fixed, thanks.

On May 18, 2016, at 12:20 PM, Mike Rice <mrice0118@gmail.com> wrote:

I got it from the following URL.


---------- Forwarded message ----------
From: Michael Jerris <mike@jerris.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Tue, 17 May 2016 16:45:16 -0400
Subject: Re: [Freeswitch-users] Disable annexb for g729
It was spread about pretty widely when we went to 1.4 but thats a couple years ago now.  It should be documented on the wiki but may have fallen off during the transition to confluence as 1.2 is quite old.  Where did you see docs for the sip_ var?

On Tue, May 17, 2016 at 3:51 PM, Mike Rice <mrice0118@gmail.com> wrote:
Thanks Michael! That did the trick. Did I miss that somewhere?

From: Michael Jerris <mike@jerris.com>

<action application="export" data="rtp_append_audio_sdp=a=fmtp:18 annexb=no"/>

between 1.2 and 1.4 the var got renamed from sip_ to rtp_


On Tue, May 17, 2016 at 10:57 AM, Mike Rice <mrice0118@gmail.com> wrote:
We have a carrier the mandates that annex b be disabled on the invites to them. I have added the following to the default dialplan but nothing seems to change the invite on the B leg to the carrier. 

<extension name="disable-annexB" continue="true">
  <condition field="${switch_r_sdp}" expression="/(.*)(m=audio \d+ RTP\/AVP)(.*)( 18 )(.*)/s">
     <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>
  </condition>
</extension>

<action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>

and

<action application="bridge" data="{sip_append_audio_sdp=a=fmtp:18
annexb=no,absolute_codec_string=^^:G729:PCMU:PCMA}sofia/gateway/carrierGW/$1"/>


inbound-late-negotiation is set to true. The logs show that it is exporting:

[DEBUG] switch_channel.c:1267 EXPORT (export_vars) [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable]
EXECUTE sofia/internal/1000@10.10.X.X bridge(sofia/gateway/lo7f/XXXXXXXXXX)
[DEBUG] switch_channel.c:1221 sofia/internal/1000@10.10.X.X EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to event
[DEBUG] switch_channel.c:1221 sofia/internal/1000@10.10.X.X EXPORTING[export_vars] [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable] to event

The SDP does not reflect:

Local SDP:
v=0
o=FreeSWITCH IN IP4 10.10.X.X
s=FreeSWITCH
c=IN IP4 10.10.X.X
t=0 0
m=audio 20022 RTP/AVP 18 0 8 101 13
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Any help would be greatly appreciated. Thanks!