*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=r...@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:fr...@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:fr...@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:fr...@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: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 !
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