Hi Jonathan,
can you be a little more clear ? What is your test configuration? How
do you expect to have voice if you use only one FXO of the card (maybe
you use regular phones too ..)
Regards,
Ciprian
On Wed, 01 Dec 2004 15:51:24 -0500, Jonathan Bartlett
[EMAIL PROTECTED] wrote:
I'm setting
On Mon, 6 Dec 2004 22:45:16 -0800, Randy MacKay
[EMAIL PROTECTED] wrote:
I have a few Cisco 7905G phones and I having a little trouble configuring
them. They are working with Asterisk. I'm able to get the sip image
loaded, but I can't get the phones to blind transfer.
Does the Cisco 7905G
I want to use PBXware but I've found that the version we need is around
$1,000. I found quite a few other solutions at
http://www.voip-info.org/tiki-print.php?page=Asterisk+GUI . Does anybody
have any specific suggestions ? I need a product that's similar or better
than PBXware.
Best Regards,
Good day all
We got the cvs yesterday,and it seems as if transfer does not work.We
are using mitel 52205055 and the Grandstream bt-100,using the transfer
buttons.
If you transfer it just goes to the next step?
please Help
Thanks
Altus
___
Yes I'm working with 7905G phones. There's no problem in transfer calls.
Here's the regarding entry in my sip.conf:
[garage]
type=friend
username=7905g_1
secret=**
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
defaultip=1992.168.1.7
callgroup=1
Hi all of you.
I am trying to configure voice mail in asterisk and i am facing problems.
I have found following warning message in /var/log/asterisk/messages
--
No application 'Voicemail' for extension (macro-mainmenu, s, 5)
I have configured voice mail accordingly
in extention.conf
No application 'Voicemail' for extension (macro-mainmenu, s, 5)
Did you load = app_voicemail.so in your modules.conf? Our simply set
autoload=yes?
Jens
--
Jens Lentföhr
http://www.jens-it.de
___
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[EMAIL PROTECTED]
I have a potential customer who has an existing PBX with analogue FXS
ports connected to phones. He wants to allow a single remote worker
to be connected to one of the analogue extension ports using VOIP.
I know I could do it using Asterisk with an X100P card, but that seems
a bit overkill. Does
Now here is strange problem i experience. Setup is easy, IAX line out
with SIP softphone registered to Asterisk. All work fine except for one
client. When using Sjphone the other end can not hear a thing. When
using X-pro the opposite happens, local user can not hear a thing. These
softphones work
Hmm, I managed to get callerid working last night!
That is calls coming in from POTS on my X1000P card show up correctly
at ASterisk.
I noticed on by BT102 phone that the number was displayed! Great!
However when I dialled in and withheld my number, the Bt102 showed something
which resembled '
Currently I am creating .wav
files and then converting them via SOX to .au file format, then running them
througha gsm codec convertor which all works fine except that it sounds
like the recording was made with a sock in my mouth !!
Could someone in * land help me
to get a good sound
Steve Underwood wrote:
Albania, I think :-)
Cite your source.
--
I am seeking part or full time employment in the Greater Toronto
Area, My preference is part time employment with some
telecommuting, but all offers will be considered.
Contact eric at fnords.org.
I have a potential customer who has an existing PBX with analogue FXS
ports connected to phones. He wants to allow a single remote worker
to be connected to one of the analogue extension ports using VOIP.
I know I could do it using Asterisk with an X100P card, but that seems
a bit overkill.
Hi,
So if I want NT mode, I need layer 2 and 3 in user
space ?
Exactly.
How can I use the mISDNuser library to works with
asterisk ? I have compiled chan_misdn with mISDNuser.
That should be enough.
A nother question, to connect asterisk to a classic
pbx, what I need ? NT or TE mode ?? ptp mode ?
Hello,
If one would like to build a redundant Asterisk setup, would it be possible
to exchange the locationdb for the SIP users between then?
IE, the following setup:
SIP Phones -- Asterisk SIP carrier
| |
On Wednesday 01 December 2004 19:44, Stephen R. Besch wrote:
Exactly. Would those people who respond from the mailing list digest
-PLEASE-PLEASE-PLEASE- do the following simple things:
1)Strip out the digest messages that have nothing to do with your reply.
