on speaker phone does not hang up?
That fixed it! THANK YOU.
-Cassius
From: Peder pe...@networkoblivion.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wed, 24 Nov 2010 07:42:52 -0600
To: 'Asterisk Users Mailing List - Non-Commercial
It is the phone itself: go to Regional tab and scroll down to Reorder Delay
and make it 255. That tells it not to play re-order tone and just hangup.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith
Sent: Wednesday,
Why are the sip latencies so high? And is it a problem? And if so, how
do I fix it?
Not a problem at all. Just a goofy Cisco thing. Polycom and Linksys and
Grandstream are all a lot lower, but Cisco has always been high. We've seen
that for 4-5 years and never had issues.
--
Use Polycom, but if you really must use cisco phones, downgrade to 7.5.
I've got a lot of 79xx phones out there and 7.5 is the last stable release
as far as I'm concerned. It just seems to work, no periodic reboots
needed,
or any other quirkiness like with the newer firmware's. The feature set
I am not aware of any way to do that. My question is if you are using
realtime, why are you doing a sip reload? If you change the settings on a
device in the realtime DB, just prune it and it will grab the new config the
next time they re-register.
-Original Message-
From:
A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone. If the reg is set for a short period, say 60
seconds, then in 60 seconds it will re-register and work fine. Yes, it is a
total pain, but this is the way it has worked since day 1 for realtime. I
But it doesnt explain why the phones that are hard coded in the sip.conf
file don't lose registration.
On a reload, it re-reads the sip.conf config file and sees the users in
there, so it doesn't flush them. It doesn't pull down the whole SIP table
on a reload, it only loads a realtime peer
qualify=2000 does not mean it sends a qualify every 2000ms, 2 seconds. It
means that the qualify timeout is 2000ms, so if it receives a response at
2600ms, it counts that phone as down. I believe the timing of qualifies is
still every 60 seconds, unless explicitly changed by the system admin:
My best advice would be don't do it, it will only cause headaches. It is
completely different than * with different terminology, design
considerations, etc.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent:
As far as Dallas, it completely depends on where you are. The only provider
that blankets an area with fiber is Verizon and that is really only 2-3
cities around Dallas and it is usually residential, not business. They
aren't in Dallas itself. Time Warner and Cogent have a lot of coverage in
would affect that.
Peder
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I am using T1's and didn't think the spill would take that long.
PRI no, EM yes.
Some PRI take that long too because the telco sends the name in a followup
message, not in the initial call setup.
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No, not until Microsoft builds a compatible soft phone. Microsoft
built software that only speaks SIP over TCP. Most SIP stacks work
over RTP.
I suspect you meant UDP, not RTP. They use TCP or UDP for SIP signaling and
RTP for the actual voice traffic.
--
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Monday, July 12, 2010 3:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk and cisco 2800
Hi Peder,
thanks for the advice
The first 6 digits of a mac address are the vendor ID and the 2nd 6 are the
unique device ID. Some vendors use more than 6 digits of device IDs, so
they have multiple vendor IDs. So 00:0E isn't Linksys, it is 00:0E:xx that
is Linksys. Some devices use CDP or LLDP to request voice vlan
: Friday, July 09, 2010 4:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk and cisco 2800
Hi Peder,
it seems to work, thank you!
Now I've got a problem with the cisco 2800 which is resetting every 5
minutes but I do not think it is related
That's not right. Should be 1245 - 4512:
http://www.voip-info.org/wiki/view/crossover+T1+cable
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Tuesday, July 06, 2010 2:35 AM
To:
days of month = daynum
| daynum'-'daynum
| *
It's either a range of days, e.g. 29-30, or * for don't care. So do 29-30.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent:
It depends on the issue. If you have a carrier that has say 5-10 different
routes, they may want to confirm that the issue occurs on the same route
every time, or see if it is hitting the same box on their end.
Theoretically they could gather all the info on their end given the
caller/called
I believe crtp is only for point to point links as it compresses the header
and there would be no way to route the packets over the Internet after being
compressed, so there is no way to do that in *.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
sip.conf and extensions.conf would be helpful as well as knowing what
version you are running. Based on what you went, I would say you have a
config error, but I can't tell where without seeing the config.
