I am testing DISA but can not dial out after getting local dialtone. The *
server takes accepts my password, gives me dialtone and allows me to dial
the digits then promply hangs up on me. CLI log shows
-- Executing DISA(Zap/1-1, 1234|contextname) in new stack
-- Accepting call from
They have a free version coming out that raises the limit to 8 gig I
believe.
Gary
Does Oracle have a decent-featured free version of their db software? That
was my original point, and where MS SQL 2005 is quite in the lead (limited
only to 1GB of RAM, 4GB DB, and 1 CPU).
-Michael
Anyone have a example of how to setup RDNIS in *?
To date we have been giving each voicemail user a individual DNIS but would like
to consolidate all the numbers into one and just use RDNIS to route the
call.
Thanks,
Gary
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Asterisk-Users
I tried to call the mexico city airport and got the following
-- Executing Dial(SIP/9104044010-541d, IAX2/[EMAIL PROTECTED]/57644910
@guest|90.Tf) in new stack
-- Called [EMAIL PROTECTED]/57644910 @guest
Jan 13 10:20:59 WARNING[1142135600]: chan_iax2.c:5339 socket_read: Call
rejected
by
I made a few test calls to the airport but don't speak spanish so I had no
idea what they were saying :)
Anyone have Marco Antonio Barrera's phone number :-()
Gary
I just made three calls. stayed on the phone about 5 to 10 minutes per
call
works great! Crystal Clear!!
Juan
9105551212 = 1234,Gary Carr,[EMAIL PROTECTED],attach=yes
Should'nt that work?
Thanks,
Gary
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http
the call to
the proper voice mailbox based on that information.
If you can configure this for us please email me directly.
Thanks,
Gary Carr
President/CEO
705A Wesley Pines Rd.
COL Networks, Inc.
Lumberton, NC 28358
Phone: 910-402-5011
Fax: 910-618-9027
Check us out at: www.carolina.net
Anyone have a copy of the DB_areskicc.psql file mentioned in the AGI tar
file for this new application?
Thanks,
Gary
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that actually work with T.38
Hi Gary,
Aren't those all tied to service providers now?
Regards,
Steve
Gary Carr wrote:
We use the PAP-2NA with fax machines and have not had any problems.
Gary
Hi,
I am implementing T.38, and finding a problem getting boxes that work
with T.38 for testing. A lot
That site is correct. You have to be authorized by Linksys to order the
product from a distributor but they will work with any VoIP service. We use
them with our * service.
Gary
Quoting Gary Carr [EMAIL PROTECTED]:
You might want to tell that to these guys:
http://www.voipsupply.com
-this is very true, however, the current version of the Axxess software
(9.0) supports SIP trunking natively on the IPRC. I just got my Axxess
upgraded and am salivating to get * connected to it.
Hmm, so 9.0 is out and it supports SIP natively. How did you plan to
integrate the 2?
Gary
Rad's TDMoIP uses DSP chips on each end of the link to compress the data.
Gary
Just a Question. I would like to know if TDMoE follows specifiaciones of
TDMoIP RAD protocol that says that there is a compression of 16/1 when
you do TDMoIP.
Manuel Marin Garcia
TRANSTELCO S.A. DE C.V.
I am not sure of your answer but we are looking to integrate * with our
class 5 switch to provide Voice Mail services to our subscribers. If anyone
here has a interest in performing this integration from a
consultant/contract basis please email me offline.
Thanks,
Gary Carr
President/CEO
705A
Hi, which IP Centrex setup are you using?
Gary
I am using asterisk as a voicemail server for our IP Centrex SoftPBX.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten
Sent: 09 July 2004 22:46
To: [EMAIL PROTECTED]
Subject: Re:
I'd like to have a system like that as well. I
would be willing to chip in on the development.
Gary
- Original Message -
From:
Paul Rodan
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Cc: 'Commercial and Business-Oriented
Asterisk Discussion'
Why is IAX termination better?
Gary
So they offer termination via SIP for $0.013/minute?
Even better-- IAX termination :)
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So they offer termination via SIP for $0.013/minute?
Gary
Any since the per minute rates can be as low as $0.013/minute last time
I looked, you have to use a LOT of minutes before you spend as much as
you would have with that unlimited plan...
Regards,
-Dorn
p.s. I use both NuFone and VoipJet
We are testing some PAP2-NA ATAs from Linksys and I can use the device to
make outbound calls as well as receive inbound calls when the ATA first
connects to the * server. After a couple of minutes of being idle the ATA
disconnects from the * server and will not take calls, but can continue to
They show as a active item on Tech Data's website but they don't have any in
stock at the moment. They are available as drop ship from linksys.
Gary
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
I am running RH9 with a 4 port and 1 port ISDN
cards. Not problems that I am aware of yet.
