[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
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>
>>
>
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