From: asterisk-users-boun...@lists.digium.com
Silly. My guess is that someone that doesn't know anything about phones
decided to install it and failed. Lots of erroneous statements:
Asterisk because it required a custom-built server - Nope. You can
pretty much use any old server or really even a desktop machine for an
install this small.
In PBX1, where are you actually dialing the phone? The first line of the
macro just says goto dialstatus with no Dial statement.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Thursday, April 29, 2010 2:03 PM
Is this an inbound call to that number? Or are you calling out from that
number? I understand the need to obfuscate the numbers, but it says Call
from '551234' to extension '551234', so are you calling yourself?
Or did you just change both numbers to the same number. Maybe just change
Like the poster below said, do a sip debug on a call and see which end sends
the bye message or ends the call and go from there. That should give you
some sort of clue as to who is having a timer issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Monday, March 15, 2010 6:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
exten=07028XX,1,Dial(SIP/${ext...@pccw-kpn)
You
exten=07028XX,1,Dial(SIP/${ext...@pccw-kpn)
You aren't sending an outbound DID with just SIP/PCCW-KPN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit Saxena
Sent: Monday, March 15, 2010 12:42 PM
To:
Since it is sporadic, my guess would be network latency / packet loss
/jitter to ITSP. You may have lots of capacity and they may claim to have
lots of capacity, but what about the links between you and them. Who knows
when/if there is loss and latency and jitter there. Setup wireshark to grab
-Commercial Discussion
Subject: Re: [asterisk-users] Robotic sound sometimes
On Fri, 12 Feb 2010, Peder wrote:
Since it is sporadic, my guess would be network latency / packet loss
/jitter to ITSP. You may have lots of capacity and they may claim to
have lots of capacity, but what about the links
Don't use Grandstream if you want quality and stability. Also check out the
Cisco SPA504G. They are the newer versions of the SPA922, support multiple
lines and are fairly cheap too.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
I had the exact same issue and it was caused by a crappy firewall at the phone
site. Once they swapped it out with a box that did NAT correctly, the issue
went away. I don't think you said if the phone site is being NAT'd or
firewalled and when you mentioned the debugs below, you said
Upgrade the phone. I ran into the same issue a year or so ago. There was
some setting that was screwed up in the config file and upgrading to the
newest version at the time fixed it. It was something like the call waiting
tone being 30 seconds of dead air.
-Original Message-
From:
Discussion'
Subject: Re: [asterisk-users] Polycom Mute Problem
By upgrade the phone I assume you mean upgrade the bios, not purchase a
newer phone?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Wednesday
there was a more generic way to do it (my dialplan
is a lot more complex than that listed above and each user has 3-4 lines
like 799-BOB, 798-BOB, 797-BOB, so a query to find all of the calls for BOB
gets ugly very quickly).
Peder
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I'm pretty sure it only pulls the background image during a reboot.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan
Thurman
Sent: Monday, November 16, 2009 9:20 AM
To: Asterisk Users Mailing List -
can't
specify what they hear (this is all assuming calls are within the same *
box). Any ideas how to set that?
Peder
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Polycom has a softphone? Is it any good? I've never seen it on their site
before.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Browning
Sent: Friday, October 23, 2009 3:26 PM
To: Asterisk Users Mailing List -
The lowest end that you can use are 2600, 2600xm, 2800 or 3600. Then like a
previous poster said, you need the DSP's and T1/E1 modules, but not all of
them support it. NM-HDV2-2T1/E1 are relatively cheap, but you need to make
sure that it actually has the t1/ei VWIC in it and it has DSP's in it
Cisco PIX and/or ASA work great. Buy them used on eBay.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Wathen
Sent: Tuesday, October 13, 2009 11:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
On 10/02/2009 08:36 AM, Martin wrote:
if a user calling you hears echo of himself then it's the fault of
your sip device/sip phone.
The manufacturer must be using a cheap or an open source echo canceller
...
try getting a different sip device made by some 'normal' company like
polycom or
I've got PBXNSIP running on a windows box and it is trying to register with
my Asterisk box. I can set up one trunk and it works fine, however if I try
to setup a second trunk from the same box, there is some sort of
authentication issue where Asterisk appears to be confusing which trunk is
Discussion
Subject: Re: [asterisk-users] 1.2 AGI Deadlock
Peder wrote:
I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an
AGI, I get the avoided deadlock message below.
