Hello, I'm trying to build a page system using a Dell Desktop PC optiplex
170L,
My sound card is working fine under /dev/snd/
exten = s,1,Dial(Console/snd/,20,A(trek))
exten = s,2,Hangup
But won't work! I get the following error
[Apr 27 11:44:46] WARNING[2950]: chan_oss.c:377 find_desc: could
I am little confused on load balancing, when asterisk server is also a sip
client.
Based on these,
XO Communications one of the largest US DID Provider, now offer SIP
Orignation Services for wholesale.
Verizon Communications One of the largest US Teleco, now offer SIP
Orignation Services.
That
You could use heartbeat http://www.linux-ha.org (or ultramonkey
http://ultramonkey.org). With this you set up a director that shares
the load to multiple servers. You can even set it to have consistent
connections so a originating IP will return the the same server. I
have hearbeat running on
I have a problem with SIP on my * box.
The * server with a private IP address is behind a NAT
modem-router with a public one.
I try to connect to a SIP provider which has a *
server with public IP but it doesnt works.
When I try making a call, the provider answers to the
SIP INVITE with
On Wed, 2006-01-18 at 15:07 +0100, amaury BOSSE wrote:
I have a problem with SIP on my * box.
The * server with a private IP address is behind a NAT modem-router
with a public one.
I try to connect to a SIP provider which has a * server with public IP
but it doesn’t works.
When I try
Port forwarding might work. My preferred method would be to bridge the
connection from the broadband modem to the * box, thus giving it the
public IP address. Then add a 2nd NIC to multi-home it and turn it into
the router/asterisk/dhcpd/firewall box. Run that connection to a switch
and out
Hi, I am pretty new to all of this but was able to
set up an asterisk server and have been able to
succesfully connect to asterisk with x-lite as sip client.
I have also connected asterisk to FWD (using iax2) and
to voipjet (also using iax2).
Now I am trying to connect asterisk to Stanaphone.
It
Hi, I am pretty new to all of this but was able to
set up an asterisk server and have been able to
succesfully connect to asterisk with x-lite as sip client.
I have also connected asterisk to FWD (using iax2) and
to voipjet (also using iax2).
Now I am trying to connect asterisk to Stanaphone.
It
: Jon Lawrence [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 14, 2004 10:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk to sip client
behindFirewall/NAT-
cancall but cannot receive calls ?
On Tuesday 14 December 2004 15:19, Shoval
: Robert Rozman [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 14, 2004 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk to sip client behind
Firewall/NAT-
cancall but cannot receive calls ?
Hi,
I hope I won't bother too much if I ask
on Firewalls...
Thanks in advance,
regards,
Robert.
- Original Message -
From: Shoval Tomer [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, December 14, 2004 11:30 AM
Subject: RE: [Asterisk-Users] Asterisk to sip client behind
On Tuesday 14 December 2004 15:19, Shoval Tomer wrote:
As far as I can remember I only opened sip and tftp ports for the phone.
For some reason (didn't look into it too much) the call stays with sip
and doesn't use RTP.
SIP is what sets up the session (ie it does session handling)
RTP is the
Hi,
I have following setup:
BT100 Firewall/nat 1 (www.ipcop.org) Internet Firewall/nat2
(Vigor) Asterisk .
I'd like to use BT100 as local extension to Asterisk. I've done simple setup
and BT100 can call Asterisk and place outgoing calls. However I cannot set
him to qualify,
- Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk to sip client behind Firewall/NAT -
cancall but cannot receive calls ?
Hi,
I have following setup:
BT100 Firewall/nat 1 (www.ipcop.org) Internet
Firewall/nat2
(Vigor) Asterisk .
I'd like to use BT100
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