Hello,
ChanSpy is not completely working for me.
Dialplan :
/exten => _*XXX***,n,ChanSpy(${SIPACC}) ; var $SIPACC has SIP peer
account name/
Verbose :
/[Jan 30 16:25:47] -- Executing [*204***@from-ITEL:10]
ChanSpy("SIP/itel0-00002f21", "itel1") in new stack
[Jan 30 16:25:48] -- <SIP/itel0-00002f21> Playing 'beep.alaw'
(language 'nl')/
But the spying IP-phone itel0 does not hear a thing. It should here the
conversation between SIP/itel1-00002f10and SIP/ITELin-00002f0d
These are the 2 channels which are talking to each other :
/SIP/itel1-00002f10 & SIP/ITELin-00002f0d/
Any idea which setting I'm missing ?
Kind regards,
Jonas.
On 01/25/2012 11:10 AM, Ishfaq Malik wrote:
I use ChanSpy successfully all the time. You do not have to specify the
full channel, just the prefix which is the peer name. As you can see it
also states 'This includes the audio coming in and out of the channel
being spied on.'
Have you tried giving it a go?
-= Info about application 'ChanSpy' =-
[Synopsis]
Listen to a channel, and optionally whisper into it.
[Description]
This application is used to listen to the audio from an Asterisk
channel.
This includes the audio coming in and out of the channel being spied
on.
If the 'chanprefix' parameter is specified, only channels beginning with
this
string will be spied upon.
While spying, the following actions may be performed:
- Dialing '#' cycles the volume level.
- Dialing '*' will stop spying and look for another channel to spy on.
- Dialing a series of digits followed by '#' builds a channel name to
append
to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing
the
digits '1234#' while spying will begin spying on the channel
'Agent/1234'.
Note that this feature will be overridden if the 'd' option is used
NOTE: The<X> option supersedes the three features above in that if a
valid
single digit extension exists in the correct context ChanSpy will exit
to
it. This also disables choosing a channel based on 'chanprefix' and a
digit
sequence.
[Syntax]
ChanSpy([chanprefix][,options])
[Arguments]
options
b: Only spy on channels involved in a bridged call.
B: Instead of whispering on a single channel barge in on both
channels
involved in the call.
c(digit):
digit - Specify a DTMF digit that can be used to spy on the
next available channel.
d: Override the typical numeric DTMF functionality and instead use
DTMF to switch between spy modes.
4 - spy mode
5 - whisper mode
6 - barge mode
e(ext): Enable *enforced* mode, so the spying channel can only
monitor
extensions whose name is in the<ext> : delimited list.
E: Exit when the spied-on channel hangs up.
g(grp):
grp - Only spy on channels in which one or more of the groups
listed in<grp> matches one or more groups from the ${SPYGROUP}
variable set on the channel to be spied upon.
NOTE: both<grp> and ${SPYGROUP} can contain either a single group
or a colon-delimited list of groups, such as
'sales:support:accountin
g'.
n([mailbox][@context]): Say the name of the person being spied on
if that person has recorded his/her name. If a context is specified,
then
that voicemail context will be searched when retrieving the name,
otherwise
the 'default' context be used when searching for the name (i.e. if
SIP/1000
is the channel being spied on and no mailbox is specified, then
'1000'
will be used when searching for the name).
o: Only listen to audio coming from this channel.
q: Don't play a beep when beginning to spy on a channel, or speak
the selected channel name.
r([basename]): Record the session to the monitor spool directory.
An optional base for the filename may be specified. The default is
'
chanspy'.
s: Skip the playback of the channel type (i.e. SIP, IAX, etc) when
speaking the selected channel name.
S: Stop when no more channels are left to spy on.
v([value]): Adjust the initial volume in the range from '-4' to
'4'. A negative value refers to a quieter setting.
w: Enable 'whisper' mode, so the spying channel can talk to the
spied-on channel.
W: Enable 'private whisper' mode, so the spying channel can talk
to the spied-on channel but cannot listen to that channel.
x(digit):
digit - Specify a DTMF digit that can be used to exit the
application.
X: Allow the user to exit ChanSpy to a valid single digit numeric
extension in the current context or the context specified by the
${SP
Y_EXIT_CONTEXT} channel variable. The name of the last channel that
was
spied on will be stored in the ${SPY_CHANNEL} variable.
On Wed, 2012-01-25 at 10:15 +0100, Jonas Kellens wrote:
This could work, yes.
But the context is not always the same.
Also ${CHANNELS(miq8) will return nothing...
Jonas.
