We are going to have people in our office who do not sit at the same desk
throughout the day (or week), and have Cisco 7940 phones using the SIP
image.
Is it possible to easily set up the phone so that they can enter their
extension number and password on the phone, and thus have their extension
Hi all,
I am to come up with a proposal to setup a network of
over 15,000 lines. I would like to scale down the
costs by using Asterisk as the main switching
equipment. Let me give u the full scenario.
1. Fiber optic cables are to run from the central
exchange to over 2 kilometer radius at
Julian wrote:
We are going to have people in our office who do not sit at the same desk
throughout the day (or week), and have Cisco 7940 phones using the SIP
image.
[..]
I really want to find the extension
Isn't this a case for Queues with callback login?
Just a thought
rgds
pos
As usual, you sit for hours thinking of how to implement something, and send
an email asking for help. Seconds later, you think of a potential solution:
Thinking that the extension is a user, not a phone:
Admin:
1) Record is created with a 4 digit UserID
2) Context to use is stored against this
Oh! Man! The simplest solution. Now I feel really stupid.
That may well solve the follow me issue.
Julian
- Original Message -
From: Peer Oliver Schmidt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, November 13, 2004 8:30
I'm confident asterisk can manage such a setup, but you will need a damn
good consultant to set it up. :)
(You cannot buy just a huge asterisk machine, you will need some kind of
cluster to do this).
Joachim (zoa)
jafar mohammed wrote:
Hi all,
I am to come up with a proposal to setup a network
--On Saturday, November 13, 2004 00:11 -0800 jafar mohammed
[EMAIL PROTECTED] wrote:
Hi all,
I am to come up with a proposal to setup a network of
over 15,000 lines. I would like to scale down the
costs by using Asterisk as the main switching
equipment. Let me give u the full scenario.
I have
With a bit of money and hard work - many things are possible.
Brandon
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-- PhoneBoy
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4. Fiber will run to the main Telecommunication
provider(PSTN) and 2 mobile providers.
[snip]
Keep in mind that their is no need for T1/PRI or any
other type of external lines. Asterisk is to switch
the voice data only.
How are you linking to the PSTN referenced in (4) above then? How many
Hi Brian,
One goal is to get BRI support in Zaptel if possible. I'm right now in the
planning stage :P Plus BRI is much cooler than pots.
Why invent the wheel again, what's wrong with bristuff from junghanns.net?
bye,
aa
_
Listen to
Hi,
There is a lot of talk about Cisco phones, SIP firmware and Contracts to
download same.
Does using a 7940/60 or other with SIP firmware offer better
features/compatibility with Asterisk over using the [default?] Call-Manager
firmware and chan_sccp? A lot of people here must have started
Hell yes!!!
The SIP firmware offers so much more and is better supported with *
On Sat, 2004-11-13 at 06:58, Derek Conniffe wrote:
Hi,
There is a lot of talk about Cisco phones, SIP firmware and Contracts to
download same.
Does using a 7940/60 or other with SIP firmware offer better
Hi,
One goal is to get BRI support in Zaptel if possible. I'm right now in the
planning stage :P Plus BRI is much cooler than pots.
Why invent the wheel again, what's wrong with bristuff from junghanns.net?
US bri (afaik) is not EuroISDN, but NI or something like.
funny mode
Of course
I must be missing something with the GPL...
Nowhere does it say you need to advertise the open source
product in your sales literature.
All (from what I gather) is necessary, is to make available
the source or instructions to retrieve the source to the end
user.
This could be on a CD or a
First, I'm really new to asterisk and I'm testing it in order
to improve my first steps...
Recently I installed * asterisk on a FreeBSD Box (5.2.1)
I got it working on my internal LAN (it works fine !).
I was trying to connect my * box through FWD using SIP
but it is not working an
The reason these threads end up rambling on far too much is people post
without reading anything pertinent in the previsious messages.
