Maybe this could be used with the Internet Repeater trunking system I
primarily use VHF... But would be interested in setting that up on my asterisk
with the Internet 2M Repeater trunking system inter-connect
Robert A. Huddleston, KF4BYY
Cavalier Telephone LLC.
(Desk) 804.422.4401
(Cell)
If nic is loaded using modprobe - you can set options for duplex -
depending on the nic...
See /etc/modules.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Monday, August 29, 2005 11:13 AM
To: Asterisk Users Mailing
U joke - duh!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Innocent Evil
Sent: Wednesday, August 24, 2005 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI + Ruby
What IDE are you
Y'see it? There it goes! Right over his head.
Huddleston, Robert wrote:
U joke - duh!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Innocent
Evil
Sent: Wednesday, August 24, 2005 3:53 PM
To: Asterisk Users Mailing List
I say start small and then go big... Oh I don't know a Proliant 1500 or
3000 should work nicely -- if you can handle the noise =)~
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Latham
Sent: Monday, August 22, 2005 1:40 PM
To: Asterisk
Is it my imagination or did I just drop off the list for several days
somehow... I didn't get any posts since Friday...
rhuddleston.vcf
Description: Binary data
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I would think that question would be as silly as me asking you a) how many
people can I fit in a vehicle or b) how many web users could I have access my
apache web server...
Need more details to make that judgement.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I've got a handful of ATAs Innomedia that support two ports... I have one
plugged in for voice for the house and the other I use for dialup internet..
ONLY for testing newly built dial-up computers that they can get online and
surf... Gotten some pretty good speeds out of them too
Robert A.
Do you know where to get one of these?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
chawki hammoud
Sent: Thursday, June 30, 2005 4:35 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] wi-fi phone advice
Hi:
I want to
- Non-Commercial Discussion
Subject: Re: [Asterisk-Users] wi-fi phone advice
This unit is vaporware from what I can tell.
Cory Andrews
Purchasing / EVP
VOIPSupply.com
v - 716.630.1555 X22
e - [EMAIL PROTECTED]
Huddleston, Robert wrote:
Do you know where to get one
Well maybe look at the support that mysql has... I've been wanting to try it -
but don't have the hardware available right now to do it..
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JD Austin
Sent: Thursday, June 30, 2005 3:05 PM
To: Asterisk Users
PROTECTED] 1.0
my mp3 is called
mp3
it has nothing before it
it is 0 bytes
does my mp3 of 0 bytes need to have a .mp3 or does it need to be called
anything?
thanks
hank
- Original Message -
From: Huddleston, Robert [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial
Ummm are you sure about this... I've seen people outpulse on PRI before
It's dependent on the carrier - was my understanding.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryce Chidester
Sent: Wednesday, June 29, 2005 12:28 PM
To: Asterisk Users
Worked for me with a different stream... I ran into this same problem before -
but it was my own fault for not RTM... Both the manual and ast install advised
of verifying correct version of mpg123... I had wrong version and thus got no
noise...
If you follow the directions explicitly laid out
Ummm this would be the purpose of using hunting or circular
hunting... The premise it sounds like is you want to always ensure there is a
phone number people can call you on...
so...
123-4567
123-5678
123-6789
roll in a circular hunt group... give customers the
123-4567 number and all calls
As an employee in the technical operations of a CLEC this information is easily
obtainable by anyone that has access to the Class 5 switch servicing that
PRI... A Q.931 trace in the Class 5 Switch will tell the whole story
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
hmmm Caller ID ? that sounds like a modem as a quick
burst
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
CityTechs.NetSent: Tuesday, June 21, 2005 2:58 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk
answers with high pitch sound
Hi,I've
Site down again?? Voip-info.org? or maybe really slow?
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I had the same question... A portion of code is on there that I need to hack
openh323
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcel van
Kaam, Fonetica
Sent: Tuesday, June 14, 2005 10:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial
I know this sounds stupid but sounds like it's not in a ground-start mode...
When we mis-program a line in our Lucent 5E as loop and it should be ground
-- the customer is always able to rx calls just not break dialtone (tx)
calls...