On 01/24/2012 08:47 PM, Danny Nicholas wrote:
Did a little research on this using my Asterisk 10.0. This should
work for you.
exten => 1246,1,answer()
exten => 1246,n,set(inuse=${CHANNELS(miq8)})
exten => 1246,n,extenspy(${inuse}@default)
exten => 1246,n,hangup()
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Tuesday, January 24, 2012 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
there is very little information about the function CHANNELS().
If I know the peer name (that I always know for sure), do you see a
way of using the function CHANNELS() to get the right channel ??
If CHANNELS() gives a "space-delimited" list of active channels, and
I know miq8... how can I get SIP/miq8-00002419 ?
Thanks !
On 01/24/2012 04:46 PM, Danny Nicholas wrote:
Extenspy(miq8@default) for miq8. I would either proceed under the
assumption that I’m going to be listening to my extensions in the
default context or set up an AGI or something to load my needed
ext@context information.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Tuesday, January 24, 2012 9:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
how to use ExtenSpy(extension@context) when conversations are named
like this ? :
SIP/378680644-00002 default
SIP/rs4-00002445 sub-uitinternation
SIP/3715320168-00002 default
SIP/ibenla2-0000244 sub-uit789
SIP/372083610-00002 default
SIP/cedhou0-000024 sub-uit789
SIP/travel3-00002 pbx-routing
SIP/INTELin-00002 pbx-routing
SIP/375382280-00002 default
SIP/miq8-00002419 sub-uitGSM
SIP/3749378004-0000 default
SIP/instlpr0-00002 sub-uitinternation
Can you tell me what is the extension ? How will I know the
context ? The context is not always the same...
On 01/24/2012 04:32 PM, Danny Nicholas wrote:
You are either going to be able to listen to SIP/miq8 or you are
going to have to know the sequence number like SIP/miq8-00001.
Maybe you should just use ExtenSpy instead?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Tuesday, January 24, 2012 9:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Of course I can control the name of my SIP-peer. Why do you tell me
this ?!
Please answer my question : how do I know the channel name so I can
ChanSpy the correct channel ?
On 01/24/2012 04:13 PM, Danny Nicholas wrote:
It’s not random. The “Channel Name” is Tech/peer-sequence (sequence
is in hex). You can control (to a degree) the peer portion in
sip.conf/users.conf.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Tuesday, January 24, 2012 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
thanks. miq8 is the name of the SIP peer account.
So when I know the SIP peer name, and I strip of the numbers of the
channel, then I can use ChanSpy. So this answers my original
question.
The only problem I see : it is Asterisk that gives the channel its
name. How do I change this ??
As far as I know, Asterisk randomly gives a channel name which
consists of the technology (SIP), the peername (miq8) and some
numbers...
How to change the channel name ?
On 01/24/2012 03:53 PM, Danny Nicholas wrote:
I would try chanspy(sip/miq8,b) – the b flag denotes to only listen
to a bridged call which (it seems to me) should pick up both sides.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Tuesday, January 24, 2012 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ChanSpy : how to know channel name ?
Hello,
OK thanks. But, I want to listen to the conversation (not just 1
channel out of 2 channels). How then do I use ChanSpy ?
On 01/24/2012 03:41 PM, Danny Nicholas wrote:
Strip off the –xxxxx. Just listen to SIP/miq8 and SIP/375382280 in
your example.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Tuesday, January 24, 2012 7:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ChanSpy : how to know channel name ?
Hello list,
to use ChanSpy, one needs to know the name of the channel.
But on an incoming call from the provider, or an outgoing call to
the provider there are always numbers added. How can one then know
the channel name ??
core show channels verbose shows me for example :
SIP/378680644-00002 default
SIP/rs4-00002445 sub-uitinternation
SIP/3715320168-00002 default
SIP/ibenla2-0000244 sub-uit789
SIP/372083610-00002 default
SIP/cedhou0-000024 sub-uit789
SIP/travel3-00002 pbx-routing
SIP/INTELin-00002 pbx-routing
SIP/375382280-00002 default
SIP/miq8-00002419 sub-uitGSM
SIP/3749378004-0000 default
SIP/instlpr0-00002 sub-uitinternation
SIP/372089170-00002 default
SIP/v9q9uLT-0000 from-GFATRUNK
46 active channels
24 active calls
If I want to listen to the conversation of SIP/miq8-00002419 and
SIP/375382280-00002 (these 2 channels have been connected to 1
conversation), how do I use ChanSpy ??
Kind regards;
Jonas.
--
_____________________________________________________________________
-- 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
--
_____________________________________________________________________
-- 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
--
_____________________________________________________________________
-- 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