On Tue, Feb 23, 2021 at 4:44 PM Michael Munger wrote:
> I have a blacklisting system to deal with robocalls. Transferring a given
> call to extension *88 will add the CALLERID(num) to astdb, and when that
> number calls back, it goes straight to tt-monkeys.
>
> Problem:
>
> With Polycom phones,
I have a blacklisting system to deal with robocalls. Transferring a
given call to extension *88 will add the CALLERID(num) to astdb, and
when that number calls back, it goes straight to tt-monkeys.
Problem:
With Polycom phones, transfer -> blind -> *88 works just fine. But,
transfer -> *88
Hi
I have been using the CHANNEL variable as a way of checking if a user is
allowed to make outgoing calls, and what their source caller ID should be
(these values are in a database).
This works all of the time with SIP and most of the time with IAX, however
sometimes with IAX the channel
I believe ${IAXPEER(CURRENTCHANNEL)} should help you with the current
IAX2 name ... you can make DumpChan() to understand what kind of channel
variables you can use there.
--
razu
On 11/25/2009 09:57 PM, Nic Colledge wrote:
Hi
I have been using the CHANNEL variable as a way of checking if a
core show function SIPPEER
2009/1/6 Klaus Darilion klaus.mailingli...@pernau.at
since 1.4 you can also use
setvar=foo=bar
in sip.conf when configuring the peer. Then the channel variable foo is
automatically set to bar for calls initiated by this peer.
regards
klaus
Philipp Kempgen
Grygoriy Dobrovolskyy schrieb:
core show function SIPPEER
Does not work. Using the SIPPEER function you have to know the name of
the peer already.
regards
klaus
2009/1/6 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
since 1.4 you can also use
since 1.4 you can also use
setvar=foo=bar
in sip.conf when configuring the peer. Then the channel variable foo is
automatically set to bar for calls initiated by this peer.
regards
klaus
Philipp Kempgen wrote:
Grey Man schrieb:
On Wed, Nov 26, 2008 at 11:47 AM, Richard Brady
Thanks Grey and Philipp
Parsing the channel name is what we have been doing, but this has an
unfortunate dependence on username as opposed to peer name. The username
property of a SIP peer is not very well documented, and when using realtime
SIP, it's an immutable field once loaded into cache.
Grey Man schrieb:
On Wed, Nov 26, 2008 at 11:47 AM, Richard Brady [EMAIL PROTECTED] wrote:
Hi folks
I'm not sure what I am missing but I cannot find a predefined channel
variable to identify the SIP peer/user which has initiated a call and
established the channel.
The one option is to
On Wed, Nov 26, 2008 at 11:47 AM, Richard Brady [EMAIL PROTECTED] wrote:
Hi folks
I'm not sure what I am missing but I cannot find a predefined channel
variable to identify the SIP peer/user which has initiated a call and
established the channel.
The one option is to extract it from the
Hi folks
I'm not sure what I am missing but I cannot find a predefined channel
variable to identify the SIP peer/user which has initiated a call and
established the channel.
The one option is to extract it from the CHANNEL variable, but that is
fraught with difficulties.
Is there another
Hi all,
we are running Asterisk SVN-branch-1.4-r114299 and face following
problem: we have a main extension (102) and other extensions (104 and 107)
When extension 104 is calling extension 107 and 107 is on the phone, the
caller is parked for 5 seconds (Park-And-Announce) and the going back to
Hi,
I have set some variable in a call.
Set(testmode=1)
For some reason, such as forward the call, the follow command called.
Dial(Local/1234567)
It will go through the dial plan again but the value of variable
testmode is nothing instead of 1.
How can I maintain the value of the variable in
-
Non-Commercial Discussion
Subject: [Asterisk-Users] Channel
Variable
Hi all,
Does anyone know how to get the channel ID on the other side
of the call?
For example: When SIP/50 calls SIP/21, and the call is
answered by SIP/21 I get:
SIP/21-6735 answered SIP/50-b456
(212)
302-5790F: (646) 201-9418C: (516) 805-7981
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
SeddonSent: Saturday, January 08, 2005 4:21 AMTo:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-Users] Channel Variable
Does anyone know
Hi
all,
Does anyone know how
to get the channel ID on the other side of the call?
For example: When
SIP/50 calls SIP/21, and the call is answered by SIP/21 I
get:
SIP/21-6735 answered
SIP/50-b456
${CHANNEL} will show
me SIP/50-b456.
Is there a parameter
or a workaround to get the
16 matches
Mail list logo
