in
spoofing Vonage into believing your Asterisk server was one of their ATA
186's? If I could do that, we would probably switch our phone lines
over to Vonage.
Thanks!
Steve Meyers
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On Thu, 2003-07-31 at 10:07, Ricardo Villa wrote:
There is no way for you to know the vonage password associated with your
account. Even if you sniff out the tftp download, its encrypted.
Is there any comparable service that isn't as anal? Or even better, is
there any service that uses IAX
On Thu, 2003-07-31 at 10:25, nathan wrote:
Iconnecthere (www.iconnecthere.com) works without any problems here,
even behind NAT.
I looked into them, but there are a couple of problems with them.
First, they don't seem to have numbers in my area. They have my area
code, but only for a city
On Thu, 2003-07-31 at 10:20, Humberto Atristain wrote:
8x8 is the only one I know (or packet8) a little less important
What specific information do I need to get from them in order to get
Asterisk to connect directly? I assume I'll need the following:
* SIP id
* SIP password
I just found this link:
http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat
It suggests that your username is your phone number, and your password
is the 10 digit activation number.
Steve
On Thu, 2003-07-31 at 15:23, Joe Cooke wrote:
I haven't tried it yet, but I believe the
Our office is set up with Budgetones internally. Occasionally, someone
will be on the phone, and their phone will ring. How can I make it so
that it will go straight to voicemail?
Thanks!
Steve
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On Fri, 2003-08-01 at 13:50, Dan wrote:
I think that you must disable Call Waiting functionality.
I can't find where to disable it... I set callwaiting=no in zapata.conf
and sip.conf, but neither seemed to help. I grepped for callwaiting in
/etc/asterisk and couldn't find anything helpful.
At least any way I've tried. I put callwaiting = no in sip.conf in
the [general] section and in the section for my specific phone, and it
still sends through calls even though I'm already on the line.
How can I disable it?
Steve
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On Mon, 2003-08-04 at 14:31, Brian West wrote:
What type of phones?
Grandstream BudgeTones. Is it a function of the phones? Is there any
way to limit them in sip.conf to one channel each?
Steve
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Where can I find a good tutorial on how channel banks work? I need to
get a 6 port (or so) channel bank for FXO. I need to find some
information on which ones are supported well under Linux and with
Asterisk, how to configure them, what specifically to look for in a
channel bank, etc. I'm
On Thu, 2003-08-07 at 01:56, Brian Capouch wrote:
2. This phone does not act like all my others do when I am talking and a
call comes in. Instead of the jarring ADSI !!!BOING!!! followed by a
series of call waiting beeps, instead I get a ringing tone in the
earpiece which is audible to the
On Thu, 2003-08-07 at 10:01, Justin Carlson wrote:
unsubscribe
Has anyone ever been on a mailing list where you could unsubscribe
simply by sending a message with unsubscribe in it to the mailing
list? I swear, every list I've been on, people try to do that, but it
doesn't work on any of them.
On Wed, 2003-08-06 at 16:20, Andy Powell wrote:
It's just a proxy service like fwd it will work with asterisk... The phones they are
selling
with the deal are Grandstreams.
Perhaps that explains why nobody can get to the site to order
Grandstreams right now. :)
On Sun, 2003-08-10 at 21:31, Steven Critchfield wrote:
On Sun, 2003-08-10 at 21:25, Andy Hester wrote:
Perhaps there is another way to cut down on increased traffic...
Specifically, I would go back to the suggestion of a collaborative website
for documentation. Collecting info and
On Tue, 2003-08-12 at 11:45, WipeOut . wrote:
The Cisco is from what I have heard a good phone but is VERY expenisve..
My suggestions would be to go with either a SNOM 200 or a Grandstream Bugetone..
Where can one get a SNOM 200 for less than a Cisco 7960? The Cisco's
are about $300 on eBay
On Wed, 2003-08-13 at 09:46, Dave Cotton wrote:
I've had a few problems with my system holding the line after a call has
been made, just now I rebooted and noticed the following in
/var/log/messages
When you say holding the line, do you mean that asterisk still
believes a channel is in use
On Wed, 2003-08-13 at 11:13, Emmanuel Bergmans wrote:
In order to test CTR21, I was forced to comment the line in the source file as I did
not find a define or a
zaptel.conf directive. It's really bad but... In my case this change has not solved
the problem (see previous
posting)
Well, I'm
On Mon, 2003-08-11 at 10:19, Jim Friedeck wrote:
Our CSR people need to be informed when a call is ringing in when they
are on the phone. Is there a mechanism for informing an off-hook target
channel of an incoming call? We have a guy who should get first shot at
all incoming calls on our
On Fri, 2003-08-08 at 12:25, Steven Critchfield wrote:
With the increased traffic as of late, I'm wondering if it is time to
split the list again. Specifically I am wondering if it should be split
along the various VoIP protocols and zap hardware, then leave a general
list that does
On Sun, 2003-08-17 at 17:55, Nathan wrote:
Does anyone have any recommendations for a cordless phone that uses SIP
(or IAX)? It doesn't have to use 802.11b, but that would be appreciated.
I think you're only solution is going to be the Cisco ATA-186, an
analog-to-SIP device. Or, you could use
On Wed, 2003-08-20 at 07:58, Mark Spencer wrote:
The FXO ports will only allow you to connect phone lines, not actual
phones, but since FXO ports are more expensive in general than FXS ones,
it's likely you could find someone to trade. We probably should have a
list dedicated to
On Wed, 2003-08-20 at 11:09, Ian Blenke wrote:
Brian West wrote:
I would use the latest CVS for one. And try again.
