I am new to asterisk but I think that with gtalk things are different!!!
In my extensions.conf I have:
exten => 2103,2,Dial(SIP/sip.generic,20); twinkle client
exten => _33XXXXXXXXX,1,Set(CALLERID(num)=01215063743); that doesn't work
exten => _33XXXXXXXXX,2,Dial(gtalk/asterisk/+${EXTEN}@voice.google.com)in my sip.conf I have nothing in relation to gtalk: [sip.generic] context=google ;context to be defined in extensions . conf type=friend ;use defined context for both inbound and outbound calls disallow=all ;enable gsm audio codec allow=gsm ;host could be dynamic host=dynamic canreinvite=no in my gtalk.conf: [general] context=google ; Context to dump call into bindaddr=0.0.0.0 ; Address to bind to allowguest=yes ; Allow calls from people not in list of peers callerid=2121212 [guest] ; special account for options on guest account disallow=all allow=ulaw allow=gsm context=google connection=asterisk callerid=2121212 in my jabber.conf: [general] ; debug = yes ; uncomment to Enable debugging ( disabled by default ). autoregister=yes ; Auto register users from buddy list . [asterisk] type=client serverhost=talk.google.com ; Route to server [email protected]/Talk ; Username with Talk resource . secret=*************** ; Gmail Password usetls=yes ; Use tls usesasl=yes ; Use sasl statusmessage="I am available" ; Google Talk status message timeout=100 ; Timeout ( in seconds ), default is 5 On Thu, Jul 28, 2011 at 3:27 PM, Alex Balashov <[email protected]>wrote: > On 07/28/2011 09:22 AM, A.H. Jos wrote: > >> Hi list, >> I have Asterisk speaking with google talk, is there any way to set or >> at least hide my google voice number when I call others? >> > > Set a different 'callerid' on either your outgoing sip.conf peer? > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
