On Sun, 8 Jun 2008, Matt Florell wrote:
Hello,
We routinely run meetme with over 140 ULAW channels connected to 70
meetme rooms with no issues on an Intel Core 2 Quad core CPU.
The major factor in capacity would be your CPU and RAM capacity. If
you have at least a base-level P4 you don't
Also try putting Asterisk in the audiopath by setting canreinvite=no in
sip.conf
Regards
Ian
On Sat, Jun 7, 2008 at 4:07 PM, Michiel van Baak [EMAIL PROTECTED]
wrote:
On 08:36, Sat 07 Jun 08, Russell Bryant wrote:
On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote:
i have this on my
well, I'm from snom, I would be interested how you measured that a snom370
takes 7 Watts :), My PoE switch tells me something below 2 Watts (1,5
Standby).
As a cheap, quit alternative for Europe, Allnet our distributor has an 8
Port Switch with 4x PoE, price is something below 100 Euros. As it's
2008/6/6 Ron Wellsted [EMAIL PROTECTED]:
Kevin Smith wrote:
Hi everyone,
Perhaps I am just mis-reading the documentation, but for call recording,
is it possible to record the conversation over a SIP channel? We have
call recording preformed on all of our ZAP connections, but I was
wondering
On Mon, Jun 9, 2008 at 12:36 AM, Sanjoy Rath [EMAIL PROTECTED] wrote:
I have installed Asterisk. I want friends to connect to my asterisk server
from their SIP Phones are talk to me. I have tried two ways 1.) Have the
Asterisk server run within the firewall, opened all the ports for that
Ed Nunez wrote:
I have found the answer to my question.
It's also worth noting (I'm sure you spotted it), That you have 2
priority 1 entries for 8484 in your extensions.conf.
extensions.conf
exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)
Hi
One of my SIP providers need me to send the Remote-Party-ID with privacy=on to
withhold CLI and privacy=off to show CLI. I want the option to withhold CLI
selectable by my users. I have set sendprid=yes in the sip.conf but I cant find
a way to toggle the privacy between on and off on a per
The Asterisk development team has released Asterisk version 1.4.21-rc2.
This release is a release candidate for the upcoming official release of
1.4.21. A few bugs have been fixed since 1.4.21-rc2. Please continue
to assist in testing before we release 1.4.21!
The release candidate is
I looked at quite a few options over the course of somewhere around 9
months.
We ended up going with Polycom VSX7000 series units. These units are
really designed to use H.323, but they have a SIP option that works
almost just as well. The only thing I seem to be missing that I've
noticed is
On Thu, Jun 5, 2008 at 9:19 PM, Guillermo Salas M.
[EMAIL PROTECTED] wrote:
El vie, 06-06-2008 a las 00:24 +0200, Matias Surdi escribió:
At the company I work for, we use Asterisk to communicate with our
offices all around the world. Recently, I've been asked to implement
a
video conference
Moises, we've already set debug level at 255 on unicall.conf and at
logger.conf we've enabled full log (notice,warning,error,debug,verbose).
Has anyone experienced with a Siemens EWSD switch?
Anyone knows about to change R2 timers at unicall.conf ?
Please any comment is welcome, thank you..
Sherwood McGowan wrote:
Gentlemen,
I have a particularly strange problem, just started happening. One of
my clients is running Asterisk 1.2.28 and has mysql realtime queues.
We log in a member, and then place a test call to the 0 queue but
since joinempty is set to no, and Asterisk thinks
Quoting Jon Farmer [EMAIL PROTECTED]:
Hi
One of my SIP providers need me to send the Remote-Party-ID with
privacy=on to withhold CLI and privacy=off to show CLI. I want the
option to withhold CLI selectable by my users. I have set
sendprid=yes in the sip.conf but I cant find a way
Sherwood McGowan wrote:
Sherwood McGowan wrote:
Gentlemen,
I have a particularly strange problem, just started happening. One of
my clients is running Asterisk 1.2.28 and has mysql realtime queues.
