Hello,
I am running Asterisk 11 on CentOS 6.x using the DAHDI module with 8x PSTN
analog phone lines for outside connectivity. Internally, I am using several
models of Yealink SIP phones (e.g SIP-T32G) on a dedicated VoIP network,
192.168.0.0/24. I have a few of these Yealink SIP phones
Hi list
I am having trouble getting asterisk to perceive the firewall's ip address as
outside localnet (setting in sip.conf). The situation is this:
- phones inside lan work fine when localnet is set to 192.168.0.0/255.255.255.0
- phones outside the lan can't ack the invite from asterisk because
Hello
Can anyone suggest sip phones with headset for use in call centers . They
should be fully inter operable with Asterisk over sip .
Thanks
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Hi,
recently I changend a few things in the configuration of the Asterisk
1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that
different groups of SIP-Phones are using different trunks to the outside
worls, so I moved some of them to a Support context.
However, dial out from this
It might help to show your Support context in outbound.conf.
MARK.
Alexander Topolanek wrote:
Hi,
recently I changend a few things in the configuration of the Asterisk
1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that
different groups of SIP-Phones are using different
On Mon, Jul 02, 2007 at 10:54:14PM +0200, Alexander Topolanek wrote:
Hi,
recently I changend a few things in the configuration of the Asterisk
1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that
different groups of SIP-Phones are using different trunks to the outside
worls,
Over my experience with 1.0 and 1.2 branch, if you register both phones
the same SIP account and you will call it then both phones will ring,
however, from reading here and there I heard mixed feedback about it so
I just dedicated an account for each phone and I dial both of them at
the same
Each employee has a Polycom phone at his desk at the real office as well
as a Polycom at his home office.
I'd like a call to the employees extension to ring both phones. I'd
also like one entry in the buddy list for each employee, and the buddy
list to indicate he was on a call no matter
Setting up a new system, have two sip phones that give dial tone and appear to
dial, but do not complete,
giving a busy.
Watching the CLI thing, get this message,
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/200-0825b648,
recordingcheck|20060929-195420|1159574059.5) in new
Chris Bagnall wrote:
Okay, so assuming I've got to drop the re-registration to a much shorter
time than the default of every hour, what are the implications of doing so
(in terms of network traffic, load on the asterisk box, etc.)? What's the
lowest one can reasonably take it? 10 minutes? 1
I would also recomend that you upgrade to the latest firmware 1.0.2.13
(contact grandstream) as it does fix some registeration issues and have
extra NAT/STUN features.
On Wed, 2006-05-03 at 17:15, Chris Bagnall wrote:
Greetings list,
I'm coming across an issue with some of the GXP-2000 phones
the problem.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Gareth Blades
Gesendet: Donnerstag, 4. Mai 2006 09:59
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] SIP Phones behind dynamic IPs
I would also
'recognize'? The phone cannot know that the external IP has
been changed, unless it is using a STUN server and
periodically re-doing the STUN queries (which I doubt any phones do).
Thanks for clearing up my misunderstanding as to the point of STUN. :-) I
thought the phone would query the
Greetings list,
I'm coming across an issue with some of the GXP-2000 phones we have out in
the wild at clients' employees' homes. In most cases they're behind consumer
ADSL NAT routers on a dynamic IP from their ISP.
In a nutshell, the phone is unable to be called unless it's restarted first,
Chris Bagnall wrote:
I think what's happening is that the ADSL router is reconnecting after a
break in the connection (as it should), getting a different IP, but the
phones don't seem to be recognising they've got a different IP and updating
the asterisk server with the good news.
I'm looking for a confirmed list of SIP phones that have support for BLA.
