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
