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Its free and supports forums with gateways to mailing lists and newsgroups.
From their feature list:
a.. NNTP Mailing List integration, allowing FUDforum to be used to archive
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lists, as well as allow forum members to
incominglimit is already implemented for SIP. Just specify under the
endpoint how many incoming connections are allowed.
For example,
[cisco]
type=friend
username=cisco
secret=blah
nat=yes; This phone may be natted
host=dynamic
canreinvite=no ; Cisco
Apparently you didn't read the README.. Please read that over again.. it
tells you exactly what to do.
bkw
On Sun, 10 Aug 2003, Serge Mankovski wrote:
Hi
I am using inAccess channel driver.
Compiled, installed. This is what I get when I am trying to start *
Hi list...
I have already installed a small PBX ( 1 FXO (E100P) and 4 FXS (TDM400P) )
but now i want to know how to build a bigger one... maybe 8 FXO and 24 FXS
something like that o bigger. But i dont know witch hardware i need.
And also any of you have installed Asterisk only for VoIP
I am attempting to set up an Asterisk box which I am only
concerned with getting a single T1 working. I have this T1 connected to my PBX and I
am looking at using Asterisk as a conference bridge.
Here is my zaptel.conf:
span=1,1,0,d4,ami
em=1-24
loadzone=us
defaultzone=us
Here
On Fri, Aug 08, 2003 at 01:06:32AM +0200, Armand A. Verstappen wrote:
On Thu, 2003-08-07 at 22:17, Wade Weppler wrote:
Any idea if these fixes will get added to CVS?
check http://bugs.digium.com/bug_view_advanced_page.php?bug_id=35
I've signed a release. See ticket.
Jayson
An advantage of a newsgroup is Google's ability to sort newsgroup results by
date. Many issues are resolved by the * community. So if I search for BRI
echo search by date, I'll learn that the problem has been solved how to
solve it. If I just search for BRI echo, I to sift thru partial
I've been working in the VoIP industry for just a bit over a year now...
Mostly taking care of the underlying systems. I've now reached the
point where I'm being drawn more and more into the call processing side
of things. My background is in computer and classic telephony systems
Hi,
On Fri, 2003-08-08 at 00:24, [EMAIL PROTECTED] wrote:
I recently purchased the Asterisk Developer's Kit (TDM) to try out
Asterisk. After following the directions in the Digium's FAQ topic entitled
Q. How do I configure my TDM40B and X100P?, I'm receiving the following
error:
You're looking for libncurses-dev and in libpri you can remove -Werror
from libpri/Makefile or cvs update libpri (it should be fixed)
regards
Martin
On Wed, 6 Aug 2003, Rhys Hopkins wrote:
Hi,
I am having trouble building and installing libpri and asterisk on my
system. Zaptel seemed to
I agree as well... phpbb is much better solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James Taylor
Sent: 09 August 2003 15:08
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] list proposal
I'd still vote for phpbb.
Then we could have
After updating asterisk from cvs today (08/07/03). i have been unable to Asterisk to
connect to mysql. cdr_mysql.so compile with no issues.
- And yes, i have mysql running on the same box as asterisk.
- And yes, cdr_mysql worked prior to update from cvs
Has anything change from cdr_mysql.c?
Thanks anyway
On Fri, 8 Aug 2003, WipeOut . wrote:
Hmm.. then i don't know.. If you were using the GS IP phones I may have had an
idea.. sorry.
ATT 957 analog sets
AJ
On Fri, 8 Aug 2003, WipeOut . wrote:
What phones are you using?
In my recent new Asterisk
[Posted here becasue your mail server is rejecting my direct reply to you.]
Hi Martin,
AFAIK SIP can run on both UDP and TCP but I have only seen it used
over UDP.. :)
To setup the GS phones you need to open up the following ports (If
its still set at the defaults)...
UDP/5060
UDP/5004
Hi,
I just recieved a new TE401P card from digium, I put it into a PC with
asterisk and did a fresh CVS update and compile.
However, when I load the driver (modprobe wct4xxp) for this card it
detects the card and loads the driver but then I keep getting all these
warning messages, such as:
Tried
You need to cvs update your Asterisk source tree.
