H.450 is currently not supported in the ooh323c stack. 

Regards, 

Ed Day
Objective Systems, Inc.
http://www.obj-sys.com


  ----- Original Message ----- 
  From: Manish Varma 
  To: ooh323c-devel@lists.sourceforge.net 
  Sent: Wednesday, May 16, 2007 2:08 PM
  Subject: [ooh323c-devel] call conference and call transfer


  Hello everyone,

         I am using ooh323c 0.8.3.. I would like to know whether this version 
supports Call conference and Call transfer feature as supplemental service 
using H450?

  Many Thanks in Advance,
  Manish.

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    Today's Topics:

    1. Re: Port usage (Avin Patel)
    2. Re: ooh323: early audio problem (Avin Patel)


    ----------------------------------------------------------------------

    Message: 1
    Date: Fri, 20 Apr 2007 16:51:27 -0500
    From: Avin Patel 
    Subject: Re: [ooh323c-devel] Port usage
    To: Tim King 
    Cc: ooh323c-devel@lists.sourceforge.net
    Message-ID: <[EMAIL PROTECTED]>
    Content-Type: text/plain; charset=ISO-8859-1; format=flowed

    Hi Tim,
    Thank you for notes. I have added this in FAQ, and added 
    h225portrange=10000,20000 configuration option for ooh323.conf file.
    So you can change to need range from configuration file.

    Can be downloaded from asterisk-addons-1.2 branch.

    Regards,
    Avin Patel
    Objective Systems, Inc.


    Tim King wrote:
    > Well after a day or so here's another gotcha unveiled.
    > 
    > Compiled-in port range used by ooh323 in H2250 setup is undocumented and 
    > is from 12030 to 12230.
    > 
    > The default range for RTP in Asterisk's rtp.conf file is to use 
10000-20000.
    > 
    > So after you have been using this in a production environment for some 
    > time the RTP port usage clashes with the H2250 setup which subsequently 
    > fails.
    > 
    > Workaround: Alter rtp.conf. Solution: Make the default range something 
    > under 10000 please.
    > 



    ------------------------------

    Message: 2
    Date: Fri, 20 Apr 2007 18:17:26 -0500
    From: Avin Patel 
    Subject: Re: [ooh323c-devel] ooh323: early audio problem
    To: George Melika , ooh323c

    Message-ID: <[EMAIL PROTECTED]>
    Content-Type: text/plain; charset="iso-8859-1"

    Hi George,
    I have checked the tcpdump for last call. The Cisco router is stopping 
    the fast start setup, as there was no fast start element in CALL 
    PROCEEDING message. Usually the Call Proceeding message was ignored. 
    ooh323 sends fast start response in Alerting and Setup message only.

    I have enabled the fast start response in Call Proceeding message. That 
    should fix the early audio problem. To apply this change, You can 
    replace attached file with asterisk-ooh323c/ooh323c/src/ooq931.c file.
    Rename ooq931.zip.txt to ooq931.zip, than unzip it.

    Let me know it works for you.

    Regards,
    Avin Patel
    Objective Systems, Inc.


    George Melika wrote:
    > Avin,
    > 
    > Here's a tcpdump of the h323 traffic with the GW. Let me know if you need 
anything else.
    > 
    > George
    > 
    > ----- Original Message ----
    > From: Avin Patel 
    > To: George Melika 
    > Cc: ooh323c-devel@lists.sourceforge.net
    > Sent: Friday, March 23, 2007 4:00:37 PM
    > Subject: Re: [ooh323c-devel] ooh323: Faking Ring Problem
    > 
    > 
    > Hi George,
    > 
    >> Me: Using an IAX2 phone
    >> GW1: Asterisk 1.4 w/ OOH323 and SIP
    >> GW2: CISCO router with H323 and SIP
    >>
    > Is this configuration you are testing?
    > 
    > IAX2 --- IAX channel ----> (IAX[GW1 asterisk]ooh323) ----H.323 channel 
    > ---> (H.323[ GW2 Cisco ]) ----> PSTN
    > 
    >> Calls are originating from GW1 and routed through GW2 to the PSTN and I 
am listening on my IAX2 phone. When I use SIP between GW1 and GW2 I get the 
proper ringtone as generated by the PSTN. When I switch to OOH323 I get a 
locally generated (North American) ringtone, and they are generated locally 
before the destination is actually ringing.
    >>
    > You can add one more change at ooq931.c file:
    > if(pq931Msg->messageType == Q931SetupMsg){
    > 
    > /*Add progress indicator IE */
    > if(pq931Msg->messageType == Q931SetupMsg ||
    > pq931Msg->messageType == Q931AlertingMsg ||
    > pq931Msg->messageType == Q931CallProceedingMsg) {
    > 
    > msgbuf[i++] = Q931ProgressIndicatorIE;
    > msgbuf[i++] = 2; //Length is 2 octet
    > msgbuf[i++] = 0x80; //PI=8
    > msgbuf[i++] = 0x88;
    > }
    > 
    > If you still face the problem, Can you provide us captured messages 
    > between GW1 and GW2 using tcpdump ?
    > 
    > Regards,
    > Avin Patel
    > Objective Systems, Inc.
    > 
    -------------- next part --------------
    An embedded and charset-unspecified text was scrubbed...
    Name: ooq931.zip.txt

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