Try Vovida's Vocal, i think it does it.
Mireia Munoz de jesus wrote:
Hi!
I am looking for a software that can work as h.323 - sip gateway other than
asterisk and free. Someone can help me?
Thanks.
Mireia
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As you see, * generates no busy tone, it hangs up the channel. It's your
client which generates the tone. This is not something to be done from *.
Regards,
Doichin Dokov
Ryan Courtnage wrote:
Hello,
I find that when 2 extensions are connected, and one of the extensions
hangs up on the call,
Yes that's right, it's the TDM400P which generates it.
Ryan Courtnage wrote:
On 29-Mar-04, at 3:23 PM, NetOne Administrator wrote:
As you see, * generates no busy tone, it hangs up the channel. It's
your client which generates the tone. This is not something to be
done from *.
Thanks,
So
Thomas Schroeter wrote:
Hi,
I just got started with Asterisk. Installation was OK, no errors.
But how do I activate the IAX and SIP channels now? I loaded the
modules, but nothing happened, there's no connection to the
relavant ports.
Anything I forgot to do...?
Yes, you forgot to read
Our telco here in Bulgaria has this option - you can get what number of
lines you want on a primary (E1 here).
You choose how much lines - from 1 to 30 - go to the primary. Price is
according to number of channels, but signalling is primary!
Check out at your telco if this is not possible out
[EMAIL PROTECTED] wrote:
SSH
Nice :)))
On Mon, 8 Mar 2004, hank smith wrote:
is there a program that I can install on my linux box so I can configure the
pbx from the internet from my windows box so I don't have to work with
config files?
thanks
hank
- Original Message -
From:
CW_ASN wrote:
People:
1) Some guy wrote a app_dial modification to start to count time when answer
arrives. Interested? (thanks to Luciano!)
AbsoluteTimeOut counts time since this statement is executed... If you have
a long ring time (without answer), it is counted!
With the proper patch, the
CW_ASN wrote:
This is wrongs. It's me who wrote the patch, it's available in CVS
Are you Klaus? If you're not Klaus, you wrote another patch. If you're
Klaus, as you see, works in that way.
Nopez i'm not
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Soren Rathje wrote:
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
HMM - This wont work :(
exten = 10,1,Dial(SIP/hha1,20,S(10))
exten = 10,2,VoiceMail,u10
exten = 10,102,VoiceMail,b10
When did you checkout your version of Asterisk from CVS ??
This feature
CW_ASN wrote:
Are you Klaus? If you're not Klaus, you wrote another patch. If you're
Klaus, as you see, works in that way.
Nopez i'm not
In that case, exists another patch from a guy called Klaus. I'm using this
patch since Dec2003.
Maybe helps, I don't know, but this is other
v.3.0 works fine too
James Coberly wrote:
v2.16.2 ata18x
Works Fine for me.
- Original Message -
From: Erick Weber V. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 03, 2004 1:18 PM
Subject: [Asterisk-Users] Best ATA 186 Firmware
Hi:
Someone know wich is the best
If you are not using Digium hardware, then you don't need libpri and zaptel.
Asterisk WILL build on its own.
SamW wrote:
I want to build a stable asterisk to run, if some one can guide through
how to compile will be useful. Currently available documentation do not
show any good information about
Angel Gabriel wrote:
I have 5 BT phone lines coming into my office. We use four for
international calls, and one for local/mobile calls. We have just
obtained another call carrier, and now we would like to be able to
make calls from any phone to any carrier, without having to remember
what
Kelly Murphy wrote:
Follow all the instructions on
http://www.asterisk.org/index.php?menu=download You still need to
checkout libpri and zaptel. If you want more information checkout
http://www.voip-info.org This is the main repository if information on
Asterisk.
No you don't need to
FreeBSD asterisk port is *NUTS*
Don't use it!
Asterisk compiles just fine on BSD, if you are using 4.x-RELEASE, and
not using chan_h323, chan_oss, zaptel libpri.
Darren Wiebe wrote:
Sorry to just come on line now. Have you tried the FreeBSD port?
net/asterisk is the place to look. It
What is your * app?
What should the frontend do?
Greetings,
Doichin Dokov
reseaux wrote:
Dear ALL
i need to develop a web frontend for my * app i need only manage data from
MySQL db, i will pay to develop it (not much :-) )
Thanks in advance
Dimitri
Hello!
Chan_H323 is not working under FreeBSD, at the moment.
Dmitry Mishchenko wrote:
Hello!
I've got the latest asterisk sources from cvs dated Jan 26 2004. And was
trying to build it under FreeBSD 4.8
Main part was built fine.
I face problems only while compiling h323 channel.
Its
Do not compile openh323 and pwlib from cvs.
Use the versions described in the README of chan_h323 so (in
channels/h323 dir).
Good luck!
Doichin Dokov
Mike Bentley wrote:
Hi can anyone help me with this
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH
Hi all!
If i purchase the G.729 codec for *, can Asterisk use it for convertion,
or just pass-through only?
I need to be able to convert from G.729 to iLBC (or GSM maybe) and vice
versa. Is it possible with *?
Greetings,
Doichin Dokov
NetOne - Bulgaria
a good QoS control, which I now have on BSD thanks to ALTQ queing.
Any suggestions comments?
Doichin Dokov
NetOne - Silistra, Bulgaria
Tilghman Lesher wrote:
On Sunday 11 January 2004 16:18, NetOne Administrator wrote:
I'm trying to set up Asterisk on FreeBSD 4.9 to route
calls to H.323 GK.
I have
PHi all!BRBRI'm trying to set up Asterisk on FreeBSD 4.9 to route
calls to H.323 GK.BRBRI have installed asterisk using the
ports.BRBRIt seems to be running OK, but when i try to dial through
h323, it segfaults.BRI'm using X-Lite as SIP client, i have set up my
h323.conf:BR[general]BRport =
Hi all
I'm trying to set up Asterisk on FreeBSD 4.9 to route
calls to H.323 GK.
I have installed asterisk using the ports.
It seems to be running OK, but when i try to dial through
h323, it segfaults
I'm using X-Lite as SIP client, i have set up my
h323.conf
[general
port = 1721
bindaddr =
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