--Request For Bug Reports--
I'm working on the next release of IAX Phone. Please let me know what, if
any, issues you who use it may have run into. I hope to be able to release
a new version in the next two weeks.
Some fixes/features:
- Conferencing
- Proper handling of 'qualify'
- Intercom
The Hype Is Back
Ok. Everybody who has ever been to a trade show knows that the
majority of what you hear is marketing hype. That's to be expected. But
usually you expect to see that on the exhibit floor or in companies'
hospitality rooms. Unfortunately, the VON show seems to have deci
If you are using IAX between the servers, you can simply add a codec line to
the iax.conf file to provide the specific codecs you want to use in the
order you want to use them. Here's a sample
disallow=all
allow=ulaw
allow=alaw
If you want GSM as a fall back, tack on:
allow=gsm
The situation f
Not currently. There is a bounty for the development of working Wildcard
drivers for Free/Net/Open BSD. Care to write them?
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Moran
Sent: Monday, March 29, 2004 6:46 PM
To: [EMAIL PROTECTED]
Subj
Call your local independent computer
retailer and find either a retired PIII box or a low end Celeron box (I buy
them in single-unit quantities for $300 each). Order the X100P from Digium.
Configure. Call. Repeat if necessary.
What kind of ideas are you looking for?
Thanks,
nd.
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven M. Sokol
Sent: Monday, March 29, 2004 11:36 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source
Gro
The VON show has started off with a number of interesting announcements.
First among these is a big announcement from Pingtel that they have created
a not-for-profit corporation called SIPFoundry. This new company includes
Pingtel (which has recently open sourced their SIPExchange PBX), Vovida and
Or, you could apply my patch, that I've been upgrading on the asterisk bug
site.
Check http://bugs.digium.com/bug_view_page.php?bug_id=752
Steve (and anybody else who may know about this code),
I have the code for your privacy enhancements compiled and installed. My
one question is, how do I
I have 1.0.3.81. How do you execute the transfer?
Thanks,
Steven
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Brown (CV)
Sent: Monday, November 03, 2003 8:58 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Transfer from Grandstream BT100?
:[EMAIL PROTECTED] On Behalf Of Steven M.
Sokol
Sent: Tuesday, October 28, 2003 10:39 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP Calls Don't Properly Connect (Continue
Ringing) After CVS Update
Hi,
I just updated my image from CVS, compiled and reinstalled it. Now
whenever I make
Hi,
I just updated my image from CVS, compiled and reinstalled it. Now
whenever I make calls from my Grandstream phone to my X-Lite soft-phone,
the call does not complete correctly.
Scenario:
1. I take the GS off hook and dial 1100 (the extension of the
x-lite phone).
2. The x-lite ph
-Users] Compiling gastman under Win32
Download binary with java , works fine with 2000 + Xp
regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven M.
Sokol
Sent: Friday, October 24, 2003 4:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
Can anybody tell me what I need to have in order to make/compile gastman
on Windows using VC++? Or do I want to download and install gcc for
Windows? I have never worked with anything designed to be
cross-platform compilable.
Thanks,
Steve
___
Aster
Hi. I have just finished Googling the list archive for information on
doing CTI with Asterisk. I never found a definitive "here's how it's
done" answer, but I did see a number of postings indicating that it was
at least a possibility. (For those of you who already know about CTI,
please pardon t
> hell i just got a quote today for D240JCT-1T1 for $4500ish
Yup. That's the one with the fancy DSP on-board that supports echo
cancellation ("continuous speech processing") and is generally used for
speech recognition apps.
By the way -- has anyone in the * community looked at adding ASR to th
Coming from the [evil] Dialogic world (where even the drivers cost
money) the prices Digium is charging seem very reasonable. New
single-span Dialogic T1 interfaces cost at least three times ($1225 USD
was the best price I could find on the D/240PCI-T1) what the single span
Digium card costs. THE
Has anyone tried installing * on a box with two eth interfaces which is
acting as a NAT box? I have only one IP at this point and I would like
to get * working without all of the NAT issues. My idea is to run * on
my gateway (which is also running the firewall and masquerade services).
All of my
Strange. I have a simple extension set up to do some Festival testing.
(Festival 1.4.3 /w Asterisk patch). My extension looks like:
exten => 1239,1,Answer()
exten => 1239,2,Festival(Welcome to the asterisk system!)
exten => 1239,3,Wait,1
exten => 1239,4,Hangup
Some times it work right (sounds c
Hi,
NOTE: Sorry if this gets duplicated. I sent a copy prior to joining
the list.
Newbie here with a repetitive question. How do I make the Festival TTS
system work w/ *? I have festival installed (it came w/ RedHat9) but *
complains:
WARNING[1226530096]: File app_festival.c, Line 327 (fes
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