On Tue, 8 Sep 2009, Alex Balashov wrote:
A deadlock? In 1.2? Really? :)
Well, that was helpful
I should be concerned about, or is it no big deal? I
am worried that if I put this into production with 200+ phones, it will
cause Asterisk to die.
Peder
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You are thinking IP (layer3), not mac address (layer2 - ethernet switching).
Bonding is general a poor choice of wording for multiple Ethernet
connections as an individual connection won't use both links. The way most
NIC's and switches do bonding is that they hash the source and destination
mac
More info is needed. Can you send relevant portions of config, version,
etc? Also, are you using Macro's? I know there was an issue with call
pickup when the calls were using macros, but I don't know when/if that was
fixed.
From: asterisk-users-boun...@lists.digium.com
Try upgrading the firmware on it. They have all sorts of goofy bugs.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese
Sent: Tuesday, June 30, 2009 4:56 PM
To: Asterisk Users Mailing List -
You have to still have all of the line2 entries in the config file and they
have to be set to .
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, June 24, 2009 4:12 PM
To:
Does anybody know of a way to tell the Polycom phones to stop trying to
download their config? We have some setup for tftp and some for ftp and if
they cannot reach the server, they just keep rebooting over and over and
over and never stop. I would think it should try once or twice and stop,
but
the syncinfo.xml file with a future time. This should tell the phone
to stop polling.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Wednesday, June 17, 2009 10:27 AM
To: 'Asterisk Users Mailing List
are more powerful, but last saved config will remain to the next
meeting with tftp. Some phones however will lost backgrounds downloaded from
the server.
Jacek
- Wiadomość oryginalna -
Od:: Peder pe...@networkoblivion.com
Data:: środa, 17 Czerwiec 2009 21:43
Temat: Re: [asterisk-users
I had the same issue with my Windows Mobile phone for a couple of years. I
finally realized that if I had the phone use IMAP instead of POP3, I could
open the attachments. No clue why as I received lots of attachments on the
phone and they always worked. It was only * attachments that didn't
Decent product, but their support and development are horrible. I showed
them that their SIP over TCP implementation was broken and their reply was
use udp
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PHP/AGI/SetVar Issue
Steve Edwards schrieb:
On Thu, 4 Jun 2009, Peder wrote:
Is there a limitation to the number of variables you can set from a PHP
agi script?
Not that I've found yet :)
One of my AGIs sets almost
: [asterisk-users] PHP/AGI/SetVar Issue
Peder schrieb:
Here is the part from the agi that sets the variables:
echo ' EXEC SetVar ISLOCALCONTEXT='.$row['context'].'';
echo ' EXEC SetVar ISLOCALDID='.$row['did'].'';
If I run it is as, ISLOCALCONTEXT gets set, but not ISLOCALDID:
-- Executing
-boun...@lists.digium.com] On Behalf Of Peder
Sent: Friday, June 05, 2009 11:15 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] PHP/AGI/SetVar Issue
Of course I just figured it out. If I send a print with \n, it works now.
Not really sure why though
Is there a limitation to the number of variables you can set from a PHP agi
script? I have a simple example and I can't get it to let me set more than
1. I am pretty sure I am just missing something, but I've searched all over
an can't find the answer. Here is the extensions.conf part:
exten =
Can you setup a second parking extension? In features.conf, it lists one,
but I don't know how you would add a second one. It seems that * just kind of
makes the parkedcalls context and there isn't a way to create another one. I
could be wrong though.
[general]
parkext = 700
Just set up 6 conference
rooms and transfer the callers to the room instead of the lot. Use hints to
monitor the available rooms with a web page or asterisk managere.
I thought about that, but is there a way to pull a user out of a conference
room? Or once you parked them, would you just have
patch/test. Keep this in mind; The parking lot is
just a timed hold with music that lets another extension pick up the
call.
Hope that helps.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Thursday, April 30, 2009 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2nd Parking Lot
Just set up 6 conference
rooms and transfer the callers to the room
with music that lets another extension pick up the
call.
Hope that helps.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Thursday, April 30, 2009 9:15 AM
To: Asterisk Users Mailing List - Non
it anywhere. Does
anybody have an idea where I might get parking.h? Or what should be in it? Or
is there a newer better version of app_valetparking.c? Thanks.
