[Asterisk-Users] Native Bridge killing audio, sending dtmf

[Tim Connolly]

I'm seeing this same issue. The following message will popup on the console:
    -- Attempting native bridge of Zap/1-1 and Zap/74-1
At the same time my call is briefly muted, I hear a quick DTMF tone, then it
unmutes. The whole process takes about 1.5 seconds. Is there any way to stop
these attempts? My call was an inbound PRI to zap te411p call, back out
another span on the same card to a legacy pbx.

Any suggestions?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anton Krall
Sent: Wednesday, June 22, 2005 11:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Native Bridge 

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

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