<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 May 17, 2016, at 2:37 PM, Michael Jerris <mike@jerris.com> wrote:

Interesting, https://tools.ietf.org/html/rfc7261 says annexb=yes is default if omitted.  Brian West, did you know that?



On 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!
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