SysMaster has been vehemently denying their systems are based on
Asterisk, so they have *not* been making any source available, or
telling customers where you
[snip]
If someone believes that they are contributing software to a GPL'd
software project, and does not realize that the nature of your disclaimer
allows Digium to release their changes under a non-GPL'd license, then
that is breaking with the spirit of the GPL.
If that is true, then
On Saturday 13 November 2004 09:11, jafar mohammed wrote:
I am to come up with a proposal to setup a network of
over 15,000 lines. I would like to scale down the
costs by using Asterisk as the main switching
equipment. Let me give u the full scenario.
Wow, you gained 5000 lines between typing
On Fri, 12 Nov 2004, Paul Fielding wrote:
Hmmm... Interesting that you mention it's not a problem with VOIP
companies as they use PRI. The analog trunk I'm connecting to is
actually a Vonage line. Thing is, it seems to me that by connecting via
the Zap channel to the Vonage ATA I'm
On Sat, 2004-11-13 at 06:03, Joe Greco wrote:
[SNIP]
However, the specific item that stopped me was the second paragraph of
the short disclaimer, because our lawyers would never allow signing of
a blanket statement such as and will do nothing to undermine it in the
future. (As it was,
Hi Brian,
One goal is to get BRI support in Zaptel if possible. I'm right now in the
planning stage :P Plus BRI is much cooler than pots.
Why invent the wheel again, what's wrong with bristuff from junghanns.net?
US BRI is alien. It's not the same as BRI elsewhere. (sigh)
... JG
--
The reason these threads end up rambling on far too much is people post
without reading anything pertinent in the previsious messages.
SysMaster has been vehemently denying their systems are based on
Asterisk, so they have *not* been making any source available, or
telling customers
Thank you for making this clear for me.
Is there any solution for the mentioned phones?
Assaf Benharoosh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Saturday, November 13, 2004 1:41 AM
To: Asterisk Users Mailing List -
So, why not use SER to register all the SIP phones, as it doesn't handle
the
media-streams, just keeps track of the phones and does the 'handshake'.
SER is supposed to be able to handle over 50.000 calls at a time, so one
SER
server would be enough.
Then interface this with one (or more)
Is anyone else experiencing a lot of busy signals after this patch? ie
Broadvoice becomes disassociated with asterisk..
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Hi,
Am I correct in saying that the wcfxs kernel module is something of the
past, and is now replaced by wctdm ?
Regards,
Thomas
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On Sat, 13 Nov 2004 10:37:27 -0500, Raymond McKay
[EMAIL PROTECTED] wrote:
So, why not use SER to register all the SIP phones, as it doesn't
handle the
media-streams, just keeps track of the phones and does the 'handshake'.
SER is supposed to be able to handle over 50.000 calls at a time, so
Has anybody tried out the new TA from Uniden? DTA200
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A SIP phone *could* normally send its media stream directly from phone to
phone, if no transcoding is required, but when using Asterisk the media
stream will always pass through the server, causing a pottential
bottleneck.
So, why not use SER to register all the SIP phones, as it doesn't handle
I didn't see anything like that on their website...
Is there something written somewhere that they say this?
besides, I read all of these posts
Greg
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL
Thomas:
Correct. Actually wctdm *is* wcfxs. They just renamed it.
Robert
Thomas Andrews wrote:
Hi,
Am I correct in saying that the wcfxs kernel module is something of the
past, and is now replaced by wctdm ?
Regards,
Thomas
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Asterisk-Users mailing
This is my second asterisk server but the first one with a T100P card.
I connected it to the phone company(SBC) jack but have only a busy
signal when calling the T1's number and nothing in the asterisk log
files to indicate a connection.
Do I need to use a crossover cable?
