Tomislav Parčina ha scritto:
>> For what we do with Asterisk(lots of meetme and Zap -> IAX2) It
>> does spread the load across both cores. In our initial
>> comparisons for equal call traffic, the P4-D had half or the
>> average loadavg for a 6 hour time period of the P4 of the same
>> speed.
>> MA
> For what we do with Asterisk(lots of meetme and Zap -> IAX2) It does
> spread the load across both cores. In our initial comparisons for
> equal call traffic, the P4-D had half or the average loadavg for a 6
> hour time period of the P4 of the same speed.
>
> MATT---
Hi Matt!
Thank you fo
We offer a management GUI for both options - multi-tenant (multiple
companies within the same instance of Asterisk) or multi-instance (multiple
instances of Asterisk).
Best regards,
Alex
Alex Epshteyn
Third Lane Technologies, LLC
http://www.thirdlane.com
-Original Message-
From: [EMAIL
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> Well, it would seem to me that with a little attention to processor
> affinity, you could run your Asterisk and DBMS code on one processor,
> and let the other one handle the device interrupts; ie: that sounds to
> me like a feature, rathe
Thirdlane PBX Manager multi-instance can be used to manage/configure
multiple instances of Asterisk. If you have any questions please contact me
at [EMAIL PROTECTED]
Best regards,
Alex
Alex Epshteyn
Third Lane Technologies, LLC
http://www.thirdlane.com
-Original Message-
From: [EMAIL PRO
Upon investigating a call quality complaint with a conference room, I
discovered this error repeated several times in the log. Looking at the
source, frametype 5 is "An empty, useless frame". Does this indicate an
actual problem?
app_meetme.c: Got unrecognized frame on channel
Local/[EMAIL
Has anyone a workin setup between asterisk and an ericsson md110 pbx?i need asterisk to do the billing and voicemail work, so i think it should be connected directly to the pstn and pass the calls to the md110 and viceversa.any recomendations?thxPatricio Bruna V.Red Hat
I have used both Sangoma and Digium cards. As far as TDM2400p with echo canceller is concerned, we had issues with static and choppy voice on the SIP side. However PSTN receivers had no issues. We called digium and they logged in remotely and found no issues in configuration. We are forced to move
On Mon, Sep 25, 2006 at 09:27:12PM -0400, Dave Fullerton wrote:
> I never really considered one. I've never used one for that matter. This
> system is really only a testbed. If it works out, then in 6 months to a
> year we'll put asterisk in main site and all the phones in this location
> will s
I am running today's SVN of the 1.4 branch, on Ubuntu dapper.
I compiled a custom kernel (2.6.15.7). Created modules of the rct and
the rtc modlue loads fine.
As soon as I load ztdummy the syslog fills up with:
rtc: lost some interrupts at 1024 Hz.
Any ideas what may be causing this?
thanks,
Try vm-intro.
On 9/25/06, unplug <[EMAIL PROTECTED]> wrote:
In the function of voicemail, the default greeting is:Please leave your message after the tone. When done, hang up, or press
the pound key.It is what I want to replace.On 9/26/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:> Wyatt:
Dave Fullerton wrote:
Anthony Cennami wrote:
What's wrong with a channel bank? Can be a much cleaner solution with
greater room for expansion in the future.
Not to mention cheaper.
I never really considered one. I've never used one for that matter. This
system is really only a testbed. If
In the function of voicemail, the default greeting is:
Please leave your message after the tone. When done, hang up, or press
the pound key.
It is what I want to replace.
On 9/26/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:
Wyatt: Try dialing *97
unplug: Are you referring to the gree
Thank you for your response!That was exactly the problem - the 841s use a deprecated identifier for G.726. Behavior and solution is described here:http://www.voip-info.org/wiki/view/Sipura+SPA-10011) To use G726 with asterisk 1.2.6 or later you must edit the rtp.c and either dfine USE_DEPRECATED_G7
Wyatt: Try dialing *97
unplug: Are you referring to the greeting that says welcome to asterisk mail or something similar to that? If so, that file is called vm-login and is in your sounds directory.
On 9/25/06, unplug <[EMAIL PROTECTED]> wrote:
Thanks.Are you talking about the customization o
Thanks.
Are you talking about the customization of voicemail greeting? In the
function of voicemail, there is a default greeting. I want to replace
it by another prompt so that everybody can get the greeting instead of
the default one.
On 9/25/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
unplug
Anthony Cennami wrote:
What's wrong with a channel bank? Can be a much cleaner solution with
greater room for expansion in the future.
