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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<BR>
<BR>
"Unsupported ptime of 2 on write Audio codec G.729{sw} for connection [0xb4c0ee10]"<BR>
<BR>
and call fails to connect. If I use ulaw codec call fails to connect with<BR>
<BR>
"Write PDU fail: no control channel"<BR>
<BR>
If I make a straight SIP to SIP call though same provider all work ok. I have posted log of call in <A HREF="http://pastebin.freeswitch.org/14216">http://pastebin.freeswitch.org/14216</A> because it is rather large.<BR>
<BR>
my h323.conf.xml file looks like this<BR>
<BR>
<configuration name="h323.conf" description="H323 Endpoints"><BR>
<settings><BR>
<param name="trace-level" value="10"/><BR>
<param name="context" value="public"/><BR>
<param name="dialplan" value="XML"/><BR>
<param name="codec-prefs" value="PCMA,PCMU,GSM,G729"/><BR>
<param name="use-rtp-timer" value="true"/> <!-- enable RTP timer - should always be enabled --><BR>
<param name="rtp-timer-name" value="soft"/> <!-- Timer name, soft is default --><BR>
<!-- <param name="ptime-override-value" value="20"/> --> <!-- Override negotiated ptime value with this value --><BR>
<param name="gk-address" value=""/> <!-- empty to disable, "*" to search LAN --><BR>
<param name="gk-identifer" value=""/> <!-- optional name of gk --><BR>
<param name="gk-interface" value=""/> <!-- mandatory listener interface name --><BR>
<param name="gk-retry" value="30"/> <!-- optional GK register retry timer --><BR>
<param name="faststart" value="true"/> <!-- optional --><BR>
<param name="h245tunneling" value="true"/> <!-- optional --><BR>
<param name="h245insetup" value="true"/> <!-- optional --><BR>
<param name="jitter-size" value="60"/> <!-- optional --><BR>
<param name="progress-indication" value="8"/> <!-- optional - PI value in progress message--><BR>
<param name="alerting-indication" value="8"/> <!-- optional - PI value in alerting message--><BR>
<param name="endpoint-name" value="fs"/><BR>
<param name="fax-old-asn" value="true"/><BR>
</settings><BR>
<listeners><BR>
<listener name="default"><BR>
<param name="h323-ip" value="$${local_ip_v4}"/><BR>
<param name="h323-port" value="1720"/><BR>
</listener><BR>
</listeners><BR>
</configuration><BR>
<BR>
And this is the log of the xml_curl dialplan that is executed<BR>
<BR>
<?xml version="1.0" encoding="UTF-8" standalone="no"?><BR>
<document type="freeswitch/xml"><BR>
<section name="dialplan" description="php dialplan"><BR>
<context name="public"><BR>
<extension name="normal"><BR>
<condition field="destination_number" expression="^(\d+)$"><BR>
<action application="set" data="bypass_media=false"/><BR>
<action application="set" data="proxy_media=true"/><BR>
<action application="bridge" data="sofia/gateway/arbinet-o/$1"/><BR>
</condition><BR>
</extension><BR>
</context><BR>
</section><BR>
</document><BR>
<BR>
Any help will be greatly appreciated <BR>
<BR>
regards, John Carpenter
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