[Freeswitch-users] FSClient and sip_auto_answer

Paul Mateer Paul.Mateer at outlook.com
Wed Apr 26 17:44:14 MSD 2017

Hi folks.

I have a FreeSWITCH server with a couple of FSClient softphones attached. I've been experimenting with conference calls using the following extension configuration (based upon the Mad Boss example):

  <extension name="Broadcast Announcement">

    <condition field="destination_number" expression="^0911$">

      <action application="set" data="conference_auto_outcall_caller_id_name=Broadcast"/>

      <action application="set" data="conference_auto_outcall_caller_id_number=0911"/>

      <action application="set" data="conference_auto_outcall_timeout=60"/>

      <action application="set" data="conference_utils_auto_outcall_flags=mute"/>

      <action application="set" data="end-conf-grace-time=0"/>

      <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/>

      <action application="set" data="sip_exclude_contact=${network_addr}"/>

      <action application="conference_set_auto_outcall" data="user/5006@$${domain}"/>

      <action application="conference_set_auto_outcall" data="user/5007@$${domain}"/>

      <action application="conference" data="SystemBroadcast at default++flags{deaf|moderator|endconf}"/>



I've noticed a couple of issues which seem to be tied to the sip_auto_answer setting.

  1.  The phone making the call is providing a live radio feed, so the callees should receive audio, but they do not
  2.  When the caller disconnects from the conference, the callees remain connected despite the fact that the moderator and endconf flags are set.

If I change the sip_auto_answer setting from true to false and then place the call, each phone rings until answered but

  1.  Once a phone is answered you hear the radio feed.
  2.  When the caller disconnects, so do all the callees.

Can anyone explain why I am seeing the difference in behaviour between the true and false settings for sip_auto_answer?
I was expecting the same behaviour in both cases and that setting the sip_auto_answer flag would simply mean that I didn't have to explicitly answer each phone added to the conference.

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