Not to mention cheaper.
I never really considered one. I've never used one for that matter. This
system is really only a testbed. If it works out, then in 6
Wherever you have your exten => s,1,Answer statement, replace with:
exten => s,1,Wait(30) ; or however long you want to wait to give someone else the chance to answer
exten => s,n,Answer
then continue on.
Asterisk will then wait 30 seconds before it answers the phone. You would probably want
Be sure that it is looking in the right place. If it is running as non root, then the ctl file would be in a different directory.
It looks as though Trixbox does run as non-root. The ctl is actually /var/run/asterisk/asterisk.ctl.
Did you install from scratch, or was a previous version of Ast
Ahh, I assume your talking about mohinterpret and mohsuggest I assume.
Thanks.
-Original Message-
From: Jason Parker [mailto:[EMAIL PROTECTED]
Sent: Mon 9/25/2006 6:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Su
Most people here don't give a damn about anything but their own personalproblems. Sounds like you are probably one of them,
Thanks for noticing!
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- Douglas Garstang <[EMAIL PROTECTED]> wrote:
> Oops I got that a bit mixed up. I meant to say that when party A put
> party B on hold, the moh class used should be party A's moh.
> Essentially the moh class used should be what's defined for the person
> putting the OTHER party on hold.
>
> Do
Lacy Moore - Aspendora wrote:
FYI, this is an asterisk mailing list, NOT trixbox. Most people on
here don't care when trixbox is going to do something. Try their list.
Most people here don't give a damn about anything but their own personal
problems. Sounds like you are probably one of them
Sorry if I'm stating the obvious, but I'm not sure if Trixbox runs
asterisk as root or not. I have to "sudo asterisk -r" on mine, but I'm
not running Trixbox, I'm running Asterisk 1.2.
Moj
Ken D'Ambrosio wrote:
I've set up a bunch of plain-jane Asterisk systems, but had heard good
things abo
Douglas Garstang wrote:
-Original Message-
From: Eric "ManxPower" Wieling [mailto:[EMAIL PROTECTED]
Sent: Monday, September 25, 2006 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk
Asterisk does no
I've set up a bunch of plain-jane Asterisk systems, but had heard good
things about the more recent incarnations of [EMAIL PROTECTED] errr, Trixbox.
So I installed it, and fired it up, and it works fine.
Until I try to do an "asterisk -r". I get the "does /var/run/asterisk.ctl
exist?" question,
Hi guys!
Can someone give advice on nice H323 IP phones brands?
??
I'm looking for some H323 IP phones for a customer. Diving in the
internet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard
about them. Can someone give feedback experince about it??, config
ease, sound qua
Oops I got that a bit mixed up. I meant to say that when party A put party B on
hold, the moh class used should be party A's moh. Essentially the moh class
used should be what's defined for the person putting the OTHER party on hold.
Doug.
> -Original Message-
> From: Douglas Garstang
That error message is almost always because the two sides cannot agree
on a codec. HOWEVER, if you are using SIPura and G726, there is a
Makefile option for Asterisk to make it work.
Guy M Guyadeen wrote:
Our pbx is a Fedora Core 4 box running Asterisk 1.2.6. It has a public
IP and is publicl
I made a post to the list on 8/7 and said that music on hold in Asterisk 1.2
seemed broken. It seemed like the moh class that should be used when party A
puts party B on hold, should be the class defined for party A. Asterisk 1.2
does it the other way around and uses party B's moh class.
Olle J
On Mon, Sep 25, 2006 at 06:09:02PM -0400, Alex Robar wrote:
>This is a non-commercial discussion list, hence the name "Asterisk
>Users Mailing List - Non-Commercial Discussion". Post this to the -biz
>group.
He does this every month or so
Steve
--
NetTek Ltd UK mob +
On 9/25/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:
FYI, this is an asterisk mailing list, NOT trixbox. Most people on here don't care when trixbox is going to do something. Try their list.
On 9/25/06, Christopher Corn <
[EMAIL PROTECTED]> wrote:
I know asterisk 1.4 has t.38 pass throug
This is a non-commercial discussion list, hence the name "Asterisk Users Mailing List - Non-Commercial Discussion". Post this to the -biz group.On 9/25/06,
Sam Tam <[EMAIL PROTECTED]> wrote:
Hello AllThis month we would like to offer our GSM Gateway range for less to clear upsome spaces.CT-GSM-100
I've noticed sla.conf in Asterisk 1.4. I'd love to test it, but how does it
work? There's bupkiss docs, and until I have a clue how to use it, I can't test
it.
