On Wed, 23 Mar 2005, GP wrote:
I've read that someone was able to do it by contacting vonage and getting
instructions for clearing the router of the vonage information. Does anyone
have the instructions for completing this or is this something that only the
vonage people can provide.
I've
On Mon, 14 Mar 2005, nik martin wrote:
Matt Riddell wrote:
Hmmm...I've had 2 problem with my NuFone service in the year or more
I've used them. Each time I've treated them professionally when
reporting the issue and received the same treatment in return. The
issues were also resolved
On Sun, 13 Mar 2005, Jess Coburn wrote:
So you basically want an SMS or IM callback app right?
One way to do this would be send an email to an address like
([EMAIL PROTECTED]) and have a cronjob query/pop this email
address for your specific message and then when it finds it have it
create
On Fri, 11 Mar 2005, Frank Abernathy wrote:
I am new to the mailing list, but I am very interested in running my small
home business office phone system using Asterisk. However, Broadvoice, a
VoIP provider of choice based on my research, is not available in my area.
I currently use Vonage
On Wed, 9 Mar 2005, Michael Graves wrote:
On Wed, 09 Mar 2005 12:54:34 +0400, Jean-Michel Hiver wrote:
Leo Ann Boon wrote:
Another question... Are you aware of a SIP ATA or phone that has some
kind of VPN (i.e. PPTP) client embedded in? This would make the NAT
problem go away
On Fri, 4 Mar 2005, Randy Johnson wrote:
I set up an asterisk box with a broadvoice sip connection for incoming
connections
it works great when I use a cell phone, vonage line, calling card to
call the asterisk box, but when I try to call it from our verizon land
line it is busy and
On Sun, 27 Feb 2005, C. Tomlinson wrote:
I have just setup a DISA setup whereby people can dial in, authenticate, are
given a dialtone and can then call out.
Everything works however there is a 10 second delay after the user enters
the number and presses #, until the system does anything.
On Fri, 25 Feb 2005, James Taylor wrote:
I have two Broadvoice lines and there's three people in the office.
Any way to:
1) Pool the connections for trunking, where any one can get a free
line?
2) Prevent more than 1 simultaneous call per line? (So I will not get
hit for 3.9 cents a
On Thu, 24 Feb 2005, John Bohman wrote:
Is there any way to hard code an ip in the IAXY then re-direct that ip
to a dynamic ip via dns.. Some sort of ip forewarding...??
I'm don't think I understand exactly what you're trying to accomplish.
Maybe a good place to start would be the DNS howto
On Thu, 17 Feb 2005, Oswaldo Arratia wrote:
Has anyone figured out how to make a Sipura to dial an extension
automatically as soon as you pick the the handset?
I need to make all my users go thorugh a menu to place a call. Users should
not be able to dial directly, only through the menu.
On Tue, 15 Feb 2005, Max Clark wrote:
I have experimented with several configs based on different pages and
threads but nothing is working. How do I properly configure my
broadvoice account?
[general]
register = [EMAIL PROTECTED]:pass:[EMAIL PROTECTED]
the register I'm using looks like
On Sun, 13 Feb 2005, Malcolm Taylor wrote:
I'd be grateful if someone could point me in the right direction.
I have a Broadvoice trunk attached to Asterisk which I use for frequent
calls to the UK using the following in extensions.conf
exten = _0[1-68].,1,Ringing
exten =
On Tue, 1 Feb 2005, Randall Shimizu wrote:
I keep recieving multiple digests per day. Need to find out if there is
a way to limit the number digests that are being sent to me. Tried
contacting the list administrator, but I have not recieved a response.
Does anyone have a alternate email for
Did you try BroadVoice's site yet? www.broadvoice.com, click Support,
click Installation, click Asterisk, follow instructions there.
Greg
On Tue, 11 Jan 2005, Vitalie Apostu wrote:
Following links says: HTTP 404 - File not found . Is it a right link
I had a little billing problem once.. my credit card on file had expired,
and it took a bit of work to finally find somebody there who could get
their system to acknowledge my updated card number and quit sending me
pay up or be disconnected!! warnings.