Thank you for any info you can provide
-Tim
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Citel Handset Gateway phones support BLA (http://www.citel.com).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tim
Ferguson
Sent: 26 April 2006 10:51
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sip Phones with BLA Support
I'm looking
On Sunday 15 January 2006 12:23, Kerry Garrison wrote:
I have an install with the Digium TDM2400 with the EC module and even
though Digium techs have spent well over 10 hours tweaking and tweaking the
call quality is so bad we are ready to chuck it. I think that you were on
Is this FXS or FXO
@lists.digium.comSubject: [Asterisk-Users] SIP phones unbeatable echoHey all again, I'm wrestling with echo problems on our sip extensions.I've set these items in zapata.conf but tweaking these values doesn't
seem to make much differenceechocancel=yesechocancelwhenbridged=yesechotraining=2500rxgain=8.0txgain=1.0are
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, January 15, 2006 12:27 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] SIP phones unbeatable echo
Hello Dan,
I was fighting with echo
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, January 15, 2006 12:27 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] SIP phones unbeatable echo
Hello Dan,
I was fighting with echo
,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Elder
Sent: Thursday, January 12, 2006 2:53 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP phones unbeatable echo
Hey all again, I'm wrestling with echo problems on our sip
On 1/12/06, Dan Elder [EMAIL PROTECTED] wrote:
Hey all again, I'm wrestling with echo problems on our sip extensions. I've
set these items in zapata.conf but tweaking these values doesn't seem to
make much difference
I assume from this that you are referring to SIP extensions making
calls out
Hey all again, I'm wrestling with echo problems on our sip extensions. I've
set these items in zapata.conf but tweaking these values doesn't seem to
make much difference
echocancel=yes
echocancelwhenbridged=yes
echotraining=2500
rxgain=8.0
txgain=1.0
are there other settings that can help me
A very good day to you all,
We can't get the phones to pick up on an incoming call on analog trunks.
We're using the digium products in the box, with snom phones internally.
This is the output from the asterisk console:
linux*CLI zap show channels
Chan Extension Context Language
Hello all,
Is there a list of phones that reliably support SIP early dial? One of the
really nice things I've noticed about the 7960 (SCCP) is that each digit is
sent straight to asterisk, so when the number has been completed, connection
is almost instantaneous. I've tried early dial on both the
Chris Bagnall wrote:
Hello all,
Is there a list of phones that reliably support SIP early dial? One of the
really nice things I've noticed about the 7960 (SCCP) is that each digit is
sent straight to asterisk, so when the number has been completed, connection
is almost instantaneous. I've tried
On Fri, Nov 04, 2005 at 09:34:29AM -0600, Eric ManxPower Wieling wrote:
Chris Bagnall wrote:
Hello all,
Is there a list of phones that reliably support SIP early dial? One of the
really nice things I've noticed about the 7960 (SCCP) is that each digit is
sent straight to asterisk, so when
Hi,
I wish to set up a simple network of about 20 SIP phones. This will be
a stand alone VoIP network, without any links to the internet or
standard PSTN networks.
For SIP phones to work, one needs a SIP server so I thought that
Asterisk might be a good choice.
Does anyone have a list of SIP IP
Check out voip supply.com. All their SIP phone have been tested with
Asterisk.
Asterisk can work in 2 ways when handling calls. It can set up the call
and then step back and let the phones go peer to peer or it can stay
involved in the call until its terminated.
Obviously the latter
On Tuesday 04 October 2005 00:42, Rajesh kumar wrote:
I am using Kphone which works great for my purposes! You can look at
twinklephone as well at http://www.twinklephone.com/
Hi, thanks all for the info, kphone does really wierd stuff and I can't get
twinkle to compile. I'm looking into that
me too looking for softphone...not able to enable kphone
Can anyone please highlight more on it.
ThX
/Gurmi
On 10/4/05, Wayne Gemmell [EMAIL PROTECTED] wrote:
On Tuesday 04 October 2005 00:42, Rajesh kumar wrote:
I am using Kphone which works great for my purposes! You can look at
Hi all
Can anyone recommend a good soft phone that can compile on x86_64 (linux)
platform?