So I would think that either the card is not right or the line
Anyone paying over $450 for a T1 is being ripped off...
If you are in VA,MD,DC,PA,DE,NJ you can get an integrated VoIP T1 for $300 -
$400 and a flat internet t1 for about $400.
The integrated VoIP T1 is great because it's handed off as an ethernet - no
need for a csu/dsu
-Original
Darn, and here I was thinking small town Melbourne,
FL, USA =(
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
jurgenSent: Thursday, June 09, 2005 11:16 PMTo:
Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and
Business-Oriented Asterisk
What about mysql cluster... I think it's open source...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rusty
Shackleford
Sent: Thursday, June 09, 2005 2:29 PM
To: 'Mark Musone'; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
As an isp I can tell you that we run into this problem on a daily basis. We
see three troubles:
A) No reverse (PTR) record at all
B) Reverse (PTR) but no forward to match
Which a) and b) usually cause the most troubles and...
C) A+B but with a domain of the mail server trying to send mail.
Awesome global variables good choice!!
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Dunham
Sent: Tuesday, May 31, 2005 2:21
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
AreskiCC - DOES IT
We too are a carrier / clec and our lucent iMerge for the PTSN is all
g711-ulaw
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Friday, May 27, 2005 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Can you post your conf file for the musiconhold??? Sounds like you haven't
defined a default class / context - I could be wrong
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
Sent: Wednesday, May 25, 2005 8:32 AM
To:
Wow I found a fellow pico user... I'm constantly receiving ridicule for my
use of pico... I cannot stand vi... If I don't have pico sometimes the ol'
emacs...
But vi is garbage.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent:
FYI - We have a solution here provided by Lucent that allows us to play /
review voicemails left on the Octel attached to the 5ESS switch... While
this is a simple webpage - doing a refresh every 3 to 4 seconds, it does
actually work.
The only loss of course would be if someone hung up on first
Now for the fun one - change ring pattern?? Like distinctive ringing? Is
this supported by asterisk or the end-point
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Umair Bari
Sent: Tuesday, May 24, 2005 10:20 PM
To: Tim P; Asterisk Users Mailing List -
I thought the HDSL device acts as an American SmartJack to terminate T1/E1
type services.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: Tuesday, May 24, 2005 2:33 PM
To: Asterisk Users List
Subject: [Asterisk-Users] Re: Red Alarm
Not sure if anyone has played with one of these gems - ToneCommander 7210 --
but they do ISDN over H323 - and seem to be proprietary to Lucent / AG
iMerge.
I'm trying to find a way to reverse engineer it to work on a standard
asterisk setup...
The first thing I found with a tcpdump / ethereal is
I funny one is when our IT manager accidentally supplies the power supply
power into one of our voip phones and also feeds it the POE =)
Melted outlet, flames in the RJ connector and melted cat5 cable
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 / NCS
1.0 -- is the channel_mgcp compat with NCS -- anything I can do to make it
compat?? This is what happens - below
*CLI mgcp reload
Reloading MGCP
== Parsing '/etc/asterisk/mgcp.conf': Found
Use EXIT or QUIT to exit
: Huddleston, Robert [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, May 20, 2005 2:52 PM
Subject: [Asterisk-Users] MGCP 1.0 / NCS 1.0
I'm trying to use a 4 port ATA that was designed for use with MGCP 1.0 /
NCS
Well here's a suggestion - a little crazy - but works... Most equipment is
taking the 120vac and converting it into DC voltage. So why not just feed it
DC voltage directly???
We had a situation where our field techs needed to test dsl circuits and
voip ata from the demarcation point outside a
Anyone tried to build * + h323 to rhel3...
I have to problems in the process...
a) Zaptel would not build - a whole bunch of errors about kernel...
b) make progdocs failed with reference to dot - check your installation.
Do I need the zaptel ?? I will not be using any interface cards..
I'd like
Think I'm doing something wrong here... followed what the wiki said but no
luck
Trying to do moh with streaming audio...
