[asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

[Elpidio Ramos]

This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux:
   
  My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC)
   
  I got the ports open in fedora core 3 (5060 and 10000 thru 30000) for both interfaces.
   
  I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP. 
   
  When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop.
   
  This is my sip.conf file:
   
  [general]
context=ramosoft  
  allowguest=no
  realm=ramosoft.com 
  bindaddr=0.0.0.0  
bindport=5060   
srvlookup=yes   
pedantic=yes   
tos=184    
tos=lowdelay   
maxexpirey=3600   
defaultexpirey=120  
disallow=all   
allow=ulaw   
allow=ilbc   
allow=gsm  
musicclass=default  
language=es   
relaxdtmf=yes   
rtptimeout=60   
rtpholdtimeout=300  
useragent=RamoSoftPBX  
regcontext=ramosoft
localnet=10.10.10.0/255.255.255.0 
rtcachefriends=yes   
   
  [authentication]
   
  [311]
type=friend
regexten=311
username=311
secret=311
callerid="Elpidio Ramos" <311>
host=dynamic
nat=yes
canreinvite=no

  Is there anything I am missing here to get two way voice?
   
  Thank you  in advance all
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