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----- 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 registered > > In your...

...;: 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 existing file, b...

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 implement custom devic...

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 Thu, Feb 13, 2020...

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 > have a mailbox...

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]." > ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2})) T...

...ration? > > [general] > dateformat=%F %T > > [logfiles] > console => notice,warning,error,dtmf > messages => security,notice,warning,error,fax > verbose => verbose > > > > Thank you! > > ------------------------------ > *From:* John Kiniston <johnkiniston at gmail.com> > *To:* idemkovitch at yahoo.com > *Sent:* Thursday, November 15, 2018 3:17 PM > *Subject:* Re: [asterisk-users] Queue not dialing out to cell phone for > some reason > > OK. > > So it looks like asterisk can't ring FF1565AABB2D-SLS because it's inva...

...s ? 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 endpoint natively....

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? --

...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 test context and h...

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's present, if i...

...ibed in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to be a real shortcoming in app_queue. Any ideas, suggestions, anyone want to work with me to sort this ? Paddy _____ From: John Kiniston [mailto:johnkiniston at gmail.com] Sent: 28 November 2018 21:17 To: paddy at wizaner.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues and penalties This should work, How are you defining your timeouts in the queues.conf ? And to verify, in your extensions.conf you...

...ing at the wiki won't STRFIME just give me what I need based on the unix time that I put in? What I am actually looking to do is convert over from 12 hour format to 24 (unless strftime does just that and I don't kow what am I am doing?). On Thu, Feb 13, 2020 at 12:03 PM John Kiniston <johnkiniston at gmail.com> wrote: > Try using the STRFIME function instead of doing this by hand. > > https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME > > *%H* > > The hour as a decimal number using a 24-hour clock (range 00 to 23). > > *%I* > > The hour as a dec...

...t -- <SIP/pbx2-000004b2> Playing 'beep.gsm' (language 'en') ... -- User entered nothing. -- Executing [s at macro-myconnector:3] GotoIf("SIP/pbx2-000004b2", "1?REJECT,1") in new stack Any idea? 2016-06-30 21:50 GMT+02:00 John Kiniston <johnkiniston at gmail.com>: > I think a simpler way to do this would be to define an member in your > queues.conf that points to a local channel that calls the remote users cell > phone. > > You can use the M option in your dial to run a macro to prompt the user to > accept the call. >...

...725][C-00000002]: pbx.c:4991 pbx_extension_helper: No application 'MYSQL' for extension (IncomingDial, 6951921078, 2) == Spawn extension (DialIn, 6912345678, 2) exited non-zero on 'Dongle/dongle0-0100000002' Any ideas please? On Fri, Apr 21, 2017 at 10:22 PM, John Kiniston <johnkiniston at gmail.com> wrote: > You can use func_odbc to do this. > > https://wiki.asterisk.org/wiki/display/AST/Getting+ > Asterisk+Connected+to+MySQL+via+ODBC2 > > There is a good chapter in the Asterisk book about using ODBC for > hotdesking that may help you understand ODBC as w...

...e point, and the strange thing: DTMF is set to rfc2833, but is working both on incoming and outgoing calls, it is not working only on calls generated with the Originate AMI command, or with the queue member that point to Local dialplan, as you suggested 2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>: > Looking at your logs it looks like you may need to modify your sip.conf, > Check with your provider as to what kind of DTMF they support and configure > sip.conf to use that type of signalling. > > > > On Thu, Jun 30, 2016 at 1:18 PM, nik600 <nik600 at g...

On Monday 06 Jul 2015, Luca Bertoncello wrote: > 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...

...me know how it goes. Alternatively, 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 test context and have put in a wildcard pattern...

Howdy, Is there a way to use realtime with phoneprov.com and pjsip? I've got a working pjsip realtime config currently but I have to add a phoneprov section to my pjsip.conf for each phone I want to provision. I was hoping the Sorcery page in the wiki would help possibly but it's blank :( https://wiki.asterisk.org/wiki/display/AST/Sorcery -- A human being should be able to change a

Is there an equivalent to ${SIPPEER(${peer},status)} for PJSIP? The closest I've been able to get is to use AST_SOURCERY to see if they have a contact ${AST_SORCERY(res_pjsip,aor,${peer},contact) but I'm not certain if I'll still have a contact entry after a phone has gone unreachable? -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a