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[Freeswitch-users] mod_opal - call charged before H.225 connect

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anthony.minessale at g...
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PostPosted: Tue Oct 06, 2009 2:37 pm    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

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 (tculjaga@gmail.com)> wrote:
Quote:
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

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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tculjaga at gmail.com
Guest





PostPosted: Tue Oct 06, 2009 4:01 pm    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:


Quote:

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.







_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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diego.viola at gmail.com
Guest





PostPosted: Tue Oct 06, 2009 4:55 pm    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

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 (tculjaga@gmail.com)> wrote:
Quote:
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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:


Quote:

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.







_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






_______________________________________________
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jason at jasonjgw.net
Guest





PostPosted: Tue Oct 06, 2009 6:11 pm    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

Tihomir Culjaga <tculjaga@gmail.com> wrote:

Quote:

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.

Is there any evidence in support of the above assertion (e.g., survey results
of VoIP traffic)? I've heard of H323 but I don't know anyone who uses it, or
any phones that implement it.

The lack of interest in this forum and the absence of financial support to
improve the H323 support in FreeSWITCH suggest that the level of demand for
this is quite low, relative to SIP.

Of course, improvements are always welcome, so if you're interested in funding
better H323 support, or helping with the module I'm sure the FreeSWITCH
community would welcome your efforts.


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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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tculjaga at gmail.com
Guest





PostPosted: Tue Oct 06, 2009 6:49 pm    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

Diego,

what i'm pointing here is the situation where you have a great product that lacks in one of most common protocol. It is true H323 is going to disappear (eventually), it is true that the community prefers SIP/IAX instead ... but the reality still remains. H323 is going to be used for quite a long time to exchange a lot of traffic while FS will be left aside. Today, when you setup an IP peering interconnection 80% of carriers will prefer H323.

Of course, developing something costs "time" (and we all know what time stands for...) and as i said, i understand the financial point of view and i really understand if nobody is going to work on that, but let's face it FS doesn't have any usable module to reliably handle H323 protocol.


said that, i don't intend to offend anyone... just facing the reality.


regarding the h323 module, we don't have any issue fixing the existing or developing a new one... but before we go developing something it is always better check if the thing you want already exists in an usable state or not... that's what i did today.


So, I'm interested in a reliable module handling H323v4... anyone else?


T.





On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:
Quote:
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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:


Quote:

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.







_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






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tculjaga at gmail.com
Guest





PostPosted: Tue Oct 06, 2009 7:21 pm    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

thanks for your e-mail,

H323 is mainly used for trunking purpose, inter-carrier traffic exchange... it is not used to control IP phones Razz

well, believe me, I've heard enough of H323 that i'm sick of it Razz
What i can tell you comes from my own experience on daily activities i'm doing for living... Of course, there might be part of the world where H323 dispersed completely but over here in Europe things tend to stick on tradition Razz

Yep, you are right... the forum wants SIP and that's understandable...

anyhow you might check this:
http://www.dailypayload.com/content/3111


T.


On Wed, Oct 7, 2009 at 12:58 AM, Jason White <jason@jasonjgw.net (jason@jasonjgw.net)> wrote:
Quote:
Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:


Quote:
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.


Is there any evidence in support of the above assertion (e.g., survey results
of VoIP traffic)? I've heard of H323 but I don't know anyone who uses it, or
any phones that implement it.

The lack of interest in this forum and the absence of financial support to
improve the H323 support in FreeSWITCH suggest that the level of demand for
this is quite low, relative to SIP.

Of course, improvements are always welcome, so if you're interested in funding
better H323 support, or helping with the module I'm sure the FreeSWITCH
community would welcome your efforts.



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diego.viola at gmail.com
Guest





PostPosted: Tue Oct 06, 2009 7:32 pm    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

Yeah I understand your point of view, but saying "I want a H.323 module" or "I want a Ferrari" wont magically make it happen.

We need to work on it ourselves or pay to the people that knows how to do it to do it for us.

There is no other way I think.

Diego



On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:
Quote:
Diego,

what i'm pointing here is the situation where you have a great product that lacks in one of most common protocol. It is true H323 is going to disappear (eventually), it is true that the community prefers SIP/IAX instead ... but the reality still remains. H323 is going to be used for quite a long time to exchange a lot of traffic while FS will be left aside. Today, when you setup an IP peering interconnection 80% of carriers will prefer H323.