My recommendation: call them on the phone.
On Thu, 6 Jan 2005, John Voss wrote:
Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this
posting.
http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html
I've tried copying the config in this listing with no success.
One thing that I have
On Fri, 31 Dec 2004, Adi Linden wrote:
BroadVoice sells a wireless SIP phone for $149. Does this phone as sold
by BroadVoice work with Asterisk or is it a locked down device like the
Vonages ATA186?
You'd probably have to ask them that. Just so you know, you can buy that
phone elsewhere. It
On Sat, 25 Dec 2004, Ronald Wiplinger wrote:
To complete my project, I would like to setup DIDs in several areas.
What do I need to do that? Another Asterisk box or can I use gateways
instead? Which hardware can I use? Who has experience?
You either set up your own points of presence, or buy
On Sat, 18 Dec 2004, Anders F Eriksson wrote:
I've never tried softphones on Linux, but my guess is that since you run
kphone and asterisk on the same server you get a port conflict. If the
client uses port 5060 (default sip port) it would defenitely have
problem connecting to an asterisk on
On Sat, 18 Dec 2004, Doug Langley wrote:
I see from reading the mailing list theres a way to set audio levels on
the zap channels but I'm wondering if there's a way to set audio levels
on either sip or iax channels. I'm using some BT-100's and people are
saying the audio levels are a little
On Fri, 17 Dec 2004, Joe Greco wrote:
Joe Greco schrieb:
Don't forget, you ought to have a conventional phone line for E911
purposes, including what happens when a hurricane goes through and my ISP
becomes toast. VoIP is a neat technology but it lacks the resiliency of
the
On Fri, 17 Dec 2004, Michael Graves wrote:
On Fri, 17 Dec 2004 09:42:28 -0600 (CST), Joe Greco wrote:
www.Covad.com
I have their TeleSoho dedicated loop DSL. It costs the same as the
bundled loop.
ADSL or SDSL? (I haven't looked at Covad's pricey offerings for a while)
ADSL
On Fri, 17 Dec 2004, Patrick Campbell wrote:
I don't have a great grasp as to what Asterick is capable of, but my
thoughts were that perhaps with VoIP telephone lines (either hooked up
to the company's network or just using a 3rd party VoIP provider such as
Packet8, which is whatI have for
On Fri, 17 Dec 2004, Patrick Campbell wrote:
Come to think of it since the DTA310 uses DNS to find the SIP server,
you could setup a DNS cache and override the DNS entry for what packet8
uses (proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP
of your own SIP server? Kind of
On Fri, 17 Dec 2004, Brent Goran wrote:
We have an application which is primarily DTMF driven (automated on both
sides), which we are trying to deploy over VOIP and Asterisk (using some
Sipuras and some IAXY's).
We are finding that in around half the cases, the Asterisk server can't
decode
On Fri, 17 Dec 2004, Jon Bebeau wrote:
HI all - I know, slightly off list, but.. I'm looking for a NPA NXX
database with City and State. Actually it's for an Asterisk routing app
I'm working on. I see several vendors that want a few bucks to those
that want an arm and leg. I expect this is
On Fri, 26 Nov 2004 [EMAIL PROTECTED] wrote:
*sigh*
Ok, I have fought and fought with this. I have read all the FAQ's, I have
scanned the list archives. I can receive calls on * from my Broadvoice
acct, but I cannot place calls...
I have the 'World Unlimited' plan, and like 5 area codes
On Fri, 12 Nov 2004, Paul Fielding wrote:
Hmmm... Interesting that you mention it's not a problem with VOIP
companies as they use PRI. The analog trunk I'm connecting to is
actually a Vonage line. Thing is, it seems to me that by connecting via
the Zap channel to the Vonage ATA I'm
On Wed, 10 Nov 2004, Stanley Cline wrote:
Has anyone else had issues with Asterisk rejecting calls from X-Lite
softphones when the dialed number contains the * or # keys (e.g., dial #86 on
X-Lite keypad and then press send, and Asterisk rejects the call with a
404 error)?
