Hi everyone
I just ordered a Sangoma A20001 with 2FXO ports - Does anyone have
suggested reading pointers for what I'll need to do to get it
working? I've only used VoIP in the past so don't know much about
Sangoma drivers or Zaptel. I opted for the non-echo canceling card
so I may need
While I don't see anything wrong with this, I'm no expert. I took my
instructions from the following URL and they worked fine... I have
the subscribecontext in General and it works fine. What is the
firmware on the GXP? old firmware may be related
-t-
Hello everyone! I'm planning on setting up a new system shortly and
can't pick the right card... We will have 2 or 3 lines coming in and
7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do
I need the Sangoma A20200 or even the A20200D (Echo cancelation)...
I was
The card will let you interface with a regular telephone line instead
of VoIP. If you want to use a regular phone instead of the computer
softphones, look into the Grandstream handytone devices - they'll
make it so your regular telephones can talk to Asterisk. You can
make the system
Speaking of the X100P, I am going to setup an asterisk server next
week for a friend's business to replace his aging system. He
currently has two voice lines and another line for the fax machine.
I was looking at the Sangoma A20200D but that's pretty expensive...
We're going to use
As Paul Hales said, I doubt that modem is supported. To interface
with a regular phone line you'll need to get a supported card. You
can read about it online. To just get started with playing, I
recommend you go ahead with the sophistocated VoIP stuff.. Perhaps
sign up with IPKALL or
short version: me too
long version: The same thing happens on my asterisk boxes - both
built with the latest trixbox image... perhaps that's a factor? My
history is always restart now, although I typically connect and run
sip show peers. I haven't typed restart now in a long time, but
I've been looking for this as well.. I need to support up to 20 VOIP phones over Internet as the Asterisk server is off-site. We'll have multiple cable modems or DSL routers. I found this device which looks promising - does anyone have any experience with this?
Thanks- they did respond. I got a new template, but was asked to not
share it for now - it'll be on their website in a few days pending
committee approval
thanks
Todd
On Nov 14, 2006, at 12:50 PM, Gordon Henderson wrote:
On Fri, 10 Nov 2006, Todd- Asterisk wrote:
I'm preparing
I see the Grandstream website now has the new config templates posted
with all the happy P commands...
http://grandstream.com/y-configurationtool.htm
Todd
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To
Perhaps you want to look at
http://tanesha.net/Wiki/GratissipTftpd.html
You can keep the P-codes in a mysql database and build all the
configs you want. For me, it was a little too much work for the few
phones I have, but if you need more.
Todd
Is there a quicker way to
I'm preparing to deploy a small number of Grandstream BT101's and
GXP2000's to a remote location (which I won't have access to). I'd
like to have them pull a config file from my server - I'm almost
there...
The phones are looking for the config file on my webserver which is
good. I
- this is a great list. ToddOn Nov 6, 2006, at 10:28 AM, Todd- Asterisk wrote:I've got my asterisk server in the DMZ of my local LAN - I've used my Budgetone and GXP2000's from the Internet- on direct IP connections with no problems. However, I'm about to deploy about 5 phones (either budgetone
I've got my asterisk server in the DMZ of my local LAN - I've used my
Budgetone and GXP2000's from the Internet- on direct IP connections
with no problems. However, I'm about to deploy about 5 phones
(either budgetone or GXP2000's) all on a LAN behind a NAT- on a
different network than
I have the Budgetone 101 and GXP2000 and thought the sound quality
was excellent. Even over the internet... I agree with Joe that
something else may be the factor...
Todd
Zeeshan Zakaria wrote:
Hi all,
I have to buy some IP phones. Previously I have used Grandstream
GXP-2000,
This might be a newbie question... I'm using a SIP trunk and trying
to get DID line information on an incoming call. All I hear is a
nice lady saying 'Zero' - then the call continues... Any suggestions?
thanks
Todd
exten = s,n,Set(DIDID=(${FROM_DID}))
exten = s,n,SayNumber(DIDID)
Thanks for the help Jerry - I'm getting closer, but still no luck...
Now, I hear the lady say S. I think what is happening is that the
GoTo command is setting the extension to 's' when it transfers
control to the context defined in the IAX.conf -where I have the
trunk line defined...
Hi Eric- It wasn't typo, it was "truncated for posting" :) Below are the complete relevant files. I'm getting 'S' when I want to hear the DID number.. This machine was a trixbox about a two weeks ago, but I've since tossed away the GUI and do everything by hand now. I just used the trixbox
When dialing from internal extension, it gives me the number I dialed
( in my case..). When dialing from AIX or SIP from didww.com
nothing comes through on the SayAlpha thanks for the thought
though... What I want is the number that the user dials to get my
serverI'm
I'm a Certified Apple Sys Admin - lots of experience with Macs and
Mac servers. However, when setting up an asterisk server, I'm still
thinking a Dell box with linux is the best direction - to get the
full reliability and full support of this group. Am I mistaken? Or
is using a Mac box
I'm setting up an asterisk server where an administrator will not
always be available in case of problems. While I expect problems to
be rare, I need to be prepared. We're thinking of VoIP DID's and SIP
phones so it's an all TCP/IP network. We could get a second server
to substitute -
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