Of course, developing something costs "time" (and we all know what time stands for...) and as i said, i understand the financial point of view and i really understand if nobody is going to work on that, but let's face it FS doesn't have any usable module to reliably handle H323 protocol.


said that, i don't intend to offend anyone... just facing the reality.


regarding the h323 module, we don't have any issue fixing the existing or developing a new one... but before we go developing something it is always better check if the thing you want already exists in an usable state or not... that's what i did today.


So, I'm interested in a reliable module handling H323v4... anyone else?


T.






On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:
Quote:
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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:


Quote:

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.







_______________________________________________
FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

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anthony.minessale at g...
Guest





PostPosted: Tue Oct 06, 2009 8:08 pm    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

I didn't mean to start anything.  I'm just saying we work very long hours and barely get anybody asking about h.323.
I wanted to support it and that's why we took up a collection to get funding for mod_opal but when only 1 donor showed any interest we were forced to proceed in our spare time which is very limited.

The developers of opal are not part of our project and they need financial compensation to be motivated to work on it.  Its not even related to me its only fair that an outside developer who makes his living as a consultant would want money to integrate his work into our project.

Like I said, I will do my best to point your issue to the opal devs but I cannot force them to work on it.




On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:
Quote:
Yeah I understand your point of view, but saying "I want a H.323 module" or "I want a Ferrari" wont magically make it happen.

We need to work on it ourselves or pay to the people that knows how to do it to do it for us.

There is no other way I think.

Diego




On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:
Quote:
Diego,

what i'm pointing here is the situation where you have a great product that lacks in one of most common protocol. It is true H323 is going to disappear (eventually), it is true that the community prefers SIP/IAX instead ... but the reality still remains. H323 is going to be used for quite a long time to exchange a lot of traffic while FS will be left aside. Today, when you setup an IP peering interconnection 80% of carriers will prefer H323.

Of course, developing something costs "time" (and we all know what time stands for...) and as i said, i understand the financial point of view and i really understand if nobody is going to work on that, but let's face it FS doesn't have any usable module to reliably handle H323 protocol.


said that, i don't intend to offend anyone... just facing the reality.


regarding the h323 module, we don't have any issue fixing the existing or developing a new one... but before we go developing something it is always better check if the thing you want already exists in an usable state or not... that's what i did today.


So, I'm interested in a reliable module handling H323v4... anyone else?


T.






On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:
Quote:
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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:


Quote:

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.







_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________
FreeSWITCH-users mailing list
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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
Back to top
tculjaga at gmail.com
Guest





PostPosted: Wed Oct 07, 2009 3:45 am    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

Anthony,

of course, nobody wants to start anything... we are all here to help making FS a better product.

so, regarding the founding for mod_opal ... what is the amount you need?


Tihomir.



On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote:
I didn't mean to start anything.  I'm just saying we work very long hours and barely get anybody asking about h.323.
I wanted to support it and that's why we took up a collection to get funding for mod_opal but when only 1 donor showed any interest we were forced to proceed in our spare time which is very limited.

The developers of opal are not part of our project and they need financial compensation to be motivated to work on it.  Its not even related to me its only fair that an outside developer who makes his living as a consultant would want money to integrate his work into our project.

Like I said, I will do my best to point your issue to the opal devs but I cannot force them to work on it.





On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:
Quote:
Yeah I understand your point of view, but saying "I want a H.323 module" or "I want a Ferrari" wont magically make it happen.

We need to work on it ourselves or pay to the people that knows how to do it to do it for us.

There is no other way I think.

Diego




On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:
Quote:
Diego,

what i'm pointing here is the situation where you have a great product that lacks in one of most common protocol. It is true H323 is going to disappear (eventually), it is true that the community prefers SIP/IAX instead ... but the reality still remains. H323 is going to be used for quite a long time to exchange a lot of traffic while FS will be left aside. Today, when you setup an IP peering interconnection 80% of carriers will prefer H323.

