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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