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. 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