Of course, developing something costs "time" (and we all know what time stands for...) and as i said, i understand the financial point of view and i really understand if nobody is going to work on that, but let's face it FS doesn't have any usable module to reliably handle H323 protocol.


said that, i don't intend to offend anyone... just facing the reality.


regarding the h323 module, we don't have any issue fixing the existing or developing a new one... but before we go developing something it is always better check if the thing you want already exists in an usable state or not... that's what i did today.


So, I'm interested in a reliable module handling H323v4... anyone else?


T.






On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:
Quote:
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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:


Quote:

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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claudiu at globtel.ro
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PostPosted: Wed Oct 07, 2009 7:55 am    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

Hi Tihomir,


I've done some tests to see how suitable is freeswitch as a
SIP/H323 translator and you are right about the fact that H323
'alert+open logical channel' will generate a SIP '200 OK'. I was
able to fix that with a couple of changes in mod_opal.cpp, however
some things were changed on mod_sofia in the latest svn. (on this
particular issue, open_logical_channel is processed BEFORE the
alerting, so the call is in SetupPhase when the proc
OnOpenMediaStream is triggered)


The most important problem I'm having right now is that G729 is
still not working (poor quality due to high buffering). Even with
the latest Opal, which includes the last week patch for jitter
buffer. If you dont need G729, I could send you a patch for the
latest svns (freeswitch, opal, ptlib), ofc no founding needed.

There are a couple of bugs in opal itself and h323ing freeswitch
with opal will bring them in. On the other hand, mod_opal is
already there, it just needs a few adjustments.


Best wishes,

Claudiu Filip




Wednesday, October 7, 2009, 11:37:50 AM, you wrote:
Tihomir> Anthony,

Tihomir> of course, nobody wants to start anything... we are all here
Tihomir> to help making FS a better product.

Tihomir> so, regarding the founding for mod_opal ... what is the amount you need?


Tihomir> Tihomir.


Tihomir>
Tihomir> On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale
Tihomir> <anthony.minessale@gmail.com> wrote:
Tihomir> I didn't mean to start anything.  I'm just saying we work
Tihomir> very long hours and barely get anybody asking about h.323.
Tihomir> I wanted to support it and that's why we took up a
Tihomir> collection to get funding for mod_opal but when only 1 donor
Tihomir> showed any interest we were forced to proceed in our spare time which is very limited.
Tihomir>
Tihomir> The developers of opal are not part of our project and they
Tihomir> need financial compensation to be motivated to work on it. 
Tihomir> Its not even related to me its only fair that an outside
Tihomir> developer who makes his living as a consultant would want
Tihomir> money to integrate his work into our project.
Tihomir>
Tihomir> Like I said, I will do my best to point your issue to the
Tihomir> opal devs but I cannot force them to work on it.





Tihomir> On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola
Tihomir> <diego.viola@gmail.com> wrote:
Tihomir>
Tihomir> Yeah I understand your point of view, but saying "I want a
Tihomir> H.323 module" or "I want a Ferrari" wont magically make it happen.
Tihomir>
Tihomir> We need to work on it ourselves or pay to the people that
Tihomir> knows how to do it to do it for us.
Tihomir>
Tihomir> There is no other way I think.

Tihomir> Diego




Tihomir> On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
Tihomir>
Tihomir> Diego,

Tihomir> what i'm pointing here is the situation where you have a
Tihomir> great product that lacks in one of most common protocol. It
Tihomir> is true H323 is going to disappear (eventually), it is true
Tihomir> that the community prefers SIP/IAX instead ... but the
Tihomir> reality still remains. H323 is going to be used for quite a
Tihomir> long time to exchange a lot of traffic while FS will be left
Tihomir> aside. Today, when you setup an IP peering interconnection
Tihomir> 80% of carriers will prefer H323.
Tihomir>
Tihomir> Of course, developing something costs "time" (and we all
Tihomir> know what time stands for...) and as i said, i understand the
Tihomir> financial point of view and i really understand if nobody is
Tihomir> going to work on that, but let's face it FS doesn't have any
Tihomir> usable module to reliably handle H323 protocol.
Tihomir>

Tihomir> said that, i don't intend to offend anyone... just facing the reality.


Tihomir> regarding the h323 module, we don't have any issue fixing
Tihomir> the existing or developing a new one... but before we go
Tihomir> developing something it is always better check if the thing
Tihomir> you want already exists in an usable state or not... that's what i did today.
Tihomir>

Tihomir> So, I'm interested in a reliable module handling H323v4... anyone else?


