I don’t’ appear to have an jabber [] OUTGOING packets?
I get just 1 incoming packet, and it just sits there, until it rings to voicemail. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Friday, February 11, 2011 1:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue 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] 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> wrote: Sorry, Asterisk Build 1.6.2.7 From: 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/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/Talk - Connected ---- Number of users: 1 ---- CLI on incoming Call ---- bannana*CLI> JABBER: jb_jabber INCOMING: <iq from="+1*********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2" to="******@gmail.com/TalkD876FAA0" id="jingle:10.218.14.137-17447266:1:03800E94" type="set"><ses:session type="initiate" id="SIP1007753261@10.218.122.83" initiator="+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" to="******@gmail.com/TalkD876FAA0" id="jingle:10.218.14.137-17447266:1:03800EB9" type="set"><ses:session type="terminate" id="SIP1007753261@10.218.122.83" initiator="+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
