search for: kiniston

My settings for the queue.log are in the [general] section of logger.conf I'm running 13, I didn't see what version you said you were running. If I wanted to add a LOCAL channel to my queue I'd do it as member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com> wrote: > John, > > FF1565AABB2D-SLS is probably invalid because it's not registered/lost > registration. This client is connected via VPN to our network, it...

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. > > > On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis <jean.aunis at prescom.fr > <mailto:jean.aunis at prescom.fr>> wrote: > > Sorry, I forgot to write that the...

Should setting a namedcallgroup & namedpickupgroup supersede numeric callgroups and pickupgroup ? I've got 5 peers on my 13.7.0 box, Three of them have a namedcallgroup & namedpickupgroup of 'kiniston' and Two of them have a namedcallgroup & namedpickupgroup of 'sanday'. I'm not specifying a numeric callgroup or pickupgroup so all the peers are defaulting to a value of '1' for both it appears. Peers: 7001kiniston 7002kiniston 7003kiniston 7001sanday 7002sanday I...

Sorry, I forgot to write that the SIP peer must keep ringing while the announcement is being played. Le 22/08/2016 ? 17:42, John Kiniston a ?crit : > This seems like the obvious answer but maybe I'm misunderstanding the > question. > > exten => s,1,Dial(SIP/alice,20) > same => n,Playback(myannouncement) > same => n,NoOP(Whatever else you want to do goes here) > > On Mon, Aug 22, 2016 at 8:3...

----- Original Message ----- > From: "John Kiniston" <johnkiniston at gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Tuesday, July 28, 2015 12:12:05 PM > Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are registe...

...ymous.txt': No such file or directory [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:949 file_write: File '/home/logs/anonymous.txt' not in line format Asterisk is running as root (yeah, I know!), and has permissions on that directory. Hmmm.... On 4 November 2016 at 21:50, John Kiniston <johnkiniston at gmail.com> wrote: > I'm able to use the FILE function to create files just fine. > > Set(FILE(${CALLFILE},,,al,u)=Extension: s) > > On Fri, Nov 4, 2016 at 2:26 PM, Jonathan H <lardconcepts at gmail.com> wrote: >> >> Seems I can write to an...

No idea on how to write it in my system. On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston <johnkiniston at gmail.com> wrote: > There's some example code in the Dial-Users context of the basic-pbx > samples that might be of use in implementing it. > > They are checking a DEVICE_STATE to see if a phone is BUSY, You could > change it to be a database call or imple...

John, That is correct. I am trying to figure out why Asterisk is executing the set part of the execif, if it's coming back as false. On Thu, Feb 13, 2020 at 2:10 PM John Kiniston <johnkiniston at gmail.com> wrote: > My Apologies Dovid, I think I misunderstood your request. > > You don't have the time you need to convert in the format of date string, > Instead you have your users entering via DTMF when they want something to > happen? > > On T...

On Wed, 2019-02-20 at 11:46 -0700, John Kiniston wrote: > Use the IF function to evaluate and change the dial command directly. Thanks for taking the time, but that doesn't actually answer the question I asked. It in fact answers the caveat I specifically mentioned: > Granted the particular above example could probably be better >...

...e idea. The problem with RetryDial, is that it will cancel >> the first call, play the announce and then dial the SIP peer once again, so >> the telephone will display a missed call. I would prefer to do everything >> in a single call. >> >> Le 22/08/2016 ? 17:57, John Kiniston a ?crit : >> >> You could try using RetryDial() instead of Dial, It supports playing an >> announcement. >> >> >> On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis <jean.aunis at prescom.fr> >> wrote: >> >>> Sorry, I forgot to write that the...

On Fri, 12 Jun 2015 11:49:05 -0700 John Kiniston <johnkiniston at gmail.com> wrote: > Try this for CHAN_SIP: > > same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer > same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the > mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we &g...

John Kiniston <johnkiniston at gmail.com> schrieb: > The easiest solution may be to strip the leading zero's off your caller ID > before your caller enters the Voicemail app to leave you a message. > > > ExecIf(REGEX("^[0][0]." > ${CALLERID(NUM)})?Set(CALLERID(num)=${CALL...

...the asterisk built in database Both of those seem difficult as the process is split between Asterisk and an external script. > - mv the file to temporary folder _before_ faxing (would be the most > easy solution as you already > know how to mv a file via asterisk...) True. This or John Kiniston's idea of lock files could work. I guess I would need to have some process to move it back if it is still there after an hour or so in case something went wrong. The same sort of thing would be needed for John's solution as well. It sure would be nice if I could query Asterisk to see if...

...device status ? From an external program ? When a user leaves VoiceMailMan application ? Using externnotify ? 2. What is MWI:101 at default expression for (see [2] ? Cheers [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+button+to+Monitor+a+Voicemail+Box 2017-11-21 17:58 GMT+01:00 John Kiniston <johnkiniston at gmail.com>: > Hello Olivier, > > I may be incorrect but I don't believe you can hint on a mailbox like > that. > > I've always used custom device states and dialplan logic for my shared > voicemail boxes that are not being watched directly by a e...

I'm implementing a SBC with my Asterisk PBX but the keeps disabling the trunk group I've configured and I think it may be because Asterisk is returning a 4r04 to the OPTIONS. I've created a test context and have put in a wildcard pattern match to try and catch those options but it doesn't seem to work. Is there a way to have asterisk respond with an 200 OK instead of a 404? --

20.07.2018 23:35, John Kiniston пишет: > > On Fri, Jul 20, 2018 at 11:41 AM Saint Michael <venefax at gmail.com > <mailto:venefax at gmail.com>> wrote: > > ​The community would benefit if a non/licensed version of G729 > would be included with Asterisk​, since the license expired. &g...

...you may also be able to configure your SBC > (kamailio/opensips? if so check dispatcher docs for *_reply_codes modparam) > to accept a 404 reply to a SIP:OPTIONS as a valid response. > > > Hope it helps. > > Cheers, > Joel. > > > On Thu, Jul 16, 2020 at 5:04 PM John Kiniston <johnkiniston at gmail.com> > wrote: > >> I'm implementing a SBC with my Asterisk PBX but the keeps disabling the >> trunk group I've configured and I think it may be because Asterisk is >> returning a 4r04 to the OPTIONS. >> >> I've created a t...

More testing. Test test test. :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

I'm trying to address a problem with users transferring to invalid destinations. In my sip peer I'm setting both __FORWARD_CONTEXT and __TRANSFER_CONTEXT to a context with a extension defined below to set some CDR variables before the call is transferred. [customer-forward] exten => _X.,1,Progress() exten => _X.,n,Gosub(do-billing,s,1${EXTEN})) exten =>

That is the general idea. But how do i make it work? is there somewhere ready? On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkiniston at gmail.com> wrote: > Define your *72 and *73 extensions in your internal context, Have them set > a value in the ASTDB that you then check when dialing your handsets. > > The same can be done for call forwarding, store a number in the ASTDB and > check if it...