Tihomir> T.






Tihomir> On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola
Tihomir> <diego.viola@gmail.com> wrote:
Tihomir>
Tihomir> Instead of complaining and demanding things for free, people
Tihomir> should start to put their money where their mouth is.
Tihomir>
Tihomir> Diego


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

Tihomir> it is somewhere here:

Tihomir>          switch_status_t
Tihomir> FSConnection::receive_message(switch_core_session_message_t *msg)
Tihomir>

Tihomir> anyhow, i will open an issue jira of course.


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


Tihomir> T.



Tihomir>  


Tihomir> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale
Tihomir> <anthony.minessale@gmail.com> wrote:
Tihomir>
Tihomir> pcap is not as useful as FS console log on debug with:
Tihomir> sofia profile internal siptrace on
Tihomir>
Tihomir> you should be reporting issues to jira under mod_opal not to the mailing list.
Tihomir> http://jira.freeswitch.org
Tihomir>
Tihomir> FYI
Tihomir> There is little financial support from the community for
Tihomir> h323 which prevents the mod_opal from getting much attention.
Tihomir> We actually have to contract the author of opal to help with
Tihomir> these issues including the original writing of the module
Tihomir> that he did with very little funding and nobody ever wants to pay him to improve it.
Tihomir>
Tihomir> That does not mean your issue will not be addressed but
Tihomir> there is no promise how fast it will be.



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


Tihomir> i was playing with mod_opal to see if i can make it working
Tihomir> ... well it seems SIP-H323 interworking is not tuned at all.
Tihomir>
Tihomir> I have a call from a registered sip user (1001) to PSTN via mod_opal
Tihomir>

Tihomir> <include>
Tihomir>   <extension name="EMERGENCY">
Tihomir>     <condition field="destination_number"
Tihomir> expression="^0(112|9[23456])$">
Tihomir>       <action application="set"
Tihomir> data="effective_caller_id_number=1001282122"/>
Tihomir>       <action application="set" data="NCX_IP=10.4.4.254"/>
Tihomir>       <action application="set" data="call_timeout=30"/>
Tihomir>       <action application="set" data="hangup_after_bridge=true"/>
Tihomir>
Tihomir>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
Tihomir>     </condition>
Tihomir>   </extension>

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

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

Tihomir>   <extension name="ENYTHING_ELSE">
Tihomir>     <condition field="destination_number"
Tihomir> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
Tihomir>       <action application="set"
Tihomir> data="effective_caller_id_number=1001282122"/>
Tihomir>       <action application="set" data="NCX_IP=10.4.4.254"/>
Tihomir>       <action application="set" data="call_timeout=30"/>
Tihomir>       <action application="set" data="hangup_after_bridge=true"/>
Tihomir>
Tihomir>       <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
Tihomir>     </condition>
Tihomir>   </extension>
Tihomir> </include>



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

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


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



Tihomir> T.


Tihomir>

Tihomir>
Tihomir> _______________________________________________
Tihomir> FreeSWITCH-users mailing list
Tihomir> FreeSWITCH-users@lists.freeswitch.org
Tihomir> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Tihomir>
Tihomir> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Tihomir> http://www.freeswitch.org
Tihomir>




Tihomir> --
Tihomir> Anthony Minessale II

Tihomir> FreeSWITCH http://www.freeswitch.org/
Tihomir> ClueCon http://www.cluecon.com/
Tihomir> Twitter: http://twitter.com/FreeSWITCH_wire

Tihomir> AIM: anthm
Tihomir> MSN:anthony_minessale@hotmail.com
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Tihomir> IRC: irc.freenode.net #freeswitch

Tihomir> FreeSWITCH Developer Conference
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Tihomir> googletalk:conf+888@conference.freeswitch.org
Tihomir> pstn:213-799-1400
Tihomir>
Tihomir> _______________________________________________
Tihomir> FreeSWITCH-users mailing list
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Tihomir>
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Tihomir>


Tihomir>

Tihomir> _______________________________________________
Tihomir> FreeSWITCH-users mailing list
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Tihomir>


Tihomir>

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Tihomir>

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Tihomir>

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bottleman at icf.org.ru
Guest





PostPosted: Wed Oct 07, 2009 8:18 am    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

On 2009-10-07 01:41 +0200, Tihomir Culjaga wrote freeswitch-users@lists.fre...:

We are developing module to handle h323 proto now, we try to use mod_opal and try improve it, but no luck,
there is many issues in libopal, and finaly we now move to h323plus library.

