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 <v...@mikhelson.com> 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 <murdo...@gmail.com> 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 >>> <will...@stillwellsoft.com> wrote: >>>> You must have 1.8+ its already been posted the 1.6 didn’t get a backport >>>> fix >>>> in the jabber protocol. >>>> >>>> >>>> >>>> >>>> >>>> From: asterisk-users-boun...@lists.digium.com >>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro >>>> Dardini >>>> Sent: Saturday, April 16, 2011 3:57 AM >>>> To: asterisk-users@lists.digium.com >>>> 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="+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" >>>> to="ldard...@gmail.com/asterisk438D86E0" >>>> id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session >>>> type="initiate" id="SIP784359174@10.177.37.1" >>>> initiator="+17174695...@voice.google.com/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 >>>> username=ldard...@gmail.com >>>> secret=********** >>>> priority=1 >>>> port=5222 >>>> usetls=yes >>>> usesasl=yes >>>> buddy=ldard...@gmail.com >>>> status=available >>>> >>>> ####### gtalk.conf >>>> >>>> [general] >>>> context=default >>>> bindaddr=0.0.0.0 >>>> allowguest=yes >>>> >>>> [guest] >>>> disallow=all >>>> allow=ulaw >>>> context=google-in >>>> >>>> [ldardini] >>>> username=ldard...@gmail.com >>>> 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