[asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

[Rich Adamson]

Elpidio Ramos wrote:
> 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

If you have two working nic's, then when the soft phone is on the inside 
of the network, it should register with the IP address of the inside nic.

When the soft phone is on the outside (eg Internet), then it should be 
registering with the IP address of the outside nic.

Any other combination is going to give you problems and particularly if 
you are using a firewall. The problems will be associated with basic 
layer-3 stuff and nating.