Displaying 16 results from an estimated 16 matches for "azhelp".
2004 Apr 29
5
Start recording during call by pressing button sequence
Does anyone configure that or is that possible ?
Thanks in advance
--
Best regards
Vlad
2004 Sep 06
1
cvs server problem
Today morning cvs server checkout problem:
cvs server: Updating asterisk-addons/format_mp3
cvs server: failed to create lock directory for
`/usr/cvsroot/asterisk-addons/format_mp3'
(/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied
cvs server: failed to obtain dir lock in repository
`/usr/cvsroot/asterisk-addons/format_mp3'
cvs [server aborted]: read lock failed -
2004 Aug 08
3
iconnect inbound - so do we know how to fix it
Just wondering whether we have a resolution to iconnect incoming problem,
which started few days ago.
Cheers
SW
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2004 Aug 10
0
iconnect inbound - FIXED (kinda)
...;> fix it
> >>
> >>
> >> May be you can find the solution in my post:
> >>
> >> http://lists.digium.com/pipermail/asterisk-users/2004-August/
> >> 058014.html
> >>
> >> Raj
> >>
> >> --- Vladyslav <vladk@azhelp.net> wrote:
> >>
> >>> Try to comment out in your sip.conf
> >>> ;qualify=yes
> >>>
> >>>
> >>> On Mon, 2004-08-09 at 06:52, Sathya Weerasooriya wrote:
> >>>> Just wondering whether we have a resolution to iconne...
2004 May 26
1
dialplan AGI DTMF
Good day All.
Is there a way to pass DTMF signals to AGI script during conversation ?
Actually here what I want to make:
Users are usually dial using dialplan and when someone press *4 (during
conversation) I want to have agi script to deal with that, but those
users should keep talking and even didn't notice that one of them press
something.
Is there a way to do that or it's complete
2004 Jun 14
0
pulse dialing
Good day,
does anyone have pulse dialing working ?
http://voip-info.org/tiki-index.php?page=Asterisk+zaptel+pulse+dialing
At the link above there is a statement:
configuration for European telephone lines will look like:
make_time=63
break_time=37
pause_time=800
So where these pamameters should go to ? zapata.conf ignore them.
I have tried on both version of asterisk (cvs and stable) with
2004 Jul 05
1
*8# into invalid extensions
Hi All!
Have a problem with remote call pickup via sip.
When 1 sip phone is ringing and I'm trying to pickup a call from another
sip phone by dialing *8#
I'm getting:
-- Sent into invalid extension '*8#' in context 'from-sip-post' on
SIP/ciscok-8d39
such configs:
zapata.conf
------
context=inbound-analog
callgroup=2
channel=2
------
sip.conf
------
[ciscok]
type=friend
2004 Aug 19
0
fax output from Asterisk into file
Good day ALL.
Could anyone tell me is there a way to get fax debug output into the
file when running safe_asterisk ?
------------------------------------------------------
V.8 capable
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter, V.29 and V.17
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm or 255mm
Recording length: A4 (297mm)
2004 Sep 23
1
video via IAX or SIP
HI ALL.
Please help.
Problem: video calls drop after 15-20 seconds all the time.
Use * latest cvs.
from sip.conf
[1102]
type=friend
username=1102
host=dynamic
callerid=Veo webcam<1102>
canreinvite=no
disallow=all
allow=gsm
;allow=ulaw
allow=h261
allow=h263
from iax.conf
[peer2] ; 192.168.0.7
type=friend
port=4569
auth=md5
secret=second2
context=local
host=dynamic
qualify=yes
trunk=yes
2004 Dec 21
1
HELP: agi-test.agi does not return any DTMF!
Hi all,
Tiny, but very important question for me: what it can be when standard
agi-test.agi script, like:
print STDERR "Testing 'waitdtmf'...";
print "WAIT FOR DIGIT 10000\n";
my $result = <STDIN>;
&checkresult($result);
on the call from KPhone application does NOT return any DTMF code back (I use
dtmfmode=inband in sip.conf):
Testing
2005 Feb 01
2
Feature automon
There is option automon => *1 in features.conf
As I understand when *1 pressed during conversation => recording should
begin. But unfortunately it doesn't work for me.
I use CVS-HEAD-01/27/05
Does anyone has that feature working?
Thanks
--
Best Regards
VladK
2005 Feb 27
0
ATA 286 downgrade failure
Good day list,
I have a problem with ATA Handy Tone 286. It has been unsuccessfully
downgraded via HTTP. Seems like during downgrade there was a problem
with connection, because now it's not responding at all. There is no way
to get to it's voice menu via phone (by pressing button on it). The
button is just flashing ...
Could anyone suggest any way to bring that device back to life.
2004 Aug 04
4
rxfax killed asterisk
HI All.
I'm using tiff-v.3.5.7 and spandsp-0.0.1k with latest * cvs on
Slackware-10.0.
Here is debug messages from * console.
Please advise.
Can receive fax
Selected data signalling rate: V.29, 9600bps
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Minimum scan line time: 0ms
Get at 9600
Changed from phase 3 to 5
Fast carrier up
Training failed (sequence failed)
Fast
2004 Jul 01
2
Grandstream HT286 1.0.4.63 & Meetme
Good day!
Have a weird problem with HT-286 and Conference room. I use Asterisk
CVS-HEAD-06/04/04.
Here it is:
When HT-286 get into the conference room first and nobody in that room
everything seems ok (with any codec HT286 allowed), but when HT-286 get
into conference room when somebody already there, have got different HT
behavior:
1. When HT use GSM codec => it connects to conference room,
2005 Mar 16
3
TxFAX problem
Hi Ppl.
Once, couple weeks ago when I have updated * from CVS-HEAD something
happen and I could not send a fax anymore.
After that I have tried previous * CVS versions with different versions
of spandsp (0.0.1, 0.0.2pre4, 0.0.2pre10) but without any changes.
I have tried that on Fedora Core 2 with libtiff-3.5.7-16.1 and
libtiff-devel-3.5.7-16.1. Everything compiles smoothly, but when I try
to send
2005 Sep 05
9
Asterisk Follow ME
Hi All.
I have notice a problem with FM feature (screen macros) on Asterisk CVS
version.
When call goes via IAX and calling part "accept the call" on Dial
command with option M, in macros context it's setting
MACRO_RESULT=CONTINUE, but anyway it hangups both channels.
If anyone faced with such problem please let me know. I need to know
whether it's bug or just configuration