Hello,

I'm trying to use "bypass_media_after_bridge" in my dialplan.
In this scenario I have
a) softphone client using g729
b) FS version: FreeSWITCH Version 1.5.13b~64bit ( 64bit)
c) Dialogic HMP 3.0 (g729 + g711)

I'd like to use FS in passthrough mode so calls from a) will be rerouted to
c).

If I use the "bypass_media_after_bridge" signalling of calls is ok but i
have no audio on a).

If I use "bypass_media" audio is ok but I don't have the answer 200 OK back
to my softphone a)

This is my dialplan:

<extension name="calls_to_ivr">
      <condition field="destination_number" expression="^9999">
        <action application="export" data="jitterbuffer_msec=60:200:40"/>
        <action application="set" data="bypass_media_after_bridge=true"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="ignore_early_media=true"/>
        <action application="set" data="effective_caller_id_number=1234"/>
        <action application="bridge"
data="{execute_on_answer=start_dtmf_generate}sofia/internal/9000@10.1.1.1"/>
        <action application="pre_answer"/>
      </condition>
</extension>

In my sip profile the "inbound-late-negotiation" param is set to true.

Is there any mistake in my dialplan?

Thanks,

Stefano F.




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