Jeremy McNamara
Kelvin Chua wrote:
thanks for the tip...
i already downloaded and compiled the required versions.
now i'm having a couple of errors while compiling h323 from
asterisk/channels/h323/
this only happend today from cvs, here is the
Does anyone have Asterisk working with Iconnect here for incoming and/or
outgoing calls? If you would be so kind as to share with me the
configuration you have used, as I cannot seem to get my SIP service to
work although it does seem to be registered with the other end:
hm6*CLI sip show registry
Just got my new Budgettone phone, and I've got a couple of issues.
Most important, it doesn't seem to be querying for the time via NTP. I
put a sniffer on the line, and once it boots up the only outbound
traffic it generates is an attempt to contact a TFTP server, which is
programmed in as
From FWD, you can now dial 1010-666, followed by the Dutch toll-free
number or IAXTel-number you wish to reach, as you would have dialled it
from the dial-tone at FWD-number 42442.
I've tried dialing the following from my FWD-client (X-lite):
1010-666-800-0402
1010-666-0800-0402
Hi Dave,
On Tue, 2003-08-05 at 14:53, Dave Wilson wrote:
I can't seem to find any info on this anywhere on the web, except that BT
caller ID doesnt use the standard bellcore system in use in the US. So, if
anyone here in the UK is onlist and using the x100p successfully, please let
me know.
As far as I understand you do not want to open this DB while Asterisk
has it open.
Even though it is in standard db1 format.
A project _way_ down my list was to write some perl DBI commands that
allowed access to the asterisk db via the manager interface.
However, for now, I am looking at
At 15:13 10-8-2003 +0200, you wrote:
If i want to call the sjphone from the ata or call the ata from de sjphone
everything is ok.
My problem is ,that i can't call the voicemail or any other phone number
..as 600 for exemple from the ata or the jphone.
I don't know why but i looked after a long
That helps a lot. But now I get this message when I try to dial any
number
NOTICE[5126]: File pbx.c, Line 1089 (pbx_extension_helper): Cannot find
extension context 'default'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armand A.
Verstappen
Sent:
And further to Dan's message I will add that I was able to help because
a colleague and I are working on identifying all callerid variants with
a view to patching * to work with as many as possible.
If anyone has specific examples of countries/networks which don't
currently work or partially
i disagree, instead of thinking 'fallback' how about 'order' the agents
(by effecting the 'metric') so you 'target' the agent you want first
then if fail they go right to the next one in the 'ordered' list.
Brian West wrote:
leastrecent suffers the same fait as fewestcalls onlying ringing the
On Wednesday 06 August 2003 02:27 pm, William Flanagan wrote:
Why is this the case? Is it that it uses TCP? Is it the protocol?
Is it that it works better through NATs? Is it just more mature
than the SIP or MGCP implementation? What's the reason for it? A
protocol is a protocol in my
On Tue, Aug 05, 2003 at 11:07:22AM -0500, Steven Critchfield wrote:
On Tue, 2003-08-05 at 06:59, Samy Touati wrote:
Hi,
I've been browsing for FXO devices, and I'm really surprised at their
costs.
Why such devices are so expensive and somehow hard to get ?
Because most people that
I've canceled my Vonage service because of the requirement to prefix
every call with a 1.
As a side note, this doesn't sound like an overwhelming obstacle. In
fact, it sounds like a good ide.
Vonage has charged me $42 will refund this when they get my ATA186
back. I'm thinking I should keep
try the app. WaitMusicOnHold(time)
Just an example:
exten = 555,1,Answer
exten = 555,2,WaitMusicOnHold(30)
exten = 555,3,Hangup
--
Stefano
From: Dan [EMAIL PROTECTED]
How can I play Music On Hold on a channel for just a limited period of
time.
The Musiconhold application plays
My concern with wanting to split the list into so many mini-lists is that some will
die from lack of membership and it will mean that people with problems in that area
will be left in the cold.. Similar to how so many news groups just sit there unused
now days..
If there is really a need to
Maybe you could open a bug for it, and attach the specs / a link to
those specs? Also, I suggest you reply to this message:
That's a great idea. The other thing is that we have to detect polarity
reversal or we'll constantly be scanning for CID.
Mark
Hi Mark,
On Sat, 2003-08-09 at 17:44, Mark Spencer wrote:
The other thing is that we have to detect polarity
reversal or we'll constantly be scanning for CID.
Indeed. I'm not familiar with the internals of the hardware, could you
give some hints on how this could be achieved?
Then, looking
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