This sounds like a codec issue. Set your verbose to 10 and retry the incoming call.
-----Original Message----- From: Frank [mailto:fr...@efirehouse.com] Sent: Tuesday, January 22, 2013 1:26 PM To: Danny Nicholas Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Google voice with no voice That's idle. If I call from D70 (working scenario) the result of the command is the same. gtalk show channels shows this when I call from D70 (again, working scenario): Channel Jabber ID Resource Read Write Gtalk/+1xxxxx@voice.googl +1xx...@voice.google.com srvres-MTAuMjI3 ulaw ulaw When I call google voice, gtalk show channels shows the following: While ringing: *CLI> gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw slin 1 active gtalk channel Once I pick up *CLI> -- SIP/D70-00000004 answered Gtalk/+xxx-2c8e gtalk show channels Channel Jabber ID Resource Read Write Gtalk/+xxx-2c8e +x...@voice.google.com srvres-MTAuMTIu ulaw ulaw 1 active gtalk channel The only difference is the WRITE column that changes from SLIN to ULAW On 1/22/13 2:22 PM, Danny Nicholas wrote: > This is incoming, outgoing or idle (no call)? > > > -----Original Message----- > From: Frank [mailto:fr...@efirehouse.com] > 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] r...@gmail.com - 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: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? 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