[asterisk-users] AMI manager logins - omitting from logging output?

2018-06-06 Thread Antony Stone
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

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

2018-06-06 Thread Eric Wieling
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 =>

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

2018-06-06 Thread David P
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

[asterisk-users] Documentation for media caching

2018-06-06 Thread Dovid Bender
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 The documentation there seems to be a bit out of date (e.g. the commands).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

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

2018-06-06 Thread George Joseph
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,

[asterisk-users] Function CHANNELS

2018-06-06 Thread Matt Hamilton
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

Re: [asterisk-users] Using ControlPlayback with AWS S3

2018-06-06 Thread Antony Stone
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. > > > >

Re: [asterisk-users] Using ControlPlayback with AWS S3

2018-06-06 Thread Dovid Bender
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

Re: [asterisk-users] Using ControlPlayback with AWS S3

2018-06-06 Thread Antony Stone
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

[asterisk-users] Using ControlPlayback with AWS S3

2018-06-06 Thread Dovid Bender
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 --

[asterisk-users] pjsip doesn't function

2018-06-06 Thread Marko Tirs
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 script  contrib/scripts/sip_to_pjsip/sip_to_pjsip.py and

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

2018-06-06 Thread Olivier
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