We also run asterisk in a virtual environment, VMWare specifically, along side
of web, database, email and DNS (virtual) servers. As far as I'm concerned, it
runs as well as it ever did in a real environment. We are using HP Proliant
DL360 G5's (3gz Xeon 5160 dual core processors). In our
Me too.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria
Sent: Saturday, October 30, 2010 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Under heavy attack
My main
We moved a 1.4 installation to a VMWare environment some time ago and it was
fairly uneventful. Still, if it were me, I wouldn't change too many things at
once and I would first wait until what I currently run is stable under VM.
Once stable, I wouldn't hesitate to upgrade and that's one of
, just plain SIP and IAX.
Zeeshan A Zakaria
--
www.ilovetovoip.comhttp://www.ilovetovoip.com
On 2010-08-24 10:07 AM, Bruce Komito
bru...@wpti.netmailto:bru...@wpti.net wrote:
We moved a 1.4 installation to a VMWare environment some time ago and it was
fairly uneventful. Still, if it were me, I
://www.ilovetovoip.com
On 2010-08-24 10:45 AM, Bruce Komito
bru...@wpti.netmailto:bru...@wpti.net wrote:
We now run VMWare ESXi 4.0 on HP Proliant DL360 G5 and have not had any issues.
A couple of years ago, we tried OpenVZ, but did not have good results. Don't
ask to me explain what the problem
FWIW, we recently moved a 1.4.29 Asterisk system onto a VMWare guest machine
and with 40+ call legs (20+ calls), it isn't even breaking a sweat. We have
had no complaints from users nor have we noticed any degradation in voice
quality, be it live, voicemail or conference bridge (with six
As does ZeroShell (www.zeroshell.net/eng).
Bruce Komito
WPTI Telecom
(775) 236-5815
On Tue, 26 May 2009, Michael Graves wrote:
m0n0wall and pfsense both do traffic shaping, which forcibly allocates
bandwidth for your VoIP traffic.
Michael
On Tue, 26 May 2009 04:32:59 -0700 (PDT), bilal
phones (Cisco 7960's and Linksys 942's). I don't recall seeing
any settings anywhere than have anything to do with echo cancellation on
non-ZAP devices. Anyone have a clue where I should start looking?
TIA
Bruce Komito
WPTI Telecom
(775) 236-5815
and a workaround for it?
TIA
Bruce Komito
WPTI Telecom
(775) 236-5815
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That was the silver bullet...thanks!
Bruce Komito
WPTI Telecom
(775) 236-5815
On Mon, 23 Feb 2009, Tzafrir Cohen wrote:
On Mon, Feb 23, 2009 at 12:26:09PM -0800, Bruce Komito wrote:
Is there some magic to compiling asterisk-addons-1.6.0 under Debian 2.6.18
and mysql 5.0? I am unable
the phones on a LAN segment that does not pass through the Sonicwall.
So, now that's our going in position. If it works, great, but if it
doesn't, our solution is to take the Sonicwall out of the picture.
My $.02 .
Bruce Komito
WPTI Telecom
(775) 236-5815
On Thu, 23 Oct 2008, Bill Michaelson wrote
You're absolutely right. I only mention Sonicwall, because those are the
ones we see most often and there is a perception out there that, because
Sonicwall is the (disputed) leading firewall, it should work.
Bruce Komito
WPTI Telecom
(775) 236-5815
On Thu, 23 Oct 2008, Kristian Kielhofner
is unintelligable. There doesn't seem to be any pattern to this.
It happens with equal frequency on incoming calls from both SIP trunks and
PRIs. I am *not* experiencing any sound breakup on live calls, either on-
or off-net.
Has any else seen anything like this?
TIA
Bruce Komito
WPTI Telecom
(775) 236
Sounds more like a hunt group than a ring group.
Bruce Komito
WPTI Telecom
(775) 236-5815
On Thu, 31 Jul 2008, Ruddy G. wrote:
Why don't you just call the Dial application for each user, one after
another ??
The ones that are busy will just go through. So, on the next priority,
you dial
,
although the cards are not.
Bruce Komito
WPTI Telecom
(775) 236-5815
On Thu, 17 Apr 2008, mark morreny wrote:
Dear all,
A quick question on deploying Asterisk over E1. I am looking for a low-cost
solution for bridging my E1 line and Asterisk with reasonable stability
suppoing both voice and fax
For those CLECs out there, if you know of a contract AOCN that you have
personal experience with and would recommend, please reply. For those who
don't know what an AOCN is, please delete this message.
