[Freeswitch-users] H323 to Sip conversion problem

Jan Riedinger riedinger at sns.eu
Wed Oct 13 15:38:32 PDT 2010


  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 at 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 at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
Südwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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