2)Copy the appropriate subject
On Wednesday 01 December 2004 20:31, Steven Critchfield wrote:
I am glad it solved the problem. Now if only someone knew what it was
about the stock RH or FC kernel that makes it happen you could get RH or
FC to stop using that patch. That or maybe more people will be like me
and always cast
I just noticed something when I 'sip show peers' from the CLI, I get the
following:
6113/6113 x.x.x.x D N 255.255.255.255 62927OK (66 ms)
6112/6112 x.x.x.x D N 255.255.255.255 50079OK (160 ms)
6111/6111 x.x.x.x D N 255.255.255.255 60810OK
On Tue, 7 Dec 2004, Julien Goodwin wrote:
On Mon, Dec 06, 2004 at 07:43:24AM -0600, Rich Adamson arranged a set of bits into the following:
I don't think its an argument as much as it is folks expressing opinions
without giving you a clue why they've formed that opinion. Here's another
one.
SCCP
Hi,
I would like to offer you the following specialized embedded
Mini-ITX Mainboard:
Samples: $390
50 pcs: $270
100 pcs: $255
The Technical Specification is:
Dimension: Mini-ITX, 170x170mm
System Processor:Intel Mobile Celeron 733MHz (Fanless)
Thats normal when it cant discover the ID
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Dent
Sent: Tuesday, December 07, 2004 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Budgetone 100 Caller ID
Greetings!
Version 1.0.3 has been released of Asterisk, Zaptel, and libpri. As
usual, the tarballs can be downloaded from the Digium ftp server. For
more detailed download instructions, please see
http://www.asterisk.org/index.php?menu=download.
The changes to Zaptel and libpri are minor.
I use stable CVS asterisk and it is working without
problems.
But now i am trying to compile chan_capi 0.3.5
module and i get following error
/usr/src/chan_capi-0.3.5# makegcc -pipe -Wall
-Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT
-D_GNU_SOURCE -O6
Peter, thanks for educating this ISDN-ignorant American! The ASCOM and the
problem are in Germany. This is definitely overlap dialing in the extreme,
from looking at the PRI debug output of asterisk. I set overlapdial=yes in zapata.conf, with no
difference observed in the behavior.
Sorry this doesn't answer your question. Any reason to not leave them as wav's?
On Tue, 7 Dec 2004 10:42:58 +0100, Matthew Oulton
[EMAIL PROTECTED] wrote:
Currently I am creating .wav files and then converting them via SOX to .au
file format, then running them through a gsm codec convertor
On Monday 06 December 2004 22:59, Rich Adamson wrote:
Inline...
I know that VoIP providers can supply their customers with a local
number and/or virtual numbers, and then that number/account can be used
with Asterisk (well, it depends on the provider and whether or not their
service is
In article [EMAIL PROTECTED],
Rich Adamson [EMAIL PROTECTED] wrote:
I have a potential customer who has an existing PBX with analogue FXS
ports connected to phones. He wants to allow a single remote worker
to be connected to one of the analogue extension ports using VOIP.
I know I could
Hi,
I cannot see cid for incomming call from PSTN (Quintum gateway) to IAX
client (FireFly). Client displays blank but when I look into cdr's
/var/log/asterisk/cdr-cvs/Master.cvs, the callerid is registered
properly. Why it's not displaying?
L.
___
Dear List
im try to look if IAX2 protocol is able to transport an hangup cause
from a
TDM PRI line, as i can see from this link http://www.cornfed.com/iax.pdf
seems support only few message like congestion,busy,call progress,
answer,ring,ringing but i cannot transport the Cuase of PRI
On Tuesday 07 December 2004 04:36, Erick Perez wrote:
Hi people,
question one
i see that asterisk is now in 1.x release. having tried it in the past
i want to know if i can use a voice modem as an outgoing line.
i know in the past that was not possible/supported so im just asking
in case
I cannot get the transfer button to work on a Snom
190, I cannot get the # to work either.
Any ideas?
Regards
Thorben
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Julien Goodwin [EMAIL PROTECTED] writes:
Otherwise you can let us know what's missing for you and we'll see
what we can do.
