[asterisk-users] Google voice with no voice
Danny Nicholas
danny at debsinc.com
Tue Jan 22 13:22:08 CST 2013
This is incoming, outgoing or idle (no call)?
-----Original Message-----
From: Frank [mailto:frank at 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] root at 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: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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