[Al Bochter wrote on 15/03/2007 12:25 PM]:
So does anyone know when Voip-info.org will be back up?
There is a message on the list from James Thompson with the subject
voip-info.org status update saying it suffered a major hard drive
crash and should be back tomorrow.
Looking at the headers that
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Paul wrote:
Anyway, bumping him is not extreme at all. IIRC - some lists are setup
to automatically unsubscribe people after N days of delivery failures.
We only see this individually when we post but the list server is
probably getting this for
Hey All,
Our upstream provider requires the use of H323 and after several months
(6!) of having problems with OH323 I've decided it might be worth biting
the bullet and getting a cisco device that can gateway up to
approximately 50 calls from SIP to H323.
Would a 2500 or 2600 series do the job?
Hey All,
Our upstream provider requires the use of H323 and after several months
(6!) of having problems with OH323 I've decided it might be worth biting
the bullet and getting a cisco device that can gateway up to
approximately 50 calls from SIP to H323.
Would a 2500 or 2600 series do the job?
Paulo Adriano wrote:
I need some help from you. I´m using Isdn4linux with Asterisk and
incoming calls are working but anytime I whant to make an outgoing call
I also use isdn4linux for interfacing with my BRI line. I have a macro
set up for the actual dialling:
-- start
On Thu, 2 Sep 2004 14:51:10 +0200, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
Is there anyone who has experience with ISDN BRIDDI?
I'm currently using a BRI with ISDN4Linux.
I want to know if asterisk can distinguish between the different numbers?
Yes, it can differentiate between
Hey All,
Just wondering if there is a version of the G729 Codec available for Mac OSX? I can see almost
all the x86 infrastructures ...
Regards
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up
steve wrote:
I'm baffled. All I want is a simple 1x1 PBX to keep telemarketers from
ringing my phone.
If I can't get this working I'm having my phone disconected. lol
Surely the shrill tone would be good for keeping telemarketers away? ;)
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email
it can't find
it.
If you don't have Zaptel installed when you build Asterisk, it might not build
chan_zap.so.
On my system the asterisk modules are in /usr/lib/asterisk/modules/.
Regards
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office
users 0 Aug 19 16:28 tmp.raw
-rw-r--r--1 darryl users 23411 Aug 19 16:28 utt.feats
I've tried setting a number of different voices. Some don't give any audio and some only seem
to give a small 'blip' of sound.
Does anyone have any ideas?
Thanks in advance
Darryl
--
Darryl
Simon Brown wrote:
In my zapata.conf, I have
callerid=unknown
That doesn't look right to me. Try:
callerid=Unknown
Cheers
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up to the whole There is more
be happy to take
them :)
Cheers
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up to the whole There is more than one way to
do it slogan, you have to give someone a swiss army chainsaw
Woody?
Thanks in advance
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up to the whole There is more than one way to
do it slogan, you have to give someone a swiss army chainsaw
Sebastian Sporleder wrote:
Darryl Ross wrote:
Assuming that the debian packages are not compatible, which version of
Festival do I need? The Wiki page mentioned above says to grab the
tarball of 1.4.3, which is no longer available from the website. Only
1.95 is available. Will that work? Does
, then assign the template to the
channel. Eg, the channel = line needs to be the last line for that configuration. Try putting
the callerid = before the channel =
Regards
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
at the default files.
Hope that helps.
Regards
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up to the whole There is more than one way to
do it slogan, you have to give someone a swiss army chainsaw
or not, but couldn't you do a
#include vars.conf
in the relevant places in extensions.conf, oh323.conf and phones.conf
and then define the variables in vars.conf ??
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED]
Phone: 08 81228363
Office: 08 81226361
If you want to live up
. And again, Asterisk supports this.
Oh, so I how does Asterisk knows when to start dialing out the
numbers, if there are no rules?
Have a look at http://www.voip-info.org/wiki-Asterisk+Extension+Matching
Regards
Darryl
--
Darryl Ross
Senior Network Engineer
OEG Australia
Email: [EMAIL PROTECTED
Thanks for the reply Freddi,
The problem I am having is I'm trying to work out how to link the CDR
records into a single 'call stream', rather than having separate records
per machine the call passes through.
I had the same problem. One solution is to include the ip-adr or uname
of the gateway
Hey All,
We are running a small SIP/IAX termination service at the moment
(planning on growing it) with 2 asterisk machines. One terminates the
SIP/IAX calls from our customers and one is our gateway to our upstream
provider. Both machines are logging CDR data to the same postgres table
using the
Hey All,
Having a bit of a problem with a Wildcard X100P card. When I try to make
an outbound call using the card, it picks the line up and then only
dials a single digit. I've confirmed it's only dialling a single digit
by listening on a phone plugged into a parallel socket.
Incoming calls
Mark Elkins wrote:
going to i4l means... incoming sound sometimes gets interpreted as DTMF
- and when your caller humms a '#' - transfer kicks in... Outgoing DTMF
simply does not work. (Don't do i4l!)
It doesn't? Funny, no one must have told my NetJet card that -- it works
fine!
Regards
Darryl
Hi List
I have also saved a copy, available at
http://mirror.afoyi.com/asterisk/, which should be very quick for anyone
in Australia.
Regards
Darryl
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Steven Critchfield wrote:
On Thu, 2004-07-08 at 09:16, [EMAIL PROTECTED] wrote:
I do not recall telling anyone 6 weeks, My book located at
www.saww.net/asterisk/ is being shipped to everyone that has not received
their orders as of next week. maybe next time you should get your facts
straight
HI Mike,
2) I could add an isdn card to the Linux box. This seems to me to be the
cleanest solution, I'd make my firewall also be the asterisk server, and
hopefully gain some control of tcp flows that way to more highly
prioritize voice traffic
+apparent simplicity, maybe fax support
-s it
Hey All,
I'm still (since April) having problems getting RxFax to work over an
ISDN4Linux channel. Just wondering if anyone has had any luck getting it
to work?
I have done a CVS update today (about half hour ago) and made sure I have
the latest version of spandsp according to Steve's website
Hey All,
I've been trying to get SpanDSP / RxFAX to work in order to set up a
soft-fax machine on my asterisk system.
I have asterisk CVS-04/08/04-10:06:15 and spandsp 0.0.1. This is on a
Fedora Core 1 with 2.4.26 kernel.
I have tried to look for a newer version of the spandsp stuff, but
in
sip.conf.
Regards
Darryl
Darryl Ross wrote:
Hey All,
I am setting up a network of Asterisk servers using IAX2. I am wondering
if it is possible to disable the handoff feature?
At the moment I have 4 asterisk machines, 3 are at SOHO offices and 1 is
centrally hosted in a data centre. In addition
I'll try again using my subscribed address... Apologies to the moderators...
---
Hi All,
I've successfully got SIP and IAX2 working on Asterisk using X-Lite as a
SIP phone and talking to a remote Asterisk over IAX2 using G729.
I'm now trying to get an ISDN BRI connection working. I have a
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