Since you ask... :-) I'm using chan_sccp with an old 12SP+, and it's
working fine except that no ring or busy signal is heard when dialing
out from the phone. On
Hi! I've got a Comdial PBX that I would dearly love to replace with an
Asterisk box. However, for various reasons, it appears not to be in the
cards. Regardless of what management does, or does not, want, our
current VM solution -- some Dialogic card with a KeyVoice application
-- is dying.
Hi all I am beginning in asterisk and am making tests with an ata-186.
For the time being the tests are going well, however have a doubt.
I am thinking about using a canal e1 with plate digium.
Assuming that the company of telecommunications supplies e1 with 30 canals
and numeration to me
I have a potential customer who has an existing PBX with analogue FXS
ports connected to phones. He wants to allow a single remote worker
to be connected to one of the analogue extension ports using VOIP.
I know I could do it using Asterisk with an X100P card, but that seems
a
Hi All
I've done some reading on the wiki and read some of the mailing list
archives, but can't see anything on this. I guess this means I'm either
searching on the wrong thing, or have totally the wrong idea... Can anyone
suggest if the following is possible?
Currently, our office has a 24
Try setting the codec settings for each peer instead of under the general
heading.
On Tuesday 07 December 2004 05:39 am, Paul Fielding wrote:
I'm in the middle of getting g729 to work on my server and running into odd
stuff. The issue revolves around what appears to be a much talked about
On Tue, 2004-12-07 at 11:36 +0100, Milos Kocbek wrote:
[snip]
chan_capi.c:23: asterisk/features.h: No such file or directory
chan_capi.c:24: asterisk/utils.h: No such file or directory
[snip]
Iirc you don't have the asterisk header files installed. They are
installed when you do make install in
I've done some reading on the wiki and read some of the mailing list
archives, but can't see anything on this. I guess this means I'm either
searching on the wrong thing, or have totally the wrong idea... Can anyone
suggest if the following is possible?
Currently, our office has a 24
On Tue, 7 Dec 2004, Nick Burch wrote:
Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines
providing it with external connectivity. We have several analogue
extensions spare, but no capacity to add fancier connectors to link to an
asterisk system (as most of the PBX linking
.. and from a newbie no less :-)
I have configured my BT101, and hooked it up to my * box. All is well.
I have entered the following in externsions.conf, and this bit works:
exten = 613,1,Answer
exten = 613,2,Playback(demo-echotest)
exten = 613,3,Echo
exten = 613,4,Hangup
If I pick up the
http://www.freedomphones.net/polycom/files/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Sent: Sunday, December 05, 2004 4:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Polycom IP500
Does anyone
On December 7, 2004 07:51 am, Nick Burch wrote:
Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines
providing it with external connectivity. We have several analogue
extensions spare, but no capacity to add fancier connectors to link to an
asterisk system (as most of the
Kevin Walsh wrote:
Robert Rozman [EMAIL PROTECTED] lazily top-posted:
do you have info in what countries g.729 is not valid... ?
You could start with the whole of Europe and can also add the UK.
I'm sure there are lots of other countries who don't feel the need to
acknowledge US-based
On Tue, 2004-12-07 at 10:54 +0100, E. Versaevel wrote:
Hello,
If one would like to build a redundant Asterisk setup, would it be possible
to exchange the locationdb for the SIP users between then?
Basically I would start with building redundancy in the the primary
server, e.g. a ton of fans
Eric Wieling aka ManxPower wrote:
Steve Underwood wrote:
Albania, I think :-)
Cite your source.
I might be wrong. I'm working from second hand knowledge. Someone told
be they never introduce copyright legislation and their patent
legislation is almost non-existant. I think you would be in the
On Mon, 2004-12-06 at 17:40, Alan Ingleby wrote:
exten = 1000,2,Dial(Zap/1:555-1234,20,tr)
Change this to exten = 1000,2,Dial(Zap/1/5551234,20,tr)
Oh, and what extension do I use to reference an incoming call on my
FXO port? exten = 1 ??
You want the s extension.
Hi list!
I'm getting these errors in the log:
Dec 7 11:08:04 NOTICE[442388]: No available lines on: [EMAIL PROTECTED]
Dec 7 11:08:04 NOTICE[442388]: Unable to create channel of type 'Skinny'
What does this mean?
Cheers!
