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=XXXXXXXXX at 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: xxxxxx at 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 at 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 at 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/08cedabe/attachment.htm>
Sorry, Asterisk Build 1.6.2.7 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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=XXXXXXXXX at 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: xxxxxx at 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 at 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 at 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/6b5fc711/attachment.htm>
William, It still looks like something is not properly set with your account on Google Voice. Have you had a chance to follow the recommendations I gave you earlier in the thread? If the account is properly set the dial string will need to look like this, "gtalk/<jabber-conf-section-name>/+$OUTNUM$@voice.google.com" where $OUTNUM$ is a called number in the international format. On the receiving end the call will come with an empty CID Number, but with the CID Name which looks like this: +15555551212 at voice.google.com/srvres-MTAuMjE4LjIuMTk3Ojk4MzM Just cut all prior to "@" as a CID Number. See https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google Also you do not need to wait 5 seconds. 1 or 2 is sufficient. -Vladimir On 2/20/2011 10:40 PM, William Stillwell wrote:> >> -----Original Message----- >> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- >> bounces at lists.digium.com] On Behalf Of Vladimir Mikhelson >> Sent: Sunday, February 20, 2011 10:16 PM >> To: asterisk-users at lists.digium.com >> Subject: Re: [asterisk-users] Gtalk/Jabber Issue >> >> "Unknown Caller" most likely refers to the CID Name, CID Number should >> be provided as your Google Voice number. >> > > I ended up doing the following: > > Outbound rule: > > Exten => > _NXXXXXXXXX,1,Dial(gtalk/value_in_jabber.conf/+1(myGoogleVoice#)@voice.googl > e.com,30,D(wwwwww2www${EXTEN}#ww)) > > This will call the gv service, and then dial out that way, and the remote > receiving party will see my gv # > > And on inbound to bypass call screening question: > > exten => s,1,NoOp(The Current Time is..: > ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) > exten => s,n,NoOp(The CallerID(num) is.: ${CALLERID(NUM)}) > exten => s,n,NoOp(The CallerID(name) is: ${CALLERID(NAME)}) > exten => s,n,Answer() > exten => s,n,Wait(5) > exten => s,n,SendDTMF(1) > exten => s,n,Dial(....... > > > > > > > > > -- > _____________________________________________________________________ > -- 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 >