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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>>>
>>
>