http://www.voip-info.org/wiki/ here got alots of example but you need to find it. You can start with http://www.trixbox.org/ that install everything. Good luck!
On 9/11/06, Panagiotis Zikos <[EMAIL PROTECTED]> wrote:
Hi all, I am new in the asterisk company. I need to set up a small voip syste
Hi all, I am new in the asterisk company. I need to set up a small voip system for about 60 phones ( a small enterprise organization). The system must support voip calls (calls inside the enterprise) but must be able to send calls over isdn (24 channels). Thus the asterisk server must oper
Rudolf Ladyzhenskii wrote:
Is there a better than include way to route calls between contexts?
[internal-extensions]
include => from-sip
include => iax-users
[from-sip]
include => internal-extensions
[iax-users]
include => internal-extensions
Doug
_
Rudolf Ladyzhenskii wrote:
Hi, all
I have couple of SIP phones and they are in [from-sip] context.
I also have an IAX2 phone. I have put this one in [iax-user] context.
I want to make calls between SIP and IAX2 phones. If I put them all in
same context all is fine, however when they are in di
Hi, all
I have couple of SIP phones and they are in [from-sip] context.
I also have an IAX2 phone. I have put this one in [iax-user] context.
I want to make calls between SIP and IAX2 phones. If I put them all in same
context all is fine, however when they are in different contexts they will
n
http://www.google.com/search?q=asterisk+iax
> -Original Message-
> From: Rudolf Ladyzhenskii [mailto:[EMAIL PROTECTED]
> Sent: Saturday, July 16, 2005 5:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Beginners question --
Hi
As a general note: if you want to start a new thread, don't reply to an
existing message: Write a new message. Otherwise your message will
appear as a reply and be buried somewhere down a thread that nobody
cares about.
On Sat, Jul 16, 2005 at 08:27:56PM +1000, Rudolf Ladyzhenskii wrote:
> Hi,
Hi, all
Can someone point me to a good resource on IAX?
I am trying to do two things at the moment:
1. I want to learn
2. I want to conenct MozPhone to my * (wiki does not have much on it)
3. I want to connect two * servers at different locations.
I have looked at asterisk wiki and dis not find
> [2203]
> type=friend
> username=2203
> secret=2203
> host=dynamic
> defaultip=192.168.0.2
> dtmfmode=inband
> canreinvite=yes
Add here:
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=inband only works with ulaw (g.711), so better use a different
setting here.
> the console on * when r
On Monday 15 December 2003 15:33, Jon Creasey wrote:
> Hi all,
>
> New user to asterisk having just got it compiled and installed.
>
> Running with no digium hardware (yet) and no soundcard in asterisk
> box.
>
> Problem is using the sample configs with a sip phone added as
> follows
>
> [2203]
> t
Hi all,
New user to asterisk having just got it compiled and installed.
Running with no digium hardware (yet) and no soundcard in asterisk box.
Problem is using the sample configs with a sip phone added as follows
[2203]
type=friend
username=2203
secret=2203
host=dynamic
defaultip=192.168.0.2
d
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