Remco
___
Asterisk-Users mailing
Hi,
Could someone help me on configuring a H.323 trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
On Tue, 2004-12-07 at 10:54 +0100, E. Versaevel wrote:
Hello,
If one would like to build a redundant Asterisk setup, would it be
possible
to exchange the locationdb for the SIP users between then?
Basically I would start with building redundancy in the the primary
server, e.g. a ton of fans
there was a website on the list recently that
allowed you to enter text (up to 50 words) and it would create a wav file with
various voice options. does anyone remember what it was? rapsody
something or another.
___
Asterisk-Users mailing list
E. Versaevel wrote:
Which app do you use for monitoring the primary box and if it fails
taking over the IP address by the backup one? I haven't found a suitable
(active-active) app so far.
Thinking of using heartbeat or something.
VRRP, Virtual Redundancy Router Protocol, an option?
Stefan de
I seem to be missin the save dialplan command in
asterisk 1.0.2, I have been searching for info
but all I get is how to use it.
Anybody have any info on this?
Regards
Greg Cirino
___
___
Asterisk-Users mailing list
[EMAIL
On Tue, Dec 07, 2004 at 09:44:59AM -0500, Steve Totaro wrote:
there was a website on the list recently that allowed you to enter text (up to
50 words) and it would create a wav file with various voice options. does
anyone remember what it was? rapsody something or another.
I think it was an
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323 trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
http://www.rhetorical.com/cgi-bin/demo.cgi
Darren Wiebe
[EMAIL PROTECTED]
Steve Totaro wrote:
there was a website on the list recently that allowed you to enter
text (up to 50 words) and it would create a wav file with various
voice options. does anyone remember what it was? rapsody something
That would lead more to keepalived I think
Would be an option, but I would have to use fixed IP addresses for the IP
Phones (that should not be a problem)
Erik
E. Versaevel wrote:
Which app do you use for monitoring the primary box and if it fails
taking over the IP address by the backup one?
Steve Totaro wrote:
there was a website on the list recently that allowed you to enter text
(up to 50 words) and it would create a wav file with various voice
options. does anyone remember what it was? rapsody something or another.
http://www.babeltech.com/Demos.php?s=48m=3f=95
On Tuesday 07 December 2004 14:39, E. Versaevel wrote:
Which app do you use for monitoring the primary box and if it fails
taking over the IP address by the backup one? I haven't found a suitable
(active-active) app so far.
Thinking of using heartbeat or something.
Take a look at
Title: Message
Hi
all,
I have
a problem starting the ztdummy. Here is what happens:
[EMAIL PROTECTED] /]# modprobe ztdummyNotice: Configuration file is
/etc/zaptel.confline 0: Unable to open master device
'/dev/zap/ctl'
1 error(s) detected
FATAL: Error running install
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent: Wednesday, 8 December 2004 1:48 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Website that reads text recently on the list?
On Tue, Dec 07, 2004 at 09:44:59AM -0500, Steve
Henry Devito schrieb:
On Sat, 4 Dec 2004, Cian O'Sullivan wrote:
They have a pizza box server as their asterisk server with a T1 card. No
more slots, so if I want to use the existing infrastructure I will need
to build a second server with an FXO port. Kinda stupid having a second
server just to
Asterisk and it works fine untill the following
situation:
- one of the telco lines occasionally becomes mute after call is
completed, would not provide dial tone, (not sure about ringing on that
line) - both via old and new PBX.
- zap show channel n would show that line as 'Offhook', though no
I'm trying to setup a Cisco ATA 186 which has a public IP address but
sits behind a firewall and connects to an Asterisk server with a NAT IP
address sitting behind a BSD firewall. The Cisco registers with the
Asterisk server without any problems, and I can place calls without any
problems and
This is a good question that the OP posted. Let say you have installed an
Asterisk box at a customer location because they have 50 extensions and all
talk to eachother alot. If their asterisk box fails, how can you re-direct
them to your master box downtown?
Matthew
- Original Message -
Title: OpSign
I'd plug four telephones in these lines and test if the lines are
really engaged or not and in case it is busy, the other will ring or it
will bring you to the voicemail. I ha a similiar problem, the telco had
no engaged the lines properly, after this was solved , I also had a
Stojan Sljivic - Pamet wrote:
Hi all,
I have a problem starting the ztdummy. Here is what happens:
I have used following command to make the ztdummy:
make clean
make linux26
make install
I use Fedora Core 3.