Doug.
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asterisk-users mailing lis
Our pbx is a Fedora Core 4 box running Asterisk 1.2.6. It has a
public IP and is publicly accessible (no NAT, firewall, etc.).
Our Sipura 841 phones are in several locations, all NAT'ed behind PIX
firewalls.
The system used to work flawlessly, but now any extension we dial
comes back as "busy
Coulda stated that a little nicer...
Christopher,
Go to the trixbox site...or in the forums...should show
you how to join the mailing list for [EMAIL PROTECTED] or Trixbox...not sure what they call
their list now.
I don't use trixbox so I can't tell you...heck I won't
even be using asterisk
Hello All
This month we would like to offer our GSM Gateway range for less to clear up
some spaces.
CT-GSM-1000 Basic GSM Gateway (RJ11) Single Sim
£69
CT-G01 GSM Gateway with SMS Feature (RJ11) Single Sim £99
CT-G04 GSM Gateway (RJ11) Quad Sims
I'm putting together a plan for a new Asterisk system and I'm trying to
decided on an interface card to use. I was originally planning on using
a Sangoma A200 but now I'm considering a Digium TDM2400P. The server is
large enough to accommodate the full sized TDM and I'll be using 8 FXO
channe
show dialplan in the CLI should give you some clues.
On 9/25/06, Eric Rousse <[EMAIL PROTECTED]> wrote:
Hello,
Just wondering if there's a simple way to display the hierarchy of the
includes within the extensions file ? Currently I have the sample file
extension.conf in my Asterisk machine.
Bu
I don't see a problem here. Using includes you dedicate every company
their own directory of configs. Macros are eithere system wide, or
each comapny can create their own. I don't see why this is any harder
than mutilple instances of asterisk.
On 9/25/06, Douglas Garstang <[EMAIL PROTECTED]> wrot
- Netem (http://linux-net.osdl.org/index.php/Netem)
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FYI, this is an asterisk mailing list, NOT trixbox. Most people on here don't care when trixbox is going to do something. Try their list.
On 9/25/06, Christopher Corn <[EMAIL PROTECTED]> wrote:
I know asterisk 1.4 has t.38 pass through, but I don't think trixbox does. i run trixbox. looks like fo
Check the archives. This has been covered multiple times.
Rick Smith wrote:
you didn't listen. SIP only. Anyone can understand that multiple
instances on the same machine can't touch the same hardware.
I can see how this would be very easy - dedicate an IP to an instance,
and it'll play nic
What's wrong with a channel bank? Can be a much cleaner solution with greater room for expansion in the future.Not to mention cheaper.On 9/25/06,
Dave Fullerton <[EMAIL PROTECTED]> wrote:
Greetings List,I'm putting together a plan for a new Asterisk system and I'm trying todecided on an interface
Could the problem is this: "Content-Type: unknown"?
Reliably Transmitting (NAT) to 192.168.1.228:5060:
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK4bf724c9;rport
From: ;tag=as744e33c0
To: "test guy" ;tag=6583e0d3a15652bd
Contact:
Call-ID: [EMAIL PR
If you download the manual you will see that the Aastra 480i CT as well as
the 480i and the 9133i (not sure about the 9112i) have dedicated HOLD,
REDIAL, XFER, ICOM, and CONF hard keys. A soft key can easily be programmed
for Voicemail.
http://www.aastratelecom.com/downloads/41-000124-00-08%20480i
Hi,Which network impairment tool would you use to teach QoS and VoIP in a lab, with limited budget ?Ideally, I would set different network conditions with it (jitter, packet loss).I've seen those tools :
- NIST Net (http://www-x.antd.nist.gov/nistnet/)- UDP Packet Reflector/Forwarder (
http://www.
Hello,
Just wondering if there's a simple way to display the hierarchy of the
includes within the extensions file ? Currently I have the sample file
extension.conf in my Asterisk machine.
But its kinda hard to search through the file to get the idea of the
context hierarchy. Like which conte
Jay R. Ashworth wrote:
On Mon, Sep 25, 2006 at 12:28:30PM -0600, Douglas Garstang wrote:
How easy do you think the management of the configuration files is
going to be if your trying to host several dozen companies on the one
Asterisk instance? Sure, you can split things into contexts, but just
Hi,
we're writting interface module for our speech recognition system. We would
like to export stream of audio samples to external app, but to preserve dtmf
recognition and dialplan progress.