It turns out that
On Wed, 10 Nov 2004, Nathan Bowyer wrote:
I have a problem which I've found quite strange, to say the least. I
have a client who uses long distance access codes from their LD
provider. The codes are 4-digits, nothing extraordinary there. The
problem is, if you dial the digits quickly,
On Thu, 4 Nov 2004, Luciano Macedo Rodrigues wrote:
That's problaby a easy question to solve but I couldn't figure out how to do
what I need.
My PSTN line is connected to a phone and a FXO card. What I need is when
someone calls me, and I don't answer in 3 or 4 rings, * makes a VoIP call to
On Wed, 27 Oct 2004, Steve Totaro wrote:
I have had several experiences where certain providers or COs could not
call other providers. When dialed I would get a fastbusy or similar
message were sorry, this number
I have just realized that this is the case with Voicepulse. Many
different
On Sat, 23 Oct 2004, Terry Evans wrote:
I just signed up for the BroadVoice service a few hours ago, but for
the life of me I can't get any incoming voice. The incoming
connection is fine as it rings my extension from outside, but I can't
hear anyone talking. Outgoing voice is working fine
On Sat, 23 Oct 2004, Tim Jackson wrote:
We just got setup with Broadvoice yesterday for LD. This isn't something
I REALLY need (No local numbers avail so we just got a Houston number),
but I'm just curious. I can make outbound calls to Broadvoice and they
work great, but I can't do inbound. I
On Sun, 10 Oct 2004, Rajeev Sharma wrote:
Yeah, thanks, I was thinking of doing something similar to that.
Actually, I was gonna spice a cable in my computer's power supply and
use that. Why? Because if it's on a UPS, then the switch will throw at
the same time as the computer looses off. I
to their hardware,
or bogus soft-phone ripoff. Broadvoice has unlimited plans that work with
Asterisk. I have it, it works great.
Digium rules, and DOES deserve everybody's support.
On Sun, 10 Oct 2004, Wolf Paul wrote:
Greg Hill [EMAIL PROTECTED] wrote:
You probably mentioned
On Sat, 18 Sep 2004, vrushank wrote:
thanx andrew
first of all
your messages are in Plain Text format!
plain text format is the preferred format for this (and most?) mailing
lists. Replying to a digest and not trimming the unrelated portion before
posting isn't a good way to earn points
On Wed, 15 Sep 2004, Marconi Rivello wrote:
On 15 Sep 2004 06:26:29 -, Murali [EMAIL PROTECTED] wrote:
Hi friends,
I tried to dial 111 from CLI without any hard/soft phones.
Well, the ellegant solution is to disable OSS/ALSA and use a softphone :)
I suggest SJphone if you want a
On Tue, 14 Sep 2004, Evert Meulie wrote:
Hi all!
I'm trying to have asterisk route all outgoing calls out via my VOIP
provider.
exten = _NXXX,1,Dial,SIP/[EMAIL PROTECTED] seems to have them to in
the direct direction. However, debug shows that my asterisk doesn't
identify itself
On Tue, 14 Sep 2004, Kuniyoshi Murata wrote:
I'm thinking of introducing Asterisk on Linux for IP PBX.
Now I'm using ISP that has VoIP service and I have VoIP terminal box for
that ISP and a SIP account for SIP server of the ISP.
Now, what I would like to do is the following.
A. Setup IP
On Sat, 11 Sep 2004, John Stegenga wrote:
[sarcasm on]
Thank you ALL for your warm welcome to this list. I posted this message
yesterday, and since I'm only getting Digest I figured I'd see a response in
a day...
[sarcasm off]
C'mon. This is the Asterisk Users mail list, isn't it? This
On Fri, 10 Sep 2004, Jon Miron wrote:
I have a question that I'm curious about. I want to set up a 4 phone
system in my home with 2 actual lines coming into the house. Both or
just regular lines (not sure of this matters?), one being VoIP and the
other just a regular analog line. For now
On Fri, 3 Sep 2004, Bill Andersen wrote:
I just ran across the * site. Looks great. I do not need a PBX at this
time, but DO need to replace an old voice mail system. I'll do my
homework and figure out the specifics, but before I dive into it all and
spend a bunch of time only to find out
On Sun, 29 Aug 2004, Johannes van Hulst wrote:
For asterisk I am using more than one sip providers.