You need to read the udev.README file in the zaptel make directory.
Doug
On Tue, 2004-12-07 at 15:47 +0100, Stefan de Konink wrote:
E. Versaevel wrote:
Which app do you use for monitoring the primary box and if it fails
taking over the IP address by the backup one? I haven't found a suitable
(active-active) app so far.
Thinking of using heartbeat or something.
Asterisk can work with ADSI phones,
What I have in mind is a pci card with zap-like-driver that supports digital
phones. This eliminates (is compairable to using channel bank) additional
delay and a primary echo source when both haves of a conversation are carried
on the same pair as found
I've been struggling with a test * install for a couple months now in a
small office and am just about ready to give up on it. It's not that the
system itself is a problem. I've got everything (attendant, voicemail,
FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers)
working
Ok,
I have had problems with calls dropping repeatedly today,
does anyone have any suggestions on what to make sure is not running? I have
x-windows disabled and Apache disabled. I noticed that mpg123 always seems to
have 2 processes running, is there any way to drop this down to just 1?
Thats it. Thanks!
- Original Message -
From: Darren Wiebe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, December 07, 2004 9:52 AM
Subject: Re: [Asterisk-Users] Website that reads text recently on the list?
However when I dialled in and withheld my number, the Bt102 showed something
which resembled ' tr1' ?
That's its babytalk for asterisk!
When we get calls with no CID, I do a setCallerID(000) for those phones
hth
___
Asterisk-Users mailing list
On Tue, 2004-12-07 at 09:18 -0600, Matthew Boehm wrote:
This is a good question that the OP posted. Let say you have installed an
Asterisk box at a customer location because they have 50 extensions and all
talk to eachother alot. If their asterisk box fails, how can you re-direct
them to your
--- Michael Manousos [EMAIL PROTECTED]
wrote:
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323
trunk.
I am trying to set up the following scenario:
Title: OpSign
Have any of you tried to disable ACPI on the kernel?
Rich Adamson wrote:
On Wed, 2004-12-01 at 13:03 -0700, Michael Welter wrote:
Steven Critchfield wrote:
On Wed, 2004-12-01 at 13:36 -0600, Rich Adamson wrote:
[EMAIL PROTECTED] wrote:
I've been struggling with a test * install for a couple
months now in a small office and am just about ready to give
up on it. It's not that the system itself is a problem.
I've got everything (attendant, voicemail, FXS extensions,
Cisco and Polycom hard-IP phones,
Loosing calls wouldn't be to much of a problem I think, and it would be
impossible to make a gracefull takeover if asterisk is in the mediastream.
keepalived implements vrrp2 so that might be good enough.
The problem lies in the registration data, but that could be solved by
using fixed ip
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.
Does anyone know if the renegotiation setting for the
polycom phones will cause any existing/current calls to be dropped when the phone
tries to renegotiate? I believe this might actually be what is causing my calls
to be dropped. Like I said in my previous email I am not seeing any errors
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: December 7, 2004 9:45 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Website that reads text recently on the
list?
there was a website on the list recently that
allowed you to
Hello all,
We've been using our Asterisk system live for about a month now and I'm
looking to tuning a few things. First, is echo, I receive a fair amount of
echo during the first 10-15 seconds of incoming calls.
Next is a very weird problem. We have serveral Polycom IP300's and one
Budgetone
I'm trying to setup a Cisco ATA 186 which has a public IP address but
sits behind a firewall and connects to an Asterisk server with a NAT IP
address sitting behind a BSD firewall. The Cisco registers with the
Asterisk server without any problems, and I can place calls without any
I have been trying to use the Avaya 4606 IP Telephone (with support for
H323) with Asterisk.
Has anyone else attempted this? Any success or definite failure?
I know I must also use a gatekeeper with it, and I have tried both GNU
Gatekeeper 2.0.8 and Open GK. However I have no success so far.
I see that the 100p is a modem with an Ambient chipset. Why does it
sell for 80$ in some places? i can get Ambient pci modem down here for
9 dollars. Any difference?
