William, I have gone through the similar frustration recently. Everything works as of early morning yesterday. The big difference, I am on 1.8.2.3.
Have you seen this ticket on the tracker https://issues.asterisk.org/view.php?id=10512 ? Anything applicable to your case? The messages are identical to yours on the outgoing call. -Vladimir On 2/11/2011 12:32 AM, William Stillwell wrote: > > Still no dice.. > > > > This make no since.. ive gone over the config a million times now.. > > > > The windows gtalk /voice client works just fine. (incoming and > outgoing calls) > > > > > > > > *From:*asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of > *Vladimir Mikhelson > *Sent:* Friday, February 11, 2011 12:51 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue > > > > 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:*asterisk-users-boun...@lists.digium.com > <mailto:asterisk-users-boun...@lists.digium.com> > [mailto:asterisk-users-boun...@lists.digium.com] *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" > <will...@stillwellsoft.com <mailto:will...@stillwellsoft.com>> wrote: > > Sorry, Asterisk Build 1.6.2.7 > > > > *From:*asterisk-users-boun...@lists.digium.com > <mailto:asterisk-users-boun...@lists.digium.com> > [mailto:asterisk-users-boun...@lists.digium.com] *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 > > username=xxxxxx...@gmail.com > <mailto:username=xxxxxx...@gmail.com>/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: xxx...@gmail.com <mailto:xxx...@gmail.com>/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="SIP1007753261@10.218.122.83 > <mailto:SIP1007753261@10.218.122.83>" > 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="SIP1007753261@10.218.122.83 > <mailto:SIP1007753261@10.218.122.83>" > 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
-- _____________________________________________________________________ -- 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