here my setups that work as an asterisk-to-gtk gateway. it works perfectly when connecting to/from original gtalk on windows but not with linux clients as jabbin or psi or tapioca. in those cases the codec negociation fails somehow... here my dingaling.conf: <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="1"/> <param name="codec-prefs" value="PCMU@8000"/> </settings> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="myuser@gmail.com/talk"/> <param name="password" value="mypasswd"/> <param name="dialplan" value="XML"/> <param name="message" value="Jingle-to-SIP Gateway via Freeswitch"/> <param name="rtp-ip" value="85.126.xxx.xxx"/> <param name="auto-login" value="true"/> <param name="auto-reply" value="Press *Call* to call me via FreeSWITCH and visit me at http://pekka.kurki.at"/> <!-- 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:5222"/> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <!-- or --> <!-- <param name="rtp-ip" value="my_lan_ip"/> --> <param name="ext-rtp-ip" value="85.126.xxx.xxx"/> <!-- default extension (if one cannot be determined) --> <param name="exten" value="1004j"/> <!-- VAD choose one --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <param name="vad" value="both"/> </profile> </configuration> and here the relevant parts of default.xml: <context name="default"> <extension name="gtalk"> <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@gmail.com"/> <action application="bridge" data="dingaling/gtalk/$1@gmail.com"/> </condition> </extension> <extension name="1004j"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^1004j"> <action application="set" data="no_media=false"/> <action application="bridge" data="sofia/sip.intellectics.com/mysipuser@sip.intellectics.com"/> </condition> </extension> </context> - sip calls with target address "gtalk=usrname" are forwarded to gtalk - all calls from gtalk are forwarded to my sip phone on asterisk server this works ok for me but i am still looking for solution for linux clients... br, pekka Shiladitya Sircar wrote:
Hello, I was wondering if any of you can point me towards some documentation in configuring FreeSwitch with Google Talk, using dingaling.conf ? A sample dingaling.conf and a dail plan would be very helpful.
thanks -ssircar ------------------------------------------------------------------------
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-- (Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden, Austria Tel. +43 2256 815811 Mobile +43 699 12532539 Fax +43 2256 815819 (Tel. US: 360-2267729 UK: 0870 4787064) sip:pekka.kurki@sip.intellectics.com skype: pkkurki, pkkurki-sip (sip gateway) gtalk: pekkis50@gtalk.com (sip gateway - experimental) gizmo: pekkis50 h323:432256815811@sip.intellectics.com email: Pekka.Kurki@intellectics.com www: http://pekka.kurki.at