On Wed, Jul 18, 2018, at 7:23 PM, Benjamin Marty wrote:
> Hello
>
> I'm currently using Asterisk 13 with the chan_sip sip driver. The
> extensions are offloaded via realtime module to a MySQL database (via
> ODBC). So basically I have a MySQL Table with the SIP users + SIP passwords
> and the
In article <007401d41e7a$edc7e190$c957a4b0$@verishare.co.za>,
Stefan Viljoen wrote:
> Hi Guys
>
> If I recompile Asterisk (on a Centos 7 test box, Asterisk 1.8.32.3) multiple
> times in a row, e. g.
>
> make clean;configure;make menuselect;make
>
> I note that the asterisk binary in the /main
On Wed, Jul 18, 2018 at 4:37 AM, Stefan Viljoen
wrote:
> Hi Guys
>
> If I recompile Asterisk (on a Centos 7 test box, Asterisk 1.8.32.3)
> multiple
> times in a row, e. g.
>
> make clean;configure;make menuselect;make
>
> I note that the asterisk binary in the /main folder in the source tree,
Hello
I'm currently using Asterisk 13 with the chan_sip sip driver. The
extensions are offloaded via realtime module to a MySQL database (via
ODBC). So basically I have a MySQL Table with the SIP users + SIP passwords
and the other stuff from the standard Asterisk database schema.
Now I want to
> On Jul 15, 2018, at 11:37 PM, Naftoli Gugenheim wrote:
>
> Crickets...
>
> I've tried this now on 15.5.0. Still completely broken.
>
>
I suspect you’re encountering behavior that is working as intended.
Normally, when Asterisk plays back a file, it scans the file system for all
files
I am having one of those days. We just replaced an old Asterisk
1.8 server with a new Asterisk 13.21.1 (both using Freepbx) and almost
everything is working except for some incoming calls directed to a Cisco
SPA-8000. The PSTN trunk is SIP. Only calls coming from the PSTN to a
direct
Hi Guys
If I recompile Asterisk (on a Centos 7 test box, Asterisk 1.8.32.3) multiple
times in a row, e. g.
make clean;configure;make menuselect;make
I note that the asterisk binary in the /main folder in the source tree, has
a different SHA256 hash each time I recompile Asterisk using the
On Fri, Jul 20, 2018 at 11:41 AM Saint Michael wrote:
> The community would benefit if a non/licensed version of G729 would be
>> included with Asterisk, since the license expired. The current codec
>> source code posted still requires licensing.
>>
> I am sure Digium would not prefer to
>
So, that's not quite a debug log, but just the console log with Verbose+
output.
A debug log will show a lot more information, including what the media
cache modules are trying to do when they go to get the file.
You can find information on getting debug information on the Asterisk here:
I just finished installing a brand new server with CentOS 7.5 and
Asterisk 13.22.0 and the minute I a call from the PSTN (from a SIP
trunk) bridges with any SIP phone Asterisk crashes:
Jul 20 10:59:53 localhost kernel: asterisk[2819]: segfault at 188 ip
7f158b9e047c sp
>
> The community would benefit if a non/licensed version of G729 would be
> included with Asterisk, since the license expired. The current codec
> source code posted still requires licensing.
>
I am sure Digium would not prefer to
acknowledge this, but the phenomenal growth of Asterisk is
On Fri, Jul 20, 2018, at 3:18 PM, Carlos Chavez wrote:
> I just finished installing a brand new server with CentOS 7.5 and
> Asterisk 13.22.0 and the minute I a call from the PSTN (from a SIP
> trunk) bridges with any SIP phone Asterisk crashes:
>
> Jul 20 10:59:53 localhost kernel:
> On Jul 20, 2018, at 1:39 PM, Naftoli Gugenheim wrote:
>
> I've tried it with .wav. Same result. It doesn't even hit my server.
>
Can you provide a debug level 5 log (including all higher level verbose+
messages) from Asterisk that shows the playback operation?
>
> On Fri, Jul 20,
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