I know it's OT here ... but anyone looking for a Radius server should
consider getting Radiator - http://www.open.com.au/radiator/.
It's not free, but at a very decent price you get the full source code
(it's written in Perl), but you seldom will need to tweak the code, as
it is flexible by
On Tuesday 03 August 2004 19:44, Chris Shaw wrote:
QoS isn't going to help you get to talk in a crowded CSMA/CD network.
I might be misunderstanding you about QoS, but I know for a fact that it
does help greatly because whether you use DSL or Cable, your bridge device
(it's not a modem no
Andres wrote:
The issue I'm having problems with is having the SPA-3000 automatically
forward all incoming PSTN calls to the Asterisk mainmenu context (or
ext I guess).
Configure an auto-dial number in the SPA to that it corresponds to
something in the mainmenu context. Like:
At 07:08 PM 8/3/2004, you wrote:
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
Gary
- Original Message
On Tue, 2004-08-03 at 14:17, Mike Benoit wrote:
My SPA-3000 finally arrived and I'm trying to get the FXO port on it to
work as if it was a X100P card as far as Asterisk is concerned.
I have Asterisk dialing out over the SPA-3000 FXO port no problem.
The issue I'm having problems with is
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On Tuesday 03 August 2004 08:06 pm, Andrew Kohlsmith wrote:
And yes I know all about huge queues... The cure for that (at least with
DSL) is to get a Sangoma S518 -- it's a PCI ADSL modem with drivers for
everything... I just prioritise packets
AJ Grinnell wrote:
Is anyone else having problems with Sipuras not being able to re-register to
Asterisk after applying the cvs update last night? Just curious if I need to
roll back or take all of my Sipuras out back and beat them.
I just updated to CVS a few moments ago and now one of my
Title: Logging into Multiple Call Queues on two * Servers and Voice Mail option.
Dear All,
I have two objectives that I need to meet;
1. I need to be able to log into two separate call queues on two different Asterisk servers, servicing two data centers. I seem to have problems
Does anyone know how to do the following:
1. Caller calls in
2. Asterisk answers.
3. Asterisk rings nominated extensions
4. Caller keys in certain digits while extensions are ringing
5. Caller is directed to another extension based on the digits keyed in
I can achieve this if I have Asterisk
- Original Message -
From: Steve Szmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 6:16 PM
Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL
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On Tuesday 03 August 2004 08:06 pm, Andrew Kohlsmith
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 5:06 PM
Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL
On Tuesday 03 August 2004 19:44, Chris Shaw wrote:
QoS isn't going to help you get to talk in
[EMAIL PROTECTED] (Chris) writes:
The thing that really kills you on the ISP end is RED... it may be great for
large traffic but it just KILLS voip... and there's not thing 1 you the
customer can do about it... :(
Interesting and somewhat disheartening. RED was really meant to put
Anybody use them... I signed up for $20 to see how there system works..
They're at $.02 per minute for US Termination and their other ITX rates
aren't too shabby.
Sadly my IAX registration is rejected... maybe a glitch, wondering if
anyone's had a similar issue.
Luke
Hello
All,
Does anyone know the
state of AstMan? I found some information and source code in the archive
but it is from November of 2003. There is mention of a lgpl release but
nothing else after. I would like to code in some of the features that were
lacking like setting this in system
- Original Message -
From: Steve Szmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 7:04 PM
Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL
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On Tuesday 03 August 2004 09:47 pm, Chris wrote:
The
Is there any way to have asterisk set CID to Private or Unknown instead
of asterisk when a call comes in that is either blocked or not available?
While the number is unavailable either way, it would still be nice to
know if the number is being withheld or if it is in fact not available.
Thx,
Hi
There is a device called a parlay made by a crowd called
voxtream which will route the ISDN calls based on the DID
and/or the callerid, before the call is answered.
It would be nice if this feature could be done in Asterisk
as well, but at this point in time, it first answers the
call.
- Original Message -
From: Chris [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 7:48 PM
Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL
- Original Message -
From: Steve Szmidt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday,
Nevermind... the microfiche username was wrong... PBAK
Works... but call quality is a little weak...
This mailbox protected from junk email by MailFrontier Desktop
from MailFrontier, Inc. http://info.mailfrontier.com
-Original
yes change your dial macro to use SetCallerID and SetCIDName
and it will use that instead
On Tue, 03 Aug 2004 19:50:26 -0700, Trevor Peirce [EMAIL PROTECTED] wrote:
Is there any way to have asterisk set CID to Private or Unknown instead
of asterisk when a call comes in that is either blocked
On Tuesday 03 August 2004 12:07, Chris Shaw wrote:
Are you using the double ## transfer patch or just the regular
single # that comes with CVS?
Hi Chris,
Where to get the 'double ##' transfer patch? We are having the same
problem and I was thinking of a similar patch.
Thanks,
-- sudhir
On Wed, 2004-08-04 at 01:46, john lawler wrote:
Finally, can I turn off the '#' to transfer, since we're using the
hook-flash (albeit manually) instead? ISTR an option to do this but have
spent the morning trying to find it again unsucessfully...
I think you might want to look at the 'T'
- Original Message -
From: Sudhir Kumar [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 8:46 PM
Subject: [Asterisk-Users] Re: problems with'#' transfer after hold
On Tuesday 03 August 2004 12:07, Chris Shaw wrote:
Are you using the double ## transfer patch or
Title: Very decent book - VoIP Telephony with Asterisk
Just received my copy of this book today. Based on a cursory examination, VoIP Telephony with Asterisk (VTwA) by Paul Mahler looks like a very decent book for anyone starting out with Asterisk. More knowledgeable readers may also
John Harragin
I have a PRI comming into each of 2 buildings. How do I
redirect an incomming call on PRI_A of particular DIDs to
arrive at PRI_B instead?
Not a real standard way of doing it in my experience.
Each switch type seems to have a different mechanism. For instance the
give me a call tomorrow i could help you with your issue
52(55) 150054 54
GonzaloWayde Nie [EMAIL PROTECTED] wrote:
Wayde Nie wrote: I can get a Cisco MC3810 with a mixture of FXO and FXS ports, the MC3810 comes with a built in Ethernet port and I believe it does SIP too... Will this mean that I
Hi all,
Don't know if anyone can help me. We just set up a CAC Access
Bank 1
with Asterisk. Everything works great except, when we ring a Zap
interface, the analog phone does not actually ring. The light blinks,
and if you answer it you are connected to the person, but the actual
phone
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