Thanks
Malcolm
Citat Joe Greco [EMAIL PROTECTED]:
[SNIP]
There are a number of competing theories on whether or not the author of a
public domain bit of code could be liable, with varying amounts of case
law,
as I understand it:
1) One theory is that you may place code in the public domain, with
Citat Joe Greco [EMAIL PROTECTED]:
[snip]
If someone believes that they are contributing software to a GPL'd
software project, and does not realize that the nature of your
disclaimer
allows Digium to release their changes under a non-GPL'd license, then
that is breaking with the
Citat Robert Lawrence [EMAIL PROTECTED]:
Thomas Andrews wrote:
Hi,
Am I correct in saying that the wcfxs kernel module is something of the
past, and is now replaced by wctdm ?
Thomas:
Correct. Actually wctdm *is* wcfxs. They just renamed it.
It would be confusing to continue
Brian Capouch wrote:
I would like to see you say out loud, just once, that those of us who
know all of that and disclaim our work to Digium are not necessarily
idiotic boobs who don't know what we're doing.
As Joe already pointed out, he doesn't believe this to be the case :-)
However, this
all i have is random echo
I have already 4 TA750 with full FXO
echocancel=yes and echo training=800
- what should i do?
- could it be solved with tweaking echo params on *?
- is there any additional devices that can be added between Channel
Bank and * to get rid off echo forever?
any help would
Martin List-Petersen wrote:
Surely, but if you contribute to a project, shouldn't you allways check the
license ? Would LICENSE, COPYRIGHT or README be the first places to look, if
not
on the website ?
Absolutely, but all license and copyright files in the GPL Asterisk
distribution are pure GPL,
On Sat, 13 Nov 2004 10:59:35 -0700, Kevin P. Fleming
[EMAIL PROTECTED] wrote:
Brian Capouch wrote:
I would like to see you say out loud, just once, that those of us who
know all of that and disclaim our work to Digium are not necessarily
idiotic boobs who don't know what we're doing.
As Joe
On 12/11/2004 16:08 Matteo Brancaleoni said the following:
I too demand sysmaster either pay Digium for a non-gpl license or
publicly admit the fact that they have repackaged Asterisk and
contribute enhancements to Asterisk back to the GPL.
*if they have made any enhancements* :)
actually, the
On 11/11/2004 06:08 Steven Sokol said the following:
The patch is necessary because (I think I have this correct -- forgive
me if I scramble any of the details) the Asterisk SIP channel was not
caching the MD5 result of the original authentication dialog, and was
instead forcing the BroadVoice
Citat Kevin P. Fleming [EMAIL PROTECTED]:
Martin List-Petersen wrote:
Surely, but if you contribute to a project, shouldn't you allways check
the
license ? Would LICENSE, COPYRIGHT or README be the first places to look,
if
not
on the website ?
Absolutely, but all license and
[snip]
Really? Wouldn't it be nice, then, if Digium explicitly stated that this
was their intention, in their little agreements?
Why aren't YOU stating your own intention with this whole thread, or do you
even realize it fully yourself?
Your intent, whether you realize it or not, is to
the asterisk suport NAT as ser?
or need modules from modules or special cofiguration?
_
Do You Yahoo!?
Información de Estados Unidos y América Latina, en Yahoo! Noticias.
Visítanos en http://noticias.espanol.yahoo.com
Anybody else having Broadvoice registration problems today?
--
Gary White [EMAIL PROTECTED]
Network Administrator Internet Pathway
105 D East Church Street Voice: 601-776-3355
P. O. Box 777
yes.. started around 12:00 noon EST
I get "sip_reg_timeout: Registration for '[EMAIL PROTECTED]"
Does anyone know if this is related to the channels patch?
Doug
Gary White (Network Administrator) wrote:
Anybody
else having Broadvoice registration problems today?
After the broadvoice patch I am getting busy messages
also on call in.
Is anyone else experiencing a lot of busy signals after this patch? ie
Broadvoice becomes disassociated with asterisk..
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[EMAIL PROTECTED]
Same here...
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sat, 13 Nov 2004, Doug Shubert wrote:
yes.. started around 12:00 noon EST
I get sip_reg_timeout: Registration for '[EMAIL PROTECTED]
Does anyone know if this is related to the channels patch?