Unfortunately, I've tried numerous times to get a current CVS trunk
snapshot to talk to *anything*, to no avail. Even getting my Grandstream
phones to register with it was
On Fri, 2003-08-29 at 23:27, Lubomir Christov wrote:
we made available this patch few weeks ago:
http://lists.digium.com/pipermail/asterisk-dev/2003-July/001202.html
Any chance of this making it into the main source?
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On Tue, 2003-09-09 at 11:41, Hielke Christian Braun wrote:
I have one problem with the BudgeTone phones and early dial. When i
dial a long external number with 9+, * starts to dial to early with
just a few digits. The outgoing call is placed through the SIP provider
Nikotel. Is there some
On Tue, 2003-09-09 at 16:03, Tilghman Lesher wrote:
Why not just use DISA:
exten = 9,1,DISA(no-password|outgoing)
Because I didn't know about it. :) I'll try it out.
Steve
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On Wed, 2003-09-10 at 21:06, Tilghman Lesher wrote:
Odd, I've found CVS-current to be extremely stable, so I run it on all
of our production machines. No machine is ever more than a couple
weeks out of sync with CVS (except for a few machines in the field
which I can't get to right now).
The
On Wed, 2003-09-10 at 20:19, Anderson Clayton wrote:
Where can i find a instalation guide for asterisk? is there anyone?
This is about the best you'll get:
http://www.digium.com/handbook-draft.pdf
http://www.wwworks-inc.com/asterisk/ also has some links.
Steve
P.S. Anyone want to take bets on
On Mon, 2003-09-29 at 13:23, Jeff Dodge wrote:
So -- If you don't distribute the compiled app to me -- I have no right to
ask you for the source. Even if I pay
you for your custom application and you must provide me with the source
(Upon request!) I have no redistribution rights
to that
On Thu, 2003-10-02 at 12:04, Jan Rychter wrote:
I'm also hearing this, with an analog phone (connected to an
S100U). Rather annoying.
Incoming calls have an entirely different problem for me, a disastrous
5-8 second crackling/clicking sound, which seems to go quiet a while
after you start
On Thu, 2003-10-02 at 07:51, Josh Roberson wrote:
Ok, see, now you're confusing what I said. Nowhere did I say I had the
102D. I said he never mentioned that it was the 102, irregardless of
the D. I *DO* have the 101, which is what he was talking about. No, it
doesn't mention it's the
On Sat, 2003-10-04 at 15:09, Jan Rychter wrote:
Any chance you could describe the hardware? Was it a Via-based board?
I have a setup where I use two *'s, both on Via boards. One is a
Mini-ITX and the other is a full-form motherboard.
Would interrupt-sharing between the X100P and another
On Tue, 2003-10-21 at 11:36, James Sizemore wrote:
9Fix the tftp configs so that I can host my own provisioning server.
Or make a command prompt based tool kit, so that I can use
Gaps with out writing a http screen scraper.
So I'm not the only one who wrote an http screen
On Mon, 2003-10-20 at 20:38, John Brown (CV) wrote:
So please rate your ideas on a scale of 1-10
10 - Fix the TCP/IP stack. The phones don't work with certain switches
(i.e. the one at my house), and occasionally do other weird things
(although they fixed the MAC address takeover bug,
On Wed, 2003-10-22 at 07:44, Andrew Kohlsmith wrote:
Can you _please_ trim the quoted text? There's absolutely no reason to
quote the entire post you're replying to, signature lines and all... +2
points for bottom-posting though. :-)
No, -10 points for bottom-posting but not trimming. If
On Tue, 2003-10-21 at 17:13, John Brown (CV) wrote:
Can you provide more specific information. Saying Its Broke Jim
doesn't provide enough content :)
True that. :) My biggest complaint was how they used to sometimes take
over the server's MAC address, confusing the crap out of my switch. We
I recompiled Asterisk with the aggressive echo cancellation on. That's
all I changed, honest. After recompiling, it refused to run. I tried
updating the source, etc, and eventually went back to no echo
cancellation. Every time, I got this error while starting Asterisk.
Please help! I have no
On Sat, 2003-10-25 at 18:49, Ken Godee wrote:
You did do a make clean first before recompiling?
Yes. Not only that, I tried deleting the zaptel, libpri, and asterisk
directories and re-checking them out.
Then I decided it might be a heat issue, so I turned it off for 6 hours
before trying
On Sun, 2003-10-26 at 08:41, Steve Meyers wrote:
On Sat, 2003-10-25 at 18:49, Ken Godee wrote:
You did do a make clean first before recompiling?
Yes. Not only that, I tried deleting the zaptel, libpri, and asterisk
directories and re-checking them out.
Then I decided it might be a heat
I have an order for an SPA-2000 through them, and they won't respond to
any email I send them. I've also tried calling them, but I can never
get a human. I've left voice messages, but they haven't responded.
Does anyone know any other way I can get in contact with them?
Thanks!
Steve
On Wed, 2003-12-03 at 03:26, Aaron Martin wrote:
Sorry to everyone on the list, but for some reason this is the only
reliable way to get hold of John.
John Brown of Chagres Technologies, please contact me! I have been
trying for weeks now to get hold of you via email and phone after wire
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