[asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

[Mark Coccimiglio]

Try setting in sip.conf:

nat=route

This tells asterisk to send all responses back to where the inquiry came 
from rather then from the info contained in the sip packet. 

Good luck,
Mark Coccimiglio
IS Director
Payroll Services Hawaii, Inc.
http://www.psh-inc.com

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
>
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