We log in a member, and then place a test call to the 0 queue but
since joinempty is set to
Hi
Thanks for that, got it working selectable by user now.
I did know about the SetCallPres() but it had temporarily slipped my mind :-)
Regards
Jon
- Original Message
From: Phil Reynolds [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, 9 June, 2008 4:09:34 PM
My employer has recently moved from a Checkpoint firewall to MS ISA, or
so I'm told. Does anyone have and advice on configuring this to pass SIP
to/from a hard phone inside the LAN? They have one Polycom IP430 that
they need to register with an external hosted provider.
Michael Graves
mgraves at
2008/6/9 Sherwood McGowan [EMAIL PROTECTED]:
Sherwood McGowan wrote:
Gentlemen,
I have a particularly strange problem, just started happening. One of
my clients is running Asterisk 1.2.28 and has mysql realtime queues.
We log in a member, and then place a test call to the 0 queue but
since
On Mon, 2008-06-09 at 00:26 -0500, Sherwood McGowan wrote:
Members:
9001 (Invalid) has taken no calls yet
It appears that there are no valid members of the queue, which at first
glance would seem to me to be your problem.
--
Jared Smith
Training Manager
Digium, Inc.
I've been thinking about something around these lines that I'd like feedback
on. What I'd like to d,o if it works, is have a fax machine in St. Louis
connected up to my asterisk box in Atlanta via Internet/SIP so that anytime
the fax machine in St Louis sends a fax it actually goes out through
Drew Gibson wrote:
Tilghman Lesher wrote:
My fxotune.conf:-
13=7,255,251,251,2,255,255,1,255
14=9,254,251,255,2,0,1,0,0
15=9,254,251,255,2,0,1,0,0
16=5,0,0,0,0,0,0,0,0
17=9,254,251,255,2,0,1,0,0
18=9,254,251,255,2,0,1,0,0
19=9,254,251,255,2,0,1,0,0
20=9,254,251,255,2,0,1,0,0
You should not expect FaxOverVoiceOverIPOverInternet to work well. If
you stick to ulaw codec for the entire call, it might work well enough
for your use, but it might not.
John Morey wrote:
I've been thinking about something around these lines that I'd like feedback
on. What I'd like to
Hi All,
Can someone provide me a step by step guide to install and configure Asterisk
1.2 with Radius using agi scripts. I have currently installed andconfigured it
but it is not disconnecting the call after the credit_time returned by radius.
So I am guessing I may have missed some
On 6/9/08, John Morey [EMAIL PROTECTED] wrote:
I've been thinking about something around these lines that I'd like feedback
on. What I'd like to d,o if it works, is have a fax machine in St. Louis
connected up to my asterisk box in Atlanta via Internet/SIP so that anytime
the fax machine in
John Morey wrote:
actually goes out through the asterisk box in Atlanta. Something if I
understand it correctly like : Fax-SIP(long
distance)-Asterisk-FXO-Customer Fax. Would something like this work?
Not reliably,
if you have a VPN connection to the remote site (We do with our remote
Hi All;
fring that used in the mobile phones, does it support
g729? Anyone can advise?
Regards
Bilal
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To UNSUBSCRIBE or update options
It should work.
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Times Telecom
Tel: 604-279-8787 ext 1586
Fax: 604-278-2793
Mobile: 604-780-2668
Click this button now and leave your phone number. Talk to me for free.
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Abid Saleem wrote:
Hi All,
Can someone provide me a step by step guide to install and configure
Asterisk 1.2 with Radius using agi scripts. I have currently installed
andconfigured it but it is not disconnecting the call after the
credit_time returned by radius. So I am guessing I may
Hi Philippe,
On Wed, Jun 4, 2008 at 7:36 AM, Philippe Sultan [EMAIL PROTECTED]
wrote:
Hi Matt,
On Wed, Jun 4, 2008 at 1:05 AM, Matthew Gibson [EMAIL PROTECTED]
wrote:
I'd be interested to know more about the status abilities as well, we've
tried to test jabberstatus application, but it
I'm looking for reports of recent experience with redfone fonebridge2
(with echo can) TDMoE gizmos.
