The audio is carried on two RTP streams: one for each direction. Is it possible those streams are being blocked by a firewall or something of the sort?

The "attempting native bridge" message means that Asterisk is bridging the two calls together without doing any codec translation... uLaw to uLaw, for instance.

If the two phones were using "reinvite" you wouldn't see this message because there would be nothing for Asterisk to bridge: the two phones are chatting to one another.

On Oct 15, 2004, at 3:34 PM, Brian Weaver wrote:

I have two fo the Sipura-2000 boxes, one at a friends house, one
here. It used to be working but now we are not getting any audio when
the call is picked up.

I'm seeing this message when he answer the phone.

-- Attempting native bridge of SIP/2204-2b1b and SIP/2203-783a

As far as I can tell, it shouldn't be doing this because I have

canreinvite=no

in the sip.conf for these extensions since we are behind NAT firewalls
and the two Sipura boxes cannot talk directly to each other.

What am I missing?

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