-- Forwarded message --
From: Celia Einhorn
Date: Wed, Feb 17, 2010 at 8:15 PM
Subject: Fwd: Erika DeBenedictis-Recommendation
To: drew einhorn
-- Forwarded message --
From: David H. Kratzer
Date: Tue, Feb 16, 2010 at 9:24 AM
Subject: Fwd: Erika DeBenedictis
oops, sent to wrong address by mistake!
Sorry!
On Mon, Mar 1, 2010 at 9:00 AM, drew einhorn wrote:
> -- Forwarded message --
> From: Celia Einhorn
> Date: Wed, Feb 17, 2010 at 8:15 PM
> Subject: Fwd: Erika DeBenedictis-Recommendation
> T
space.
DD-WRT has a good description of the hardware in various models.
See:
https://www.dd-wrt.com/wiki/index.php/Supported_Devices#Linksys_.28all_the_rest_that_is_not_re-engineered_til_today.29
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Drew Einhorn
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on a new server box.
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> So an out-of-the-box thing would be better, but I was recommende the pfsense
> before and will take a look at it.
>
> Mike
>
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.
tag the voip packets.
Need other hardware to actually manage the queues.
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k ignore that false answer and allow the other
lines to continue simultaneously ringing until we get a real answer, or it
goes to voicemail?
> Gordon
>
>
>>
>> That would work for you and still give callers the audible ,essage they want.
>>
>> Steve
>>
>> On 3/20/09
re illegally war dialing cell phone numbers, I don't think they listen
>>>
>> for
>>
>>> the Special Information Tones.
>>>
>>>
>>>
>>
>>
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>&g
implement this. How difficult would it be to configure asterisk to
handle this as I suggest?
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On Thu, Mar 19, 2009 at 5:39 PM, Phil Reynolds
wrote:
> Quoting "drew einhorn" :
>
>> This sort of message is usually preceded by some magic tones that
>> allow direct marketing application to immediately drop a call to a
>> dead phone number.
>>
>>
This sort of message is usually preceded by some magic tones that
allow direct marketing application to immediately drop a call to a
dead phone number.
What is the proper terminology for the tones?
Where can I find information about how this is implemented?
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Drew Einhorn
2009 at 9:20 PM, Tilghman Lesher
wrote:
> On Monday 16 March 2009 21:49:53 drew einhorn wrote:
>>
>
>
> What does this have to do with Asterisk?
>
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> Tilghman
>
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iginal Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of drew einhorn
> Sent: Monday, March 16, 2009 8:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Problem with Verizon
b3R0bGVzGgjBPl5rQmDuSygBUhMI8aH_lPOomQIVBE6DCh2MHDrk&hl=en&kw0=BPA+Free+Water+Bottles&kw1=Plastic+Water+Bottles+7&kw2=Bisphenol+a+Bottles&kw3=Plastic+Sippy+Cups&okw=BPA+Free+Water+Bottles
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etwork, they pick up the call and answer with a
voice error message (sometimes after only one ring), before anybody
has a chance to answer on a sip device.
Or, am I misunderstanding you suggestion.
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Maybe just one PSTN line per DSL connection to avoid paying a sip provider to
terminate some local calls, and supporting some backup functionality, if the
Asterix box has crashed, but it will be a while before things get that
complicated.
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> From: asterisk-use
the cell lines are busy they generate busy
signal.
I need to know the right incantation to use with
Verizon to get them to just let the cell lines
ring until either some picks up a voip line,
or the voip voicemail picks up the call.
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Drew Einhorn
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ontentID=17633
These kids were wondering why they were having trouble getting the
expected performance from their model, and discovered it was a
limitation of their multi core processor.
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inal mainstream version of the firmware that
I should be running on the ATA186. And where would I get it.
I'm assuming I can't get it from Cisco anymore.
Any known, serious issues, that were never addressed
because they were discovered after the box was past its
end
The ATA generates it's own dialtone, and waits for
the user to dial a number, before sending anything
to the * box. So one of the first examples in the
in the Brief Introduction to Dialplans from
Vol. 1 of the Asterisk Documentation Project.
[incoming]
exten => s,1,Answer()
>>>After dialing the extension it takes about 10 sec
>>>for the other phone to ring. A little more the
>>>Cisco to ATCOM, a little less in the other direction.
>>Look at the CLI when you dial a number. How does the dialplan look? It
>>sounds like asterisk is waitng for more digits.
>Or maybe the
>> Has anyone succeeded in connecting to iaxtel.com
>> with a low bandwidth codec? So far all I've
>> been able to connect with is adpcm, and it
>> overwhelms my isdn line. breaks up, stops, starts,
>> plays for a while, etc.
>
>Haven't tried it lately, but iaxtel has been mostly unusable
>for the
My first asterisk box is up an running!
I have 2 ATAs on my local net.
One is a Cisco ATA-186 with SIP 3.1.0 software.
The other is an ATCOM AG-168V with V1.42.028 IAX2
firmware.
After dialing the extension it takes about 10 sec
for the other phone to ring. A little more the
Cisco to ATCOM, a li
Has anyone succeeded in connecting to iaxtel.com
with a low bandwidth codec? So far all I've
been able to connect with is adpcm, and it
overwhelms my isdn line. breaks up, stops, starts,
plays for a while, etc.
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