Hi.
Is there any way to eliminate AMI manager logins from the logging output
(without just turning the log level down and thereby losing lots of other stuff
as well)?
I'm running Asterisk 13.14.1 as a backend service to LVS/IPVS, and using the
AMI login as the "service alive" check to see whic
Keep it as simple as you can.
​[test]
exten => 12345678900,1,Noop( Set callER hangup handler
CHANNEL(name)='${CHANNEL(name)}' SIPCALLID='${SIPCALLID}')
same => n,Set(CHANNEL(hangup_handler_push)=test_caller_hangup,${EXTEN},1)
same =>
n,Dial(SIP/pbx-nyigc/4408,60,gb(test_pre_dial^${EXTEN}^1)F
FYI, we found that our peers don't hangup properly. But we would still like
to know how to get the peer's hangup handler to fire upon peer hangup,
because right now it corrupts our globals by firing after the caller's
hangup handler.
On Tue, Jun 5, 2018 at 5:40 PM, David P wrote:
> FWIW, I added
Hi,
Is there any documentation for media caching other then
https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Project+-+URI+Media+Playback
? The documentation there seems to be a bit out of date (e.g. the commands).
Also I see that there are files that have expired that are still on the
box.
On Wed, Jun 6, 2018 at 1:51 AM Olivier wrote:
>
>
> 2018-06-05 20:29 GMT+02:00 George Joseph :
>
>>
>>
>> On Tue, Jun 5, 2018 at 10:59 AM Olivier wrote:
>>
>>>
>>>
>>> 2018-06-05 15:27 GMT+02:00 George Joseph :
>>> Thank you very much, George for replying.
>>>
On Tue, Jun 5, 2018
I appreciate it if someone can post an an example for function CHANNELS showing
the usage of the regular expression filter.
Basically I would like to get a count of active channels having a certain
criteria. Is it possible to search for a channel having a custom variable set
to specific value
On Wednesday 06 June 2018 at 16:30:08, Dovid Bender wrote:
> On Wed, Jun 6, 2018 at 6:18 AM, Antony Stone wrote:
> > On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote:
> > > Hi,
> > >
> > > I have tested ControlPlayback and grabbed files via an apache server
> > > with no issue.
> >
> >
On Wed, Jun 6, 2018 at 6:18 AM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote:
>
> > Hi,
> >
> > I have tested ControlPlayback and grabbed files via an apache server with
> > no issue.
>
> ControlPlayback is an Asterisk dial
On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote:
> Hi,
>
> I have tested ControlPlayback and grabbed files via an apache server with
> no issue.
ControlPlayback is an Asterisk dialplan function.
How have you integrated this with Apache?
> I want to be able to grab files via aws S3 wh
Hi,
I have tested ControlPlayback and grabbed files via an apache server with
no issue. I want to be able to grab files via aws S3 which would require me
to add some headers to authenticate. Is there any way to have Asterisk add
headers or would I need a http proxy in the middle?
TIA.
Dovid
--
Hi All,I tried to switch from SIP to PJSIP but I can't make any calls.
Asterisk 15.4.0Clients: MicroSIP (based on the pjsip SIP stack)
With sip.conf all functions OK (SIP instead of PJSIP in extensions.conf)
I converted SIP to PJSIP with the scriptÂ
contrib/scripts/sip_to_pjsip/sip_to_pjsip.py and
2018-06-05 20:29 GMT+02:00 George Joseph :
>
>
> On Tue, Jun 5, 2018 at 10:59 AM Olivier wrote:
>
>>
>>
>> 2018-06-05 15:27 GMT+02:00 George Joseph :
>> Thank you very much, George for replying.
>>
>>>
>>>
>>> On Tue, Jun 5, 2018 at 3:35 AM Olivier wrote:
>>>
Hi,
After a long dis
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