TC>Diego,
TC>
TC>what i'm pointing here is the situation where you have a great product that
TC>lacks in one of most common protocol. It is true H323 is going to disappear
TC>(eventually), it is true that the community prefers SIP/IAX instead ... but
TC>the reality still remains. H323 is going to be used for quite a long time to
TC>exchange a lot of traffic while FS will be left aside. Today, when you setup
TC>an IP peering interconnection 80% of carriers will prefer H323.
TC>
TC>Of course, developing something costs "time" (and we all know what time
TC>stands for...) and as i said, i understand the financial point of view and i
TC>really understand if nobody is going to work on that, but let's face it FS
TC>doesn't have any usable module to reliably handle H323 protocol.
TC>
TC>
TC>said that, i don't intend to offend anyone... just facing the reality.
TC>
TC>
TC>regarding the h323 module, we don't have any issue fixing the existing or
TC>developing a new one... but before we go developing something it is always
TC>better check if the thing you want already exists in an usable state or
TC>not... that's what i did today.
TC>
TC>
TC>So, I'm interested in a reliable module handling H323v4... anyone else?
TC>
TC>
TC>T.
TC>
TC>
TC>
TC>
TC>
TC>On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola@gmail.com> wrote:
TC>
TC>> Instead of complaining and demanding things for free, people should start
TC>> to put their money where their mouth is.
TC>>
TC>> Diego
TC>>
TC>>
TC>> On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga <tculjaga@gmail.com>wrote:
TC>>
TC>>> hi Anthony,
TC>>>
TC>>> it is somewhere here:
TC>>>
TC>>> switch_status_t
TC>>> FSConnection::receive_message(switch_core_session_message_t *msg)
TC>>>
TC>>>
TC>>> anyhow, i will open an issue jira of course.
TC>>>
TC>>>
TC>>> I understand your financial point of view, but anyhow while the entire
TC>>> world is wants sip and trying to move to sip, the reality is quite
TC>>> different. The majority of voice traffic exchanged via IP is still H323.
TC>>> This means a working SIP - H323 interworking is really needed... pity nobody
TC>>> wants/has time to work in this direction to produce a decent mod_h323.
TC>>>
TC>>>
TC>>>
TC>>> T.
TC>>>
TC>>>
TC>>>
TC>>>
TC>>>
TC>>>
TC>>> On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale <
TC>>> anthony.minessale@gmail.com> wrote:
TC>>>
TC>>>> pcap is not as useful as FS console log on debug with:
TC>>>> sofia profile internal siptrace on
TC>>>>
TC>>>> you should be reporting issues to jira under mod_opal not to the mailing
TC>>>> list.
TC>>>> http://jira.freeswitch.org
TC>>>>
TC>>>> FYI
TC>>>> There is little financial support from the community for h323 which
TC>>>> prevents the mod_opal from getting much attention.
TC>>>> We actually have to contract the author of opal to help with these issues
TC>>>> including the original writing of the module that he did with very little
TC>>>> funding and nobody ever wants to pay him to improve it.
TC>>>>
TC>>>> That does not mean your issue will not be addressed but there is no
TC>>>> promise how fast it will be.
TC>>>>
TC>>>>
TC>>>>
TC>>>> On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga <tculjaga@gmail.com>wrote:
TC>>>>
TC>>>>> hello guys,
TC>>>>>
TC>>>>>
TC>>>>> i was playing with mod_opal to see if i can make it working ... well it
TC>>>>> seems SIP-H323 interworking is not tuned at all.
TC>>>>>
TC>>>>> I have a call from a registered sip user (1001) to PSTN via mod_opal
TC>>>>>
TC>>>>>
TC>>>>> <include>
TC>>>>> <extension name="EMERGENCY">
TC>>>>> <condition field="destination_number"
TC>>>>> expression="^0(112|9[23456])$">
TC>>>>> <action application="set"
TC>>>>> data="effective_caller_id_number=1001282122"/>
TC>>>>> <action application="set" data="NCX_IP=10.4.4.254"/>
TC>>>>> <action application="set" data="call_timeout=30"/>
TC>>>>> <action application="set" data="hangup_after_bridge=true"/>
TC>>>>>
TC>>>>> <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
TC>>>>> </condition>
TC>>>>> </extension>
TC>>>>>
TC>>>>> <extension name="SPECIAL_SERVICES">
TC>>>>> <condition field="destination_number"
TC>>>>> expression="^0(9[01789]\d{3,4})$">
TC>>>>> <action application="set"
TC>>>>> data="effective_caller_id_number=1001282122"/>
TC>>>>> <action application="set" data="NCX_IP=10.4.4.254"/>
TC>>>>> <action application="set" data="call_timeout=30"/>
TC>>>>> <action application="set" data="hangup_after_bridge=true"/>
TC>>>>>
TC>>>>> <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
TC>>>>> </condition>
TC>>>>> </extension>
TC>>>>>
TC>>>>> <extension name="ENYTHING_ELSE">
TC>>>>> <condition field="destination_number"
TC>>>>> expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$">
TC>>>>> <action application="set"
TC>>>>> data="effective_caller_id_number=1001282122"/>
TC>>>>> <action application="set" data="NCX_IP=10.4.4.254"/>
TC>>>>> <action application="set" data="call_timeout=30"/>
TC>>>>> <action application="set" data="hangup_after_bridge=true"/>
TC>>>>>
TC>>>>> <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/>
TC>>>>> </condition>
TC>>>>> </extension>
TC>>>>> </include>
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> One of the many issues i sow is that FS connects the call on SIP leg
TC>>>>> before it actually receives H.225 connect from H323 leg... as it is
TC>>>>> configured to send 200 OK on the 1st H.225 message containing a FastStart
TC>>>>> element/OLC.
TC>>>>>
TC>>>>>
TC>>>>> Attached is the tcpdump i took on FS machine... just use this filter:
TC>>>>> "h225 or h245 or q931 or sip"
TC>>>>> Also, you can check the attac CDR,,,, this is an unanswered call i
TC>>>>> placed to PSTN and FS billed it 23 seconds.
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> Can anyone tell where i can do correct SIP - H323 message mappings to
TC>>>>> avoid this?
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> T.
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>>
TC>>>>> _______________________________________________
TC>>>>> FreeSWITCH-users mailing list
TC>>>>> FreeSWITCH-users@lists.freeswitch.org
TC>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
TC>>>>> UNSUBSCRIBE:
TC>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
TC>>>>> http://www.freeswitch.org
TC>>>>>
TC>>>>>
TC>>>>
TC>>>>
TC>>>> --
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PostPosted: Wed Oct 07, 2009 8:31 am    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

