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