[asterisk-users] using SIP to connect remote other VoIP server

[Elpidio Ramos]

Hi,
   
  This is a sample file I am currently using on my server.
  My server has a public IP address and an internal IP address (duan NIC).
  It runs Fedora Core 3 running iptables firewall already configured with ports 
  4569, 5060, 10000-20000 open (udp and tcp)
   
  [general]
context=default
allowguest=no
realm=your.hostname.ext
bindaddr=0.0.0.0
bindport=5060
externip=your.server.ip.address
srvlookup=no
maxexpirey=3600
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
musicclass=default
language=es
rtptimeout=120
rtpholdtimeout=300
useragent=asterisk
localnet=10.10.10.0/255.255.255.0
rtcachefriends=no
qualify=yes
   
  [311]
type=friend
regexten=311
username=311
secret=311
callerid="User on extension 311" <311>
host=dynamic
nat=yes
canreinvite=no
   
  [312] 
type=friend
regexten=312
username=312
secret=312
callerid="User on extension 312" <312>
host=dynamic
nat=yes
canreinvite=no
  

tengulre <tengulre at megamail.com.cn> wrote:
              How to using SIP to connect remote other VoIP server? is it only running one line voice if I registered a one SIP account?
     anybody can give me some sample configuration files? thanks a lot!
   
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                  Elpidio Ramos 
President 
RM International Services SA CV 
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