Anybody? Good? Bad?
smime.p7s
Description: S/MIME Cryptographic Signature
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Hello there,
I have just finished the Asterisk setup for 3G video
calls and tried to test with my Samsung SGH-G800 but
no success.The phone says Dialing for 20-30 seconds
and call is disconnected to the end.
Any tips/suggestion to get it working are most
aprreciated.I have asterisk 1.4.19.2,
Jared Smith wrote:
On Mon, 2008-06-09 at 00:26 -0500, Sherwood McGowan wrote:
Members:
9001 (Invalid) has taken no calls yet
It appears that there are no valid members of the queue, which at first
glance would seem to me to be your problem.
Thank you to all who
On Monday 09 June 2008 12:27:39 Drew Gibson wrote:
I thought I had found something, all of the lines were patched in with
Cat 5 patch cords except Port 16 which had a telephone cable (which
would flip the polarity). After changing all the patch cables to
telephone type, I re-ran fxotune but
We have 2 T1's coming from our phone switch to a digium TE220B. We have
managed to get CPN and the extension outpulsed from the switch, but call
setups are really slow.
Our T1's are set up as EM Wink, and they send us the last 5 digits
dialed followed by the 10 digit calling party number (we
Apologies - I know this isn't either Polycom or ISC support, but if
anyone would have an answer to my problem, I'm certain they would be on
this list.
I'm experiencing odd behavior with Polycom handsets obtaining DHCP
addresses. It always worked fine for me up until a few months ago.
Hello everybody!
Is it possible to set call on hold via dialplan application,then call other
commands,then call Dial() and connect call on hold to called channel?
Thank you!
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Denis V. Gudtsov wrote:
Hello everybody!
Is it possible to set call on hold via dialplan application,then call other
commands,then call Dial() and connect call on hold to called channel?
Yes:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce
Doug
--
Ben
On Mon, 9 Jun 2008, Eldon Koyle wrote:
We have 2 T1's coming from our phone switch to a digium TE220B. We have
managed to get CPN and the extension outpulsed from the switch, but call
setups are really slow.
Our T1's are set up as EM Wink, and they send us the last 5 digits
dialed followed
Hello all,
I've got an Asterisk system I'm working on here, and we often dial
remote IVR systems, where our end must enter an extension to get to a
remote user. We're using Polycom hardphones here, speaking SIP, and
Asterisk sends these out over a PRI line with Zaptel hardware.
I've used rtp
Hola a todos, estoy creando una comunidad de asterisk en español que
se dividira en un blog y un foro, estoy buscando gente que quiera
ayudarme a escribir articulos para el blog, y claro, pueda participar
en el foro.
Si a alguien le interesa saber mas escribanme un mail.
[EMAIL PROTECTED]
http://67.169.112.100/openmeetings/ Its OSS, runs on Linux and is not
buggy
- Original Message -
From: Sanjoy Rath
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk video alternatives
Date: Mon, 9 Jun 2008 04:41:06 +
Hello,
To add, here's one weird difference (how am I missing VLDTMF events?):
Broken:
sur-pbx-1:/home/martins# grep -i dtmf rfc2833-broken | grep -i chan_zap
[Jun 9 16:26:21] DEBUG[11028] chan_zap.c: Started VLDTMF digit '2'
[Jun 9 16:26:21] DEBUG[11028] chan_zap.c: Ending VLDTMF digit '2'
Working:
John Morey wrote:
I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran
fxotune to tune the lines. fxotune.conf ended up looking exactly the same
as before the change. Since I was expecting/hopping to see a change but did
not I switched everything back to the way it was. Is
Steve Totaro wrote:
I have consulted on so many systems with poor audio, the first thing I
check is IAX or SIP. If IAX, I move over to SIP and the calls are
prefect.
