ax.russo at gmail.com Guest
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Posted: Thu Mar 12, 2009 10:46 am Post subject: [Freeswitch-users] Freeswitch and OPAL/H323 |
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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
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freeswitch@atest> module_exists mod_opal
API CALL [module_exists(mod_opal)] output:
true
freeswitch@atest>
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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
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<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>
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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
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<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>
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but it fails: when a FS user calls 1500, FS returns this message
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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]
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I don't understand why....
Any suggestions...
Thanks
Alessandro R. |
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