You must first purchase a license from the G723 patent holders. A
license costs about US$10,000.
On Tue, 2003-05-27 at 01:52, [EMAIL PROTECTED] wrote:
hi!
From where do I get the source code for G.723 for asterisk. And how do I
compile it (is there any specialities other that make make
I've a problem with my X100P card.
I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I
call an PSTN phone number, some digits are duplicated, so I'm unable to
call the right person.
Not very clear ? I'll try to do better (sorry, I'm french...)
example :
I use ohphone (with
Fabrice Tereszkiewicz wrote:
I've a problem with my X100P card.
I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I
call an PSTN phone number, some digits are duplicated, so I'm unable to
call the right person.
Not very clear ? I'll try to do better (sorry, I'm french...)
Though I think .mp3 encoding is an interesting idea, even though
inefficient at some levels. One benefit is that there are dozens of
.mp3 players on the market, with extremely sophisticated library
functions. Being able to archive calls in one of the .mp3 library
programs might be of some
I've had the same thing happen, only on the single port T1 card and a
channel bank, and one of the FXO channels also having a phone attached
elsewhere...
I just wound up putting that channel in a different context and running
Exten = s,1,Hangup
(I'm just using the line for outbound dialing)
On Tue, 2003-05-27 at 11:24, John Todd wrote:
Though I think .mp3 encoding is an interesting idea, even though
inefficient at some levels. One benefit is that there are dozens of
.mp3 players on the market, with extremely sophisticated library
functions. Being able to archive calls in one
Sorry, I seem to have responded to the wrong thread there. Should have
been part of the Duplicate numbers thread.
On Tue, 2003-05-27 at 12:36, Joe Antkowiak wrote:
No popping/bad audio on this one, clear as can be, asterisk just decides to
pick up the channel after about a minute and use the s
Can anyone share any links regarding packages to do Call Detail Record (CDR)
analysis from the CDR Master file?
Login-distance reconciliation, billback, and data presentation are three primary
areas of interest.
Thanks in advance for your help!
--Nick
--
Nick Eggleston
Consultant
Data
I have to get some ModTaps (converters) too.
My question, is do they need to be Master, Slave or PBX type ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut .
Sent: 27 May 2003 16:21
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] TDM400P
Hello there
I have a serious issue with the AVM Fritz PCI V2
I have the following setup and the problem is, that the kernel freezes hard after
about 16 hours. The second problem is, that the S-Bus gets jammed as well, so
you can't even use a analog phone! on the NT
Kernel 2.4.21rc2 with ACPI
On Tue, 27 May 2003 11:48:19 -0700 (PDT), Brad Bergman wrote:
I think SayDigits will say anything for which there is a sound file in the
digits directory. So if you put a S.gsm file there, SayDigits,S98 should
say Star Nine Eight. I realize that's not exactly what you're looking
for.
Close,
On Tue, 2003-05-27 at 18:51, Steve Bourg wrote:
I noticed someone mentioning PGSQL in this mailing list. I found an older
message about it being used for CDR. Is there any info about the extent
to which Asterisk and PostgreSQL can interact? I see that there is an
Application in the source.
Ah, now has anyone got a gsm of thevoice for start and hash ??
On Tue, 27 May 2003 17:06:30 -0700, Jim Gottlieb wrote:
On 2003-05-28 at 09:57, Gary ([EMAIL PROTECTED]) wrote:
how the heck can you have a asterisk(star).gsm file ???
I was able to create one with
touch \*.gsm
so this should
Turn on h.323 debug and then possibly h.323 trace 4 (for the hardcore)
in the Asterisk CLI
Unfortunately a tcpdump tells me nothing.
Jeremy McNamara
Nick wrote:
I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk.
Has anyone gotten a Webswitch running? When I try to
Heck, I am not that fussy !!
Actually, if we could actually get festivel to be fully understandable
and using thevoice I think we could all be a lot happier :-)
On Tue, 27 May 2003 20:52:58 -0400, Richard Alexander wrote:
I suspect that the (American) voice would have called the hash pound
in
On Tue, 27 May 2003 17:06:30 -0700, Jim Gottlieb wrote:
On 2003-05-28 at 09:57, Gary ([EMAIL PROTECTED]) wrote:
how the heck can you have a asterisk(star).gsm file ???
I was able to create one with
touch \*.gsm
so this should work. I doubt asterisk is doing any globbing.
Title: Message
My asterisk machine
has a dual port adaptec ethernetcard in it, one port configured with a
public IP and the other with a private (10.0.0.5).
To get my Cisco 7960
phone to sucessfully registerwith
asterisk, I have to configure
bindaddr to 10.0.0.5 and not the default
With a X100P, I would be able to get the user's caller id here in Japan?
Isamar
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Hi All,
Is is possible to bridge two outbound calls via iconnecthere, so that two
people can be connected. If so
what do I need to put in the extensions.conf. Thanks for all the help.
Pradeep
_
Himalayan holiday! Waiting to be
1.
It would be very helpful if the voicemail module included the caller-ID
info (name and number) in the Subject line. Otherwise, looking at the
list of messages is like looking at an INBOX without a From column.
I patched app_voicemail.c quickly, but it would be even better if this
were a
There seems to be no way to distinguish between Out of Area and
Private/Withheld in extensions.conf. I see in the zaptel source code
that the distinction is available when the data is being read from the
line. Since those really are different situations, it would be great to be
able to
Hi Klaus,
Yes - el-cheapo BRI means the cheapest ISDN BRI card I could find at the
local hardware pusher, absolutely no-name locally (Malaysia) manufactured
Hisax type 20 card from isdn4linux point of view and yes - chan_modem_i4l.
I primarily bought the card just to check out how Asterisk
3. Dual MB won't help much in pure telephony.
In pure telephony, you are basically dealing with serial line
IO. A T1 is little more than I long distance serial line. 8 T1s
is just 11.7megs per second each way, or 23.4 megs in and out.
Not too much for a good machine to
On Tue, 2003-05-27 at 22:52, Miguel Cruz wrote:
2.
When calls come in to an ATA-186 with a garden-variety caller-ID display
attached, the date and time always show up as 01/01 12:00. I am not sure
whether this is because nothing is getting sent, because a mis-formatted
date is getting sent,
Hi All,
I wonder if Asterisk can be setup as a Call Centre and do Call Logging? I
want to log all incoming and outgoing calls. What sort of logging mechanisms
does it supports?
Any help is greatly appreciated.
Thanks in advance!
Regards,
Denzel
___
Hello ,
I am a newbie to * and have just been able to call
a sip User Agent on a different machine thru *. I was
trying to set up conferencing between 3 sip useragent
on different macines at my worplace but was not able
to figure out the procedure. I made the changes in
meetme.conf and
greetings!
We are planning to deploy Asterisk PABXs between several offices. The
connection between the offices are going to be on IAX channels. Currently we
use G.729 for the IAX channel. But for sending FAXes between offices over
IP, this method is not gonna be satisfactory since with this
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