How to call from SIP to H323 using freeswitch?

Question

I tried to integrate calls from H323 to SIP using freeswitch so these functions could be achieved:

1. sip can call h323
2. h323 can call sip

Endpoint for each signalling protocol:

1. H323 : Avaya and Polycom Realpresence
2. SIP : Eyebeam / X-Lite

I am using mod_h323 inside freeswitch. Both realpresence desktop and avaya could call each other successfully using voice and video.

Something strange happened when I tried to make a call from SIP(Eyebeam) to H323 (Realpresence). Realpresence show an incoming call from SIP, but after a few second it says that call already hanged up. When I tried to grep inside my server, it seemed that freeswitch already crashed and force closed, then I needed to restart freeswitch again.

Below is my h323.conf:

<configuration name="h323.conf" description="H323 Endpoints">
  <settings>
    <param name="trace-level" value="10" />
    <param name="context" value="public" />
    <param name="dialplan" value="XML" />
    <param name="codec-prefs" value="PCMA,PCMU,GSM,G729" />
    <!-- <param name="codec-prefs" value="PCMA,PCMU"/>-->
    <param name="use-rtp-timer" value="true" /> <!-- enable RTP timer - should always be enabled -->
    <param name="rtp-timer-name" value="soft" /> <!-- soft is default -->
    <param name="gk-address" value="172.16.0.97" />    <!-- empty to disable, "*" to search LAN -->
    <param name="gk-retry" value="30" /> <!-- optional GK register retry timer -->
    <param name="endpoint-name" value="fs" />
    <param name="fax-old-asn" value="true" />
    <param name="ptime-override-value" value="20" />
    <param name="faststart" value="true" /> <!-- optional -->
    <param name="h245tunneling" value="true" /> <!-- optional -->
    <param name="h245insetup" value="true" /> <!-- optional -->
    <param name="jitter-size" value="60" /> <!-- optional -->
    <param name="progress-indication" value="8" /> <!-- optional - PI
value in progress message-->
    <param name="alerting-indication" value="8" /> <!-- optional - PI
value in alerting message-->
  </settings>
  <listeners>
    <listener name="default">
      <param name="h323-ip" value="$${local_ip_v4}" />
      <param name="h323-port" value="1720" />
    </listener>
  </listeners>
</configuration>

Below is my last log before it crashed

      2023-10-05 11:08:56.382 [DEBUG] Reading PDUs: callRef=3272
          2023-10-05 11:08:56.382 [DEBUG] PThread::PXBlockOnIO(82,0)

If anyone has experience setting up call between H323 and SIP using freeswitch, please help me.

UPDATE: I changed realpresence to another softphone that use H323 protocol called "Ekiga". Now calling between ekiga and X-Lite already okay and has voice. The problem is calling from SIP to Avaya H323, call can be established, but there are no voices from both sides. Anyone faced this issue before?