Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] Freeswitch and OPAL/H323


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
ax.russo at gmail.com
Guest





PostPosted: Thu Mar 12, 2009 10:46 am    Post subject: [Freeswitch-users] Freeswitch and OPAL/H323 Reply with quote

Hi to all,

I am a newbie in Freeswitch (FS).

I have already installed a FS machine,  following the wiki installation procedure, and I have also added the opal module following this procedure: http://wiki.freeswitch.org/wiki/Mod_opal

When I running FS

######################################
freeswitch@atest> module_exists mod_opal
API CALL [module_exists(mod_opal)] output:
true
freeswitch@atest>
######################################

I think I'have installed it correctly.
My goal is to provide a conference tool for incoming h323-calls that come from a cisco call manager: we are behind a Cisco VoIP cloud.
Every time a PSTN phone calls the number 1234-123456 the call manager knows that the extension 3456 has to be redirected
to 192.168.193.38, that is the IP address of the FS machine, where file dialplan/default/myconference.xml contains the following lines

######################################
<include>
    <extension name="no_conferences">
      <condition field="destination_number" expression="^3456$">
        <action application="answer"/>
        <action application="info"/>
        <action application="conference" data="$1-${domain_name}@wideband"/>
      </condition>
    </extension>
</include>
######################################

On the other hand, whenever a user of FS calls an local extension (like 1XXX ), what I want is that FS forward this call to the cisco call manager through opal/h323
therefore I have a file in

######################################
<include>
<extension name="1xxx">
    <condition field="destination_number" expression="^(1\d{3})$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="ringback=$${us-ring}"/>
        <action application="set" data="instant_ringback=true"/>
        <action application="bridge" data="opal/h323:$1@IP.CALL.MANA.GER"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="voicemail" data="default ${ip_server_cisco_call_manager} ${dialed_extension}"/>
     </condition>
  </extension>
</include>
######################################

but it fails: when a FS user calls 1500, FS returns this message

######################################
2009-03-12 15:55:08 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1000@192.168.193.38 (1000@192.168.193.38) [c7b69402-0f15-11de-b4dc-c11b39fce37c]
2009-03-12 15:55:08 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->1500 in context default
2009-03-12 15:55:08 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel opal/h323:1500@IP.CALL.MANA.GER:1720 [c7c010e0-0f15-11de-b4dc-c11b39fce37c]
2009-03-12 15:55:08 [INFO] h323pdu.cxx:999 H225() Read error (0):
2009-03-12 15:55:08 [NOTICE] mod_opal.cpp:591 OnReleased() Hangup opal/h323:1500@IP.CALL.MANA.GER:1720 [CS_CONSUME_MEDIA] [UNKNOWN]
2009-03-12 15:55:08 [INFO] tlibthrd.cxx:363 PWLib() Destroyed thread 0xb171a708 H225 Caller:0xa9587b90(id = 0)
2009-03-12 15:55:08 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 22 (opal/h323:1500@IP.CALL.MANA.GER:1720) Ended
2009-03-12 15:55:08 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel opal/h323:1500@IP.CALL.MANA.GER:1720 [CS_HANGUP]
2009-03-12 15:55:08 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed.  Cause: UNKNOWN
2009-03-12 15:55:08 [NOTICE] mod_dptools.c:596 hangup_function() Hangup sofia/internal/1000@192.168.193.38 (1000@192.168.193.38) [CS_EXECUTE] [NORMAL_CLEARING]
2009-03-12 15:55:08 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 21 (sofia/internal/1000@192.168.193.38 (1000@192.168.193.38)) Ended
2009-03-12 15:55:08 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1000@192.168.193.38 (1000@192.168.193.38) [CS_HANGUP]
######################################


I don't understand why....
Any suggestions...
Thanks

Alessandro R.
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services