I tested this case on 1.4 branch
http://lists.freeswitch.org/pipermail/freeswitch-users/2015-July/114358.html

I able to receive RTP stream after answer.
Looks as need make "git bisect"

Sergey

On Tue, Dec 31, 2019 at 9:11 PM Mike Jerris <mike@freeswitch.org> wrote:
Is this the case on the most recent release?

On Dec 31, 2019, at 4:18 AM, Babak Yakhchali <babak.freeswitch@gmail.com> wrote:

Hi
trying below dialplan results in no audio:
<extension name="DailyIVR" continue="false" uuid="d8037520-2a15-11ea-9a9f-9bf4580ab5b8">
<condition field="destination_number" expression="^110$">      
<action application="answer" data=""/>                
<action application="sleep" data="1000"/>                
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="ringback=local_stream://default"/>
<action application="set" data="transfer_ringback=local_stream://default"/>
<action application="set" data="ivr_menu_uuid=d8049250-2a15-11ea-9f14-1b8cf3ef5a32"/>
<action application="ivr" data="d8049250-2a15-11ea-9f14-1b8cf3ef5a32"/>
<action application="" data=""/>
</condition>
</extension>

but if I move sleep before answer everything works fine. Using tcpdump shows that when sleep is called after answer no rtp is established but if I move sleep before answer rtp is established
I'm using freeswitch 1.8.7 on debian 8
thanks

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