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/[email protected] [email protected] 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 [email protected] 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 [email protected] 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:[email protected]]
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] [email protected] - 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:[email protected]]
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
[email protected]
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:[email protected]]
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:[email protected]]
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:[email protected]]
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:[email protected]]
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 [[email protected]@gtalk_incoming:1]
Verbose("Gtalk/+1xxxxxxxxxx-2310", "0, Incoming gtalk from
"[email protected]/srvres-MTAuMTIuMTU1LjE1Ojk4MjU="
<>") in new stack
Incoming gtalk from
"[email protected]/srvres-MTAuMTIuMTU1LjE1Ojk4MjU=" <>
-- Executing [[email protected]@gtalk_incoming:2]
Answer("Gtalk/+xxxxxxxxxx-2310", "") in new stack
-- Executing [[email protected]@gtalk_incoming:3]
Wait("Gtalk/+xxxxxxxxxx-2310", "2") in new stack
-- Executing [[email protected]@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, [email protected], 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: [email protected]
[mailto:[email protected]] On Behalf Of
Frank
Sent: Tuesday, January 22, 2013 10:18 AM
To: [email protected]; 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 <[email protected]
<mailto:[email protected]>> 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:[email protected]>)
same => n,Hangup()
;Google voice incoming
[gtalk_incoming]
exten => [email protected]
<mailto:[email protected]>,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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