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
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
'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
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.
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
Would you mind sharing with the list the tellabs hardware and how you got it up and running (ie pinouts etc)?
On 1/15/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello Dan,I was fighting with echo on a number of circumstances, and came to thefollowing conclusions.If you are on a distant loop,
-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
Hello Dan,
I was fighting with echo on a number of circumstances, and came to the
following conclusions.
If you are on a distant loop, or analog lines with issues, those issues
need to be addressed or you need a workaround.
In a few cases, I converted to ISDN-BRI, which has been one of my best
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
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
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
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]
What is the Model you have? ML220(newer model, Supports SIP and H323) or
ML210A (older model, supports only H323 I guess)?
The manual at the link below is self explanatory. Just provide your Server IP,
Account and Pin/Password for your Asterisk Box and you are ready to go.
Seshu
Do you have a firewall turned?___
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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
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/
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
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
card it is most probably the busy
detect.
Doug
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Tim
JacksonSent: Friday, November 26, 2004 9:44 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] SIP
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 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
Ive had the same problem. I posted
to the list earlier about the problem, and from what I can tell, its a Polycom
issue (not a network issue as stated in the other post). It happens after the
phones have been on for about 2-3 days from what I can tell. My solution to
this was to use a
: [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
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
___
Asterisk-Users mailing list
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
___
Asterisk-Users mailing list
[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
___
Asterisk-Users mailing list
[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
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
Me and another guy are working on LCD drivers etc for Linux. The thing
Including touchscreen?
Ideally someone would tell me how to make something either a) seamlessly
convert serial/parallel/USB port to TCP and back at the other end, or b)
point me to a resource on a simple chip with TCP
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 that does the same thing as the current solution. Baby steps.
I
On Sun, Nov 21, 2004 at 06:18:04PM +1300, Matt Riddell spake thusly:
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
What would
Tracy R Reed wrote:
This is the way I want to go. A very small PC with a good touch screen.
And there are tons of extremely small systems that could do this job. I
have here in front of me a Soekris net4801 which is tiny, noiseless
computer (similar to a PC, but not quite the same) that draws
On Sun, Nov 21, 2004 at 12:21:16AM -0700, Kevin P. Fleming spake thusly:
And there are tons of extremely small systems that could do this job. I
have here in front of me a Soekris net4801 which is tiny, noiseless
I know there are plenty of small systems that would be great. The problem
is the
$400-500 device here. Not very price competitive. I would like to see less
than half that.
What is the price point you are trying to hit? Any piece of a
proprietary telecom system is by nature overpriced to begin with, and
receptionist consoles certainly fit into that category.
I agree that
Title: SIP phones
Grandstream - are pretty loud
Mitel
5055 or the conference unit are both loud
Doug ReidDirectorStormcorp Network Solutions (Pty)
LtdTel: +27 11 807 1141Fax: +27 11
807 3504Mobile: +27 83 989 0008E-Mail:
[EMAIL PROTECTED]Web:
www.stormcorp.co.za---NOTICE - This message
On Wed, 20 Oct 2004 09:39:31 -0400, Michael Di Martino [EMAIL PROTECTED] wrote:
I am looking for a loud ringing SIP phone. I am presently using the Polycom
and just cannot loud enough to hear it over the din in a collocation room.
Cisco 7905G, 7940G and 7960G phones have very loud ringers.
Title: SIP phones
Why dont
you use an ATA device with a loud regular phone and/or hook up one of those
really loud ringing devices you can get at a phone shop? J
Just a
suggestion.
S.
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Michael Di Martino wrote:
I am looking for a loud ringing SIP phone. I am presently using the
Polycom and just cannot loud enough to hear it over the din in a
collocation room.
My Cisco 7960 has the loudest ring that I have ever heard, from any
phone, period.
--
Kristian Kielhofner
Title: Message
Keep your
phone.
Get an ATA and connect an analogringer to it (there are
hundreds of different kinds of analog alerters to be had, from chimes to strobe
lamps). Just program the ATA to ring when your phone does, connect your alerter
of choice to it and you're set!
Wheelock
On Oct 20, 2004, at 10:42 AM, Kristian Kielhofner wrote:
Michael Di Martino wrote:
I am looking for a loud ringing SIP phone. I am presently using the
Polycom and just cannot loud enough to hear it over the din in a
collocation room.
My Cisco 7960 has the loudest ring that I have ever heard,
Uniden-200
- Original Message -
From: Jean-Yves Avenard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 12:17 PM
Subject: [Asterisk-Users] SIP phones recommendations
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dear all.
We are currently using either Grandstream
Uniden UIP-200
you can follow this thread in the list: Cheap (US$120 or less) SIP Phones
- Original Message -
From: shabanip [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 1:21 PM
Subject: Re: [Asterisk-Users] SIP phones recommendations
Uniden-200
- Original
Consider the Uniden UIP200, I believe it meets all your criteria.
http://www.voip-info.org/wiki-Uniden
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Jean-Yves Avenard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 14, 2004 9:47 PM
Subject:
Yeah, but how do you get one? We can't find a single dealer with any in
stock -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James H.
Thompson
Sent: Thursday, July 15, 2004 2:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP phones
-Users] SIP phones recommendations
Consider the Uniden UIP200, I believe it meets all your criteria.
http://www.voip-info.org/wiki-Uniden
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Jean-Yves Avenard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
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Tomica Crnek wrote:
Hi everyone,
I have to test few models of SIP IP phones with Asterisk. I have seen on
voip-info.org that there are lot of phones that work ok with Asterisk.
But, I want to ask for suggestion - which models are the best for Asterisk?
We have a dozen or so Polycom IP 500's
My own
personal opinion in order of preference:
Snom
200 - excellent albeit a little too expensive
Grandstream - excellent - absolutely excellentfor the
price
Cisco
7960 - wel - cool looking build like a tank but require some fiddling and WAY to
expensive
ZyXEL
Prestige 2000W - too
On Mon, Feb 09, 2004 at 02:29:50PM -0500, Jess Magnaye wrote:
have you tried this gs-102 with pppoe? verizon dsl uses
pppoe. pppoe is
No, I didn't try. Yes, pppoe is fairly standard DSL stuff (when used
with an ethernet modem).
logically like dhcp, but using ppp for added feature like aaa :)
On Mon, Feb 09, 2004 at 11:21:42PM +0100, Tomas Prybil wrote:
How would you roll out a SIP based VoIP platform to to endusers with
various connection solutions. Is there such a thing that solves the
various issues of NATting a phone?
Well, there is not one-fit-all solution. GS phones
The polycom phones (ip 500) have a simple switch built in. I say simple
because we had an issue were a network device was generating large
amounts of traffic to nonexistent mac addresses and the computer behind
the phone would not receive some packets. Probably not an issue though
as once we got
On Mon, Feb 09, 2004 at 04:36:41PM +0100, [EMAIL PROTECTED] wrote:
Some vendors sells phones with dual ethernet ports. Are these just
incorporating a hub/switch functionality? The reason for my question is
that the normal case for a DSL customer is the possibilty to use one MAC
adress from
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