Daniel Mikusa wrote:
Look in the Makefile for the variables 'INSTALL_PREFIX' and 'PREFIX'
they control where Asterisk is installed.
Dan
Jeremy Jones wrote:
Is there a way to install Asterisk from source and not stomp on your
already existing Asterisk installation? I do
Is there a way to install Asterisk from source and not stomp on your
already existing Asterisk installation? I don't see a "configure"
script and it looks like it's trying to find stuff in /etc/asterisk and
in /usr/lib/asterisk and probably other place
isk IO, nor is network IO.
Has anyone seen anything like this? Anyone have any alternatives they'd
like to propose?
BTW, I'm running Asterisk 1.0.9 on Ubuntu Breezy and have run Linphone
on Linux, SJPhone on Linux, and SJPhone on Windows all with the s
On Thu, 2005-03-31 at 02:43 -0500, Brian Capouch wrote:
> My understanding is that to an extent when we buy Sangoma we're putting
> the dagger to Digium.
Come on! If Digium started manufacturing tires, would i need to put 'em
on my car to keep on the favorable side of karma? Digium makes
tel
Paco Perez wrote:
> Hello. I would like to know if somebody did a wireles voip with Asterisk PBX.
>
> I think to deploy a wireless for about 500 potential customers, it's a 3 km
> radius maximum coverage with houses without phone lines, I work for public
> places telephony small enterprises ( a
Hi David
On Tue, 2005-01-25 at 09:10 -0800, David Shaw wrote:
> Out of the blue extension 100 will ring once. This will happen 3-4 times
> a day. I have checked the logs and no incoming calls. I have extension
> 100 and 101 going to a SPA-2000 ATA. Any ideas on this??
this is probably that little
no
answer. Anyone have insight?
--
Jeremy Jones <[EMAIL PROTECTED]>
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TYPO ALERT...
On Fri, 2004-12-10 at 14:25 -0700, Jeremy Jones wrote:
..snip...
> Using the tdm400p card, with the first port connected to a 24-port fxs
..snip...
That should read "...T400P card...", not "...tdm400p card..."
--
Jeremy
re's no ringing after the phone is hung up, but there's no dtmf
picked up by asterisk either...
Anyone with ideas?
Thanks,
--
Jeremy Jones <[EMAIL PROTECTED]>
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Hi all,
> This is phone and the ATA is available soon from
> http://www.eezeephone.com priced at $75.00 each.
>
> Both have SIP+H323 and MGCP (also Net2Phone) compatibility.
That site's a bit out of whack...
Got to the voip phone product page at
http://www.eezeephone.com/ezp_frm_productdetail
Hi Duane (et alia),
>
> YES, because you could have an MGCP gateway device (more than
> one POTS line)
> ie. ours have 4
> If so you would do something like this...
>
> [2084728800103]
> host = dynamic
> context = westcomllc
> callerid = "Jeremy Jones
> [2084728800103]
> host = dynamic
> context = westcomllc
> callerid = "Jeremy Jones" <103>
> nat = no
> transfer = yes
> callwaiting = yes
> threewaycalling = yes
> cancallforward = yes
> mailbox = [EMAIL PROTECTED]
> line
layed
properly by the called party's phone. Calls from the mgcp device to any
other device display "Asterisk" as the cid name, nothing for number.
Here's what I have in my mgcp.conf for the device:
[2084728800103]
host = dynamic
context = westcomllc
line => aaln/1
calleri
can't get zapdummy (these boxes use
usb-ohci, not usb-uhci) to compile, and zaprtc sure won't work (no rtc
module here...). Does anyone have any suggestions as to how I might
either
1. weasle out of the timer requirements for meetme
Or
2. use some as-yet-undocumented 3rd party conferencing pl
Hi,
>
> Hello,
>
> I have been trying for a while to make the oh323 channel
> working but i didnt manage, i have everything compiled
> correctly but asterisk find somethign like an "undefined
> symbol" when it loads the oh323 module...
Put the path to your openh323 libraries in your LD_LIBR
Didn't I hear a week or two ago (on this list) that someone had taken it
upon themselves to write an asterisk module for the openss7-modified
digium t1/e1 cards? Maybe soon asterisk'll do it.