The provider in Holland would like to have the international calls like
00 31 20 1234567 but the provider in the US likes it like 0011 31 20 1234567
Can I make a rule in asterisk so that I can
On Mon, 23 Aug 2004, Erik Anderson wrote:
Hello all - I'm just starting to play around w/ asterisk, and I've run
into a seemingly simple problem that has really manged to frustrate
me...
I'm running the latest cvs version of *, and am trying to dial in to
the default extention 1000 demo
On Tue, 24 Aug 2004, James Sizemore wrote:
Ex-girlfriend logic not working in latest CVS?
Incoming sip calls don't work. Anyone else seen this
problem?
Extension logic looks good:
exten = 6153248305/_931NXXX,1,Queue(queue1);
exten = 6153248305/_615NXXX,1,Queue(queue2);
;exten =
On Tue, 24 Aug 2004, Huddleston, Robert wrote:
Take mercy on me - I'm a newbie w/ Asterisks... Here's what I'm trying to do
- and please someone let me know if this can be done...
We have a commercial VoIP network... The gatekeeper supports MGCP/H.323 and
allows for calls to be made to the
On Tue, 24 Aug 2004, Erik Anderson wrote:
On Tue, 24 Aug 2004 11:46:36 -0600 (MDT), Greg Hill
[EMAIL PROTECTED] wrote:
x-lite uses the RFC2833 style for DTMF out of the box (it can be set to
transmit inband). You need dtmfmode=rfc2833 in [general] or in the section
for your x-lite user
On Tue, 24 Aug 2004, Paterson, Mark wrote:
Honestly, do you think I would ask for help on the list if I hadn't come
up with any successful results on my own??
Just asking if anyone has made this work. If so what rev of * were they
running and what do their configs look like.
Jay does have
On Tue, 17 Aug 2004, John Williams wrote:
My desire to run Asterisk is finally giving me the reason to install a
Linux box at home.
Is RH9 the only distro that Asterisk will run on, and can anyone
recommend a good source for a cheap Linux (RH9) box?
For example, walmart.com has microtel
On Tue, 17 Aug 2004, Lucas Wrenn wrote:
Hello all I have a RH8 machine running the latest CVS of asterisk
against digium hardware including a TDM400P (with two fxs modules) and a
X100P. I am having some really weird problems with m internal
extensions. (I am trying to get them working before
On Mon, 16 Aug 2004, Info wrote:
Hoping someone might know how to resolve this (probably an easy one). I
have one Asterisk PBX with a single NIC and an FXO card with PSTN line
attached, and one IP phone (Budge Tone 100) on the LAN. Via the phone I
get no dial tone, and dialing 9, number
On Mon, 16 Aug 2004, Olle E. Johansson wrote:
James Freire wrote:
Hi All,
I am trying to setup another sip trunk in addition to what I am already
using. The sip provider we are using right now gives you your username
as your email address. So IE. If my email is [EMAIL PROTECTED] that
might see this as a message to [EMAIL PROTECTED]
and re-send the message there. The supposed-closed relay would effectively
be an open relay, with all the implications that go with that.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill
Sent
On Mon, 16 Aug 2004, Mike Roberts wrote:
Is there anyway to test if this call is touching my servers? Everyone
is telling me the DID is fine. But I can't confirm that for sure. And
I don't want to go ahead and start making changes to my config when
the DID isn't even working in the first
On Mon, 16 Aug 2004, Mike Roberts wrote:
sip debug made no changes
I can makes calls with my asterisk using X-lite softphone, I can even
call the 877 number and it works perfectly! But when I call the 877
number over the PSTN, it does nothing. A busy signal.
Try calling from a different
On Mon, 16 Aug 2004, Olle E. Johansson wrote:
Greg Hill wrote:
On Mon, 16 Aug 2004, Olle E. Johansson wrote:
James Freire wrote:
Hi All,
I am trying to setup another sip trunk in addition to what I am already
using. The sip provider we are using right now gives you your username
On Sat, 14 Aug 2004, administrator tootai wrote:
Hi list,
I have SIP clients and H323 GK connected through h323 channel (Nufone).