Hi,

Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
Quote:
barely get anybody asking about h.323.


H323 may not be popular for small ITSPs or small/medium PBXes, but
it's widely used by the big players.. and freeswitch doesnt share the
same goals with asterisk.



Best wishes,


Claudiu Filip




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PostPosted: Wed Oct 07, 2009 9:27 am    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

I am not commenting on how popular it is overall. 
I am commenting on the specific demand presented to us.
I don't know what else to say to explain that I am completely neutral when it comes to this topic.

One more time:

1) We are an open source project who volunteer most of our time as well as paid commercial support.
2) We made mod_opal anyway despite only a small handful of requests.
3) The devs from OPAL are consultants for hire and like us, they will look at what they can for free but you can hire them to heighten their attention span.

This is not a discussion on the merits of h323, I am more than happy to support it.  The OPAL guys are some of the best open source telephony guys out there and they have been doing this for over a decade.  We Just can't promise a whole lot of man hours towards it when we already work day and night managing jira with only a few managers.  We also cannot make any promises on the free time the OPAL devs have to devote to it either.




On Wed, Oct 7, 2009 at 8:14 AM, Claudiu Filip <claudiu@globtel.ro (claudiu@globtel.ro)> wrote:
Quote:

Hi,

Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
Quote:
barely get anybody asking about h.323.



H323 may not be popular for small ITSPs or small/medium PBXes, but
it's widely used by the big players.. and freeswitch doesnt share the
same goals with asterisk.



