[asterisk-users] Google voice with no voice
Frank
frank at efirehouse.com
Tue Jan 22 13:52:14 CST 2013
OK, so here is the new..
By mistake, when I picked up the D70 , I pushed the 2 button.
I suddenly heard google voice saying "Okay, I'll send the caller to
voicemail". So I called again.. picked up.. I could not hear anything on
the D70.. But if I push 1 (which is the google voice option to pickup
the screened call), then the audio path works in both way.
So the real issue is that when google voice talks when I pick up to let
me know who's calling, I can't hear anything, until I press a digit.
If I press 1, I get the call connected.
If I press 2, I can hear google voice.
The question is why can't I hear google voice right away without pushing
a digit ?
I tried to go into google voice configuration and remove the call
screening, but it looks like for calls on gtalk , the screening is
always active.
So I guess I will know that I need to press 1 or 2 from the D70 for
everything to work. It slightly sucks, but I'll take it.
On 1/22/13 2:29 PM, Danny Nicholas wrote:
> 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:
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>>>>>>>>
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>>>>>>>> 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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