Musiconhold.conf
Default = custom:/var/lib/asterisk/mohmp3,http://209.51.128.160:5112/
Also tried the trick of
Default =
Nevermind - didn't RTM - mpg123 not installed..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Huddleston,
Robert
Sent: Thursday, May 19, 2005 3:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] MOH
Think I'm
Forgive my ignorance - I'm building * on an old redhat 8 box... I can't
build Zaptel - I don't need libpri - Asterisk is building though.
Do I need Zaptel - I remember once using ztdummy or something like that...
If I need Zaptel - any ideas why no build.. I'm assuming the common answer
is going
Anyone know of a Lucent EMRS PRI Card? Know where to get one? Ours went
dead.
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Anyone experimented with Calling Card support in * Am I wrong in
presuming that if I have one calling card caller call in and want to
complete a call I will use 2 lines (1 for the customers inbound and another
to complete the remote call)??
Thanks
Good answers to all...
My idea
a) Use a h323 / sip / mgcp line from the CLEC I'm employeed with...
This line will register with the * as an inbound line... Probably have
series hunting - thus I could get 50 #'s in a hunt group and you only have
to call the lead and get a line into the
Didn't see any responses...
I looked at the logs more closely and I see a message that says
releaseCompleteReason - destinationRejection.
This is showing around 60 seconds when the DRQ occurs and the connection is
broken.
-Original Message-
From: Huddleston, Robert [mailto:[EMAIL
I have a problem where an Asterisk server is sending a premature DRQ... Not
sure why..
Here's the setup - Asterisk using inAccess networks H323 replacement channel
driver
Connecting to a Lucent iMerge...
The call connects fine - I get the out of the box greeting - but after
exactly one
Minute -
Wow babelfish interpreted that French pretty bad =(
Since j'ai forgotten to do it, if you see this message, send to me your mall
so qu'on remains in contact. I opposite Marek, where were you with the table
are the preque old man? With soon!
Robert A. Huddleston, KF4BYY
Cavalier Telephone LLC.
I work for a carrier that provides flat LD Nationwide on MGCP and H323 -
soon SIP... We have been testing Asterisk and it seems to work good...
Check out www.phonom.com
-Original Message-
From: Adi Linden [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 09, 2004 1:53 PM
To: [EMAIL
Title: GUI
Anyone been able to integrate say ICECast or Shoutcast
broadcasts into their MOH... I guess if you used something like xmms (X-Winamp)
or something like that you could do it??
I'd like to be able to take a good streaming radio station
and make it my MOH..
Thanks
Is directory included w/ Asterisk or external app... I'm running older
release * so j/c
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 16, 2004 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial
I'm assuming nobody has experience with running ISDN / BRI over H.323...
-Original Message-
From: Huddleston, Robert [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 03, 2004 8:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] H323 ISDN
Anyone know where we could get a cheap free maybe would be nice sip
phone... We've been playing with an Innomedia MGCP and SIP adapters and
failing - so thinking that testing with a real phone might be good..
Robert A. Huddleston, KF4BYY
IT Support Analyst
Cavalier Telephone LLC.
(Cell)
I am having the same problem and not using the NuFone h323 but the
Asterisk-OH323...
Inbound to sip from h323 seems to be a problem with audio...
Robert A. Huddleston, KF4BYY
IT Support Analyst
Cavalier Telephone LLC.
(Cell) 804.400.3686
[EMAIL PROTECTED]
-Original Message-
From:
I've successfully got inbound/outbound calling working with our Asterisk
using the Asterisk-OH323 channel driver. We are using a parent gatekeeper
and the NuFone H323 channel driver would not work with the parent
gatekeeper...
I'm trying to determine a way to ensure that the line used for outbound
The only disadvantage we found to using the OH323 channel driver is that we
cannot now register netmeeting or other h323 directly to the * With the
nufone h323 channel driver we could register netmeeting and other h323
devices directly to the *...
So if we wanted to run internal h323
I have built latest
Asterisk w/ OpenH323 channel driver. We have a SIP softphone registered to the
Asterisk. We can place outbound calls from the SIP phone to the PSTN via
OpenH323 connection to our gatekeeper. Everything works okay - DTMF and
Audio...
