Instead of complaining and demanding things for free, people should start to put their money where their mouth is.

Diego

On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
hi Anthony,

it is somewhere here:

         switch_status_t FSConnection::receive_message(switch_core_session_message_t *msg)


anyhow, i will open an issue jira of course.


I understand your financial point of view, but anyhow while the entire world is wants sip and trying to move to sip, the reality is quite different. The majority of voice traffic exchanged via IP is still H323. This means a working SIP - H323 interworking is really needed... pity nobody wants/has time to work in this direction to produce a decent mod_h323.



T.



 


On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
pcap is not as useful as FS console log on debug with:
sofia profile internal siptrace on

you should be reporting issues to jira under mod_opal not to the mailing list.
http://jira.freeswitch.org

FYI
There is little financial support from the community for h323 which prevents the mod_opal from getting much attention.
We actually have to contract the author of opal to help with these issues including the original writing of the module that he did with very little funding and nobody ever wants to pay him to improve it.

That does not mean your issue will not be addressed but there is no promise how fast it will be.



On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
hello guys,


i was playing with mod_opal to see if i can make it working ... well it seems SIP-H323 interworking is not tuned at all.

I have a call from a registered sip user (1001) to PSTN via mod_opal


<include>
  <extension name="EMERGENCY">
    <condition field="destination_number" expression="^0(112|9[23456])$">
      <action application="set" data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="SPECIAL_SERVICES">
    <condition field="destination_number" expression="^0(9[01789]\d{3,4})$">
      <action application="set" data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
    </condition>
  </extension>

  <extension name="ENYTHING_ELSE">
    <condition field="destination_number" expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
      <action application="set" data="effective_caller_id_number=1001282122"/>
      <action application="set" data="NCX_IP=10.4.4.254"/>
      <action application="set" data="call_timeout=30"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
    </condition>
  </extension>
</include>



One of the many issues i sow is that FS connects the call on SIP leg before it actually receives H.225 connect from H323 leg... as it is configured to send 200 OK on the 1st H.225 message containing a FastStart element/OLC.


Attached is the tcpdump i took on FS machine... just use this filter: "h225 or h245 or q931 or sip"
Also, you can check the attac CDR,,,, this is an unanswered call i placed to PSTN and FS billed it 23 seconds.



Can anyone tell where i can do correct SIP - H323 message mappings to avoid this?



T.




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Anthony Minessale II

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