On Fri, Jun 26, 2009 at 9:25 PM, Artem Shiyanov <shiyanov@gmail.com> wrote:

Hello!

I got a problem with one way audio, symptoms are:
firstly play audio file to channel A (A is hears sound)
secondly bridge channel B with A (A doesn't hear B).

Environment:
- no NAT
- User Agents being used X-Lite, EyeBeam, SJphone - same result for all of them- no audio, Wireshark shows that there is no RTP-flow to A from FreeSwitch
- dialplan:
<extension name="playback_media_file">
    <condition field="destination_number" expression="playmedia">
      <action application="answer"/>
      <action application="playback" data="test.wav"/>
    </condition>
</extension>

<extension name="Local_Extension_from_SP">
      <condition field="destination_number" expression="^([0-9]{2,9})$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
      </condition>
      <condition field="${sip_to_host}" expression="^([^.]*)\..*$">
        <action application="set" data="orgname=$1"/>
      </condition>
      <condition field="destination_number" expression="^${caller_id_number}$">
        <anti-action application="set" data="ringback=${us-ring}"/>
        <anti-action application="set" data="transfer_ringback=${us-ring}"/>
        <anti-action application="set" data="call_timeout=10"/>
        <anti-action application="set" data="hangup_after_bridge=true"/>
        <anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
        <anti-action application="set" data="continue_on_fail=true"/>
        <anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
        <anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
        <anti-action application="answer"/>
        <anti-action application="export" data="sip_h_X-SPFrom=&quote;${sip_from_user}&quote;<${sip_from_uri}>"/>
        <anti-action application="export" data="sip_h_X-SPTo=<${sip_to_uri}>"/>
        <anti-action application="export" data="sip_h_X-SPCallId=${sip_call_id}"/>
        <anti-action application="bridge" data="sofia/external/${orgname}send2voicemail@$${starpound_sip_app_server}"/>
      </condition>
    </extension>
- Call routing scheme:
user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc
Exact description what's going on is:
user A -> FS -(bridge)-> my B2BUA
Then my B2BUA transfers (using re-INVITE's), on behalf of user, call to extension "playback_media_file" . After a while B2BUA transfer (re-Inviting) user to extension "Local_Extension_from_SP". This should create a new call to user B. As a result - A doesn't hear B, but B- is OK.
On the contrary, if a call is routed (by B2BUA) to the "Local_Extension_from_SP" extension (ommiting "playback_media_file" ext) - everything is OK.

What I've tried:
- set parameter "inbound-proxy-media" to "true" in Sofia profile
- set parameter "disable_rtp_auto_adjust to "true" in Sofia profile
Nothing helps.

Any help or thoughts would be MUCH appreciated!
Artem