[Freeswitch-users] H323 to Sip conversion problem

[John Carpenter]

Hi, I am trying to bridge an incoming h323 call to an external SIP
provider, I am using latest git release. If I use G729 codec I get
message

"Unsupported ptime of 2 on  write Audio codec G.729{sw}  for connection
[0xb4c0ee10]"

and call fails to connect. If I use ulaw codec call fails to connect
with

"Write PDU fail: no control channel"

If I make a straight SIP to SIP call though same provider all work ok. I
have posted log of call in http://pastebin.freeswitch.org/14216 because
it is rather large.
This is my first venture into using the mod_h323 module and maybe I am
doing something stupid but have read all docs and seem to have hit a
brick wall on this.

my h323.conf.xml file looks like this

<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="use-rtp-timer" value="true"/> <!-- enable RTP timer -
should always be enabled -->
   <param name="rtp-timer-name" value="soft"/> <!-- Timer name, soft is
default -->
   <!-- <param name="ptime-override-value" value="20"/> --> <!--
Override negotiated ptime value with this value -->
   <param name="gk-address" value=""/>    <!-- empty to disable, "*" to
search LAN -->
   <param name="gk-identifer" value=""/>  <!-- optional name of gk -->
   <param name="gk-interface" value=""/>  <!-- mandatory listener
interface name -->
   <param name="gk-retry" value="30"/>  <!-- optional GK register retry
timer -->
   <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-->
   <param name="endpoint-name" value="fs"/>
   <param name="fax-old-asn" value="true"/>
</settings>
<listeners>
   <listener name="default">
     <param name="h323-ip" value="$${local_ip_v4}"/>
     <param name="h323-port" value="1720"/>
   </listener>
</listeners>
</configuration>

And this is the log of the xml_curl dialplan that is executed

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="dialplan" description="php dialplan">
<context name="public">
<extension name="normal">
<condition field="destination_number" expression="^(\d+)$">
<action application="set" data="bypass_media=false"/>
<action application="set" data="proxy_media=true"/>
<action application="bridge" data="sofia/gateway/arbinet-o/$1"/>
</condition>
</extension>
</context>
</section>
</document>

Any help will be greatly appreciated 

regards, John Carpenter
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