[Asterisk-Users] Native Bridge

[Anton Krall]

Guys.,

How can I disable native briding on sip?

I get this but after that, the call just tries to do the bridge and freezes

== Parsing '/etc/asterisk/sip_notify.conf': Found
    -- Executing Dial("SIP/demo-3763", "SIP/demo2|20|mwtWT") in new stack
    -- Called demo2
    -- Started music on hold, class 'default', on SIP/demo-3763
Jun 22 23:31:46 WARNING[31090]: chan_sip.c:2901 find_call: Call missing call
ID from '201.129.249.85'
    -- SIP/demo2-07af is ringing
    -- SIP/demo2-07af answered SIP/demo-3763
    -- Stopped music on hold on SIP/demo-3763
    -- Attempting native bridge of SIP/demo-3763 and SIP/demo2-07af
  == Spawn extension (telefonos, 102, 1) exited non-zero on 'SIP/demo-3763'

Both phones are inside a NATted network connecting to a remote asterisk, the
eco test works so audio works great on each phone to asterisk.

My sip.conf file has canreinvite=no and nat=yes on both phones config.

Any ideas