I wonder if recording application would be a good start for that (recording
application obviously st
But if I segment my zap channels, that shouldn't be an issue, correct? I.e.
Instance 1 => Port 1, Instance 2 => Port 2, etc. Of course, you are also
assuming there is Zap channels, as I believe he is using a gateway, which
takes that out of the equation.
On 9/25/06 2:23 PM, "Eric "ManxPower" Wi
We aren't accessing ZAP channels. No Digium hardware is installed!
> -Original Message-
> From: Eric "ManxPower" Wieling [mailto:[EMAIL PROTECTED]
> Sent: Monday, September 25, 2006 1:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Runnin
Greetings List,
I'm putting together a plan for a new Asterisk system and I'm trying to
decided on an interface card to use. I was originally planning on using
a Sangoma A200 but now I'm considering a Digium TDM2400P. The server is
large enough to accommodate the full sized TDM and I'll be us
you didn't listen. SIP only. Anyone can understand that multiple
instances on the same machine can't touch the same hardware.
I can see how this would be very easy - dedicate an IP to an instance,
and it'll play nice.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On 14:31, Mon 25 Sep 06, Mailing List wrote:
>
> >My asterisk server has 2 NICs . One with a public IP and one with an
> >internal LAN IP. All the phones configure to the LAN IP so there's
> >basically nothing between them. A 3com switch and that's it.
>
> "basically nothing" is wrong. I have
On 9/25/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> [EMAIL PROTECTED] zaptel-1.4.0-beta1]# ztcfg -vvv> Notice: Configuration file is /etc/zaptel.conf
> line 235: Unable to read Zaptel version information.>> Zaptel Version: $êþP¦0> Echo Canceller:> Configuration> ==>>> Channe
Best of luck getting multiple instances of Asterisk to play nice when
accessing Zap channels.
James Texter wrote:
Doug,
I actually see this as a pretty logical way to solve the problem.
Please keep us posted if you have any luck sorting out running multiple
instances, or mail me off-list i
Mailing List wrote:
My asterisk server has 2 NICs . One with a public IP and one with an
internal LAN IP. All the phones configure to the LAN IP so there's
basically nothing between them. A 3com switch and that's it.
"basically nothing" is wrong. I have a 3com switch in front of the one
p
Doug,
I actually see this as a pretty logical way to solve the problem.
Please keep us posted if you have any luck sorting out running multiple
instances, or mail me off-list if no one else is interested.
Thanks,
On 9/25/06 1:52 PM, "Douglas Garstang" <[EMAIL PROTECTED]> wrote:
>> -Orig
I know asterisk 1.4 has t.38 pass through, but I don't think trixbox does. i run trixbox. looks like for now i will have to setup my fax machine to connect directly to my t38 provider. anyone know when trixbox may have this update?___
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On Mon, Sep 25, 2006 at 12:52:43PM -0600, Douglas Garstang wrote:
> > -Original Message-
> > From: Brian Rogan [mailto:[EMAIL PROTECTED]
> > Sent: Monday, September 25, 2006 12:40 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Running Mu
On Mon, Sep 25, 2006 at 12:28:30PM -0600, Douglas Garstang wrote:
> How easy do you think the management of the configuration files is
> going to be if your trying to host several dozen companies on the one
> Asterisk instance? Sure, you can split things into contexts, but just
> try and imagine ho
> -Original Message-
> From: Brian Rogan [mailto:[EMAIL PROTECTED]
> Sent: Monday, September 25, 2006 12:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk
>
>
> Doug,
>
> Why do you want to do this t
I'm seeing "channel.c: Nobody there, continuing..." in the asterisk
full.log. This error is repeated 20+ times per second when it occurs. I
thought this problem was specific to one PBX that performs call
recording on all the call queues, but after disabling all call
recording, the error persi
Doug,
Why do you want to do this to begin with? I think the best solution is
to use the realtime stuff, and build your own management tools, which
would allow you to do this (you could drastically cut the complexity
with the right tools). Even if you could run them together, how
would you put ev
My asterisk server has 2 NICs . One with a public IP and one with an
internal LAN IP. All the phones configure to the LAN IP so there's
basically nothing between them. A 3com switch and that's it.
"basically nothing" is wrong. I have a 3com switch in front of the one phone
that reports the
> -Original Message-
> From: Eric "ManxPower" Wieling [mailto:[EMAIL PROTECTED]
> Sent: Monday, September 25, 2006 11:24 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk
>
>
> Asterisk does not support
All my Cisco phones show less than 75ms except for one (mine of
course). I do have a switch in my cube that I use for extra ports and
that's the only real difference.
