Hi,

  I have used the function of conference set auto outcall in freeswitch 1.0.6, it is very good,

several days ago, I installed a new freeswitch with git, the version is 1.0.7, I wrote a very simple

dialplan to test the conference auto outcall function, and the problem happened:

This is my dialplan, I have tested it in my freeswitch 1.0.6 successfully:

   <extension name="Demonstrate conference_set_auto_outcall">

     <condition field="destination_number" expression="^(12345)$">

       <action application="answer"/>

       <action application="set" data="conference_auto_outcall_timeout=5"/>

       <action application="set" data="conference_auto_outcall_flags=none"/>

       <action application="set" data="conference_auto_outcall_caller_id_name=$${ caller_id_name}"/>

       <action application="set" data="conference_auto_outcall_caller_id_number=$${ caller_id_number}"/>

       <action application="set" data="conference_auto_outcall_profile=default"/>

       <action application="conference_set_auto_outcall" data="[call_timeout=15]user/1001@$${domain}"/>

       <action application="conference_set_auto_outcall" data="[call_timeout=15]user/1002@$${domain}"/>

       <action application="conference" data="$1@default"/>

     </condition>

   </extension>

Problem 1:  When I use a softphone to dial number 12345, in my old freeswitch I can hear the moh music when the phone 1001 and 1002 is ring,

But in freeswitch 1.0.7 I can’t hear anything, and I found the debug informations in freeswitch cli:

2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:1156 Send signal sofia/internal/sip:1001@192.168.2.226:5060 [BREAK]

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:369 (sofia/internal/sip:1001@192.168.2.226:5060) State INIT going to sleep

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/sip:1001@192.168.2.226:5060) Running State Change CS_ROUTING

2011-08-31 18:59:14.520108 [DEBUG] switch_channel.c:1828 (sofia/internal/sip:1001@192.168.2.226:5060) Callstate Change DOWN -> RINGING

2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:857 Send signal sofia/internal/sip:1001@192.168.2.226:5060 [BREAK]

2011-08-31 18:59:14.520108 [DEBUG] sofia.c:5128 Channel sofia/internal/sip:1001@192.168.2.226:5060 entering state [calling][0]

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:1001@192.168.2.226:5060) State ROUTING

2011-08-31 18:59:14.520108 [DEBUG] mod_sofia.c:148 sofia/internal/sip:1001@192.168.2.226:5060 SOFIA ROUTING

2011-08-31 18:59:14.520108 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1001@192.168.2.226:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA

2011-08-31 18:59:14.520108 [DEBUG] switch_core_session.c:1156 Send signal sofia/internal/sip:1001@192.168.2.226:5060 [BREAK]

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:1001@192.168.2.226:5060) State ROUTING going to sleep

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/sip:1001@192.168.2.226:5060) Running State Change CS_CONSUME_MEDIA

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/sip:1001@192.168.2.226:5060) State CONSUME_MEDIA

2011-08-31 18:59:14.520108 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/sip:1001@192.168.2.226:5060) State CONSUME_MEDIA going to sleep

2011-08-31 18:59:14.540080 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz

2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal sofia/internal/1000@192.168.2.242 [BREAK]

2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal sofia/internal/1000@192.168.2.242 [BREAK]

2011-08-31 18:59:14.540080 [DEBUG] switch_core_session.c:857 Send signal sofia/internal/1000@192.168.2.242 [BREAK]

2011-08-31 18:59:14.559978 [DEBUG] sofia.c:5128 Channel sofia/internal/1000@192.168.2.242 entering state [ready][200]

       That mean the freeswitch is sending the moh music to the caller 1000, is that right? But I can’t hear anything.

       Problem 2: when just one phone pickup, like phone 1001, and phone 1002 don’t, (I use phone 1000 to call number 12345) I can’t hear anything from phone 1001 when 1002 is ringing except phone 1002 pick up or when the phone 1002 is timeout to ring, it’s very stranger!!! I tested this situation in freeswitch 1.0.6, when just only person pickup the phone the conference will work, 1000 will hear the phone 1001’s voice, don’t need the 1002 to pickup or timeout.

By the way: I have tested this with the newest version: FreeSWITCH Version 1.0.head (git-6a5f6e5 2011-08-30 15-00-07 -0500)

The problem is still exist.

Thank you for advising me how to solve this problem.

Regards!

Dennis