asterisk users - May 2005 - NAT and sip issues

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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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.
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users@lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

G.Marshall wrote:
> 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 thought that canreinvite=no was supposed to force the audio to go via
asterisk?
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