Gary
- Original Message -
From:
Henry Devito
To: [EMAIL PROTECTED]
Sent: Sunday, September 19, 2004 6:30
PM
Subject: [Asterisk-Users] Asterisk and
Red Hat 9
I am running a P4 2.8 with 1 gig of ram and 7200
rpm IDE drives. Nobottlenecks as yet.
Gary
- Original Message -
From:
Henry Devito
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Monday, September 20, 2004 8:25
PM
Subject: RE:
Shite, I ordered some a few days ago from TD and they have my order on hold.
Gary
I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It
installed pretty easily and has worked great so I went to order some more
of these units today.
When I logged into Tech Data this morning,
This really chaps my hide. The situation as it's been explained to me
is: Apparently, too many *consumers* were accidentally buying the
PAP2-NA (unlocked) version and then complaining/returning them to
Linksys b/c they didn't understand that they need a service provider to
be able to place and
Anyone confirmed a stocking vendor we can purchase these from?
Gary
Ryan Wilkins wrote:
This begs the question, again, that someone else posted originally.. what
about loading SPA-2000 or PAP2-NA firmware in the PAP2? If it's the same
hardware, there shouldn't be any reason not to try it.
You have some contact info for Bottom Line Tech? We are a ISP/CLEC and want
to order some of these.
Gary
Eric,
I was told by Bottom Line Tech that Linksys told them to pull all units
and
stop all shipments unless there customer could prove they were and ISP,
which i am not so i can not, so no
.
Then we do not have to go with all the trouble getting to them.
- Original Message -
From: Gary Carr [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, September 22, 2004 3:59 PM
Subject: Re: [Asterisk-Users] Linksys PAP2-NA
The RPMs had errors for me
After installing RPMS and running modprobe zaptel I get
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol
register_chrdev_R07a6f6f0
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol
remove_wait_queue_Rd7b46182
/lib/modules/2.4.20-31.9/misc/zaptel.o:
Hi-
I've run extensive load testing with both single and dual P4's and Xeon's
(all at least 2.8GHz), and I've got 6 installed IVR systems of this size in
various configurations.
Hmm, I was under the impression that it was impossible to run dual P4 CPUs.
I thought Intel programmed instruction in
I don't understand your targeted market. Is your software available for
people who have their own asterisk servers and if so why a limit on the # of
usable ports?
Gary
Our already made solutuons are designed for just such scenarios.
Have a look at
I would like to see those configs as
well.
Gary
- Original Message -
From:
Tim Connolly
To: [EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 6:02
PM
Subject: [Asterisk-Users] Using
Lucent/Ascend TNT as a PSTN Gateway?
About a year ago, a
Is there any way to send a caller to a certain context when the caller is
forwarded from another pstn number. We are using * as a voicemail server for
our cusotmers and we are currently providing each vm customer a did to send
the caller to when their line is busy. I would like setup * to take
where did you get them from?
Gary
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 19, 2004 4:37 PM
Subject: [Asterisk-Users] Wonderful Success with PAP2-NA
Finally got authorized to purchase some PAP2-NA's from Linksys's.
Works
Wondering if it is possible or if something already exist to setup * to
offer Internet Call Waiting. For those that do not know what it is, it's a
small application that runs on a users computer that will pop up a window
letting them know they have a incoming call and who it is from then they
The * box would sit in a CO connected via PRIs.
Gary
Gary Carr wrote:
Wondering if it is possible or if something already exist to setup * to
offer Internet Call Waiting. For those that do not know what it is, it's
a small application that runs on a users computer that will pop up a
window
: 212.477.0990 x 810
e: [EMAIL PROTECTED]
http://www.cytexone.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr
Sent: Thursday, April 21, 2005 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] using
instead of System, which blocks, waiting
for the return code from the command I passed. Because the return code
is prolly irrelevant, you'd most likely want to use TrySystem too...
hope this helps :)
Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
You need a V92 capable modem for your client and a V92 capable access
server for you. The feature is called modem on hold, it lets you
pick up a call without loosing your internet connection, and resume
the dialup session after hangup. The only feature you need for your
telco is call waiting. It
Running CVS-HEAD-04/12/05-16:39:24 on CentOS 4.0 final installation. I am
hearing a brief echo on our Cisco 7960 phones when a incoming call is
answered. After a few seconds of conversation the echo disappears. There is
no echo on outbound calls or transferred calls. After a search of the
Running CVS-HEAD-04/12/05-16:39:24 on CentOS 4.0 final installation. I am
hearing a brief echo on our Cisco 7960 phones when a incoming call is
answered. After a few seconds of conversation the echo disappears. There is
no echo on outbound calls or transferred calls. After a search of the
mailing
What should the description be for a digium FXS card when running the
lspci -b? Mine shows the following.
00:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
Subsystem: Unknown device b100:0003
Flags: bus master, medium devsel, latency 32, IRQ
I have that exact setup and am getting a echo at the beginning of a inbound
call. What gain settings work best for you?