Bruce Komito
WPTI Telecom
(775) 236-5815
, but that's only a presumption.
Bruce Komito
WPTI Telecom
(775) 236-5815
On Wed, 28 Nov 2007, Thermal Wetland wrote:
Hello,
I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that
the MWI indicators will never go off (The blinking red light and envelope in
the LCD).
I have tried
AND Cisco NAT together, but beyond that, I
haven't a clue.
Has anyone ever seen a message like this, and/or understand the cause and,
better yet, the solution?
TIA!
Bruce Komito
WPTI Telecom
775-236-5815
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to 1.4!
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 29 Oct 2007, Matt wrote:
This question is about 1.2.x asterisk. Please no flames, or you should
upgrade to 1.4.
Does anyone know what might be the cause for 'stuck voicemail's in
1.2.6asterisk
Has anyone come up with a way of sharing a single SIP registration with
two or more line buttons on the Cisco 79x0? This is possible on a Linksys
94x, but I haven't found the magic parameter on the Cisco (assuming there
is one).
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
Does anyone have this guide and be willing to share it with me?
Thank in advance?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
___
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asterisk-users
else having this problem, and if so, is there a fix or solution?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
___
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asterisk-users mailing list
symbolic links, you name it, but unwinding the build spaghetti is beyond
my capabilities, I'm afraid.
TIA
[If you don't have any experience with FreeBSD, please don't bother
responding!]
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
cycle to do it. We are running the latest Linksys
firwmare.
My question is this. Has anyone else experienced this problem and if so,
what have you done about it? I can't believe we're alone, as there must
be a bezillion of these phones connected to Asterisk systems.
TIA
Bruce Komito
High
Try prefixing the % with a \.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 4 Dec 2006, Garth van Sittert wrote:
Hi Jon
No luck - it works with the quotes and no % sign but as soon as I add
the % it doesn't work.
Garth
Jon Farmer wrote:
Try
and/or upgradeable.
If you are interested in working on such a project, please contact me
off-list.
Thanks
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
___
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Asterisk-Users
Try
register = 7723821447:[EMAIL PROTECTED]/7723821447
That works for me.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 20 Dec 2005, Shawn Porter wrote:
Would someone be so kind as to point out what stupid little mistake I have
made. I thought I did
Has anyone used a Cisco 7940/7960 (with or without a 7914) to display busy
extensions and if so, would you mind sharing the XML code to do it?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
___
--Bandwidth
). If you use the latest firmware, you'll be fine.
BTW, if you need some ZIP2s, we have about a dozen new units that we ended
up buying but not using because the customer upgraded to multi-line phones
for some of their users.
Bruce Komito
WPTI Telecom LLC
(775) 236-5815
On Mon, 21 Nov 2005
Yo tambien.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 1 Nov 2005, Carlos Alperin wrote:
Si se?or, I AGREE.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: Tuesday, November 01, 2005
. I've tried this on three
different IP phones (Cisco 79xx, ZIP2 and Sipura) and they all behave the
same, leading me to conclude it isn't a phone config problem. Everything
(Cisco and phones) are configured for dtmfmode=rfc2833.
Anyone got any ideas?
TIA
Bruce Komito
High Sierra Networks, Inc
that works.
Thanks in advance
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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Asterisk-Users@lists.digium.com
http://lists.digium.com
If you have successfully connected MF trunks from a telco switch, please
respond. We are looking to support E911 directly from Asterisk and our
911 trunking to the LEC will be over MF trunks to their 911 tandem.
Thanks
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
The answer is, YES. We have exactly that configuration using a 3640
running SIP to *.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 8 Aug 2005, Bojan Jeremic wrote:
Have you resolved this issue,
I have a friend who has a solution that involves using
not sure when this started happening, but I assume it was sometime
after I upgraded.
Has anyone else seen a problem like this, and if so, what's the solution?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
___
Asterisk
code that made the call, and as a result, the channel simply hangs
(i.e., nothing else happens) and astcc never returns to the dialplan.