In h323 conf I gave prefix=09 so all call starting with this prefix are
send to asterisk. The context is also given their as [fromh323]
But now, in asterisk, I
On Thu, 12 Aug 2004, Jay Milk wrote:
Praytell, if you have Asterisk working, why do you even bother with the
BroadVoice voicemail? First order of business when I activated my BV
lines was to disable voice-messaging and set busy-forward to my PSTN
number (which in the meantime has been
On Wed, 11 Aug 2004, Chris wrote:
Set your sip.conf and your phone to inband as BroadVoice requires. Then
simply create an extension for BV Voicemail and use the SIPDtmfMode()
command like this
exten = *86,1,SIPDtmfMode(rfc2833)
exten = *86,2,Dial(SIP/[EMAIL PROTECTED],60)
exten =
On Thu, 12 Aug 2004, Travis Conway wrote:
I signed up for BV and entered everything, to the best of my knowledge,
in the sip.conf and extensions.conf file, but am having problems getting
it to connect. If someone could show me an example of their working BV
sip.conf that would be greatly
On Mon, 9 Aug 2004, Kevin Johnson wrote:
When dialing 8437624, I get the following output:
-- Executing NoOp(SIP/office1-b727, call for 843762 43762
6) in new stack
on the following line:
exten = _8.,3,NoOp(call for ${EXTEN} ${EXTEN:1} ${LEN(${EXTEN})})
this is really odd. I've got a
On Tue, 10 Aug 2004, AJ Grinnell wrote:
I am now at a total loss. Using Sipura spa-2000s connected to *, I get
DTMF working just fine for internal extensions, voicemail, etc. If
making an outgoing call like this spa -- * -- Cisco AS5350 -- PSTN, I
get no dial tone. I am working unsuccessfully
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Greg Hill
Sent: Tuesday, August 10, 2004 1:32 PM
To: Asterisk
Subject: Re: [Asterisk-Users] DTMF issues
On Tue, 10 Aug 2004, AJ Grinnell wrote:
I am now at a total loss. Using Sipura spa-2000s connected
On Sun, 8 Aug 2004, niko singh wrote:
Thanks for taking a look greg and hank. This seems to be getting bettre
everyday..help please My sjphone is running on the same box as
asterisk...i believe then the red hat firewall should not be a problem.
Some of these problems sound like they might be
On Sun, 8 Aug 2004, Kevin Johnson wrote:
I'm having a problem with extensions.
Any extension longer than 6 characters gets truncated to 6 characters.
For example,
exten = _7XX,3,NoOp(call for${EXTEN})
results in
call for 712345
when given
7123456
that's ${EXTEN}, not
On Sat, 7 Aug 2004, niko singh wrote:
Thanks greg , for pointing out the valuable resources for reference. I
tried SJphone in a windows environment to connect to fwd and it worked
fine(including (audio). Now have to do the same thing for linux(red hat
9 ) and hope the nat issue is resolved.
On Sat, 7 Aug 2004, niko singh wrote:
I just installed asterisk on my system with the purpose of rerouting
calls on sip channels. I don't think i need any hardware for that.
you're right, mostly. There are some asterisk features like meetme
conferences which require a timing source. This could
On Wed, 4 Aug 2004, Scott Petersen wrote:
[snip]
What I am seeing is an event every half hour exactly, on each of the two
voice lines. This causes the simple switch to kick in and ring the
extensions. Of course there is no one there. I have put a workaround in
[snip]
Since these events happen
On Wed, 4 Aug 2004, William R. Lorenz wrote:
I'm looking for U.S. providers that will provide access to the PSTN and
allow me to easily use my Asterisk box with their services. I would
prefer a provider that supports number portability, so that I can park my
existing cell number on their
On Mon, 2 Aug 2004, Areski wrote:
I have made an update to asterisk RC1 - all works well :)
but I am getting all the time an error message:
Aug 2 13:34:57 NOTICE[15375]: sched.c:221 sched_settime: Request to
schedule in the past?!?!
