[Freeswitch-users] Setting up for googletalk
Shiladitya Sircar
shiladitya at gmail.com
Tue Feb 20 19:39:47 PST 2007
Hi,
Thanks for the clarification and help. I was wondering what version of
freeswitch are you using? I have configured pretty much identical as your
sample and now I get "TLS NOT SUPPORTED IN THUS BUILD" below is the debug
message I get:
freeswitch at JSIRCAR> 2007-02-20 22:34:33 [DEBUG] libdingaling.c:1459
xmpp_connect
() io error 7
SEND[<?xml version='1.0'?><stream:stream xmlns:stream='
http://etherx.jabber.org/
streams' xmlns='jabber:client' to='gmail.com' version='1.0'>]
RECV[<?xml version="1.0" encoding="UTF-8"?><stream:stream from="gmail.com"
id="2
6571D019C85090F" version="1.0" xmlns:stream="
http://etherx.jabber.org/streams" x
mlns="jabber:client">]
RECV2007-02-20 22:34:34 [DEBUG] libdingaling.c:1096 on_stream() TLS NOT
SUPPORTE
D IN THIS BUILD!
[<stream:features><starttls
xmlns="urn:ietf:params:xml:ns:xmpp-tls"/><mechanisms
xmlns="urn:ietf:params:xml:ns:xmpp-sasl"><mechanism>X-GOOGLE-TOKEN</mechanism><
/mechanisms></stream:features>]
The freeswitch version that I have compiled and running is : FreeSwitch
Version 1.0.4296M
thanks
-ssircar
On 2/20/07, Pekka K. Kurki <pekka.kurki at intellectics.com> wrote:
>
> 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 at 8000"/>
> </settings>
>
> <profile type="client">
> <param name="name" value="gtalk"/>
> <param name="login" value="myuser at gmail.com/talk"<myuser at 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 at gmail.com"<effective_caller_id_number=myuser at gmail.com>
> />
> <action application="bridge" data=
> "dingaling/gtalk/$1 at gmail.com" <dingaling/gtalk/$1 at 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 at sip.intellectics.com"<sofia/sip.intellectics.com/mysipuser at 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 at sip.intellectics.com
> skype: pkkurki, pkkurki-sip (sip gateway)
> gtalk: pekkis50 at gtalk.com (sip gateway - experimental)
> gizmo: pekkis50
> h323:432256815811 at sip.intellectics.com
> email: Pekka.Kurki at intellectics.com
> www: http://pekka.kurki.at
>
>
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