[asterisk-users] Google voice with no voice

Frank frank at efirehouse.com
Tue Jan 22 13:03:57 CST 2013 

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 !
>>>>>
>>>>>             --
>>>>
>>>> --
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>>>>
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>>>> ____________________________________________________________________
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>>>> -- 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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