Has anyone else experienced this or anything like it?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
The Sipura SPA2000 only supports one G729 call at a time. Same with the
Linksys PAP2.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Fri, 17 Jun 2005, David wrote:
Hi All,
I have configured Line1 (2011) and Line2 (2012) in Sipura SPA-2000 (latest
Firmware
We've tried a lot of different types of boxes, but the best I've found so
far has been from SuperMicro. Contact me off list for more specifics.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 19 May 2005, Michael B. Murdock wrote:
Is there anywhere
should only need about 32-40k over a link that
claimed to guarantee 64k, and the best we got was broken sound.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 11 May 2005, Yiannis Costopoulos wrote:
Hi All,
I am investigating the deployment of VoIP
-end, I didn't expect much, since they don't offer ANY CIR.
But when they claimed 64k, silly me, I believed it.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 11 May 2005, Chad Wicker wrote:
Well there are several problems in your description of Satellite
(.03 ms is the default), and I know * will swallow whatever the
Sipura sends it. So, I know it's possible to change this in at least one
direction if you are using a Sipura.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sun, 3 Apr 2005, Matt wrote:
IAX
If you're going to promote your product, you might consider making sure
your web site is up, before giving out the URL.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 23 Mar 2005, Chris Ford wrote:
You should try Fordvoice
http://www.fordvoice.org
If you have any experience using * (or VoIP in general) with DirecWay,
please respond privately. I am particularly interested in experiences in
Latin America.
TIA!
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
at the expense of delay, by making
better use of the limited bandwidth available. The problem is not so much
that the bandwidth is limited, but that it is intermittent and
inconsistent.
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
That's your opinion, and I'm sure you have good reason for it. However,
in order to be widely accepted, any app must support mysql, simply because
many environments run mysql as their choice of database, and are not
likely to change.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
I had the same problem, and it's a database issue, not a code problem.
Use the character ^ in front of the pattern in the routes table, and I
think you will have better luck. E.g., ^1416... will match only
numbers that start with 1416.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r
Sorry if I missed the beginning of this thread, but I've never heard of
the ** transfer key sequence, nor have I found a way to make it work.
Would you mind, please explaining this further or pointing me to somewhere
where it's documented? (I checked Wiki and Google but no joy.)
Thanks
Bruce
. There is
a setting in the general section of the config file called timezone, which
defaults to -480 (minutes off of GMT), but this setting only seems to
control the value that you are prompted with when the phone boots.
If I get a solution, I'll let you know.
Bruce Komito
High Sierra Networks, Inc
That was the hint I needed. Try adding this to your dhcp.conf:
option time-offset -480
(-480 is for PST, -420 is mountain, etc.)
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 17 Jan 2005, Ronald Hartmann wrote:
I have been reading the RFC http
I've found, when upgrading from earlier releases that do not support
realtime (e.g., 1.0.1), you must first make install from the asterisk
directory before attempting to build asterisk-addons.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sun, 9 Jan 2005, Serge
I'm sure it took several hours, but, hey, he only has to sell one to get
his money back (:
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 22 Dec 2004, Luke Catranis wrote:
How much time did you waste on that?
-Original Message-
From: [EMAIL
That is correct, and the last time I checked, they sell subscriptions for
a monthly charge (depending on frequency of updates) or a one-time charge
of $750 for a single copy.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Fri, 17 Dec 2004, Dave DeChellis wrote
If you have iax.conf on /etc/asterisk, the iax configuration will be
loaded from there and not from what is specified in the realtime config.
Remove the iax.conf file if you haven't already.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 14 Dec 2004, Jason
defined, you don't have the sql table set up
properly. There is a perl script that takes any .conf file and loads its
values into the ast_config table. If you would like me to send you that
script, let me know.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 15
WARNING[8102]: Unable to open /dev/dsp: No such device
Dec 14 15:31:01 WARNING[8102]: Requested contexts didn't get merged
Dec 14 15:31:01 NOTICE[8102]: Loading Config voicemail.conf via mysql engine
Dec 14 15:31:01 WARNING[8102]: MySQL database sock file not specified. Using
default
Bruce Komito
If you do:
cvs checkout asterisk-addons
(without the -r v1-0), you'll get everything you need...including
res_mysql.conf.sample .
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 13 Dec 2004, Bill wrote:
Same here. I've deleted and re-installed
I have the same problem, and I assumed it was because MySQL voicemail
support is now accomplished through the realtime facility. But, so far, I
haven't had a chance to research it further.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 9 Dec 2004, Bill
Greetings, I tried to build astcc, but the Makefile is looking for
Asterisk/AGI.pm. Anyone have any idea where this file is supposed to be
and where it comes from? I've dragged in everything I can think of from
cvs, and * is otherwise running fine.
