http://linoxide.com/tools/install-setup-asterisk-13-pbx-centos-7/
2017-04-27 2:15 GMT+08:00 Jerry Geis :
> > yum install jansson*
>
> This works for CentOS 7 but not CentOS 6.
>
> Thanks,
>
> Jerry
>
>
>
> --
> _
> -- Bandwidth a
On Wed, 26 Apr 2017, Jerry Geis wrote:
dialplan show testing-sip
'**' => 1. Noop(Testing) [pbx_config]
2. Playback(demo-congrats) [pbx_config]
Looks like its there.
if I do ** "Dial" it works, but if I do "New Call
dialplan show testing-sip
'**' => 1. Noop(Testing)
[pbx_config]
2. Playback(demo-congrats)
[pbx_config]
Looks like its there.
if I do ** "Dial" it works, but if I do "New Call" ** then "Dial" it does
not work. Weird.
How do I get it to work for both cases. (glad
On Wed, 26 Apr 2017, Jerry Geis wrote:
I just tried this in my extensions.conf
exten => **,1,Noop(Testing)
exten => **,n,Playback(demo-congrats)
Did a reload... and the above does not happen.
I created as 12 instead of the ** and that works fine.
Is there anyway to get the ** to work? I also
On Wed, Apr 26, 2017 at 2:28 PM, Jerry Geis wrote:
> I just tried this in my extensions.conf
>
> exten => **,1,Noop(Testing)
> exten => **,n,Playback(demo-congrats)
>
> Did a reload... and the above does not happen.
> I created as 12 instead of the ** and that works fine.
>
> Is there anyway to g
I just tried this in my extensions.conf
exten => **,1,Noop(Testing)
exten => **,n,Playback(demo-congrats)
Did a reload... and the above does not happen.
I created as 12 instead of the ** and that works fine.
Is there anyway to get the ** to work? I also am using a polycom phone if
that affects
> yum install jansson*
This works for CentOS 7 but not CentOS 6.
Thanks,
Jerry
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Check out the new Asterisk community forum at: https://community.asteris
> > Anybody got an idea why the last scenario fails to work?
>
> If you turn up core debug (core set debug 2) and ensure it is going to
> the CLI then the bridge_native_rtp module will tell you why exactly it
> can't native bridge. You might also want to do a core show channel on
> both channels t
On Wed, Apr 26, 2017, at 01:25 PM, Daniel Tryba wrote:
> The request now gets routed based on a fully qualified domainname (with
> NAPTR/SRV records), which ultimately resolves to an ip that is matched in
> the
> endpoint SBC used above to originate a call. But now the asterisk stays
> in the
>
I'm trying to implement direct_media between multiple peers and an
uplink provider, all of whom have direct_media=yes configures.
For originating calls to the uplink provider direct_media=yes works like
expected. SIP flows through asterisk, rtp doesn't
SIP: enduser <-> SBC <-> asterisk 13 <-> upl
yum install jansson*
Jerry Geis 于2017年4月26日 周三下午8:32写道:
> >It can't be disabled. jansson is a required dependency for Asterisk 13
> >as JSON is used internally for things.
>
>
> Ok thanks - that is a little confusing since there are entries in the
> configure script that lead one to think it can
>It can't be disabled. jansson is a required dependency for Asterisk 13
>as JSON is used internally for things.
Ok thanks - that is a little confusing since there are entries in the
configure script that lead one to think it can be a configure time
switch.
I'll go the other route and install the
On Wed, Apr 26, 2017, at 09:24 AM, Jerry Geis wrote:
> Trying to install asterisk 13 on CentOS 6.
>
> The ./configure tells me:
> configure: error: *** JSON support not found (this typically means the
> libjansson development package is missing)
>
> I don't really need JSON so I thought I would j
Trying to install asterisk 13 on CentOS 6.
The ./configure tells me:
configure: error: *** JSON support not found (this typically means the
libjansson development package is missing)
I don't really need JSON so I thought I would just disable it.
./configure --with-jansson=no does not work
./conf
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