Aug 2 13:34:57 NOTICE[18446]: sched.c:221
On Mon, 2 Aug 2004, Carlos Arnt wrote:
I Have a problem here, if anyone know a method to avoid please tell me.
Using * with the option canreinvite=yes i can in theory tell to my *
box, send RTP Packet directly from one Sip device to another one, then
In Theory, i will not use my own internet
On Sun, 1 Aug 2004 [EMAIL PROTECTED] wrote:
exten = _34.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
The r at the end of this line tells asterisk to generate a ringing sound
for you to hear. In other words, the ringing you're hearing isn't coming
from the far end SIP device. Taking the r out will
On Sun, 1 Aug 2004 [EMAIL PROTECTED] wrote:
Hi,
I've had a look at it and the timeout error is what happens straight after the
phone disconnects:
Aug 1 04:07:13 WARNING[106511]: pbx.c:922 pbx_substitute_variables_temp: The
use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo'
Aug
On 29 Jul 2004, ShanKutti wrote:
I would like to study the asterisk source code(Program). I dont' know
from which file i've to start reading the code. can anyone helpme.
depends on what you're trying to do, I guess.. if you want to start at the
entry point of the asterisk binary, then 'grep
On Wed, 28 Jul 2004, Chris Shaw wrote:
exten = _9NXXNXXNXXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60)
exten = _9NXXNXXNXXX,102,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60)
exten = _9NXXNXXNXXX,203,Congestion()
Any thoughts? I know, I know... UGLY... but it would work I think...
so if I'm reading
On Wed, 28 Jul 2004, Bartosz Wegrzyn wrote:
[snip sip.conf]
Incomming calls still fails.
NO SOUND AT ALL!!!
okay, hang on a minute.. is it the call that fails, or is it the audio
stream that fails? If the call connects but no audio comes through then
the problems are likely to be in an entirely
On Wed, 28 Jul 2004, programmer_ted wrote:
I have an X-Lite phone on my box and I'm trying to register it with a
remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I
know it's a pain to do because of SIP not working well with NATs, but I
know there are ways to do such a
I finally decided to get a little source code dirt under my fingernails
tonight and dig through chan_sip.c to understand how registrations are
currently implemented. The hope is to perhaps at least seed some ideas
about how to make registrations to a server name, which resolves to
multiple IPs,
On Mon, 26 Jul 2004, John Fraizer wrote:
That should be
exten = 911.,1,blah
and
exten = 9911.,1,blah
You don't want to not catch a call when the user is scared to death and hits too
many 1's.
won't you need _ in it (_911.) in order to make it do pattern matching?
On Sun, 25 Jul 2004, Rich Adamson wrote:
I just started service with Broadvoice.com and everything seems to work.
However, apparently my understanding of incoming sip contexts is less
then what I thought it was. Could someone point me in the right
direction? (* on a public address,
On Thu, 22 Jul 2004, Jason Hartman wrote:
Has anyone tried using BroadVoice for VSP? I have Asterisk configured
for a home office I've been trying to decide which VoIP provider to go
there are quite a number of BroadVoice users on the list. (myself
included)
You'll find find a lot of posts
On Wed, 21 Jul 2004, Michael Wang wrote:
How do I change configuration of Asterisk so that phone B can use
aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address?
sounds like * is using reinvite to get itself out of the loop and let the
phones send RTP directly between
On Wed, 21 Jul 2004, Preeti Gopalan wrote:
[EMAIL PROTECTED]
type=user ; either friend (peer+user), peer or
user
context=default
[EMAIL PROTECTED]; usually matches the
section title
host=172.16.4.79 ; we have a static but private IP
On Mon, 19 Jul 2004, Kevin P. Fleming wrote:
Scott Laird wrote:
So $1600 for 24 ports. That's not *too* bad. HP seems to have a
similar model (2626-PWR) for a similar price. 3com also seems to have a
24-port injector for $800.