TIA
Bruce Komito
High Sierra Networks, Inc
LA seems to be down. Switch to DCA or MIA and you'll probably be OK.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 1 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
Hi,
I am having problems with Broadvoice incomming calls.
Did anybody who use broadvoice
that *
was not supported. From listening to the chatter on the list, my sense is
that most Broadvoice problems are configuration-related (on the * side),
and that was also the case with me. However, once properly set up, the
problems have been few and far between.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r
glean from reading it is that there was
something in the REGISTER request that * didn't like, but I can't tell
from the message what it is or why it doesn't like it.
Does anyone know what this message means, why it appears and what I should
do to get rid of it?
TIA
Bruce Komito
High Sierra Networks
If anyone finds the generic version of this available (i.e., not locked to
Vonage), please advise the list of where.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 22 Nov 2004 [EMAIL PROTECTED] wrote:
Has anyone tried out the Linksys RT31P2 with Asterisk
messages or having a dialog with an IVR. When we changed to
inband signalling, our problems went away.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 18 Nov 2004, Joseph wrote:
On Thu, 2004-11-18 at 12:44 -0800, Jongsuk Lee wrote:
My guess for problem
I found LAX either unreachable or non-responsive for most of yesterday. I
switch to DCA and no more problems.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 16 Nov 2004, TELUX wrote:
I have been using LAX and getting a LOT of busy signals, i have taken
I bought a few of these from PC Connection but then when I tried to order
more, they claim the product has been discontinued by the
distributor...whatever that means.
Does anyone know of a source for these that is still shipping them?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r
Same here...
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sat, 13 Nov 2004, Doug Shubert wrote:
yes.. started around 12:00 noon EST
I get sip_reg_timeout: Registration for '[EMAIL PROTECTED]
Does anyone know if this is related to the channels patch?
Doug
Could you please explain how this allows one to interogate the ALERT_INFO
sent to * by another SIP device or host?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Fri, 12 Nov 2004, Brian West wrote:
You need ot set _ALERT_INFO and yes it works.
bkw
The Busy show be at priority 102 (n+101).
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 8 Nov 2004, Eric Wieling wrote:
Nicklas Bondesson wrote:
Just like this? It doesn't seem to work though.
[wx3trunk-outgoing]
include = internal-sip-callers
I'm doing this and it works. You're right, all the calls come into the
same context, but your dialplan should match based on the dialed number.
If that doesn't help you, I'll send you a config snipet.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 2 Nov
they are re-registering often. I'm sure there are other ways to
deal with this, but that is what worked for me.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 1 Nov 2004, Paul Rodan wrote:
Just moved into a new place and it'll take 2-3 weeks for my SDSL
.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Fri, 29 Oct 2004, Peder Angvall wrote:
I am trying to get a Cisco PRI gateway to send calls to * and it doesn't
appear to be working. It is a 2610 running 12.3 IP+. I've got the
config in there, I can see calls
G711ASS NO
G729ON YES
G729PACK 20
G729SS NO
AJB_MAXDELAY 100
FJB_DELAY 40
AUTO_JB_SWITCH NO
COUNTRY USA
NTPSERVERIP 192.43.244.18
TIMEZONE -420
DST YES
RINGTONE 1
LINE1NUMBER 90055522368
LINE1AUTHUSER 9005552368
LINE1AUTHPSWD pw3268
LINE1CALLERID John Public 900-555-2368
Bruce Komito
High Sierra
For what it's worth, I have the same observation. Meetme used to work
great, but sometime in the last few (3-4) months, it seems to have
developed significant latency. Our echo test is also normal (way under a
second), as are non-meetme calls.