But in the reverse
- if we call from
Does anyone know how to do this with the OH323 channel driver?
I want the local (7 digit dialing) to go out an h323 that I have registered
to a gatekeeper...
can I do something like
exten = _7.,2,Dial(OH323/ipofgatekeeper)
-Original Message-
From: Begumisa Gerald M [mailto:[EMAIL
Due to the format of the message coming from the H323 channels included w/
Asterisk we were unable to use our gatekeeper.
For a quick solution we tried the OH323 channel drivers and can receive
inbound calls from the parent gatekeeper.
We are trying to do a dial to gatekeeper...
I am trying
exten
: Michael Manousos [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 2:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dial Out w/ OH323
Huddleston, Robert wrote:
Due to the format of the message coming from the H323 channels included w
:[EMAIL PROTECTED]
Sent: Sunday, September 05, 2004 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GRQ / RRQ
Huddleston, Robert wrote:
All - NEED HELP BADLY..
Using Asterisk with H323 Channels..
Everything is installed but we cannot register
I posted previously that my Asterisk is failing to register with parent H323
gatekeeper.
It appears that it is issuing a GRQ GatekeeperDiscoveryRequest even though
I have
the file h323.conf saying exact IP address of gatekeeper... No discovery.
I'm trying to just bypass - remove the GRQ and have
Yes it's funny... Everyone can argue about a signature - but I request
several pieces of technical assistance - and I get the ignore... Thanks!
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 07, 2004 1:49 PM
To: 'Asterisk Users Mailing List -
All - NEED HELP BADLY..
Using Asterisk with H323 Channels..
Everything is installed but we cannot register w/ Lucent gatekeeper.
We ran ethereal and found that it was making GRQ (Gatekeeper discovery
requests)..
We had provided the name of the Gatekeeper (it's IP) and cannot determine
why
it's
I've read the wiki and other resources on how to connect Vonage / Voicepulse
and all these other services to Asterisk... We are attempting a connection
to a Lucent iMerge. Lucent has told us that it won't work - but we feel
confident that it will. Has anyone worked with the Lucent iMerge - or
Any got experience w/ PWLIB - sorry I know it's somewhat off topic...
I do not have a bison.simple file located on Fedora RC2...
But when make'ing PWLIB I get
../common/getdate.y:106:1: warning: YYPURE redefined
../common/getdate.tab.c:43:1: warning: this is the location of the previous
Look
at the /etc/rc.d and init.d directories... Unix has run levels and there are
shell scripts (like batch files) that are called upon system
booting..
As far
as nightly restart - you need to use cron or atd (at daemon) these
processes allow you to schedule scripts to run...
Otherwise
I've been having troubles compiling in the openh323 on both redhat and
debian... one of the biggest problems I had w/ Debian is it couldn't find
alot of libraries like termcap etc...
Has anyone else ran into these problems?
-Original Message-
From: Tim Jackson [mailto:[EMAIL PROTECTED]
I'm trying to make the chan_h323 in /usr/src/asterisk/channels/h323
But I'm getting all kinds of errors about PWLIB... I built using the newest
PWLIB and OpenH323 from CVS
Error log from make below
make
g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG
I'm trying to make the chan_h323 in /usr/src/asterisk/channels/h323
But I'm getting all kinds of errors about PWLIB... I built using the newest
PWLIB and OpenH323 from CVS
Error log from make below
make
g++ -g -c -fno-rtti -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG
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 (we are a communications carrier)...
The gatekeeper (Lucent iMerge) supports MGCP/H.323 and
allows for calls to be made to the
Uh oh... Does that mean that my request for help - with opening statement
take mercy on me - won't be reviewed =(
-Original Message-
From: Dave Covert (Sailtech) [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 25, 2004 10:16 AM
To: [EMAIL PROTECTED]
Subject:
I've read almost everything on every site possible ever made on Asterisk =)
I've posted to a gizzillion forums and email lists...
I can understand not wanting to share proprietary information - so if
someone is just able to tell me yes/no that this is capable of doing I would
be happy...
My
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 PSTN cloud.
I would like to
build Asterisks
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