Doug
Its working here, some issues tho. All outbound calls have no CID.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce
Komito
Sent: Saturday, November 13, 2004 1:16 PM
To: Doug Shubert
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Yes. :-(
-jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary White
(Network Administrator)
Sent: Saturday, November 13, 2004 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] BroadVoice
Anybody else
Working fine for me.
I installed their patch like they asked.
I'm registering with proxy.dca.broadvoice.com
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Martin List-Petersen wrote:
No .. the README tells about the Dual License.
Not quite. The README says that Digium can grant others the right to
create modules that link with Asterisk at runtime but are not required
to be licensed under the GPL. It does not say that Digium can grant
others the
all i have is random echo
I have already 4 TA750 with full FXO
echocancel=yes and echo training=800
- what should i do?
- could it be solved with tweaking echo params on *?
- is there any additional devices that can be added between Channel
Bank and * to get rid off echo forever?
depends
Tom Lahti wrote:
Or for some other poster, because he chose to use Asterisk in his
corporation and now _somehow_ he should have some say in how its
developed/maintained just because he's using it. Well you know what,
you can always stop using it if you don't like it, or you never should
have
Well, back working now. Guess they were having problems again.
--
Gary White [EMAIL PROTECTED]
Network Administrator Internet Pathway
105 D East Church Street Voice: 601-776-3355
P. O. Box 777
Citat Joe Greco [EMAIL PROTECTED]:
[SNIP]
There are a number of competing theories on whether or not the author of a
public domain bit of code could be liable, with varying amounts of case
law,
as I understand it:
1) One theory is that you may place code in the public domain, with
By the way this was not related to the patch. I installed it
Friday and did not start having trouble until today.
Well, back working now. Guess they were having problems again.
___
[off-list]
Brian Capouch wrote:
I would like to see you say out loud, just once, that those of us who
know all of that and disclaim our work to Digium are not necessarily
idiotic boobs who don't know what we're doing.
As Joe already pointed out, he doesn't believe this to be the
Please [off-list]
- Original Message -
From: Joe Greco [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 13, 2004 4:10 PM
Subject: Re: [Asterisk-Users] Asterisk dual licensing
[off-list]
Brian Capouch wrote:
I would like to see you say out loud, just once, that those of
Citat Joe Greco [EMAIL PROTECTED]:
If so, you are allowed to fork or distribute your own patches and not sign
any of the disclaimers. This is what Klaus-Peter Junghanns does with the
bristuff patch (adds various addons, including full DSS1 BRI support to
Asterisk, Zaptel and Libpri and
From the * machine, I'm able to fax (txfax) just the first page of a
multi-page document. I've tried this on a Sharp UX-P200 as well as an
HP 5510 machines with the same result. The document was assembled using
gs from a series of .ps documents.
The fax machines are connected to an Adtran
Are you saying that those of us that are using the product should not be
allowed to voice our opinions about its licensing, development and
maintenance? That we should all just shut up and take whatever Mark co.
give us? If that's the case, then this is most definitely NOT an
open-source
[snip]
Really? Wouldn't it be nice, then, if Digium explicitly stated that
this
was their intention, in their little agreements?
Why aren't YOU stating your own intention with this whole thread, or do you
even realize it fully yourself?
I do. I don't believe you do.
Your
Andreas Anderson wrote:
Hi Brian,
One goal is to get BRI support in Zaptel if possible. I'm right now
in the
planning stage :P Plus BRI is much cooler than pots.
Why invent the wheel again, what's wrong with bristuff from junghanns.net?
bye,
aa
On Sat, 2004-11-13 at 01:48 -0800, Dameon D. Welch-Abernathy wrote:
For your testing pleasure. Feedback welcome:
http://voxilla.com/spa3kasterisk.php
-- PhoneBoy
PhoneBoy! What is the point of testing it you don't even know how to
ship whatever you sell internationally and/or resolving the
On Saturday 13 November 2004 17:55, Billy Huddleston wrote:
So, why not use SER to register all the SIP phones, as it doesn't handle
the
media-streams, just keeps track of the phones and does the 'handshake'.