I still don't understand why I can buy single-port
run safe_asterisk and then asterisk -rc (add v's to your liking). You
probably weren't able to connect to remote asterisk because none was
running. safe_asterisk is a script which re-starts asterisk in the event
that it segfaults, dies, or otherwise implodes..
Greg
On Mon, 19 Jul 2004, James
On Fri, 16 Jul 2004, Jeremy McNamara wrote:
How about top posting and improper editing?
Jeremy McNamara
yeah, that's a good one too. (I didn't write the paragraph below.. Carlos
did!)
Greg Hill (DIDN'T) wrote:
One thing I have learned is to document the questions, so people can
On Thu, 15 Jul 2004, Jeremy Kenney wrote:
I have a cisco 7960G and would like to run in on asterisk I have an issue
though for some reason it has a problem running thru NAT
Any assistance would be great!
Note that you posted your question as a reply to another thread. This
action alone is
On Wed, 14 Jul 2004, jurgen wrote:
Of course you can do this, but you will need a channel bank for all
those non-VoIP phones you want. You will probably want to use T1 channel
banks with the TE4xx cards. One E1 coming in plus 2 or three T1 channel
banks.
I guess what I'm missing here is
On Wed, 14 Jul 2004, Nik Martin wrote:
Gabriel Millerd wrote:
Is there a magic 'fan card' that has a power out that people are
using?
This may work for you.
http://www.thermaltake.com/products/subzero/subzero4g.htm
you lost me, its a processor cooling device. it doesnt
On Sun, 11 Jul 2004, Kannaiyan Natesan wrote:
When I call a SIP user, the phone should ring in more than one
extentions. Also more than one phone should be able to register with
asterisk. Right now it is not the case. The last phone which register will
be receiving the calls. This type of
On Fri, 9 Jul 2004, Jay Milk wrote:
Try dtmfmode=rtc2833 then sip reload
er, make that rfc2833 instead. :)
Greg
___
Asterisk-Users mailing list
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To UNSUBSCRIBE or update
On Mon, 5 Jul 2004, Chris Foster wrote:
I'm trying to setup a primitive announcement-paging system in my house
using the line-out from my * box to a cheap amplifier that runs to
speakers on our first and second floors from the basement. I have a
extension that connects to Console, and console
On Mon, 5 Jul 2004, hank smith wrote:
how would I do this but do it with broadvoice?
I want to give people the oppsion to call my cell phone but I use a voip
carier
stay tuned to see how he gets the thing figured out, then change
exten = 2000,2,Dial(Zap/1/5551212,10)
to
exten =
From the archive on June 18:
Date: Fri, 18 Jun 2004 19:23:57 -0500
From: Brian K. West [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
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Subject: [Asterisk-Users] #asterisk is +r now, meaning register your nick with nickserv
How do you register?
do this
On Tue, 29 Jun 2004, Andrew Elchuk wrote:
I'm trying to get asterisk to auto-dail out. I created a *.call file
did you create the file in /var/spool/asterisk/outgoing/, or did you
create it elsewhere and then move it to that directory? The docs mention
that if the file is created in the
On Sun, 27 Jun 2004, Vassilis Konstantinou wrote:
I have been struggling with my Asterisk setup for 3 days now and I think I
have done well...apart from the small detail that I cannot dial out on my
phone (PSTN) line.
[snip]
The scenario is: if I dial 9123 (for the UK clock) then output
On Thu, 24 Jun 2004, Matt wrote:
NOTICE[-1147675728]: Peer '004' is trying to register, but not
configured as host=dynamic
from all the phones I've set as host=xxx.xxx.xxx.xxx
It sounds like * is letting you know that it got a registration attempt
where none was expected.. Have you
On Thu, 17 Jun 2004, PAZ wrote:
I'm trying to implement an IVR for internal use for the
enterprise I work for, but the goal I'm trying to reach is that the
main menu of this IVR present itself to the user after 5 seconds he picks
up his extension (and only if the user doesn't press any
On Wed, 16 Jun 2004, Joe Baptista wrote:
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.
well, you could dial 911.. ;) But more seriously, I think I'd start by
calling the
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