Bruce Komito
High Sierra Networks, Inc.
www.servers
it means?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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exten = 777,1,VoicemailMain([EMAIL PROTECTED])
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sun, 26 Sep 2004, Henry Devito wrote:
I set up the pilot number to voicemail to be 777. When a user calls 777 the
voicemail answers and asks for mailbox
Try www.sipgate.de . They have DID numbers available in 14 cities in
Germany.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sat, 25 Sep 2004, Klaus-Peter Junghanns wrote:
Hi,
if i understand german telco regulations right (even for a german that's
Not true, in my experience. We have no analog lines (i.e., no FXO ports),
only PRIs, and we have consistent echo problems.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 23 Sep 2004, David Cook wrote:
I'd like a good plan for this too, however
Probably the reason you get echo on the Voicepulse calls is because the
propogation delay between the IP phone and where the call becomes analog
is much greater than over your FXO lines.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 23 Sep 2004, Shilliday
You can't run E1 on a circuit designed for T1. T1 is 24 x 64k = 1.5mb; E1 is 30 x 64k
= 2mb
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 16 Sep 2004, Andrew Thompson wrote:
Christopher Jacob wrote:
All,
This may be a stupid question, but here
, or is there a version of the Directory app that
queries the users table instead of the voicemail.conf file?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
___
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[EMAIL PROTECTED]
http
that Zultys provides don't help.
If you have an example of a tftp-loadable config file for the Zip2 that
you would be willing to share, I would sure appreciate it.
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
I could be wrong, but according to the Max documentation, drop insert
only works on a channelized T1...not a PRI.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 11 Aug 2004, Nate Carlson wrote:
On Wed, 11 Aug 2004, Martin List-Petersen wrote:
Shouldn't
Is any one else having problems with Voicepulse today? Suddenly, I can't
register and calls to my Voicepulse numbers get a fast busy.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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[EMAIL
other sources, we're about 50%. That means 50% of the
time, we get our stuff and the rest of the time the order is either lost
or significantly delayed.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Thu, 29 Jul 2004, Jean-Yves Avenard wrote:
-BEGIN PGP
I have * working with a 4x4. The only difference I can see is that you
don't have a secret configured. You might try that and see if it makes a
difference. BTW, don't even think of putting the 4x4 behind a NAT server.
It won't work.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
User
use_proxy=yes
register_w_proxy=yes
proxy_addr=1.2.3.4
proxy_port=5060
registration_expires=300
auth_password=geheim
proxy_password=geheim
call_park_extension=700
inb_im_enabled=no
session_expires=300
subscription_expires=300
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775
If you know of a good *reliable* source for Zip phones, please respond,
off-list if you prefer.
Thanks
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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Amen!
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 28 Jul 2004, AJ Grinnell wrote:
Has anyone been able to change the way that asterisk performs transfers?
Instead of using the # key, I would like to due something else, such as
flash. # is just causing
?
Thanks
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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go away.
If this isn't the problem, zttool might still give you a hint if there are
problems on the PRI itself.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 20 Jul 2004, Paul Oster wrote:
I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest
I had the same problem. Before you make install from the asterisk
directory, try removing all the files in /usr/lib/asterisk/modules . That
should resolve any potential conflicts from stuff left over from the last
build.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236
to Grandstream, is there any way to recover the phone?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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Turn off interrupt masking in your IDE driver:
/sbin/hdparm -u1 /dev/hda
That solved the problem for me.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 14 Jul 2004, Brian D'Arcy wrote:
Hello everyone,
I'm using a TE410P, no irq sharing, and all
If you have an example of a config file for a Grandstream BT101/102, I
would appreciate if you would share it with me.
Thanks
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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this happens, I check the Sipura and it is thinks it is still
registered and I check * and it shows registered. If I reboot the Sipura
or restart *, the problem clears. It also clears by itself eventually.
Has anyone seen this behaviour and/or know how to cure it?
TIA
Bruce Komito
High Sierra
either this hard-wired callerid , or the one you provide.
It sounds to me like your PRI is provisioned as the former. I would talk
to your PRI provider and see if they agree and are willing to change this.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 29 Jun
A minute is a minute, except that Vonage's plans are mostly all you can
eat (unlimited) for a fixed price.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 29 Jun 2004, Ken Wiesner wrote:
Personally I don't understand why this is a problem for them
this so that * waits for call
progress from the gateway before giving the caller the appropriate
indication, i.e., ring or busy tone? I have been told this is a result of
setting * to forced ring and this should be turned off, but of course,
on * it is probably called something else.
Thanks
Bruce
Directory only reads the number if the voicemail user has not recorded his
name. If the name has been recorded, it plays that, instead.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 21 Jun 2004, Harold Workman wrote:
Just a quick question. I setup
I'm having the same problem...nothing changed...just the CVS version.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sun, 20 Jun 2004, Aaron J. Angel wrote:
I updated from CVS yesterday and now everytime I start asterisk, I get the
following message
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