Does your install have a set of gtalk commands? GV isn't a SIP call per se, so the incoming line would be a gtalk peer. Try these commands from CLI Gtalk show peers Core help gtalk
-----Original Message----- From: Frank [mailto:[email protected]] Sent: Tuesday, January 22, 2013 1:04 PM To: Danny Nicholas Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google voice with no voice Hi, No, it's not even connecting. On the caller side, I do not see anything showing that the called party picks up. On the D70 side, when I pick up, I have the counter starting so I can see the seconds going up, but no audio at all. (and the remote party still hears ring tone) On 1/22/13 2:02 PM, Danny Nicholas wrote: > If you needed a MITM, nothing would work now. The incoming call is > connecting, but no voice or no connection at all? > > -----Original Message----- > From: Frank [mailto:[email protected]] > Sent: Tuesday, January 22, 2013 11:56 AM > To: Danny Nicholas > Subject: Re: [asterisk-users] Google voice with no voice > > I added port 5061 without success. > I am wondering if I used a man in the middle like iptel.org service, > it would work ? > > On 1/22/13 12:00 PM, Danny Nicholas wrote: >> 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? 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