[Asterisk-Users] NAT and sip issues

[G.Marshall]

The rtp audio is going phone to phone, not via asterisk.  This is one of
the reasons I am trying to set up SER with Asterisk.

>> I have an asterisk server behind NAT - no audio on the test external
>> calls
>> I
>> have tried making so far.
>>
>> Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No
>> solution
>> evident from there, sounds like I have case 9. I would have thought that
>> all I
>> would have to do is port foward and have the external IP on the asterisk
>> server,
>> which I have done
>>
>> I have fowared 5060UDP, 8000UDP, and  35000 to 37000 UDP to the internal
>> IP
>> (192.168.1.115)
>>
>> I have put 35000 and 37000 into the rtp.conf as the start/end ports
>>
>> extracts of sip.conf:
>>
>> externip = 60.234.129.154
>> localnet = 192.168.1.115
>> localmask = 255.255.255.0
>>
>>
>> [88]
>> type=friend
>> secret=**********
>> dtmfmode=rfc2833
>> nat=yes
>> host=dynamic
>> canreinvite=no
>>
>>
>> Trying with xlite at the other end
>>
>> Registered ok, can dial both ways, just no audio at all.
>>
>> In the log of xlite (cant see it at the moment as im not vnc'd in at the
>> moment)
>> it showed the xlite machines private IP address on some of the
>> transactions that
>> were logged.
>>
>> The client has a dynamic IP address so cant really be specified anywhere
>> in the
>> xlite configuration, I am also not sure on all the different firewall
>> types.
>>
>> I was under the impression that there was no need to configure any
>> portfowards
>> at the sip softphone end.
>>
>> I will hopefully be using xlite or similar from a location with a very
>> locked
>> down firewall environment. I want to check all works on a normal nat
>> router
>> before trying it behind the nasty nat/firewall at this location.
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