--
Regards
Wayne Gemmell
Tel Fax: (011) 894-4081
Cell : 072 836 4325
Email : [EMAIL PROTECTED]
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Once upon a time Monday 03 October 2005 3:11 pm, Wayne Gemmell wrote:
Hi all
Can anyone recommend a good soft phone that can compile on x86_64 (linux)
platform?
kphone compiles and is available in Fedora extras and im sure is available
for other distros. If you want to get adventurous you
I am using Kphone which works great for my purposes! You can look at
twinklephone as well at http://www.twinklephone.com/
rajesh
- Original Message -
From: Wayne Gemmell [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 03, 2005 3:11 PM
Subject: [Asterisk-Users
Hi,
I have a bunch of NetPhones that I have bought from MeritCall some time ago for
their service. How can I use this phones (supposed SIP phones) to integrate
with a Asterisk Setup.
I have seen a manual for a similar one but I don´t know If mine are hardcoded
in some way. This devices are
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francisco Paulo
Mateus Nascimento Adriano
Sent: Tuesday, July 19, 2005 10:32 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Phones with Asterisk
Hi,
I have a bunch of NetPhones that I have bought
This is the asterisk output:
-- Executing Answer(SIP/202-8236, ) in new stack
-- Executing Dial(SIP/202-8236,
SIP/203|100|tTr) in new stack
-- Called 203
-- SIP/203-3c5d is ringing
-- SIP/203-3c5d answered SIP/202-8236
-- Attempting native bridge of SIP/202-8236 and
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Hi,
I know this can be done but I guess I am not understanding the few notes
I have seen on this - can SIP phones be tied to Asterisk using a PC mac
address instead of their IP address (obviously I am using DHCP). If
someone could please point to the right Wiki or How to I would greatly
Hi,
I know this can be done but I guess I am not understanding
the few notes
I have seen on this - can SIP phones be tied to Asterisk
using a PC mac
address instead of their IP address (obviously I am using DHCP). If
someone could please point to the right Wiki or How to I
would
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giles Coochey
Sent: Monday, April 04, 2005 10:33 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] SIP phones to Asterisk using MAC
addressinsteadof IP address
Hi
: Monday, April 04, 2005 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SIP phones to Asterisk using MAC
addressinsteadofIP address
If you setup host=dynamic in sip.conf, then the registration does not
depend on ip address. It depends on sip user name
C F [EMAIL PROTECTED] wrote:
Use the latest stable or CVS HEAD and modify features.conf. You can
change it there.
FYI, only CVS HEAD (not stable) supports the new features.conf options.
--
Robert L Mathews, Tiger Technologieshttp://www.tigertech.net/
Hi list!
I have some sip phones and Sipura ATA 2000's. However after dialling a
number I need to dial a # to control a device.
When I dial # Asterisk kicks in and puts the call on hold. How can I
change this?
Thx!!
Remco
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Asterisk-Users mailing
I have had this same problem. The only way I know is to disable
transfers in asterisk. You can still use the transfer control in your
SIP device. Of course this does not work with call parking. I would
be very interested in a solution that does not require disabling of
transfers in asterisk as
Remco Barende wrote:
Hi list!
I have some sip phones and Sipura ATA 2000's. However after dialling a
number I need to dial a # to control a device.
When I dial # Asterisk kicks in and puts the call on hold. How can I
change this?
Do you have the T in your Dial statment? Remove the T and try it.
Is there a way to somehow do an escape # so that you can still use
the # key to control devices that require a #, but still keep the T in
the dial plan? We have clients that need to check external voicemail
systems that require the use of the # sign, but still want to have the
call parking
Pedro wrote:
Is there a way to somehow do an escape # so that you can still use
the # key to control devices that require a #, but still keep the T in
the dial plan? We have clients that need to check external voicemail
systems that require the use of the # sign, but still want to have the
call
Use the latest stable or CVS HEAD and modify features.conf. You can
change it there.
On Tue, 15 Feb 2005 11:35:12 -0600, Eric Wieling [EMAIL PROTECTED] wrote:
Pedro wrote:
Is there a way to somehow do an escape # so that you can still use
the # key to control devices that require a #, but
On Tue, 15 Feb 2005, Eric Wieling wrote:
Pedro wrote:
Is there a way to somehow do an escape # so that you can still use
the # key to control devices that require a #, but still keep the T in
the dial plan? We have clients that need to check external voicemail
systems that require the use of the
Your SIP device does not support attended transfers?