Gary
Hello Bryce,
Gain settings do seem to have an effect. I am going from a Cisco
7960AsteriskZap TDM CardPOTS
Thanks,
Greg
-Original Message-
From: [EMAIL
While we have not integrated the asterisk CDRs yet it should not be a
problem to do. We our building a billing system for ISP/CLECs that will do
what you want. If you want more information you can contact via email to
[EMAIL PROTECTED] or by calling 910.402.5010
Regards,
Gary Carr
President
That sigh will turn to cursing after a couple of months. We currently use
Rodopi, have for 10 years but the inflexability is too much to deal with
anymore so we are moving away from it.
Gary
- Original Message -
From: Ejay Hire [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday,
That sigh will turn to cursing after a couple of months. We currently use
Rodopi, have for 10 years but the inflexability is too much to deal with
anymore so we are moving away from it.
To what? I am also a cursed Rodopi owner. :-(
Tom
We bought the source code to wirebill and are
Are they still hurdles using Cisco phones with asterisk as mentioned at
http://www.voip-info.org/wiki-Cisco+Phones ?
We are looking for some cisco phones to test with.
Gary
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[EMAIL PROTECTED]
looks interesting. You mentioned that you are using the source
code to build your own platform, but how does it hold up on its own? Can
I ask what it can't do that requires you to build your own?
Thanks,
- Darren
On Wed, 2004-08-04 at 08:14, Gary Carr wrote:
That sigh will turn to cursing
the SIP image on them).
mitchel
On Wed, 4 Aug 2004 15:57:11 -0400, Gary Carr [EMAIL PROTECTED] wrote:
Are they still hurdles using Cisco phones with asterisk as mentioned at
http://www.voip-info.org/wiki-Cisco+Phones ?
We are looking for some cisco phones to test with.
Gary
Is it possible to set the attach= setting on a per user or per context
basis? We want to give our users the choice of no email notfiication, email
notification with no attachment, or notification with attachment.
Thanks,
Gary
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Who are the US wholesalers selling the uniden phones?
Thanks,
Gary
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Can anyone send me the sip images for the 7960g? I have 2 I want to test but
need them to be sip.
Thanks,
Gary
I've tried a *lot* of phones with Asterisk, and thus far, the Cisco's
are by far the best I've used.
Brian D'Arcy
-Original Message-
From: [EMAIL PROTECTED]
test message. No list messages received today.
Gary
- Original Message -
From: Soren Rathje [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 15, 2004 5:08 PM
Subject: Re: [Asterisk-Users] Asterisk MIBS
Alagalah wrote:
Hi,
I was wondering if there are any Asterisk
PROTECTED]/5011register
= [EMAIL PROTECTED]/5052
[cisco7960]type=friendhost=dynamicnat=yesqualify=200dtmfmode=rfc2833canreinvite=nomailbox=5052callerid="Cisco
7960"context=local
[garycarr]type=friendhost=dynamicnat=yesqualify=200dtmfmode=rfc2833canreinvite=nomailbox=5011callerid="
Did you check the dell outlet (refurbed with same warranty) or ebay? You can
typically purchase and ship same day from either of those.
Gary
I noticed the PowerEdge 750 seems to have one of each: 32- and 64-bit
PCI's,
both brought to the rear panel - nice.
BUT, I can't get the Dell's fast
on the same IP address and
port, how can that be?
Oh, I see the error message is actually coming from the sip phone, and
it's because those phones
have the same IP address, and therefore a loop is detected there. Is
this just ONE phone with two
proxy-accounts or personalities?
Gary Carr
Currently running version 1.8.16.0 and trying to manage confbridge rooms and
users. When I try to use the confbridge cli command I get a command not found
error.
CLI confbridge
No such command 'confbridge' (type 'core show help confbridge' for other
possible commands)
I've tried googling
I am trying to track down a white noise problem we are having in our conference
rooms. If there are 3 or 4 users in the conference the quality is good. After
we get more users in the conference we develop a white noise that gets louder
as more users come online. I have tried both meetme and
I am trying to setup a context to take a inbound call, hold the call,
connect to an external number, play a sound file to the external number,
then connect the inbound caller to the external number.
My thought was to accept the call and place them in a parking lot. Then call
the external
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Gary Carr
Sent: Wednesday, October 03, 2012 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call extension play
I received the same spam myself.
Regards,
Gary Carr
List users,
Did anyone else recently receive spam from DIDForSale with the subject
DIDForSale 2012 achievements? I suspect that they are using this
list to harvest email addresses and think they should be called out on
this poor business
Is it possible to issue the POKE to a end point from the CLI? Our
asterisk servers is not seeing some end points drop off and I would like
to create a script to manually check end points.
Thanks!
Gary
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