SER is supposed to be able to handle over 50.000 calls at a time, so one
SER
No .. the README tells about the Dual License.
Try reading it as though you didn't know about the dual license.
If you read it without that knowledge, it sounds as though they are trying
to provide a way within the terms of the GPL to allow linking with stuff
that isn't GPL'd. This could be
Are you saying that those of us that are using the product should not be
allowed to voice our opinions about its licensing, development and
maintenance? That we should all just shut up and take whatever Mark co.
give us? If that's the case, then this is most definitely NOT an
open-source
Here's my a section of my simple extensions.conf
exten = s,1,Answer
exten = s,2,playback(thx4call)
exten = s,3,Dial(Zap/1|15) ; Calls channel 1
exten = s,4,playback(trying_bert)
exten = s,5,Dial(Zap/4/2326932|15)
exten = s,6,Voicemail,u100
exten = s,7,hangup
exten = s,104,Voicemail,b100
exten =
On Saturday 13 November 2004 12:19 pm, Martin List-Petersen wrote:
It has been tested in city/county court in Munich (Germany) and found valid
(http://yro.slashdot.org/article.pl?sid=04/07/23/1558219tid=117), not that
that might help anybody in the US, but it is a start.
Kind regards,
Martin
hi
Il dom, 2004-11-14 alle 00:13, DB ha scritto:
Here's my a section of my simple extensions.conf
snip
exten = s,5,Dial(Zap/4/2326932|15)
exten = s,6,Voicemail,u100
snip
It works, but when the call is routed out on ZAP/4 (at priority 5),
Asterisk seems to not realize the call is answered.
On Saturday 13 November 2004 01:57 pm, Walter Willis wrote:
the asterisk suport NAT as ser?
or need modules from modules or special cofiguration?
Hmm, your English is a bit too crippled to understand. I'm guessing you are
asking if Asterisk supports NAT as something (server?)
And then if it
Ok I'm not one to beat around the bush here... so here goes.
This has got to be the most wasted energy I have ever seen. If only half of
you put as much energy into pissing and moaning on the list as we do working
to make asterisk better IT WOULD TEN TIMES BETTER!!! (Not that it isn't
already,
Havent had that problem but look here
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
I would try callprogress=yes
- Original Message -
From: DB [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 13, 2004 6:13 PM
Subject: [Asterisk-Users] Remote answer not detected
Hello
Im trying to have my normal incoming calls automatically forwarded to
my SIP phone, or even better, directly to a given number thorough my SIP
service provider.
Example: Im visiting the office in Argentina or Spain, someone call to
our office in Italy (a 'normal'
Hi all,
I'd like to know what's most reliable configuration for BudgeTone 101 in
the following setup:
PSTN
|
Legacy phones == Alcatel Omnipcx == QuadBRI-Asterisk1
|
| IAX trunk
thank you, my English is terrible, I don't usually use
it and ti is not my language.
unfortunately they don't exist clever of mail in
another language. and I don't have with the one who to
practice it.
XD I go he is necessary to have to practice it but.
thanks you for the you help me.
Walter Willis wrote:
thank you, my English is terrible, I don't usually use
it and ti is not my language.
unfortunately they don't exist clever of mail in
another language. and I don't have with the one who to
practice it.
XD I go he is necessary to have to practice it but.
thanks you for the you
yee
gracias por la ayuda
pero de todos modos tengo que practicarlo.
lo leo lo entiendo pero no lo escribo ni lo
habloo!
XD
_
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Información de Estados Unidos y América Latina, en Yahoo! Noticias.