I avoid IAX at all costs, use OpenVPN, open tons of ports on your
firewall, whatever you can do to use SIP. The only time I
Thanks all for the info. Yes I do have HylaFAX running and was thinking
about either the print-to-fax or email-to-fax route but for some reason the
remote site loves to write stuff on the faxes before they send them. It's
something, the writing on the fax, they are not used to and I've been told
Mariano:
Could you send us please the log files, and the console output... so we
can help you.
On Mon, Jun 9, 2008 at 8:01 AM, Mariano Borgognone
[EMAIL PROTECTED] wrote:
Moises, we've already set debug level at 255 on unicall.conf and at
logger.conf we've enabled full log
On Sun, 2008-06-08 at 00:45 +0200, Jan Eirik Sandnes wrote:
Actually, i do get CDR on the originate, but NOT on the Dial() in the
context provided in the originate.
It looks like this:
[callgw]
exten = _X.,1,Set(CALLERID(num)=1123)
exten = _X.,n,Set(CALLERID(name)=John Travolta)
exten =
On June 9, 2008 12:57:11 pm John Morey wrote:
I've been thinking about something around these lines that I'd like
feedback on. What I'd like to d,o if it works, is have a fax machine in
St. Louis connected up to my asterisk box in Atlanta via Internet/SIP so
that anytime the fax machine in St
John Morey wrote:
Thanks all for the info. Yes I do have HylaFAX running and was
thinking about either the print-to-fax or email-to-fax route but for
some reason the remote site loves to write stuff on the faxes before
they send them. It's something, the writing on the fax, they are not
We had an outage from our ISP this afternoon that cut prevented us from
connecting
to our SIP provider (someone physically cut a line downstream). All our phones
inside
the office stopped working as well? Why is that, and how can I set this up so
phones
can still dial each other inside the
On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote:
You should not expect FaxOverVoiceOverIPOverInternet to work well. If
you stick to ulaw codec for the entire call, it might work well enough
for your use, but it might not.
Just as an FYI - you have too many Over's in your
On Mon, 09 Jun 2008 16:51:32 -0600, Joseph L. Casale wrote:
We had an outage from our ISP this afternoon that cut prevented us from
connecting
to our SIP provider (someone physically cut a line downstream). All our phones
inside
the office stopped working as well? Why is that, and how can I set
Are they just a trunk? Or are they your full PBX? If they are the full
PBX, they handle the dialplan for dialing between phones, so there is no
way around this. You would instead have to have your own Asterisk box at
the same location as your phones, and use them for trunking if this is
what you
I don't think is possible to change the R2 timers in unicall.conf, if you
want to do something like that, maybe mfcr2.c in the libmfcr2 source will
give you that chance.
What happend to me once, is that I couldn't complete long distance calls
using telco's E1 (Avantel, Mexico). At the end the
The exact question pose I must leave for others to answer.
However, I recently completed a project that overcomes the situation
you describe. I installed a cellular gateway giving me a wireless
trunk. If I lose IP connectivity I can route calls out through my cell
carrier. Works really well.
On Mon, 09 Jun 2008 17:00:50 -0600, Joseph L. Casale wrote:
The exact question pose I must leave for others to answer.
However, I recently completed a project that overcomes the situation
you describe. I installed a cellular gateway giving me a wireless
trunk. If I lose IP connectivity I can
I'm using software echo cancellation. ztcfg says its MG2. In zapata.conf I
have echocancel=64 and echotraining=256 set. I'm going to try Digium's hpec.
I've did an online request for the free licenses, I'm using Digium TDM400
and TDM800 series cards, yesterday and am waiting to hear back.
On
Mariano Borgognone wrote:
Moises, we've already set debug level at 255 on unicall.conf and at
logger.conf we've enabled full log (notice,warning,error,debug,verbose).
Has anyone experienced with a Siemens EWSD switch?
