On Thursday, March 14, 2013, Joseph wrote:
Can someone refresh my memory how to backlist caller ID in
asterisk 1.8?
I had it working in ver. 1.4 but in 1.8 it changed.
I'm still using 1.4. In that I add a number to the blacklist with
CLI database put blacklist 0123456789 1
That is to add
Hi,
The idea was to record an ongoing call by three party bridging on the mobile
phone.
Well my problem was to halt execution of the Dialplan so the server would not
hang up the call. And I don´t want the server to say anything during the call.
Now I solved this case as well by using Answer and
hi,
the music heard by MoH is configurable... so if you want silence...
But hold could e.g. also be done by transferring a caller into a
dynamic meetme room...
yves
Am 14.03.2013 08:43, schrieb Henrik Westerberg:
Hi,
The idea was to record an ongoing call by three party bridging on the
chan_datacard was discontinnued two years ago, chan_dongle is the
current dongle driver for asterisk.
chan_mobile uses bluetooth mobile phones as FXO devices.
to send SMS chan_dongle should be used.
i.e:
asterisk -rx [ENTER]
dongle sms dongle0 0415340999 hello world
this command will send
Hi,
I need to get type of called number (TON), which is displayed in pri
debug messages:
Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xx' ]
Does anyone know how to do it?
According to documentation it is
2013/3/14 Puzankin Grigoriy gpuzan...@gmail.com:
Hi,
I need to get type of called number (TON), which is displayed in pri debug
messages:
Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xx' ]
Does anyone know
On Tue, 2013-02-19 at 02:05 +, Klaverstyn, David C wrote:
Is it possible to display the incoming calling number on a handset
when trying to pick up a call from another handset?
I currently have Call Pickup working using *8, I have also used the
PickUp application successfully but
I need to get type of called number (TON), which is displayed in pri
debug messages:
Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xx' ]
Does anyone know how to do it?
According to documentation it is
On 28/02/2013, at 6:08 PM, Richard Kenner ken...@gnat.com wrote:
Sorry for a possible retransmit: the first was sent from an incorrect
email address.
I'm trying to use the Polycom SoundStation IP 7000 with Confbridge.
But the transcoding from siren14 to slin32 is via slin. First, it
On 03/14/13 06:04, Geoff Lane wrote:
On Thursday, March 14, 2013, Joseph wrote:
Can someone refresh my memory how to backlist caller ID in
asterisk 1.8?
I had it working in ver. 1.4 but in 1.8 it changed.
I'm still using 1.4. In that I add a number to the blacklist with
CLI database put
Hi everybody,
Does any one knows how to place a call from a shell agi? I guess is something
like echo Exec Dial(DAHDI/g2/2010,,W). Algo how i get the dnid variable?
Thanks.
--
_
-- Bandwidth and Colocation Provided by
El 12/03/13 19:11, Hans Witvliet escribió:
Hi Emiliano,
thanks for your reply,
I think i might use it for a different project, I got an huawei-E1820
But at the moment i have to look at something else:
The issue is contacting people not currently in the office.
I've been trying to accomplish
Do you have transcode_via_sln set in asterisk.conf?
No, but as I said in a later email, I found the problem: when computing the
cost of a path, any downconvert has the same cost. So
siren14 - slin - slin32
is the same cost as
siren14 - slin16 - slin32
which is wrong.
I fixed this
The Asterisk Development Team has announced the first release candidate of:
DAHDI-Linux-2.6.3-rc1
DAHDI-Tools-2.6.3-rc1
dahdi-linux-complete-2.6.3-rc1+2.6.3-rc1
This beta release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
Hi all
I installed
DAHDI Version - 2.6.1
DAHDI Tools Version - 2.6.1
libss7-trunk
Asterisk 11.0.1
from source on Fedora 12 x86_64.
Now i`m unable to load chan_dahdi and libss7:
myserver*CLI module load chan_dahdi.so
ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling method
I installed
DAHDI Version - 2.6.1
DAHDI Tools Version - 2.6.1
libss7-trunk
Asterisk 11.0.1
from source on Fedora 12 x86_64.
Now i`m unable to load chan_dahdi and libss7:
myserver*CLI module load chan_dahdi.so
ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling
method
(A more specific subject may yield better answers -- better bait == better
fish.)
On Thu, 14 Mar 2013, Gustavo Salvador wrote:
Does any one knows how to place a call from a shell agi? I guess is
something like echo Exec Dial(DAHDI/g2/2010,,W).
While you can write an AGI in any language that
On 03/14/2013 11:04 PM, mohsen feyzzadeh wrote:
Hi all
I installed
DAHDI Version - 2.6.1
DAHDI Tools Version - 2.6.1
libss7-trunk
Asterisk 11.0.1
from source on Fedora 12 x86_64.
In case the 12 in Fedora 12 was not a typo, you do realize that Fedora
12 has been end-of-line for years and has
There appears to be a disagreement between the encoding given in the
sources for Siren14 that are downloaded from Polycom (and the ITU, both
are the same) and that implemented by codec_siren14.so. The latter
agrees with the actual device.
If I make a .sln32 file and run the encoder from
sorry if this is already asked?
May get any help about call blocking issue
Asterisk Version : Asterisk 1.6.2.7
FreePBX Version : 2.7.0.10
OS: CentOS
My asterisk is running basic incomming and outgoin call is working but I
can not configure call blocking and IVR.
any one can give me some hitns?
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