[Asterisk-Users] Disable IAX2 native bridging / Monitor() app

[Colin Anderson]

Answered my own question, partially from the "route call based on codec"
thread:

If u are using 1.2 there is global var SIP_CODEC or IAX_CODE

exten => 88,1,NoOP(${SIP_CODEC})
exten => 88,2,NoOP(${IAX_CODEC})

So I can modify my dialplan to check the codec. If it's anything but GSM,
route to the IAX provider. If it's GSM, dial out via the PSTN. 


-----Original Message-----
From: Colin Anderson [mailto:ColinA at landmarkmasterbuilder.com]
Sent: Wednesday, November 30, 2005 9:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Disable IAX2 native bridging / Monitor() app

I am working with a 3rd party provider who is providing me with IAX2
dialtone. I am using GSM codec end-to-end and my provider insists on ULAW
only. When my remote IAX clients attempt to use the provider for PSTN calls
by calling my primary * box, and my primary's dialplan is set to dial the
provider, native bridging is attempted. Because the codecs are different,
the call fails and the caller hears dead audio.

What I want is to disable native bridging altogether and to force my primary
to stay in the media path and transcode. My primary staying in the media
path is also essential for call recording. I have found:

> If you don't want native bridging, you need to disable it in chan_iax2.c
* by undefining BRIDGE_OPTIMIZATION.

Can anyone confirm if this is actually the case? Or is there a simpler,
undocumented Dial() switch I can use?

Running 1.0.9 on remote IAX boxes, 1.2 Beta 1 on primary.

Second question: Can someone summarize for me the circumstances under which
only a single part of the channel / one side of the conversation is recorded
using Monitor() under 1.2? For example, I have heard that natively bridged
Zap channels only record one side of the conversation. This may not be the
case in 1.2.
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