Anyone knows about to change R2 timers at unicall.conf ?
Please any
What type of PBX hardware do you have on-site? Also what make/models of
phones?
Michael/Darryl,
I do have a local asterisk box, which is why I am baffled. I am new to Asterisk
and there is lots to learn, but my config is pretty basic, my sip.conf simply
has
the phones and single sip provider
in this whole thread are we missing a subtle difference? that being the
difference between inter vs. intra office. when your wan connectivity drops
I'd expect your INTERoffice (from one office to another) calls to fail.
INTRAoffice (within the same office) calls should work though.
Eric
On
On June 9, 2008 07:49:13 pm Joseph L. Casale wrote:
What type of PBX hardware do you have on-site? Also what make/models of
phones?
Michael/Darryl,
I do have a local asterisk box, which is why I am baffled. I am new to
Asterisk and there is lots to learn, but my config is pretty basic, my
in this whole thread are we missing a subtle difference? that being the
difference between inter vs. intra office. when your wan connectivity drops
I'd expect your INTERoffice (from one office to another) calls to fail.
INTRAoffice (within the same office) calls should work though.
Eric
I've seen this behaviour from Asterisk as well... while I can't say I have
tracked it down and verified this... I've seen other talks about how Asterisk
gets rather unhappy when it can't preform DNS queries. I suspect that may be
your problem. Might want to check the archives for other issues
Hi,
I have what I think is a relatively advanced question. Any help is
appreciated, even if it's not a complete answer.
I am using Asterisk in mostly realtime fashion, specifically SIP
registrations are in a MySQL table. This works fine (mostly). I also set a
few variables in the setvar
Add your local Asterisk server hostname to your /etc/hosts.
I would also go as far as running a local DNS server and just having the
phones and server point to it. It is a small CPU load application so it
can be hosted on your own machine.
Use the tools for DNS and make sure your machine can
On Mon, 9 Jun 2008 20:32:29 -0400, Matt Watson wrote:
On June 9, 2008 07:49:13 pm Joseph L. Casale wrote:
What type of PBX hardware do you have on-site? Also what make/models of
phones?
Michael/Darryl,
I do have a local asterisk box, which is why I am baffled. I am new to
Asterisk and there
I have used ISA with out issue. Although it was configured in a very
trusting way. (ie No filters) If filters are applied you may want to
read up on iptables and its effect of Asterisk and SIP. (You can Google
for that) You will then have to translate the commands b/w iptables and
Michael Graves wrote:
On Mon, 9 Jun 2008 20:32:29 -0400, Matt Watson wrote:
On June 9, 2008 07:49:13 pm Joseph L. Casale wrote:
What type of PBX hardware do you have on-site? Also what make/models of
phones?
Michael/Darryl,
I do have a local asterisk box, which is why I am baffled. I am
Change the order of resolution (hosts first, then DNS) and add relevant
entries to your hosts table. That makes asterisk happy w/o an internet
connection.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: June 9, 2008 9:09 PM
To:
- Original Message -
From: Joseph L. Casale [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, June 10, 2008 10:47 AM
Subject: Re: [asterisk-users] Interoffice phone setup
I've seen this behaviour from Asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tzafrir Cohen wrote:
On Wed, Jun 04, 2008 at 04:06:28PM +1200, Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tilghman Lesher wrote:
On Tuesday 03 June 2008 10:12:58 Todd Reese wrote:
Hi All,
I'm stumped on this and I
On Monday 09 June 2008 20:01:53 Mike wrote:
I have what I think is a relatively advanced question. Any help is
appreciated, even if it's not a complete answer.
I am using Asterisk in mostly realtime fashion, specifically SIP
registrations are in a MySQL table. This works fine (mostly). I
They can now turn off their internet connection and everything works fine.
We left the internet down for 30mins.
I am worried that if the cache time on the DNS server runs out the problem
may come back, but this is set to 6 hours.
Hope this helps, and if anyone can shed some more light on this
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