Peder
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I'd also be more sold on it if it had half the features of the GXP2000
(which is only a little over half the price).
Sure, but if only half of the features in the GXP2000 actually work,
what is the point of them? I'd take a stable phone with less features
over one that has lots of features
: No
Auto Clear: 120
Again, if I do a sip show peer after pruning, I see the new values,
but it appears that * is still holding it somewhere that isn't updating.
Marc Smith wrote:
On Tue, Jul 15, 2008 at 12:05 PM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
I am using realtime
to figure out what the bug is. I
did some research, but couldn't find it.
Peder
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asterisk
Does anybody have the settings that you use on a Cisco 7970/79x1 to get
presence? I see the * side settings, but I can't find the Cisco side
settings anywhere.
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SIP.
Michiel van Baak wrote:
On 14:59, Wed 25 Jun 08, Peder @ NetworkOblivion wrote:
Does anybody have the settings that you use on a Cisco 7970/79x1 to get
presence? I see the * side settings, but I can't find the Cisco side
settings anywhere.
Sip or Skinny
They still have issues. If you use TCP and reboot the server, the phone
will never reconnect as it tries to use a closed TCP session. I opened
a ticket with them and after a week their answer is . use udp.
Rob Hillis wrote:
Doug wrote:
There is a bug in these units that won't let
you
They still make them. We use the CS70N with HL10 (headset lifter). They
are around $300 with the lifter, so they aren't cheap, but they work
well. The lifter fits on a Cisco 79xx phone pretty easily, but anything
else requires a little extra tape and some experimentation.
Peder
Steve
FYI, I have probably 10 Fortinet units with multiple SIP phones behind
each and all of the phones work flawlessly. As long as the Fortinet is
ver 3.0 or newer, it does NAT so that you don't need to have nat=yes on
*. No pinholes or static nat or anything, it just works.
As a side note, I
How does the g729 encoder/decoder count in regards to the total number
of licenses and how does it count an encoder/decoder? I looked on the
wiki and don't really see anything that explains it. In other words,
how do the calls below count (assume no reinvite)?
g729 phone calls into voicemail
the g729 side (
no license for g711 side of call ) . In short anytime u need to
convert g729 into some other codec ( transcoding ) you need 1 license
.
On Wed, Apr 2, 2008 at 1:59 AM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
How does the g729 encoder/decoder count in regards
to stop updating correctly.
Peder
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Enable NAT on the phone itself and leave it enabled in *.
Jerry Geis wrote:
I have a cisco 7960 phone. Worked fine in the office.
I took it home. At home I have a linksys router that the phone is
plugged into.
The linksys router has DHCP enabled. I am getting the following error on
the
Do you mean Call Manager? Unity is just their voicemail system. Yes,
you can use SIP to talk between * and CM. You can also use h.323, but
it is a big hassle.
Tony Mountifield wrote:
Has anyone here any experience in getting an Asterisk box to talk to
a Cisco Unity system? I have a
autoload=yes says to load everything, so you either need to change it
to no and then add load statements for every module you need, or leave
it as yes and then add noload for everything you don't need.
Vincent wrote:
On Wed, 20 Feb 2008 21:44:30 -0500, C F [EMAIL PROTECTED] wrote:
vi
What happens when you try it? And what do you do on the phone? We have
lots of GXP-2000 and 2020 and transfer is one feature that does work.
Gustavo Gonzalez wrote:
Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with
asterisk?. Attended and blind transfer does not
We use PRI, not BRI, with Cisco gateways and it works great. Rock solid.
Razza wrote:
Is anyone using a cisco router as an ISDN gateway with Asterisk?
As you might have seen from a couple of my threads, I have been looking
at Fritz! and Cologne cards, both of which require development
-15 minutes. If I put it at 10
minutes, it loses connectivity once or twice a day. I tried Grandstream
support and their answer was completely useless. Has anybody seen this?
Or does anybody have any ideas? Again, no NAT involved, so don't say
STUN or NAT issue.
Peder
appears to kill
registration until the Grandstream is rebooted. Has anybody else seen
this? Or maybe know how to get around it?
