ul 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 i
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 th
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 fo
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.
St
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 pretty
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
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 a
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 eBa
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,
On Thu, 2003-08-07 at 13:27, Andy Powell wrote:
> I have to say that some listserv's do allow this .. at least
> he didn't reply to 20 messages with
>
> REMOVE
>
> in them
True. I've seen that. I guess I'm just not on the right lists. :)
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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 ou
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 configu
Please disregard my last email, I didn't read it all the way through...
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On Wed, 2003-08-06 at 13:39, John Schmerold wrote:
> I've canceled my Vonage service because of the requirement to prefix
> every call with a 1.
You could set up some simple extension rules in Asterisk that will
prefix 1 plus your area code on any numbers that don't start with a 1.
Just a though
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 trading/se
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
On Wed, 2003-08-20 at 14:20, Mike Ciholas wrote:
> Are there VoIP dialtone providers? That is, could I use only my
> internet connection for voice calls and not have a separate
> T1/POTS bank for that?
packet8.com
nufone.com
iconnecthere.com
I haven't tried any of them yet. I'm considering se
On Thu, 2003-08-21 at 00:32, Brian Capouch wrote:
> I have seen two references today (don't recall whether here or on one of
> the other VoIP lists I read) to people having the .78 version of the
> firmware installed on their phones.
>
> I'm keen on getting hold of it, but their support page sti
On Thu, 2003-08-21 at 02:01, Jamie Carl wrote:
> This IS a new thread 'bonehead'.
Actually, it wasn't. He's correct in his complaint, however rude it may
have been. It appears that you simply replied to a message and deleted
the subject and body, instead of starting a new message. Email clients
On Thu, 2003-08-21 at 08:33, Steven Critchfield wrote:
> BTW, what size is size=3D2 ? It seems to be in all HTML email from
> Microsoft products.
http://www.ietf.org/rfc/rfc2047.txt
It's part of the MIME spec for encoding non-ascii text. =3d is =, I
believe, so size=3D2 is actually size=2.
Any
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 t
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).
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 Wed, 2003-10-01 at 07:46, Steven Critchfield wrote:
> Or if you wish to write closed source apps, you purchase licenses.
> Nothing in life is free. Open source software has strings attached, only
> these strings don't go to your wallet.
I've never heard it expressed so beautifully.
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 st
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 anoth
On Tue, 2003-10-14 at 19:31, Gene Kochanowsky wrote:
> Does anyone know if or when the FXO daughter boards for the TDM400P will be
> available?
September 2003. :)
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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 scr
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, apparentl
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.
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.
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 agai
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.
>
> Th
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 wi
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