1:1 nat, I even turned off iptables.. same issue.
Guess I will try install wireshark when I get back next week, im done farting with this tonight, when I get back from fort Lauderdale next week I will play with it some more. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Friday, February 11, 2011 2:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Gtalk/Jabber Issue William, Another thing to exclude is networking. Can you verify that nothing blocks the specific traffic on your network? Any chance of taking the packet trace on your gateway? -Vladimir On 2/11/2011 1:18 AM, William Stillwell wrote: 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
-- _____________________________________________________________________ -- 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
