firewalls that he is behind won't
block the connection?
You can also use OpenVPN (using port 80, I don't know if it's possible
to use TCP, I think so).
So
I've upgraded to asterisk-1.6.0-beta2.
I'm trying to get the new Telco MWI detection function working. It
doesn't appear to be working.
I have this in zapata.conf
; PSTN connected here
;immediate=no
;busydetect=yes
;busycount=8
;musiconhold=default
mwimonitor=yes
;mwilevel=512
Do you have a range of registration ports configured and forwarded through
the firewall on the server end? Ie. 5060-5065 for example.
On the Phone side you should forward 5060 to phone1 and 5061 to phone 2 etc.
and configure the phones to use that port for registration. You may need to
Dear List,
We are tearing out legacy PBX and replacing with Asterisk PBX and new
LAN for our 90+ person operation. Question: what QOS capabilities
(protocols, etc) does Asterisk support/require in a LAN switch to deliver
business grade phone service? Thanks
Test
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.
http://tools.search.yahoo.com/newsearch/category.php?category=shopping___
--
John Williams [EMAIL PROTECTED] writes:
We are tearing out legacy PBX and replacing with Asterisk PBX and new
LAN for our 90+ person operation. Question: what QOS capabilities
(protocols, etc) does Asterisk support/require in a LAN switch to deliver
business grade phone service? Thanks
Lee Jenkins wrote:
Jared Smith wrote:
On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote:
What format is the LastCall variable of QueueMember event? I'm looking at:
1201897536 for instance.
Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I
recall.
Thanks.
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Hash: SHA1
Hi All,
Just to throw in my 2 cents worth, I have been evaluating most of the
'big' distro's for my Asterisk platform and since I've used almost all
of them I kept the pre-requisites to these standard options.
1. Package management - easy to
I need to set up the sound card of a server to use in an overhead paging
system, as normal I am testing this on my home machine first (which has
slightly different Hardware).
I'm using chan_alsa with the Intel HD Audio driver on an Intel 82801G
(ICH7 Family) sound card.
I am running Asterisk
Thomas Kenyon wrote:
When calling console/dsp (using various methods) the output from the
sound card changes the voice to sound like a Dalek, which looks similar
When this happens to me, I switch. If is sounds badly with Alsa, try OSS.
Doug
--
Ben Franklin quote:
Those who would
I thought sounding like a dalek was a good thing.
PaulH
On Sun, 2008-02-03 at 23:56 +, Thomas Kenyon wrote:
I need to set up the sound card of a server to use in an overhead paging
system, as normal I am testing this on my home machine first (which has
slightly different Hardware).
I have the following situation: I drop a call-file into the Asterisk
spool directory and I get called back. That all works.
And I have this script:
#!/usr/bin/perl -w
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer();
my $i;
On Sun, 03 Feb 2008 22:11:04 +0100, Benny Amorsen wrote:
John Williams [EMAIL PROTECTED] writes:
We are tearing out legacy PBX and replacing with Asterisk PBX and new
LAN for our 90+ person operation. Question: what QOS capabilities
(protocols, etc) does Asterisk support/require in a LAN
a) The call file
Channel: Zap/g4/0409227633
MaxRetries: 0
RetryTime: 60
WaitTime: 30
Extension: 0409227633
Callerid: 0409227633
Context: barnet-callback
Priority: 1
b) the snippet of extensions.conf that this is called in
;
; dial back
;
exten = 0293353699,1,AGI(callback1.agi)
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Have you tried with AGI Debug on?
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News -
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Hash: SHA1
Edwin Groothuis wrote:
I have the following situation: I drop a call-file into the Asterisk
spool directory and I get called back. That all works.
And I have this script:
#!/usr/bin/perl -w
use Asterisk::AGI;
my $AGI = new
Have you tried with AGI Debug on?
Yes! Even before you asked :-)
This is when I use DeadAgi (for some reason):
-- Executing [EMAIL PROTECTED]:3] DeadAGI(Zap/4:103-1, callback2.agi)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/callback2.agi
AGI Tx agi_request:
The MWI detection is done using fsk modem detection within chan_zap itself.
(It does not support neon MWI detection.) The driver plays no real part in the
detection.
Doug Bailey
- Original Message -
From: Jim Duda [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Al lists wrote:
Theoretically, setting TOS value ( these days called DSCP) wont change
anything in switch behavior, unless you are using Layer 3 switches.
What makes a difference in a switch is COS bits, and i'm not sure how
asterisk sets that.
I guess to be safe, you would need to create 2
Theoretically, setting TOS value ( these days called DSCP) wont change
anything in switch behavior, unless you are using Layer 3 switches.
What makes a difference in a switch is COS bits, and i'm not sure how
asterisk sets that.
I guess to be safe, you would need to create 2 VLANS and in the
Hi all
Just trying to set up a queue and wondering if this is possible.
We have 3 agents, One of them is sort of the first point of contact
What i am looking to do is
1. Someone rings the queue.
2. It rings Agent A.. If Agent A is on the phone then put them on hold
for 120 seconds, and if
Hi
Thanks for the response
Anthony Messina said the following on 01-Feb-08 03:36 PM:
On Thursday 31 January 2008 11:52:09 pm Ian wrote:
Sorry for taking so long to reply,
This email got lost in translation, again.
Ian
Ian said the following on 30-Jan-08 03:57 PM
Thaks for the
Здравствуйте, Mindaugas.
MK Download for Pentium4
MK The output of cat /proc/cpuinfo giving a [Intel (R)
MK Pentium (R) D] so what is the g729 version I have to
MK download to work with my machine?
If it is not loaded in Asterisk successful - try for Pentium
--
Alexey
Hi
I have a Server with Centos 5,
TDM400p, HP Server ML110.
My problem is that I see IRQ Share with my TDM400P.
How can I fix that???
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Al lists [EMAIL PROTECTED] writes:
Theoretically, setting TOS value ( these days called DSCP) wont change
anything in switch behavior, unless you are using Layer 3 switches.
What makes a difference in a switch is COS bits, and i'm not sure how
asterisk sets that.
Sadly Asterisk still calls
Dears
Any one knows a standalone voip transcoder software name,not an ip pbx.
What I want is to transcode the incoming sip calls from g711 to g723 or
ilbc or g729 . and forward it to a media gateway ..
Regards
Khaled chehab
*
No
Thanks for the speedy reply
Tzafrir Cohen said the following on 30-Jan-08 12:37 PM:
On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote:
Hi all
I have a small problem here. I asked this question on another asterisk
mailing list, but nobody seemed to be able to help me there.
We are
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