Any expert on this issue? -jesse On Wed, Aug 18, 2010 at 5:05 PM, jesse zhao <chat2jesse@gmail.com> wrote:
hi, help:
I searched online pages and FS wiki , basically the doc about Gtalk and FS is pretty lousy. I still couldn't get my gtalk and FS work together.
two major issues:
1) when I call gtalk(gtalk=XYZ) from SIP phone, call is established, but no audio. 2) how could I call SIP from gtalk ? when I request chat to sip, gchat in gmail requests an invitation. suppose the voice request is : conf+1000@172.18.115.73
here are my config files: jingle_profiles/client.xml
<profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="XYZ@gmail.com/talk"/> <param name="password" value="XXXXX"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <!-- SASL "plain" or "md5" --> <param name="sasl" value="md5"/> <!-- if the server where the jabber is hosted is not the same as the one in the jid --> <param name="server" value="talk.google.com"/> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <!-- default extension (if one cannot be determined) --> <param name="exten" value="jingle2sip"/> <!-- VAD choose one --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <param name="vad" value="both"/> <!--<param name="avatar" value="/path/to/tiny.jpg"/>--> <!--<param name="candidate-acl" value="wan.auto"/>--> <param name="local-network-acl" value="localnet.auto"/> </profile> </include>
dialplan/default.xml
<extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^conf\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/default/$1%172.18.115.73"/> </condition> </extension>
<extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^gtalk=([a-zA-z0-9.-]+)$"> <action application="set" data="no_media=false"/> <action application="set" data="effective_caller_id_number=myuser at gmail.com"/> <action application="bridge" data="dingaling/gmail.com/$1@gmail.com"/> </condition>
I tried suggestion from http://lists.freeswitch.org/pipermail/freeswitch-users/2007-February/028456...., above two issues still not work.
I heard id: intralanman is good at this, if you saw the post, please share your advice.
-jesse