Do you have anything plugged into the extra network port on the phone?
Yes, I have workstations plugged into the extra ports o
At 09:23 AM 9/25/2006, you wrote:
2. Is there a soft key that can be programmed on the wireless handset?
Not really, there's a function key menu and you can set that up any
way you want, but what you can assign to the functions is very
limited. The cordless is very handy, but the functionalit
Thanks for the input
Yes I have nat=yes and qualify=yes I know in the SIP.cnf file
I have
# NAT/Firewall Traversal
nat_enable: 1
nat_received_processing: 1
nat_address: phone's public IP Address
Do I still need to set it again in SIP Configuration ?
Thanks all
Barry
Hughes, Sam wrote:
On
Bidirectional SIP trace usually helps in these situations.On 9/25/06, Mr. Jones <[EMAIL PROTECTED]> wrote:
Hi Folks,Has anyone seen these errors repeatedly in the CLI?Incoming call: Got SIP response 415 "Unacceptable Content-Type" back
from 192.168.1.209We're using GXP-2000s.TIA,Brian__
Asterisk does not support this, as it already has features for
multi-client configuration within a single Asterisk installation/process.
Douglas Garstang wrote:
I'd like to know if anyone has sucessfully managed to run multiple instances of
Asterisk on the same system.
- Did you run each inst
At 08:31 AM 9/25/2006, you wrote:
aw crap, that's a biggie but I think I can work around it, teach the user to
dial *98 for voicemail, *700 for park and hash to transfer, currently the
It has a dedicated hold button and you can easily program dedicated
Park and voice mail buttons. I've not don
At 07:48 AM 9/25/2006, you wrote:
It's excellent home phone. I wouldn't use it in a business environment. No
hold, no one-touch voicemail. However, it works great!
No Hold? Mine has a hold button and programming one touch voice mail
would be no problem at all.
Ira
one that also offers support for it. thanks.___
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Hi Doug,On 9/25/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local lookup to see if a number is available locally, in order to find out if the number is available on one of the other 9 servers, this peer has to query all 9 r
On 09:35, Mon 25 Sep 06, yrving rivas wrote:
> I would like to know if any of you have a cell phone like a pci card to
> install in one slot to my asterisk server?, I want to make a connection from
> my asterisk to the cellular network.
> Does anybody has a solution like this?
1/2/4 simslot pci
I just downloaded asterisk 1.4beta2, and did a:
./configure --prefix=/home/pbx/1.4
[11:[EMAIL PROTECTED](pbx1):asterisk-1.4.0-beta2]# ls /home/pbx/1.4
bin include lib sbin share
What happened to etc? If I do a 'make samples', the default conf files get
thrown in /etc/asterisk.
Doug.
At 07:25 AM 9/25/2006, you wrote:
Anybody using these? How's the cordless? Does it play nice with * ?
I have 3 of them here, we're very happy with them. The cordless is
fine, about the range of my old Panasonic cordless. Sound quality is
good and the speaker phone seems good. Plays fine with
Hi Folks,
Has anyone seen these errors repeatedly in the CLI?
Incoming call: Got SIP response 415 "Unacceptable Content-Type" back
from 192.168.1.209
We're using GXP-2000s.
TIA,
Brian
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aster
Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local
lookup to see if a number is available locally, in order to find out if the
number is available on one of the other 9 servers, this peer has to query all 9
remaining peers.
Is that true?
Is there a way to have 'regi
SEÑORES DIGIUMLA PRESENTE ES PARA CONFIRMAR QUE
REQUERIMIENTOS DE HARDWARE Y/O SOFTWARE SON NECESARIOS PARA CONECTAR
UNA CENTRAL TELEFONICA PANASONIC TDA620 QUE TIENE INSTALADA UNA TARJETA
(E1) SE5E18, NO SE SI SEA ESA LA REFERENCIA, CON UNA TARJETA DIGUIM TE110P.
VI EN ALGUNOS FOROS QUE TIENE QUE
On Mon, 25 Sep 2006 08:25:10 -0600, Colin Anderson wrote
> Looks good, great price:
>
> http://www.aastratelecom.com/ipphones/pro_243.asp
>
> Anybody using these? How's the cordless? Does it play nice with * ?
Very good phone. The range of the cordless unit is not the greatest but
enough t
I'd like to know if anyone has sucessfully managed to run multiple instances of
Asterisk on the same system.