Yes they do
If your devices support their own transfer feature (odd enough usually
labeled Transfer)
then there is NO REASON to use T/t transfers.
Call parking can only work with T/t transfers (at least on the version
I am running - CVS
FYI: Found the info on the wiki regarding features.conf:
http://voip-info.org/tiki-index.php?page=Asterisk%20config%20features.conf
On Tue, 15 Feb 2005 13:10:40 -0500, C F [EMAIL PROTECTED] wrote:
Use the latest stable or CVS HEAD and modify features.conf. You can
change it there.
On
sip phones
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Hello
I have one phone (phone1) in one network, the other (phone2) in public
network. both can call the other side; phone1 can be heard by phone2, phone2
can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER.
Is NAT still necessary to be set on both phones?
Thank you!
Hello
I have one phone (phone1) in one network, the other (phone2) in public
network. both can call the other side; phone1 can be heard by phone2,
phone2
can't be heard. I don't have NAT set on both end, but I use rtpproxy on
SER.
Is NAT still necessary to be set on both phones?
Thank
HendersonSent: Friday, November 26, 2004 12:11
PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] SIP phones
cutting out with Asterisk??Importance:
High
Hi
folks,
I've got a very bizarre problem
recurring when making calls with Polycom SoundPoint IP500 SIP phones
with
the technology.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ian Clough
Sent: Friday, November 26, 2004 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phones-Receptionist Setup
- Original
On Thu, 2004-11-25 at 13:24 -0600, Carmi Weinzweig wrote:
Again, note that I am not asking to display trunk status, just
extension status, and to allow a user to place a call on hold on one
phone and pick it up on another (that has that shared extension).
From another posting today (SNOM
- Original Message -
From: Gregory Junker [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion [EMAIL PROTECTED]
Sent: Thursday, November 25, 2004 11:36 PM
Subject: Re: [Asterisk-Users] SIP Phones-Receptionist Setup
I'm not saying
I'm not saying that it would compromise *'s 'PBXness'. But you are
comparing products that have DECADES of development and maturity,
building on basic features that * is just now getting stable, and that
use proprietary hardware to accomplish these features.
Kinda my point. I
Hello,
I'm certainly not an expert on this, but isn't one of the limiting
factors the functionality implemented by manufacturers in their sip
phones? Or, are we assuming the lamp field is an external device
unrelated to the current production phones?
(I do understand that at least some
Hi
folks,
I've got a very
bizarre problem recurring when making calls with Polycom SoundPoint IP500 SIP
phones and Asterisk. Sometimes when a call comes in to an IP500, one of
the sides of the conversation is cut off (i.e. the caller can't hear the callee,
or vice-versa). This isn't
Hi Dave,
I had a similar problem some time ago on one of our customers servers
and it's not an Asterisk problem,I suggest you to take a look at your
network state, we found a switch failure causing that, you can try this
tool to test the network:
http://www.cacti.net/
Cacti is designed to be a
PROTECTED]
Subject: [Asterisk-Users] SIP
phones cutting out with Asterisk??
Importance: High
Hi folks,
I've got a very bizarre problem recurring when making
calls with Polycom SoundPoint IP500 SIP phones and Asterisk. Sometimes
when a call comes in to an IP500, one of the sides
: [Asterisk-Users] SIP phones cutting out with Asterisk??
Hi Dave,
I had a similar problem some time ago on one of our customers
servers and it's not an Asterisk problem,I suggest you to
take a look at your network state, we found a switch failure
causing that, you can try this tool to test
JacksonSent: November 26, 2004 2:44 PMTo: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] SIP phones cutting out with Asterisk??
