On Mon, Jan 21, 2013 at 9:59 PM, Frank <fr...@efirehouse.com> wrote: > Actually, the funny thing is that it works randomly. >
This may be due to the fact that voice.google.com actually resolves to a range of IP addresses. When you set up your firewall, it may not be including all of the possible resolutions for voice.google.com... voice.l.google.com. 300 IN A 74.125.225.36 voice.l.google.com. 300 IN A 74.125.225.46 voice.l.google.com. 300 IN A 74.125.225.33 voice.l.google.com. 300 IN A 74.125.225.32 voice.l.google.com. 300 IN A 74.125.225.41 voice.l.google.com. 300 IN A 74.125.225.38 voice.l.google.com. 300 IN A 74.125.225.35 voice.l.google.com. 300 IN A 74.125.225.39 voice.l.google.com. 300 IN A 74.125.225.40 voice.l.google.com. 300 IN A 74.125.225.34 voice.l.google.com. 300 IN A 74.125.225.37 (ie 74.125.225.32-41 and 74.125.225.46) Since these are short TTL values (the 300 means 5 minutes) there may be a brief period where your devices and your firewall agree, before one or both change their mind about the IP address behind that hostname. > I just tried out of the blue calling from D70 through Google Voice to a > cell phone, and it worked. I hung up, redial, and no audio at all. > > > On 1/21/13 10:38 PM, Frank wrote: > >> Greetings all, >> >> I was reading the documentation tonight, and decided to try Google voice >> with my asterisk. >> >> I was able to setup iksemel, connect to google using jabber, and connect >> to google voice using gtalk. >> >> >> Here is my physical configuration: >> >> Digium D70 <-- private network 192.168.1.x --> Airport express <--> >> Internet <--> Asterisk with public IP >> >> My asterisk has the following ports open: >> 5060 tcp/udp from my Airport Express public IP and from voice.google.com >> 10,000:20,000 from my Airport Express public IP and from voice.google.com >> >> My issue is that when I place a call with google voice, I have no audio >> path at all in both way. >> >> When a call is received on google voice (and sent to the D70), if I pick >> up, nothing happen, and the caller still hear the ringing tone. >> >> >> >> My D70 is setup as follow in the sip.conf: >> [D70] >> type=friend >> nat=yes >> qualify=yes >> directmedia=no >> host=dynamic >> secret=takapoum >> disallow=all >> allow=ulaw >> context=LocalSets >> mailbox=D70@default >> >> >> my gtalk.conf is setup as follow: >> [general] >> bindaddr=0.0.0.0 >> allowguest=yes >> >> [guest] >> disallow=all >> allow=ulaw >> context=gtalk_incoming >> connection=asterisk >> >> >> >> and finally, the interesting parts in my extensions.conf are setup as >> follow: >> ;Dialing out on google voice: >> exten => _1zxxzxxxxxx,1,Dial(Gtalk/**asterisk/+${EXTEN}@voice.** >> google.com <exten...@voice.google.com>) >> same => n,Hangup() >> >> ;Google voice incoming >> [gtalk_incoming] >> exten => r...@gmail.com,1,Verbose(0, Incoming gtalk from >> ${CALLERID(all)}) >> same => n,Answer() >> same => n,Wait(2) >> same => n,Dial(SIP/D70) >> same => Hangup() >> >> >> I would appreciate if anyone could give me a hint about the audio path. >> This is a project that we I will try to setup in a small fire >> department, and before I try it, I would like to make sure that my >> Digium phones will be able to get full audio path behind private networks. >> >> Thanks a ton for the help ! >> >> -- >> ______________________________**______________________________**_________ >> -- 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<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<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248
-- _____________________________________________________________________ -- 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