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:fr...@efirehouse.com] 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:fr...@efirehouse.com] > 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:fr...@efirehouse.com] >> 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 [r...@gmail.com@gtalk_incoming:1] >> Verbose("Gtalk/+1xxxxxxxxxx-2310", "0, Incoming gtalk from >> "+1xxxxxxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <>") >> in new stack >> Incoming gtalk from >> "+xxxxxxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <> >> -- Executing [r...@gmail.com@gtalk_incoming:2] >> Answer("Gtalk/+xxxxxxxxxx-2310", "") in new stack >> -- Executing [r...@gmail.com@gtalk_incoming:3] >> Wait("Gtalk/+xxxxxxxxxx-2310", "2") in new stack >> -- Executing [r...@gmail.com@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, r...@gmail.com, 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: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank >>> Sent: Tuesday, January 22, 2013 10:18 AM >>> To: ch...@acsdi.com; 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 <fr...@efirehouse.com >>>> <mailto:fr...@efirehouse.com>> 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:exten...@voice.google.com>) >>>> same => n,Hangup() >>>> >>>> ;Google voice incoming >>>> [gtalk_incoming] >>>> exten => r...@gmail.com <mailto: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? 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