Ive had the same
problem. I posted to the list earlier about the problem, and from what I can
tell, its
On Nov 21, 2004, at 11:15 AM, Wayne Sheppard wrote:
Tracy R Reed wrote:
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
I'm not surprised. Asterisk is a PBX, not
On Nov 20, 2004, at 11:05 PM, Gregory Junker wrote:
Most customers don't want to be in a new era. They want something
they are
accustomed to. I don't need any more impediments to making money than
I've
already got. So if the customer wants a busy lamp, I am going to do my
best to give it to
I would like you to name one PBX that does not support this behavior?
Every system from Avaya including their Definity, Merlin Legend, Merlin
Magix, Partner, and their new IP based PBXes support it, as do those
from Mitel, Nortel, InteCom and every other system that I have ever
used. A typical
Carmi Weinzweig wrote:
On Nov 21, 2004, at 11:15 AM, Wayne Sheppard wrote:
Tracy R Reed wrote:
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
I'm not
On Nov 25, 2004, at 12:37 PM, Gregory Junker wrote:
I would like you to name one PBX that does not support this behavior?
Every system from Avaya including their Definity, Merlin Legend,
Merlin Magix, Partner, and their new IP based PBXes support it, as do
those from Mitel, Nortel, InteCom and
I'm not saying that it would compromise *'s 'PBXness'. But you are
comparing products that have DECADES of development and maturity,
building on basic features that * is just now getting stable, and that
use proprietary hardware to accomplish these features.
Kinda my point. I reiterate, if
I went an nosed around the Bayonne site, and looked at their devel list
archivesbased on historical trends, that project looks dormant (it
seems to be duplicating what Asterisk does already -- and better). Other
projects it links to also look either dormant or missing.
I have seriously
Hello all,
I'm new to the list, but use VoIP and * for a little while now.
Running Asterisk 1.0.2 on debian linux I'm facing the following problem:
I've got two Fritz!Box Fon Adapters (kind of ATA's) with two hardware phone
connectors each. So I'm trying to set up a PBX with four internal
I have a 200 and the hint() stuff works fine for indicating status of
any channel (including Agent channels). The Snom subscribes to
asterisk at whatever url you put in there, then * will send notify
events when the dialog state changes. It's not quite a shared-line
(at least the way I
] On Behalf Of Kevin Blackham
Sent: Tuesday, 23 November 2004 18:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Phones-Receptionist Setup
I have a 200 and the hint() stuff works fine for indicating status of any
channel (including Agent channels
On Sat, 20 Nov 2004, Brian Roy wrote:
I would look at putting a dual monitor on her desk. You can pick up a
15 flat panel and a video card for about the same cost as the SNOM.
Not to mention, you get quite a bit more benifite from the FOP
controls than you do busy lamp fields. It's a a new
You should always design an interface around a human being. A hard
I could not agree more. Usability is my focus in any software
system...including open-source, where it is typically the last thing
considered.
Greg
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Gregory Junker [EMAIL PROTECTED] wrote:
$400-500 device here. Not very price competitive. I would like to
see less than half that.
I agree that any touch screen ought to be able to do normal computer
graphics. At this point, you are into normal LCD displays with touch
capability, which I
On Sunday 21 November 2004 11:16, James H. Thompson wrote:
Gregory Junker [EMAIL PROTECTED] wrote:
$400-500 device here. Not very price competitive. I would like to
see less than half that.
I agree that any touch screen ought to be able to do normal computer
graphics. At this point, you
Hi,
Me and another guy are working on LCD drivers etc for Linux. The thing
is, the display would be run from your Asterisk Server. I.E. It will
need to be run from either Parallel, Serial or USB port. We will open
source it once finished, and are not too far off, probably just a spare
day
Not all over $500 - a quick search finds:
For purposes of replacing a receptionist console with a touch screen
(for example, replacing a 6x9 grid of buttons), that would be too small
as well.
Greg
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[EMAIL PROTECTED]
On Sunday 21 November 2004 11:42 am, Gregory Junker wrote:
Not all over $500 - a quick search finds:
For purposes of replacing a receptionist console with a touch screen
(for example, replacing a 6x9 grid of buttons), that would be too small
as well.
