Hello, Yes, the issue I am having is currently only with Google Talk. Wonder if what development will be made to fix this issue.
--Elliot On Wed, Jun 15, 2011 at 9:20 AM, Vladimir Mikhelson <[email protected]> wrote: > Elliot, > > I do not think Issue # 17993 is related. As Terry Wilson says on the > Bug Tracker, "Google Voice inbound calls still work, it is just coming > from Google Talk that doesn't." > > -Vladimir > > > On 6/14/2011 5:51 PM, Elliot Murdock wrote: >> Hello, >> >> Seems that it's been spotted and tracked at >> https://issues.asterisk.org/jira/browse/ASTERISK-17993 >> >> --Elliot >> >> >> On Tue, Jun 14, 2011 at 7:03 PM, Vladimir Mikhelson <[email protected]> >> wrote: >>> Elliot, >>> >>> You need to execute "sendDTMF(1) " >>> >>> Articles are available with detailed setup description. >>> >>> -Vladimir >>> >>> >>> >>> >>> On 6/14/2011 1:26 AM, Elliot Murdock wrote: >>>> Hello, >>>> >>>> To help clarify, Jabber is receiving the incoming packets, but >>>> Asterisk does not seem to be associating it with the gtalk >>>> configuration and the call is not routed into any context. The remote >>>> caller only hears continous ringing. However, outgoing, gtalk and >>>> jabber work fine. >>>> >>>> What could be the problem? >>>> >>>> Elliot >>>> >>>> On Tue, Jun 14, 2011 at 2:02 AM, Elliot Murdock <[email protected]> wrote: >>>>> Hello, >>>>> >>>>> I am using 1.8.4.2 and while outgoing seems to work, incoming still >>>>> does not route calls in to the appropriate context. >>>>> >>>>> Please advise. >>>>> >>>>> Thank you, >>>>> Elliot >>>>> >>>>> On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell >>>>> <[email protected]> wrote: >>>>>> You must have 1.8+ its already been posted the 1.6 didn’t get a backport >>>>>> fix >>>>>> in the jabber protocol. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> From: [email protected] >>>>>> [mailto:[email protected]] On Behalf Of Leandro >>>>>> Dardini >>>>>> Sent: Saturday, April 16, 2011 3:57 AM >>>>>> To: [email protected] >>>>>> Subject: [asterisk-users] Google Voice receiving call problem >>>>>> >>>>>> >>>>>> >>>>>> Hello, >>>>>> I have a Google Voice phone number and want to connect it to my asterisk >>>>>> box >>>>>> to have calls handled to my SIP account. >>>>>> >>>>>> When I call the number I receive the correct INCOMING request on Jabber >>>>>> portion of asterisk, but the call is not connected to the gtalk part. >>>>>> >>>>>> JABBER: asterisk INCOMING: <iq >>>>>> from="[email protected]/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" >>>>>> to="[email protected]/asterisk438D86E0" >>>>>> id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session >>>>>> type="initiate" id="[email protected]" >>>>>> initiator="[email protected]/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" >>>>>> 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> >>>>>> >>>>>> No other messages are logged. Where is my mistake? >>>>>> >>>>>> I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are >>>>>> the >>>>>> relevant files. >>>>>> >>>>>> Thank you >>>>>> >>>>>> Leandro >>>>>> >>>>>> ####### jabber.conf >>>>>> >>>>>> [general] >>>>>> autoregister=yes >>>>>> >>>>>> [asterisk] >>>>>> type=client >>>>>> serverhost=talk.google.com >>>>>> [email protected] >>>>>> secret=********** >>>>>> priority=1 >>>>>> port=5222 >>>>>> usetls=yes >>>>>> usesasl=yes >>>>>> [email protected] >>>>>> status=available >>>>>> >>>>>> ####### gtalk.conf >>>>>> >>>>>> [general] >>>>>> context=default >>>>>> bindaddr=0.0.0.0 >>>>>> allowguest=yes >>>>>> >>>>>> [guest] >>>>>> disallow=all >>>>>> allow=ulaw >>>>>> context=google-in >>>>>> >>>>>> [ldardini] >>>>>> [email protected] >>>>>> disallow=all >>>>>> allow=ulaw >>>>>> context=google-in >>>>>> connection=asterisk >>>>>> >>>>>> ######## extension.ael >>>>>> >>>>>> context google-in { >>>>>> s => { >>>>>> NoOp( Call from Gtalk ); >>>>>> Dial(SIP/************@************,60,r); >>>>>> }; >>>>>> } >>>>>> >>>>>> >>>>>> -- >>>>>> _____________________________________________________________________ >>>>>> -- 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
