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