Jeremy Jones
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[
I'd be willing to bet you have "r" in your dialout string (i.e.
something like: Dial(${TRUNK}/${EXTEN},120,r)...
Get rid of that in the outbound dialing, and you otta be ok.
Jeremy Jones
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTE
Andrew Thompson wrote:
> Are your internal-to-internal calls handled seperately from
> external-to-internal calls?
>
> I see you are accounting for local versus ld calls, but what
> about when the
> person in the next cube over calls you?
Well, I have a dual-purpose asterisk setup -- hosted bus
This:
> ; Return last call
> exten => *69,1,DBget(temp=LastCIDNum/${CALLERIDNUM}) ; read db value
> LastCIDNum for this CALLERIDNUM (i.e. the extension making the call)
> exten => *69,2,GotoIf($[${temp:0:3} = 208]?3:4)
> ; if it's area
> code 208 (my local area), go to priority 3,
69,4,Macro(dialout,${temp}) ; call my
dialout macro, using full string
exten => *69,103,Congestion ; no
key? congestion
exten => *69,104,Congestion ; no
k
e of such a configuration, or hints (i.e. what variable I'll catch
in my standard extension macro)?
Thanks!
Jeremy Jones
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If the user to whom that vm is assigned goes through the setup process &
records their name, that is played.
Jeremy Jones
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Harold Workman
> Sent: Monday, June 21, 2004 11
Hi all,
> Odd... I did a make update and how the MailboxExists works fine.
> However, it works just as the docs say: add 101 to priority if the box
> *does* exist, add 1 if not. I have tested it and this seems to be how
> it works. You may wish to test your flow and make sure it works as
y
o answer in 2 min & no vm
It's a little ugly w/all those NoOps, but I think I need those to get
the priorities right.
Jeremy Jones
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Can some share one with me/us?
>
Not aware of any scripts like that, but...
you could use the odbc support in asterisk in conjunction with some
slick odbc-ldap connectivity.
Jeremy Jones
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Type:
user$ man patch
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Ed Devine
> Sent: Tuesday, June 08, 2004 11:28 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] HOW-TO DIFF
>
> How do I patch an * file if all I have is the .diff file?
I can't speak for "general" cases, but I know when I've tried to set
videosupport=yes, my as5300 can no longer speak w/*. I wonder if it can
be set per peer - haven't tried that...
jeremy jones
> -Original Message-
> From: [EMAIL PROTECTED]
> [mai
I think what he means is this:
I can have extension 104 defined in multiple contexts, for instance if I
host virtual pbxs for multiple customers on one * box. The syntax of my
* conf files requires @ if I want to differentiate
between these extensions. If you're using the * box for one business
to
be able to include a different "parkedcalls" context in each business'
context, and prevent someone at, say, the mortgage broker's office from
picking up a parked call from the collection agency's context. Anyone
have an idea of how t
Let's say I have a third-party device acting as a sip<-->pstn gateway, a
cluster of three asterisk servers, and a teensy bit of dns knowledge.
Let's now say those asterisk servers are a1.company.com at 192.168.0.1,
a2.company.com at 192.168.0.2, and a3.company.com at 192.168.0.3.
1. If I setup ro
t; To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Transfering with Grandstream Phones
>
> On Mon, 2004-05-10 at 09:12 -0600, Jeremy Jones wrote:
> > The gs handytone 286 user manual at the gs website lists
> call transfer/call
> > forward as not yet implemented (the firm
The gs handytone 286 user manual at the gs website lists call transfer/call
forward as not yet implemented (the firmware version listed on the manual is
out of date, thoughg). I sent a query to gs tech support a month or two ago
and received this:
"Call waiting and Caller-ID are implemented in HT
You need to mount a cd before running the Registration binary.
Also, you'll need to mount a cd any time you want to start/restart asterisk
(as I discovered this morning...)
jeremy
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Mark Elkins
> S
nyone
have that diff file lying around?
Thanks,
Jeremy Jones
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Make sure you don't have "videosupport=yes" in sip.conf when using
as5300. I found mine doesn't like that much & got that codec error.
