find the dtmf.
You have to modify the code around the line 6075 to extract the information.
I think it can resolve your problem.
hope it's helps.
I have some problem detecting DTMF from GSM phone using B3G services
(cirpack node also)...
do you have also this kind of problem?
Florian Lefeuvre
find the dtmf.
You have to modify the code around the line 6075 to extract the information.
I think it can resolve your problem.
hope it's helps.
I have some problem detecting DTMF from GSM phone using B3G services
(cirpack node also)...
do you have also this kind of problem?
Florian Lefeuvre
hi everyone,
just a poll toknow if someone out there is using intensively
asterisk-HEAD version (mean the very last version
of asterisk).
I currently used asterisk.1.0.5 and sometimes I need to kill the process
because it's freezing (deadlock maybe, or something else...).
is this kind of problem
it (spent an awful lot of time last week compiling different
configurations of stable, head, patches...taking a break this week).
This is Florian said:
On Feb 15, 2005, at 1:12 PM, Florian Lefeuvre wrote:
I find the compilation option RADIO_RELAX.
this option change a threshold in DTMF detection
Hi all,
I use a GSM device to send dtmf on my asterisk system (via SIP).
the codec I use is ulaw (or a-law).
dtmf mode is INBAND.
relaxmode is on.
but most of the case, I 'missed' some DTMF or
I 'double' one.
as anybody as seen this before?
is there any way to prevent this
thanks
On Feb 15, 2005, at 10:09 AM, Florian Lefeuvre wrote:
Hi all,
I use a GSM device to send dtmf on my asterisk system (via SIP).
the codec I use is ulaw (or a-law).
dtmf mode is INBAND.
relaxmode is on.
but most of the case, I 'missed' some DTMF or
I 'double' one.
as anybody as seen this before
Hi,
What do you want to check exacly?
that * is still alive? you want to know the number of concurrent call?
Florian
Hi *,
Does anyone have a lead on a Nagios plugin that speaks IAX or a small
app to do so? I'm trying to set up remote monitoring for my Asterisk
server and only IAX2 traffic is
Hi all,
I encounter an annoying problem using Asterisk.
I 'm using SIP. I try to register an Asterisk as a SIP end user with
another Asterisk.
If I put both asterisk in the same local network, no problem to do it.
The asterisk end user registered perfectly with the other (let's call it
the
