What about "jabber show channels"? -----Original Message----- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:12 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice
*CLI> core show help gtalk gtalk show channels Show GoogleTalk channels *CLI> gtalk show channels Channel Jabber ID Resource Read Write 0 active gtalk channels And that's my jabber.conf [general] debug=no autoprune=no autoregister=yes auth_policy=accept [asterisk] type=client serverhost=talk.google.com username=r...@gmail.com secret=toor priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage="Ohai from Asterisk" timeout=5 On 1/22/13 2:06 PM, Danny Nicholas wrote: > 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:fr...@efirehouse.com] > 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: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? 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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