[asterisk-users] Call bridged, but no sound

[Norbert Zawodsky]

Brian Candler wrote:
> On Thu, Oct 12, 2006 at 02:26:16PM +0200, Norbert Zawodsky wrote:
>   
>> As soon as the connection is up and the receiver is lifted on both
>> sides, the leds of the DSL Modem between Asterisk and my ISP, and the
>> leds of the switch between Asterisk and the SNOM phone start rapidly
>> flashing. So I assume there are lots of data packets on the wire. But no
>> sound in both receivers.... Could it still be a firewall problem?
>>     
>
> tcpdump is your friend, and a lot more useful than a flashing light :-)
>
> # tcpdump -i eth0 -n -s0 udp
>
> Look at the source and destination IPs and port numbers of the packets.
>
> You might see:
> 1. packets from phone to Asterisk
> 2. packets from Asterisk to phone
> 3. packets from Asterisk to outside IP address
> 4. no packets from outside world to Asterisk
>
> Or you might see:
> 1. packets from phone to outside IP address
> 2. no packets from outside IP address to phone
>
> The message "Attempting native bridge" makes me think of this second
> possibility. If so, you could try setting nat=yes and/or canreinvite=no on
> the channel to the SIP provider, so that Asterisk proxies the RTP data.
>
> Regards,
>
> Brian.
>
>   

Hi Brian,

I've set canreinvite=no on the channel to the SIP provider and it
immediately worked. O.k., I'm happy about that but I want to
*understand* what's going on here.
.
My setup is:

Asterisk is connected on one side via eth1 to the "outside world" (IP
adress 81.223.xxx.xxx) and on the other side via eth0 to the internal
LAN (eth0 has IP 192.168.1.200, SNOM phone has 192.168.1.201, ...).

Is it right that, with canreinvite=yes, my SIP provider's RTP server
tries to connect directly to the phone passing by Asterisk? Because, in
that case, how should the firewall be able to do NAT?  I think that's
not possible.

Regards,
Norbert