On Sun, 9 Jul 2006, Andrew D Kirch wrote:
To some extent I see your point and have been on the receiving end of
one of Jeremy's tirades.
I've since decided that NuFone is an interesting study in whether your
business can survive
with only clueful customers.
Some people are into SM I
== Spawn exten
(Charity, ---0059, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
*/SNIP
Im sure I could change everything to ulaw G711 the problem would go away
but I do not want to do that.
Any Ideas?
Thanks
Scott H
--
Joe Baptista
www.joebaptista.com
can anyone recommend a plain ol fashioned telco in north america where i
can get a DID with call hunt and voicemail which can be programmed via web
interface.
thanks and pls pvt
joe
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On Sat, 27 Nov 2004 [EMAIL PROTECTED] wrote:
Does anyone knows a possibility to disable the message of the server
and only able the message of our client?
Example: client says:Im not in my office, please leave a message.
Well, after this message the sever should send the signal and record
Is there alist of codecs asterisk actually has per version number - i.e.
0.7, 0.9 etc?
On 9 Jul 2004, Wolfgang S. Rupprecht wrote:
Instead of all you may want to try listing the codecs asterisk
actually has (this is from -current):
;
; codecs: a_mu adpcm alaw g726 gsm ilbc lpc10 ulaw
;
I'm installing the new Slackware 10.0 distribution - but not sure if i
should go with the 2.4 kernal - which i think is the default install - or
the new 2.6 kernal? anyone running * and slackware 10.0 with 2.6 kernal?
thanks
joe
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-- Forwarded message --
Date: Wed, 07 Jul 2004 00:31:21 -0400
From: Declan McCullagh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Politech] House bill exports analog phone regs to VoIP
http://www.politechbot.com/docs/boucher.voip.bill.070604.pdf
There's a new bill in the
There has been a sudden increase in VoIP pro and con articles. Lots of
political discussion. I think we are seeing the first major PR attempts
by Telcos to stem the VoIP tide.
regards
joe
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On Tue, 6 Jul 2004, Kevin Walsh wrote:
People are entitled to ask questions; If no questions were asked then
this mail list would not have the volume of articles that it has.
Absolutly correct - except for Randy who has a tendancy of starting
arguments over irrelevant trivia.
My own concern
On Mon, 5 Jul 2004, Randy Bush wrote:
i did not criticize the protocol. remember, my question started
with
i am looking at iax to see if it is applicable to my needs.
i don't need nats, nat traversal, nat anything. if i did, iax
might well be one of the technologies i would consider.
On Tue, 29 Jun 2004, Jay Milk wrote:
Like I said, they just seem to be lazy and/or badly organized. If they
can do LNP, why can't they change a hardline into a softphone, break
one number out onto a different ATA, etc? I basically laid it out for
them, saying If you can't move my 2nd line
I need a provider of DIDs with multiple inbounds.
regards
joe baptista
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On Wed, 23 Jun 2004, Stefan de Konink wrote:
Hi All,
Since 21 june skype is available to be used on Linux, with a static
binary, which includes QT, of 8 meg its big.
http://www.skype.com/help_linux_faq.html
I presume, with some hacking, there could be a possibility to use the
Skype
where traditional 911 calls go.
Does anyone know how I can get information on howto contact the people at
the Public Safety Answering Points (PSAPs)? Is there alist somewhere I
can reference.
thanks
joe baptista
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On Sat, 12 Jun 2004, Jeremy McNamara wrote:
Joe Baptista wrote:
i.e.
iax1---+
iax2---|
iax3---|-- * -- MeetMe -- shoutcast radio
---|
iaxn---+
This already exists today in Asterisk: show application ICES
excellent - but i get this when
frankenstein and for good reason - to experiment.
just an idea to the * community.
regards
joe baptista
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.
regards
joe baptista
www.joebaptista.com
www.baptista.god
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I have the same situation - i.e. three different extensions scattered
about. But I don't try them each individually. When a call comes in my
asterisk attempts to ring up to four different devices at the same time.
To do this using your dial plan is easy - i.e.
exten =
I've been watching to see if this problem comes up with anyone elses
firefly - but so far i'm the only one experiencing the problem.
When I connect to either my asterisk server or FWD all goes well on the
first call. I can hear and talk. But every call after the first one I
end up with no
: QUERY, status: NOERROR, id: 44741
;; flags: qr rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 9, ADDITIONAL: 4
;; QUESTION SECTION:
;0.1.3.1.2.7.8.5.0.7.1.e164.org.IN ANY
;; ANSWER SECTION:
0.1.3.1.2.7.8.5.0.7.1.e164.org. 600 IN TXT Joe Baptista
0.1.3.1.2.7.8.5.0.7.1.e164.org. 600 IN NAPTR
On Sat, 1 May 2004, Dean Collins wrote:
Yes but no information about how this will operate, what regulation or
restrictions on joining, what connection protocols will be used etc etc
agreed - you see alot of business fluff - but the technicals are very
important and on many of these ventures
on my asterisk - but FWD only see me as an external
SIP agent and not a SIP client of the FWD network. DOn't know exactly why
- so would luv to compare your conf files.
thanks
joe
Joe Baptista: USG Portal www.joebaptista.com, Personal www.baptista.god
LOW: low cost, Low Lands. http
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