[asterisk-users] Google voice with no voice
Danny Nicholas
danny at debsinc.com
Tue Jan 22 13:29:53 CST 2013
This sounds like a codec issue. Set your verbose to 10 and retry the
incoming call.
-----Original Message-----
From: Frank [mailto:frank at efirehouse.com]
Sent: Tuesday, January 22, 2013 1:26 PM
To: Danny Nicholas
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Google voice with no voice
That's idle.
If I call from D70 (working scenario) the result of the command is the same.
gtalk show channels shows this when I call from D70 (again, working
scenario):
Channel Jabber ID Resource
Read Write
Gtalk/+1xxxxx at voice.googl +1xxxxx at voice.google.com srvres-MTAuMjI3
ulaw ulaw
When I call google voice, gtalk show channels shows the following:
While ringing:
*CLI> gtalk show channels
Channel Jabber ID Resource
Read Write
Gtalk/+xxx-2c8e +xxx at voice.google.com srvres-MTAuMTIu ulaw
slin
1 active gtalk channel
Once I pick up
*CLI> -- SIP/D70-00000004 answered Gtalk/+xxx-2c8e
gtalk show channels
Channel Jabber ID Resource
Read Write
Gtalk/+xxx-2c8e +xxx at voice.google.com srvres-MTAuMTIu ulaw
ulaw
1 active gtalk channel
The only difference is the WRITE column that changes from SLIN to ULAW
On 1/22/13 2:22 PM, Danny Nicholas wrote:
> 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
>>>>>>> -- 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
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> _________________________________________________________________
>>>>>>> _
>>>>>>> _
>>>>>>> _
>>>>>>> _
>>>>>>> -- 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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