William, I have just noticed that you have several configuration statements commented out.
I would suggest to un-comment the "status=" in jabber.conf. I would also suggest to un-comment the "timeout=", I am not that concerned of the "keepalive=". You can reload jabber, no need to restart the Asterisk. -Vladimir On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: > William, > > Have you tried outgoing calls? What happens there? > > Have you restarted the Asterisk after you fixed the typo? > > -Vladimir > > > > On 2/10/2011 10:44 PM, William Stillwell wrote: >> >> Yeah, that was a typo, but I fixed, still no dice. >> >> >> >> The incoming jabber call doesn’t fire the gtalk connection. >> >> >> >> >> >> *From:*[email protected] >> [mailto:[email protected]] *On Behalf Of >> *Warren Selby >> *Sent:* Thursday, February 10, 2011 10:16 PM >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue >> >> >> >> You've got connection=jp_jabber defined in one file, and [jb_jabber] >> defined in the other. >> >> Thanks, >> >> --Warren Selby, dCAP >> >> >> On Feb 10, 2011, at 5:55 PM, "William Stillwell" >> <[email protected] <mailto:[email protected]>> wrote: >> >> Sorry, Asterisk Build 1.6.2.7 >> >> >> >> *From:*[email protected] >> <mailto:[email protected]> >> [mailto:[email protected]] *On Behalf Of >> *William Stillwell >> *Sent:* Thursday, February 10, 2011 6:50 PM >> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' >> *Subject:* [asterisk-users] Gtalk/Jabber Issue >> >> >> >> OK, im pulling my hair out, everything looks configured right, >> deleted, and started over, etc, etc. but can’t seem to get this >> to work >> >> >> >> >> >> Gtalk.conf >> >> >> >> [general] >> >> context=google-in >> >> allowguest=yes >> >> bindaddr=192.168.xxx.xxx >> >> extenip=96.254.xxx.xxx >> >> >> >> [guest] >> >> context=google-in >> >> disallow=all >> >> allow=ulaw >> >> allow=g729 >> >> connection=jp_jabber >> >> >> >> jabber.conf >> >> >> >> [general] >> >> debug=yes >> >> ;autoprune=no >> >> autoregister=yes >> >> >> >> >> >> [jb_jabber] >> >> type=client >> >> serverhost=talk.google.com >> >> [email protected] >> <mailto:[email protected]>/Talk >> >> secret=XXXXXXX >> >> port=5222 >> >> usetls=yes >> >> usesasl=yes >> >> ;status=Available >> >> statusmessage="Connected via Asterisk" >> >> ;timeout=100 >> >> ;keepalive=yes >> >> >> >> >> >> Extensions.conf >> >> >> >> [google-in] >> >> exten => s,1,NoOp(Call from GTalk) >> >> exten => s,n,Set(CallerID(Name)="From GoogleTalk") >> >> exten => s,n,Dial(SIP/1000) >> >> >> >> jabber show connected >> >> >> >> Jabber Users and their status: >> >> User: [email protected] <mailto:[email protected]>/Talk >> - Connected >> >> ---- >> >> Number of users: 1 >> >> >> >> >> >> ---- CLI on incoming Call ---- >> >> >> >> bannana*CLI> >> >> JABBER: jb_jabber INCOMING: <iq >> from="+1*********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 >> <mailto:+1*********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>" >> to="******@gmail.com/TalkD876FAA0 >> <mailto:******@gmail.com/TalkD876FAA0>" >> id="jingle:10.218.14.137-17447266:1:03800E94" >> type="set"><ses:session type="initiate" >> id="[email protected] >> <mailto:[email protected]>" >> initiator="+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 >> <mailto:+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>" >> xmlns:ses="http://www.google.com/session";><pho:description >> xmlns:pho="http://www.google.com/session/phone";><pho:payload-type >> id="0" name="PCMU" clockrate="8000"/><pho:payload-type id="101" >> name="telephone-event"/></pho:description><transport >> behind-symmetric-nat="false" >> can-receive-from-symmetric-nat="false" >> xmlns="http://www.google.com/transport/raw-udp"/><transport >> xmlns="http://www.google.com/transport/p2p"/></ses:session></iq> >> >> bannana*CLI> >> >> JABBER: jb_jabber INCOMING: <iq >> from="+1********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 >> <mailto:+1********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>" >> to="******@gmail.com/TalkD876FAA0 >> <mailto:******@gmail.com/TalkD876FAA0>" >> id="jingle:10.218.14.137-17447266:1:03800EB9" >> type="set"><ses:session type="terminate" >> id="[email protected] >> <mailto:[email protected]>" >> initiator="+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 >> <mailto:+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>" >> xmlns:ses="http://www.google.com/session";><pho:call-ended >> xmlns:pho="http://www.google.com/session/phone";>Call >> cancelled</pho:call-ended></ses:session></iq> >> >> bannana*CLI> >> >> >> >> >> >> it doesn’t even try to fire the google-in context ? >> >> >> >> Lastest Version of iksemel Installed, asterisk was rebuild after >> installed, asterisk sees both jabber/gtalk commands. >> >> >> >> It just will NOT ring my dialplan. >> >> >> >> >> >> >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> >> >> -- >> _____________________________________________________________________ >> -- 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 > > > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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
