Google Voice w/11

Hi all,

I am trying to get asterisk to connect to my google voice account and create a sip server. I am following the documentation here:
wiki.asterisk.org/wiki/display/ … ing+Google

I have asterisk 11 and have all the dependencies installed. In my /etc/asterisk directory, I have the following:

asterisk.conf

[directories](!)
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk/asterisk
astlogdir => /var/log/asterisk
astsbindir => /usr/sbin

[options]
documentation_language = en_US	; Set the language you want documentation


[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6

extensions.conf

[viaMotif] ;
exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)


[incoming-motif]
exten => s,1,NoOp()
 same => n,Set(crazygooglecid=${CALLERID(name)})
 same => n,Set(stripcrazysuffix=${CUT(crazygooglecid,@,1)})
 same => n,Set(CALLERID(all)=${stripcrazysuffix})
 same => n,Dial(SIP/WalterWhite,20,D(:1))
 
;exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)

logger.conf

[general]

[logfiles]
messages => notice,warning,error

motif.conf

[google]
context=incoming-motif
disallow=all
allow=ulaw
connection=google

rtp.conf

[general]
rtpstart=10000
rtpend=20000
icesupport=yes

sip.conf

[me]
type=friend
username=me
secret=12345
host=dynamic
context=local

xmpp.conf

[general]
[google]
type=client
serverhost=talk.google.com
username=me@gmail.com
secret=1234
priority=1
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="Asterisk Instance - Google Talk - VoIP"
timeout=5

I am attempting to run asterisk via asterisk -vvv. It starts up, but I never see anything about sip and netstat -an | grep 5060 shows nothing as well as my sip clients are unable to connect.

What else do I need to do to get this up and running?

Thanks,

Walter

Check the logs for errors during start-up. Try to manually load chan_sip and see why it is failing to load, if it is failing.

Note the default for allowguest is yes, which most people don’t want.

I am attempting to run asterisk via asterisk -vvv. It starts up, but I never see anything about sip and netstat -an | grep 5060 shows nothing as well as my sip clients are unable to connect.

Sorry if it was a typo, but the correct command to startup Asterisk should be asterisk -rvvv

You must not have the r option when starting Asterisk; it tries to start a remote console to an existing running Asterisk.

The most common reason for having problems with chan_sip on version 11 is that you didn’t have all the dependencies. Normally it is the OpenSSL development packages that is missing. make menuconfig will tell you whether or not chan_sip has been built and any missing dependencies.