Each asterisk call uses 3 ports; 5060 is used to initiate the connection (5222 for chan_motif/google voice), then 2 consecutive ports from the 10001-20000 range are used for voice. Since GV uses TLS, I'm wondering if 5061 also comes into play. I assume you started from this link: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
-----Original Message----- From: Frank [mailto:[email protected]] Sent: Tuesday, January 22, 2013 10:51 AM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice Danny, I tried netstat -anp on a working outgoing call, and non working incomgin, and I see that the working has "CONNECTED" status, while the other one has nothing like that at all. Any other idea ? Thanks On 1/22/13 11:36 AM, Danny Nicholas wrote: > Do a "netstat -anp" during the call. This will (hopefully) show you > where the out of range condition is occurring. > > -----Original Message----- > From: Frank [mailto:[email protected]] > Sent: Tuesday, January 22, 2013 10:33 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: Danny Nicholas > Subject: Re: [asterisk-users] Google voice with no voice > > Danny, > > Thanks for the trick, that made all outgoing calls working. > Now, the issue is with incoming calls. Even if I turn off all other > phones in google voice configuration and have the calls routed to my > Google Chat only, this is what happens: > > The Asterisk receives the call. > The D70 rings. > If I pick up, nothing happens (I see on the D70 display that I picked > up) The caller still hear the ringing tone > > THat's what I see on the console: > > *CLI> -- Executing [[email protected]@gtalk_incoming:1] > Verbose("Gtalk/+1xxxxxxxxxx-2310", "0, Incoming gtalk from > "[email protected]/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <>") > in new stack > Incoming gtalk from > "[email protected]/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <> > -- Executing [[email protected]@gtalk_incoming:2] > Answer("Gtalk/+xxxxxxxxxx-2310", "") in new stack > -- Executing [[email protected]@gtalk_incoming:3] > Wait("Gtalk/+xxxxxxxxxx-2310", "2") in new stack > -- Executing [[email protected]@gtalk_incoming:4] > Dial("Gtalk/+xxxxxxxxxx-2310", "SIP/D70") in new stack > == Using SIP RTP CoS mark 5 > -- Called SIP/D70 > > *CLI> > *CLI> -- SIP/D70-00000006 is ringing > > *CLI> -- SIP/D70-00000006 answered Gtalk/+xxxxxxxxxx-2310 > == Spawn extension (gtalk_incoming, [email protected], 4) exited > non-zero on 'Gtalk/+xxxxxxxxxx-2310' > > > > > > > On 1/22/13 11:21 AM, Danny Nicholas wrote: >> You are obviously getting the call connected, so the subnet issue is moot. >> What this sounds like (pardon the pun) to me is an rtp skip issue. >> The "working" calls are generating rtp connections in the allowed >> range; the other calls have one or more ports outside of your rtp >> range. Verify that all of your ports defined in rtp.conf >> (10000-20000 by default) are open in the firewall. >> >> -----Original Message----- >> From: [email protected] >> [mailto:[email protected]] On Behalf Of Frank >> Sent: Tuesday, January 22, 2013 10:18 AM >> To: [email protected]; Asterisk Users Mailing List - Non-Commercial > Discussion >> Subject: Re: [asterisk-users] Google voice with no voice >> >> Chris, >> >> I covered the whole 74.125.225.* subnet. >> Even if I open the ports mentioned below for all (not limited to IP >> addresses) I still have the same issue. >> >> Have anyone ever succeeded in such configuration? : >> >> Digium phones on 2 different private networks (2 different buildings) >> Asterisk server in the internet with a public IP Use Google Voice >> >> Even if you have asterisk on a private network, but have the same >> kind of solution working for you, I'd love to hear your story.. >> >> >> >> >> >> On 1/22/13 9:55 AM, Christopher Harrington wrote: >>> On Mon, Jan 21, 2013 at 9:59 PM, Frank <[email protected] >>> <mailto:[email protected]>> wrote: >>> >>> Actually, the funny thing is that it works randomly. >>> >>> >>> This may be due to the fact that voice.google.com >>> <http://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 <http://voice.l.google.com>.300INA74.125.225.36 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.46 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.33 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.32 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.41 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.38 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.35 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.39 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.40 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.34 >>> voice.l.google.com <http://voice.l.google.com>.300INA74.125.225.37 >>> >>> (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 <http://voice.google.com> >>> 10,000:20,000 from my Airport Express public IP and from >>> voice.google.com <http://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 >> <mailto:[email protected]>) >>> same => n,Hangup() >>> >>> ;Google voice incoming >>> [gtalk_incoming] >>> exten => [email protected] <mailto:[email protected]>,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 >> >> >> -- >> _____________________________________________________________________ >> -- 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