Peder @ NetworkOblivion wrote:
I did post most of that. Point to point T1, no firewalls and no nat,
cisco routers, bandwidth is monitored at 30 second intervals
]: Registration from
'sip:[EMAIL PROTECTED]' failed for '192.168.2.165'
That message is from a phone that is set to register every 5 minutes.
It's been 50 minutes and it still hasn't re-registered. If I reboot the
phone, it will register right away...
Any ideas?
Peder
Andrew Joakimsen wrote:
Yes
I've got a setup where we have 100 DID's. Our default dialplan has one
line that calls a macro:
exten = _22XX,1,Macro(STDEXT,${EXTEN})
The macro is pretty basic:
[macro-STDEXT]
exten = s,1,NoOp
exten = s,2,Dial(SIP/${ARG1},15,Tt)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten =
What about the situation where there is no voicemail box for an
extension. Is there a way to tell the difference between the phone
isn't registered and there is no phone at that extension?
Doug Lytle wrote:
exten = s-CHANUNAVAIL,1,Voicemail(${ARG1}|b)
exten = s-CHANUNAVAIL,n,Hangup
Or you can prune the specific user entry and it will look it up again.
Anthony Francis wrote:
Adam Moffett wrote:
I asked this question last week and never got an answer. I also
didn't find the answer in the wiki.
I think it would be nice if asterisk would check the database again if
.
Peder
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the calling user hangs up the phone. Any idea how to make
it do this? It doesn't do it by default and I don't see any settings
that might help with that. If I plug an analog phone in, it works just
fine.
Peder
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on the original called number (x456). Any ideas? When I do a
test, it appears that the called number is 'a' and the calling number is
123. I need to be able to tell that it was a call to x456. Thanks.
Peder
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What is the issue with the Grandstream? We are getting tired of Cisco
issues, so we have started looking at Grandstream and they seem to be
pretty good. The Polycom work well, but they seem to die after about a
year or so. We bought 20 of them about 2 years ago and 7 of them have
died or
to send it. If I test it and hit * from my voicemail, I get 'a'
as the EXTEN, which doesn't help me. I need 'a' to be able to see the
called number so that I can do a db lookup and send the call to the
appropriate extension.
Peder
James FitzGibbon wrote:
On 10/26/07, *Peder @ NetworkOblivion
user as there are about 40 people that want this. They won't
all go to the same number. Thanks.
Peder
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and
then forward issue that I am having, I would like to hear it.
Peder
Jason Lixfeld wrote:
I guess what I'm asking is if there is a recipe anyone has used to
allow a voicemail message (once recorded by asterisk) to playback on
iPhone when said recorded voicemail is received as an email
Yes, you need to buy a license if you use it with ANY pbx, whether it is
Callmangler or Asterisk or whatever. If you buy one used, then you need
to pay to re-license it as well.
The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you
will need a switch that provides Cisco PoE for
:
Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can
handle the 7940G ?
The 7941G does conform to the standard but it only support SCCP (shame
on cisco).
On 9/27/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Yes, you need to buy a license if you use it with ANY pbx
Could be a mysql permission issue. Try this from the local box:
mysql -u root -p
enter asterisk as the password
use asterisk;
select * from sip_buddies;
select * from iax_buddies;
If you get that far and can see the entries in iax_buddies and
sip_buddies, you know it isn't a permissions issue.
There has to be some reasonable priced sip provider that would perform
just as well as ATT. Does it exist?
The problem is that there is no QoS control between you and the
provider, so a lot of the quality issues you have are probably not
related to the specific provider, but just the
Is there a way to decrease the volume on the native files version of MOH
in 1.4? I've had several people complain that it is too loud.
Peder
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calling Steve 1.
Peder
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You can buy a used Cisco 2600 with dual-port PRI/T1 card for VoIP for
~$1500. No worries about echo-cancelation, or IRQ issues or anything
like that. It just works. And the config for inbound/outbound calls is
maybe 20 lines total.
Alex Balashov wrote:
For a price tag that does not scale
The question I always have when someone mentions distributing the load
across multiple machines is how do you handle contexts for phones on
different machines? I want all of my phones to dial into
[companyA-phones]. I have to define it in two different places (or more
depending on the number
DB's within the one
mysql box for each * box. Thanks.
Peder
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