[asterisk-users] Google voice with no voice

Danny Nicholas danny at debsinc.com
Tue Jan 22 13:14:57 CST 2013


What about "jabber show channels"?

-----Original Message-----
From: Frank [mailto:frank at 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=root at 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:frank at 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:frank at 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:frank at 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:frank at 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 [root at gmail.com@gtalk_incoming:1]
>>>> Verbose("Gtalk/+1xxxxxxxxxx-2310", "0, Incoming gtalk from 
>>>> "+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" 
>>>> <>") in new stack
>>>>       Incoming gtalk from
>>>> "+xxxxxxxxxx at voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <>
>>>>          -- Executing [root at gmail.com@gtalk_incoming:2] 
>>>> Answer("Gtalk/+xxxxxxxxxx-2310", "") in new stack
>>>>          -- Executing [root at gmail.com@gtalk_incoming:3] 
>>>> Wait("Gtalk/+xxxxxxxxxx-2310", "2") in new stack
>>>>          -- Executing [root at 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, root at 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-bounces at lists.digium.com
>>>>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
>>>>> Frank
>>>>> Sent: Tuesday, January 22, 2013 10:18 AM
>>>>> To: chris at 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 <frank at efirehouse.com 
>>>>>> <mailto:frank at 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 at 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%7D at voice.google.com>)
>>>>>>                    same => n,Hangup()
>>>>>>
>>>>>>              ;Google voice incoming
>>>>>>              [gtalk_incoming]
>>>>>>              exten => root at gmail.com
> <mailto:root at 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
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>>>>>
>>>>>
>>>>> --
>>>>> __________________________________________________________________
>>>>> _
>>>>> _
>>>>> _
>>>>> -- 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
>>>>>
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>>>>>
>>>>
>>>
>>
>




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