Leandro Dardini
2011-Apr-16 07:56 UTC
[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="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to=" ldardini at gmail.com/asterisk438D86E0" id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session type="initiate" id="SIP784359174 at 10.177.37.1" initiator="+ 17174695631 at 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=ldardini at gmail.com secret=********** priority=1 port=5222 usetls=yes usesasl=yes buddy=ldardini at 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=ldardini at 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); }; } -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110416/09b45bd1/attachment.htm>
William Stillwell
2011-Apr-16 13:24 UTC
[asterisk-users] Google Voice receiving call problem
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-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Leandro Dardini Sent: Saturday, April 16, 2011 3:57 AM To: asterisk-users at 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="+17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="ldardini at gmail.com/asterisk438D86E0" id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session type="initiate" id="SIP784359174 at 10.177.37.1" initiator="+17174695631 at 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=ldardini at gmail.com secret=********** priority=1 port=5222 usetls=yes usesasl=yes buddy=ldardini at 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=ldardini at 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); }; } -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110416/e4c6c6f6/attachment.htm>
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 <william at 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-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Leandro > Dardini > Sent: Saturday, April 16, 2011 3:57 AM > To: asterisk-users at 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="+17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" > to="ldardini at gmail.com/asterisk438D86E0" > id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session > type="initiate" id="SIP784359174 at 10.177.37.1" > initiator="+17174695631 at 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=ldardini at gmail.com > secret=********** > priority=1 > port=5222 > usetls=yes > usesasl=yes > buddy=ldardini at 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=ldardini at 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 >
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 <murdocke at 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 > <william at 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-bounces at lists.digium.com >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Leandro >> Dardini >> Sent: Saturday, April 16, 2011 3:57 AM >> To: asterisk-users at 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="+17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" >> to="ldardini at gmail.com/asterisk438D86E0" >> id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session >> type="initiate" id="SIP784359174 at 10.177.37.1" >> initiator="+17174695631 at 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=ldardini at gmail.com >> secret=********** >> priority=1 >> port=5222 >> usetls=yes >> usesasl=yes >> buddy=ldardini at 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=ldardini at 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 >> >
Vladimir Mikhelson
2011-Jun-14 16:03 UTC
[asterisk-users] Google Voice receiving call problem
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 <murdocke at 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 >> <william at 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-bounces at lists.digium.com >>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Leandro >>> Dardini >>> Sent: Saturday, April 16, 2011 3:57 AM >>> To: asterisk-users at 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="+17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" >>> to="ldardini at gmail.com/asterisk438D86E0" >>> id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session >>> type="initiate" id="SIP784359174 at 10.177.37.1" >>> initiator="+17174695631 at 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=ldardini at gmail.com >>> secret=********** >>> priority=1 >>> port=5222 >>> usetls=yes >>> usesasl=yes >>> buddy=ldardini at 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=ldardini at 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 >
Hey Elliot; Would you mind posting your dialplan for your Google Voice config? I am having a hell of a time getting it to do *anything*. Perhaps I am just fat-fingering. Would you mind? Thanks in advance. Glen On 6/13/2011 19:02, Elliot Murdock 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 > <william at 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-bounces at lists.digium.com >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Leandro >> Dardini >> Sent: Saturday, April 16, 2011 3:57 AM >> To: asterisk-users at 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="+17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" >> to="ldardini at gmail.com/asterisk438D86E0" >> id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session >> type="initiate" id="SIP784359174 at 10.177.37.1" >> initiator="+17174695631 at 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=ldardini at gmail.com >> secret=********** >> priority=1 >> port=5222 >> usetls=yes >> usesasl=yes >> buddy=ldardini at 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=ldardini at 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