Jeremy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Radius
Sent: Thursday, April 15, 2004 2:37 AM
To: [EMAIL PROTECTE
Nice & elegant! Looks great.
jj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolas
Gudino
Sent: Thursday, April 01, 2004 1:52 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
http://sip.house.com.ar/operator
Its a
llow=ulaw
context=some-biz
[EMAIL PROTECTED]
callerid=<7145551212>
**
Anyone know what I otta be doing differently? I've told the ata's to do
dtmf "via RTP (RFC2833)". Should I change that to "in-audio"
rom the iproute2 package solved my problem. (Perhaps this is
what Doichin Dokov had going on late last week?)
Jeremy Jones
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p
Is this all wrong? Please tell me where... I have sip-speaking
Grandstream HandyTone-286 ATAs. What's the standard transfer sequence?
Flash-5 or something like that?
Thanks,
Jeremy Jones
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s ago back, and
all is well...)
Thanks
Jeremy Jones
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w=all
allow=ulaw
So far, this is a very straightforward setup - nothing fancy, just works.
This particular AS5300 also speaks h323 to another voip soft-switch, so I
have other dial-peers that match 714554, etc.
Jeremy Jones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[
:
[2085551212]
type=friend
username=2085551212
secret=1234
host=dynamic
canreinvite=yes
disallow=all
allow=ulaw
context=testme
mailbox=5551212
callerid="Jeremy Jones" <2085551212>
[pstn_gw]
type=friend
username=pstn_gw
disallow=all
allow=ulaw
context=default
canreinvite=yes
host=10.
:
[2085551212]
type=friend
username=2085551212
secret=1234
host=dynamic
canreinvite=yes
disallow=all
allow=ulaw
context=testme
mailbox=5551212
callerid="Jeremy Jones" <2085551212>
[pstn_gw]
type=friend
username=pstn_gw
disallow=all
allow=ulaw
context=default
canreinvite=yes
host=10.
st the gear. Like this:
dta-310 <--> * <--> as5300 <--> pstn
Dialing in works (i.e. pstn to dta-310), but I can't figure out the
dta-310 to pstn thing.
Jeremy Jones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hermann
Wecke
Sent:
amp; no scrolling gibberish on the * console.
Anyone have ideas?
Thanks,
Jeremy Jones
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If asterisk'll compile against uclibc, it'll go on the toaster. Most
toasters (and coffee grinders & such) don't have enough flash memory for
a full glibc...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Gomillion
Sent: Tuesday, February 03, 2004
Generally speaking, unless you're using an rtp proxy, the rtp audio
should go client<-->client. H323 does the call setup and teardown and
such, but the audio stream is usually direct.
Jeremy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marc Fargas
Se
Hi,
Yep, I got the latest firmware (and the next-to-latest, and the
next-to-next-to-latest, and one earlier yet) for SIP. The first three
(firmware versions 1228, 1227, and 1226) all have that password protected
"Advanced Configuration" page. The fourth one I found (version ) is a
bit more o
This one does mgcp... It's been used in conjunction with a hosted pbx
system called Centile that 8x8 now owns. If there's a firmware image
anyone knows of to make these do sip, I'd rather do that. But for now,
mgcp help is what I need.
Jeremy
-Original Message-
From: [EMAIL PROTECTED]
Hello folks,
I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no
dialtone & can't get it to ring. My mgcp.conf says:
;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 0.0.0.0
[172.16.2.25]
host = 172.16.2.25
context = default
line => aaln/1
And here's the inte
This one is for an mgcp phone... In my /tftpboot directory, there's one
to match each SEP*.cnf file
Default
2002
2001
3106
2427
2428
10.0.0.112
P003E302
en
1
iso-8859-1
en
0
http://10.0.0.112:8086/ciscodirectory?accent=true
-Original Message-
From: [EMAIL PROTECT
Thanks all who replied, I think you've gotten me on my way.
Over the next few days, while I fiddle with the system I'm testing at
home, I'll try to churn out some documentation regarding my setup &
configuration that may be helpful to someone. I'll submit my notes to
the wiki when I'm ready.
Any
ticular...
The configuration files & examples I've found all assume a business
environment, where you'd dial a 9 for outside lines. Anyone have an
example config where an endpoint gets dumped directly to the pstn when
they pick up the phone?
Thnaks,
Jeremy Jones
_
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