- Did you run each instance as a separate user?
- Did you have any install or config problems?
- It looks like the G729 codec registration utility doesn't work when files
aren't installed
On the 7960 with a SIP image, Press the button and go to
option 4 "SIP Configuration". Scroll down to line "24 NAT Enabled" and
set it to yes. Then set "25 NAT Address" to the external IP address.
This will need to be manually changed every time the phone's router
pulls a new DHCP lease. In you
yrving rivas wrote:
I would like to know if any of you have a cell phone like a pci card
to install in one slot to my asterisk server?, I want to make a
connection from my asterisk to the cellular network.
Does anybody has a solution like this?
Regards,
Yrving
---
Thanks for the feedback. More questions:
1. How's the range on the wireless?
2. Is there a soft key that can be programmed on the wireless handset?
3. Can I make a soft key basically do anything, any keystroke?
4. How's the call log detail?
-Original Message-
From: Mike Clark [mailto:[EMA
On 9/25/06, Colin Anderson <[EMAIL PROTECTED]> wrote:
>It's excellent home phone. I wouldn't use it in a business environment.
No
>hold, no one-touch voicemail. However, it works great!
aw crap, that's a biggie but I think I can work around it, teach the user to
dial *98 for voicemail, *700 fo
Colin Anderson wrote:
> Looks good, great price:
>
> http://www.aastratelecom.com/ipphones/pro_243.asp
>
> Anybody using these? How's the cordless? Does it play nice with * ?
> ___
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>
> asterisk-u
>It's excellent home phone. I wouldn't use it in a business environment.
No
>hold, no one-touch voicemail. However, it works great!
aw crap, that's a biggie but I think I can work around it, teach the user to
dial *98 for voicemail, *700 for park and hash to transfer, currently the
users dial fe
On 9/25/06, yrving rivas <[EMAIL PROTECTED]> wrote:
I would like to know if any of you have a cell phone like a pci card to
install in one slot to my asterisk server?, I want to make a connection from
my asterisk to the cellular network.
Does anybody has a solution like this?
Regards,
Yrving
I don't have experience using the 480i CT, only using the 9112i, so you
should take what I say with a grain of salt.
I have been nothing but impressed with this phone. In terms of being
friendly with *, they dedicate a section of their manual to asterisk
configuration, which makes things go quite
On Mon, Sep 25, 2006 at 08:25:10AM -0600, Colin Anderson wrote:
> Looks good, great price:
>
> http://www.aastratelecom.com/ipphones/pro_243.asp
>
> Anybody using these? How's the cordless? Does it play nice with * ?
Well, anyone who thinks that a 4-p,4-c modular jack *has* an RJ
designation mak
- Original Message -
From: "Steve Glaus" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, September 22, 2006 4:25 PM
Subject: [asterisk-users] Very high ping times from 7960 phones
I've asked this here before and never really got a resp
On 9/25/06, Michelle Dupuis <[EMAIL PROTECTED]> wrote:
I have a asterisk box with some queues for a call center and need help on
two points:
1. I have a scenario where if a queue has no agents logged in, an inbound
call should immediately failover to the failover destination for that queue.
Howe
Hi Steve!
The problem is following
PSTN PSTN
| |
| |
E1 E1
| |
PBX1<--E1-->Asterisk1<---SIP--->Asterisk2<--E1-->PBX2
2 office
i have all files in the same directory: c:\agi
(asterisk-java-0.2.jar, fastagi-mapping.properties, HelloAgiScript.class and
HelloAgiScript.java). My slasspath is also c:\agi
Did you mean this?
But i get still the following errors:
if i start it with eclipse:
...
INFO: Received connection.
25.09.20
I have this phone on my desk. It works very very well!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Monday, September 25, 2006 10:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Spam? [asterisk-users] OT
It's excellent home phone. I wouldn't use it in a business environment. No
hold, no one-touch voicemail. However, it works great!
/R
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Monday, September 25, 2006 10:25 AM
To: 'Aster
SEÑORES DIGIUMLA PRESENTE ES PARA CONFIRMAR QUE REQUERIMIENTOS DE HARDWARE Y/O SOFTWARE SON NECESARIOS PARA CONECTAR UNA CENTRAL TELEFONICA PANASONIC TDA620 QUE TIENE INSTALADA UNA TARJETA SE5E18 (E1) CON UNA TARJETA TE110P.
ATENTAMENTEDIEGO FERNANDO GÜIZA ARCE
_
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