This is incoming, outgoing or idle (no call)?
-----Original Message----- From: Frank [mailto:[email protected]] Sent: Tuesday, January 22, 2013 1:21 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice *CLI> jabber show connections Jabber Users and their status: [asterisk] [email protected] - Connected ---- Number of users: 1 On 1/22/13 2:14 PM, Danny Nicholas wrote: > What about "jabber show channels"? > > -----Original Message----- > From: Frank [mailto:[email protected]] > 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 > [email protected] > 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:[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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