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:fr...@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 [r...@gmail.com@gtalk_incoming:1]
Verbose("Gtalk/+1xxxxxxxxxx-2310", "0, Incoming gtalk from
"+1xxxxxxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <>") in new
stack
Incoming gtalk from
"+xxxxxxx...@voice.google.com/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <>
-- Executing [r...@gmail.com@gtalk_incoming:2]
Answer("Gtalk/+xxxxxxxxxx-2310", "") in new stack
-- Executing [r...@gmail.com@gtalk_incoming:3]
Wait("Gtalk/+xxxxxxxxxx-2310", "2") in new stack
-- Executing [r...@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, r...@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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Sent: Tuesday, January 22, 2013 10:18 AM
To: ch...@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 <fr...@efirehouse.com
<mailto:fr...@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@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...@voice.google.com>)
same => n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten => r...@gmail.com <mailto:r...@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
--
_____________________________________________________________________
-- 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
