[Freeswitch-users] H323 to Sip conversion problem

[John Carpenter]

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 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 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
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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
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> 14197 Berlin GERMANY                    Skype :  Jan Riedinger
> 
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