Greg
Another strong possibility is that
Another strong possibility is that after a while, few operators would be
willing to continue holding their arms in the air to operate a touch screen.
Why would they be holding their arms in the air? You mount the touch
panel in the same place at the same angle as the current console...
Greg
On Sunday 21 November 2004 11:42 am, Gregory Junker wrote:
Not all over $500 - a quick search finds:
For purposes of replacing a receptionist console with a touch screen
(for example, replacing a 6x9 grid of buttons), that would be too small
as well.
Greg
Another strong
On Sunday 21 November 2004 11:50 am, Gregory Junker wrote:
Another strong possibility is that after a while, few operators would be
willing to continue holding their arms in the air to operate a touch
screen.
Why would they be holding their arms in the air? You mount the touch
panel in
Tracy R Reed wrote:
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
I'm not surprised. Asterisk is a PBX, not a key system or a hybrid
system. The
I'm not surprised. Asterisk is a PBX, not a key system or a hybrid
system. The kind of functionality that is being described here is one or
both of those 'other' beasts. Now I'm not saying that this wouldn't be
nice, or even a long term requirement if you really want to open the
entire SME
Is there an open source key system, comparable to *?
If there isn't , I'd be happy to work on developing one. It is clear
that the need still exists for such a user interface paradigm.
Greg
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[EMAIL PROTECTED]
Bob Goddard wrote:
Not all over $500 - a quick search finds:
http://www.xenarcdirect.com/search_results.asp?txtsearchParamCat=6txtsearc
hParamType=ALLtxtsearchParamMan=ALLtxtsearchParamVen=ALLiLevel=1
Product ID: 700TSCategory: 7 LCD Monitor
700TS - 7' USB Touch Screen LCD Monitor with VGA
On Sat, Nov 20, 2004 at 09:11:15PM -0800, Tracy R Reed said:
On Sun, Nov 21, 2004 at 12:05:27AM -0500, Gregory Junker spake thusly:
What is the size of the current line panel on her desk? I am thinking it
might be worthwhile to produce an addon to Asterisk that drives a flat
touchpanel
Gregory Junker wrote:
Is there an open source key system, comparable to *?
If there isn't , I'd be happy to work on developing one. It is clear
that the need still exists for such a user interface paradigm.
Bayonne is supposed to act as a key system, at least that's what was
described on the
Title: SIP Phones-Receptionist Setup
I am looking at placing a system in an office with a central receptionist, and phones for each individual employee thereafter. Could I use a Snom 220 with additional keypads to view if the lines are in use by the other employees?
Fred is in sales... A call
I am looking at placing a system in an office with a central
receptionist,
and phones for each individual employee thereafter. Could I use a
Snom 220
with additional keypads to view if the lines are in use by the other
employees?
Fred is in sales... A call comes into the receptionist and they
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
Another option is the Flash Operator Panel, you can see a live demo at
http://www.asternic.com/ It is a
On Sat, 20 Nov 2004 15:58:48 -0800, Tracy R Reed
[EMAIL PROTECTED] wrote:
I proposed something like this to a client but the receptionist has other
duties for her computer and does not want to have to have the operator
panel up all the time or go searching for the window in the taskbar every
On Sat, Nov 20, 2004 at 09:25:38PM -0600, Brian Roy spake thusly:
I would look at putting a dual monitor on her desk. You can pick up a
15 flat panel and a video card for about the same cost as the SNOM.
She doesn't want another monitor.
Asterisk is not your dad's pbx.
Most customers don't
Most customers don't want to be in a new era. They want something they are
accustomed to. I don't need any more impediments to making money than I've
already got. So if the customer wants a busy lamp, I am going to do my
best to give it to them.
I agree. This is why engineers do not make good
Gregory Junker wrote:
Most customers don't want to be in a new era. They want something they
are
accustomed to. I don't need any more impediments to making money than
I've
already got. So if the customer wants a busy lamp, I am going to do my
best to give it to them.
I agree. This is why
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