Best wishes,


Claudiu Filip





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PostPosted: Wed Oct 07, 2009 9:28 am    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

I think the way to determine the funding is to get all the most important issues up on jira, try to deal with them and see if we need to put bounties on any of them to get them done faster.


On Wed, Oct 7, 2009 at 3:37 AM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:
Quote:
Anthony,

of course, nobody wants to start anything... we are all here to help making FS a better product.

so, regarding the founding for mod_opal ... what is the amount you need?


Tihomir.




On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:
Quote:
I didn't mean to start anything.  I'm just saying we work very long hours and barely get anybody asking about h.323.
I wanted to support it and that's why we took up a collection to get funding for mod_opal but when only 1 donor showed any interest we were forced to proceed in our spare time which is very limited.

The developers of opal are not part of our project and they need financial compensation to be motivated to work on it.  Its not even related to me its only fair that an outside developer who makes his living as a consultant would want money to integrate his work into our project.

Like I said, I will do my best to point your issue to the opal devs but I cannot force them to work on it.





On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:
Quote:
Yeah I understand your point of view, but saying "I want a H.323 module" or "I want a Ferrari" wont magically make it happen.

We need to work on it ourselves or pay to the people that knows how to do it to do it for us.

There is no other way I think.

Diego




On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:
Quote:
Diego,

what i'm pointing here is the situation where you have a great product that lacks in one of most common protocol. It is true H323 is going to disappear (eventually), it is true that the community prefers SIP/IAX instead ... but the reality still remains. H323 is going to be used for quite a long time to exchange a lot of traffic while FS will be left aside. Today, when you setup an IP peering interconnection 80% of carriers will prefer H323.

Of course, developing something costs "time" (and we all know what time stands for...) and as i said, i understand the financial point of view and i really understand if nobody is going to work on that, but let's face it FS doesn't have any usable module to reliably handle H323 protocol.


said that, i don't intend to offend anyone... just facing the reality.


regarding the h323 module, we don't have any issue fixing the existing or developing a new one... but before we go developing something it is always better check if the thing you want already exists in an usable state or not... that's what i did today.


So, I'm interested in a reliable module handling H323v4... anyone else?


T.






On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola <diego.viola@gmail.com (diego.viola@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:
Quote:
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 (anthony.minessale@gmail.com)> wrote:
Quote:
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 (tculjaga@gmail.com)> wrote:


Quote:

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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PostPosted: Wed Oct 07, 2009 10:21 am    Post subject: [Freeswitch-users] mod_opal - call charged before H.225 conn Reply with quote

Claudiu Filip wrote:
Quote:
Hi,

Wednesday, October 7, 2009, 3:58:20 AM, Anthony M. wrote:
Quote:
barely get anybody asking about h.323.


H323 may not be popular for small ITSPs or small/medium PBXes, but
it's widely used by the big players.. and freeswitch doesnt share the
same goals with asterisk.



Best wishes,


Claudiu Filip


I have to agree with Filip, H323 is not popular (or sometimes even
known) to the VoIP end user and small PBX community. In fact many VoIP
users that I've talked with even believe that VoIP started with SIP,
which is not true.

H323 although diminishing in usage is still used among voice carriers
because it better resembles telephony environments and is not prone to
errors and problems of SIP (example: if there is a network error in
mid-call H323 uses TCP for signaling and therefore almost immediately
detects the error to hang-up the call, while SIP _typically_ uses UDP
and cannot detect such issues via signaling but only if a media gateway
monitors RTP activity etc.).

As it is already pointed out, many carriers switch to SIP steadily, but
still H323 is important for carrier-to carrier interconnection.

FreeSWITCH is positioned/advertised as a high-capacity softswitch (not
just a SME PBX) which implies interconnection with large carriers (who
still prefer H323).

There are many SIP platforms out there, open source and closed source
that can do SIP (most of them pretty good, some of them excellent).

There are many closed source platforms that can do SIP and H323 pretty
good (and some of them excellent).

However, there is no open source platform that can do BOTH SIP and H323
REALLY good. That could be a competitive advantage for an open source
platform...

NOTE: All the above is just academic discussion. A reply to the above in
the spirit "put your money where your mouth is" is not really
constructive and will be countered by a series of valid arguments
against it.

Best regards,
Vlasis Hatzistavrou.

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