On Tue, 7 Dec 2004 10:58:55 +, Jon Lawrence [EMAIL PROTECTED] wrote:
On Tuesday 07 December 2004 04:36, Erick Perez wrote:
Hi
I feel your pain! We have had the same problem with our telco lines
but found that converting to ISDN helped. If the delay on the send
and receive two pair is to big the echo canceller is not strong enough.
Try using a Voictronix card as they seem to solve the problem to a
degree but I would
Is it possible to play a message, when user pickups a phone.
For example:
press 1 to use this provider,
press 2 to use this ...
etc..
Thanks
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To
Erick Perez wrote:
I see that the 100p is a modem with an Ambient chipset. Why does it
sell for 80$ in some places? i can get Ambient pci modem down here for
9 dollars. Any difference?
Because Digium is selling support plus the modem, not just the modem.
-Chris
--
Christopher L. Wade
Patch could not be applied to the latest cvs version
and also
http://www.voip-info.org/wiki-Asterisk+Broadvoice+patch?page=Asterisk%20Broadvoice%20patchcomments_threshold=0comments_offset=0comments_sort_mode=commentDate_desccomments_maxComments=10comments_parentId=1209#threadId1210
--
Best
For a few weeks we have been getting errors that drop our PRI. The telco
says the the line is clean and that our equipment is the problem. We're
currently running Asterisk CVS-HEAD-12/03/04 but several versions
have been tried in an attempt to fix the problem.
The * server is based on a
I'm trying to redirect the call to PSTN if no one is available in the queue
or the agents in the queue do not answer.
The following will redirect the call if no agents are logged in. But if
the agent does not answer the call will timeout and the call will be
terminated, not redirected. I've
There are some documentation about it ?
Thanks
Sergio Faulhaber
[EMAIL PROTECTED]
B. Vallet - www.acropolistelecom.net wrote:
Yes it is possible but make distinct between simultaneous channels and
phones numbers (DID) you can have for example 1000 phones numbers and 30
channels (E1) or
I'm doing this in a call centre with Budgetone 100 telephones. But, in
my case, its the Budgetones that offer the option to automatically dial
an extension when the handset is lifted (or the speakerphone button is
pressed)
Derek
PS. The latest release of the Budgetone firmware is broken and
Christopher L. Wade schrieb:
Because Digium is selling support plus the modem, not just the modem.
But when you don't need the support?
Bye!
Michael
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Hello,
We have a high volume of incoming and outgoing calls that come in via
our analog POTS lines connected to FXO cards in an Adtran TA750. This
is connected to a T100P.
We are using Polycom IP 500's. The problem we are experiencing is, on
frequent occasions, when someone dials out, there is
Hello!
Am Mittwoch, 1. Dezember 2004 14:56 schrieb Michael Graves:
I love my Polycom IP600s. However, I'm not clean on how the status
setting on the phone impacts the behaviour of *. Anyone here have the
details?
No answers so far?
Andi
--
- Andreas Roedl- Senior IT Manager /
I have been looking at moving from SIP-based DID (Illinois) providers to
one that uses the IAX protocol for DIDs. After a search, I've come up
with the following:
http://connect.voicepulse.com -- $8/month, many rate-centers
http://www.iax.cc -- $1.50/month + 0.014/min, many rate-centers
Can
Which app do you use for monitoring the primary box and if it fails
taking over the IP address by the backup one? I haven't found a suitable
(active-active) app so far.
Thinking of using heartbeat or something.
VRRP, Virtual Redundancy Router Protocol, an option?
Cisco
Michael Vogel wrote:
Christopher L. Wade schrieb:
Because Digium is selling support plus the modem, not just the modem.
But when you don't need the support?
Bye!
Michael
Exactly. Choose the level of support you want from Digium and/or the
list. Historically, Digium equipment gets support from
Nice sounding audio.
On the demo there is a button to Download wav file This sounds like
it should allow
me to save the sample?
Does not seem to work for me.
thanks
Mike
On Tue, 07 Dec 2004 07:52:29 -0700, Darren Wiebe [EMAIL PROTECTED] wrote:
http://www.rhetorical.com/cgi-bin/demo.cgi
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