[Asterisk-Dev] native bridge woes

[mjr-asterisk at ranney.com]

Native bridging sure seems like a great idea, except every time a call
is bridged by my asterisk server, the audio cuts out of a second.
Unfortunately, it cuts out right as the person is saying "hello" so
its hard to tell who answered the phone or what they said.  I'm using
Cisco 7940/7960 phones.  Is this a problem with asterisk or a problem
with the phones?  Is there any way to do native bridging without the
audio dropout?

I've tried setting "canreinvite=no" on all of my SIP peers, as well as
in the general section, but native bridges and their dropouts
continue.  Setting the t/T/r options as the call is dialed does indeed
all the calls to be transferred, etc., but I still get the native
bridge dropout.

Note that I haven't actually verified at the network level where all
the packets are going, but as messages like this show up on the
asterisk console:

    -- Attempting native bridge of SIP/3122618252-7b97 and SIP/colo-gw-481b

users get a very annoying audio dropout.

If this is unexpected behavior, I'm happy to dig deeper into the
problem with packet captures, running different code, etc., Any help
would be appreciated.
-- 
Matt Ranney - mjr at ranney.com