On Thu, Jun 6, 2013 at 10:12 AM, Matteo matteo.camp...@gmail.com wrote:
On Thu, Jun 6, 2013 at 9:56 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Jun 06, 2013 at 09:31:39AM +0200, Matteo wrote:
Hi list,
I had a problem with the dahdi XPP driver.
After this error in syslog,
Adtran MX2800 is rock solid. Save some money and use NFAS.
Thanks,
Steve Totaro
On Sun, Jun 9, 2013 at 10:11 PM, Nick Khamis sym...@gmail.com wrote:
Thank you so much for your responses!!! With this route we would have
to manage so many * boxes with T1s, not to mention, the hit we would
Not sure how I should officially report this, but I'm getting a compile
error with DAHDI-linux 2.7 when I define CONFIG_DAHDI_NET in
include/dahdi/dahdi_config.h. I am able to compile successfully when I
leave it undefined, but I need to be able to use the network support.
snipped
On Mon, Jun 10, 2013 at 11:33:16AM -0400, Dave Fullerton wrote:
Not sure how I should officially report this...
You should feel free to open issues at http://issues.asterisk.org.
but I'm getting a compile error with DAHDI-linux 2.7 when I define
CONFIG_DAHDI_NET in
Hey Philipp, I will try soon the new version and let you know.
Currently my users are pointing to a PBX in my local-private network with
no problems.
When I use wireshark I see my internal peers trying to send the ACK packets
4 or 5 times until hangup, at the same time the PBX are requesting
On 06/10/2013 11:53 AM, Shaun Ruffell wrote:
On Mon, Jun 10, 2013 at 11:33:16AM -0400, Dave Fullerton wrote:
Not sure how I should officially report this...
You should feel free to open issues at http://issues.asterisk.org.
but I'm getting a compile error with DAHDI-linux 2.7 when I define
I am trying to build Asterisk on a NetBSD system but I am running into
two problems. The first only happens on an installation built from
NetBSD HEAD. The config variable HAVE_NEWLOCALE is erroneously set
during configure but this system does not have newlocale(). I can't
seem to find where
This is the second issue I found while trying to install Asterisk on a
NetBSD box. I can't load the rtp module because HAVE_OPENSSL_SRTP
seems to be set. Is there some way to simply force this variab;e to be
unset from a configuration variable?
--
D'Arcy J.M. Cain
System Administrator, Vex.Net
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:
[Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng- to Motif/+12025551...@voice.google.com-da3c
[Jun 10 16:18:22]
Hello guys,
I looking for some dial plan which can mach on +xxx numbers instead of
00xxx numbers.
Some users of main use + instead of 00 for international dial. Is there any
solution for this problem?
As far as i readed in asterisk is some kind of replacement of characters in
dial plan command.
Hi,
On 06/10/2013 22:26, Jonson Player wrote:
Some users of main use + instead of 00 for international dial. Is there
any solution for this problem?
swap the + sign to double zeros if your provider can't handle it
; normal 00 prefix
exten = _00ZZXXX.,1,Macro(beforealldials)
exten =
On 06/10/2013 05:24 PM, Sean Darcy wrote:
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:
[Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng- to
On 06/10/2013 05:24 PM, Sean Darcy wrote:
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:
[Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng- to
https://bbs.archlinux.org/viewtopic.php?pid=920549
On Sat, Jun 8, 2013 at 9:51 AM, Jerry Geis ge...@pagestation.com wrote:
When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port
I get a motorboating sound or warble - or - just not clear audio.
When I switch that to ALSA direct
Pulse Audio 4.0 just came out and has gotten good reviews as it improves
audio quality...I installed it on the devel and support mediaports and will
test tomorrow.
http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/4.0/
On Mon, Jun 10, 2013 at 7:59 PM, Robert Krakora
Hi Satish,
I tried with sox, without any parameter, just sox filename.wav to
filename.mp3, in linux shell prompt... the file is been converted...
Now If i want to run that command using dialplan,
MixMonitor(filename.wav,m)
Monitor_Exec(sox filename.wav filename.mp3)
Or to use System command?
Hi Gopamkrishnan,
Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
facility to execute a command when recording is over.
*In my case, 'wav2mp3' is a script which gets executed and converts
recorded wav audio file to mp3. I pass ${FILENAME} as an argument to
my script.
*
*You
And yes if you want to use System application in your dialplan then have
System in your h extension
System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav
/PathToMp3FileToBE Stored/filename.mp3)
On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot satish4aster...@gmail.comwrote:
Hi
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