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
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
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
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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