Hi Jan, yep setting ptime-override-value to 20 cured the problem, thanks. regards, John On Thu, 2010-10-14 at 00:38 +0200, Jan Riedinger wrote:
Hi John,
a ptime of 2 seems to be much too less and is a strange value, the default value is 20. Maybe you should try to force its usage. I'm not sure, how this can be done, maybe by using <param name="ptime-override-value" value="20"/> or by specifying G729@20i for the codec.
BR Jan
Am 13.10.2010 16:59, schrieb 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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-- Jan Riedinger Phone : +49-30-39 73 19 66 Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64 E-Mail: riedinger@sns.eu SNS Consult GmbH ICQ : 163-237-041 Südwestkorso 49a MSN : jan@sns-consult.de 14197 Berlin GERMANY Skype : Jan Riedinger
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