Visítanos
Hi Michael,
Michael Welter wrote:
From the * machine, I'm able to fax (txfax) just the first page of a
multi-page document. I've tried this on a Sharp UX-P200 as well as an
HP 5510 machines with the same result. The document was assembled
using gs from a series of .ps documents.
The fax
On November 13, 2004 12:11 pm, Malcolm Bader wrote:
This is my second asterisk server but the first one with a T100P card.
I connected it to the phone company(SBC) jack but have only a busy
signal when calling the T1's number and nothing in the asterisk log
files to indicate a connection.
Do
Hello again,
I am reposting this issue since I realized I posted in an existing
thread ( I am sorry about that).
I am still faced with the same problem since long time:
The question is how to correctly handle failed calls.
In my application I want to make hundreds of outgoing calls
Steve Underwood wrote:
Hi Michael,
What happens with 0.0.2pre4? For most people that version gives better
results than 0.0.1k. It seems to fix most of the quirks people have had.
It didn't work with the HP3150. I have a new HP5510, and I'll try
0.0.2pre4 again.
spandsp says we have more
Uh ok...So when will Asterisk be a licensed product? Will it take the
form of a Redhat sort of platform... Fedora with Redhat the pay me money
side of the house?
Just a simple question: When can we expect to see Asterisk the licensed
as in paid for version ?
Brandon
Are you saying that
Are you saying that those of us that are using the product should not be
allowed to voice our opinions about its licensing, development and
maintenance? That we should all just shut up and take whatever Mark
co. give us? If that's the case, then this is most definitely NOT an
if i have two sip contexts for my spa3k, on inbound and
one outbound, e.g.
[spa3k-in]
type=friend
host=dynamic
port=5061
auth=md5
secret=pfui
qualify=1000
canreinvite=yes
context=ext-in42
[spa3k-out]
type=peer
auth=md5
I've got a TDM100P card with a fxo and fxs module in the US. I'm
using kewlstart for all ports. I've noticed that when I make
a call out from an analog phone out the POTS line that if after
talking to the party I called (in this case the phone company itself)
they put me on hold asterisk
Hi all,
I am new to asterisk. I was able, but not without
pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs
softphone to work with the PBX.
Three remarks:
* On the SJphone, i use the GSM and the G711 (ulaw
and alaw) codec. In the h323.conf file i enabled those
Il dom, 2004-11-14 alle 00:13, DB ha scritto:
Here's my a section of my simple extensions.conf
snip
exten = s,5,Dial(Zap/4/2326932|15)
exten = s,6,Voicemail,u100
snip
It works, but when the call is routed out on ZAP/4 (at priority 5),
Asterisk seems to not realize the call is answered. After 15
Guys, in fact we will give an Applause to Sysmaster guys that are
doing a great job in their products (world wide sales), this guys are
doing money , and in this point is ,, Mark (and/or Digium is not
receiving money for that).
But Back to basics of Open source you can sale , modify , distribute
You've got a 50/50 shot.
Try the crossover.
http://www.cisco.com/en/US/products/hw/routers/ps233/products_tech_note0
9186a00800a3f09.shtml#topic2
It would be more helpful for you to send your /etc/zaptel.conf file and
/etc/asterisk/Zapata.conf file.
You should have something like the following
Im not sure about the G711 codec on
the ATA, but I know you need to purchase the g729 from digium.
http://www.digium.com/index.php?menu=asterisk_g729
pretty inexpensive at $10 each.
Thats for concurrent connections to the server.
Tim.
-Original Message-
From:
I am having a few problems with my queue. I am using the AgentCallbackLogin
feature. When the call comes to the
user, it does not announce the call to the agent. It waits until you enter the #.
After you hit #. It will
play the queue-support announcement to the agent and tell them to
Hello,
I know the d-link units (DVG-1120 ATA and their router as well) are
supposed to work well with asterisk...does anyone know if the units
that come with ATT callvantage are locked, or can they be used
w/asterisk or SER? And if they are locked, is it linksys no way out
locking or a simple
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