[asterisk-users] Google voice with no voice
Frank
frank at efirehouse.com
Tue Jan 22 10:32:42 CST 2013
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
>
More information about the asterisk-users
mailing list