search for: tasti

Displaying 20 results from an estimated 165 matches for "tasti".

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2003 Feb 21
0
Live a healthy life with VERIUNI nutritionals. Tasty liquid multivitamin (PR#2567)
Live a healthy life with VERIUNI nutritionals. Tasty liquid multivitamin ensures you'll get nearly every daily nutrient required for prolonged health.Powerful antioxidant unleashes the power of red wine extract and polyphenols for the ultimate mind and body energizer. All-natural ingredients. No sugar or artificial preservatives. For more information, check out
2016 Aug 22
3
Dial and start music on hold after timeout
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
2008 Jul 08
3
apt vrs yum ?
I was checking out Dag's ( not dagw ;-> ) new blog... I don't know how much, if at all, this has been debated... http://dag.wieers.com/blog/using-apt-in-an-rpm-world if he is so adamant about apt over yum, why are we using yum still? Laziness? ;-) ...or are we just tasty food centric? Yum yum yum !!! - rh
2016 Aug 22
2
Dial and start music on hold after timeout
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. >
2016 Nov 04
2
Any way of creating a file to write to from the dialplan, or must I use AGI?
That's just what I'm using, John. But I'm getting (eg) [Nov 4 21:46:16] ERROR[1676][C-00000003]: func_env.c:449 file2format: Cannot open '/home/logs/anonymous.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
2011 May 13
4
unexpected results when extending methods to class Class and class Object
Hey all, There''s a core Class class and core Object class in Ruby library: http://www.ruby-doc.org/core/classes/Object.html http://www.ruby-doc.org/core/classes/Class.html First, let''s resolve the simple distinction between an Object and Class as envisioned by Smalltalk but within the Ruby context: #A class is a template used to define methods and properties class Hello
2020 Jul 16
3
Problem with OPTIONS requests.
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? --
2020 Jul 17
1
Problem with OPTIONS requests.
I've got this setup in a test context. [test] exten => s,hint,SIP/7124 exten => s,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => _x.,hint,SIP/7124 exten => _X.,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => Anonymous,hint,SIP/7124 exten => Anonymous,1,NoOP(Options to $EXTEN) same => n,Hangup() I added hints to see if that would make a difference
2017 Nov 23
2
How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?
Hi, 1. How do you then, synced then unread message presence with custom 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
2018 Jan 11
2
how do i enable call features??
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
2018 Jan 10
2
how do i enable call features??
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
2020 Feb 13
2
Help with FUNC_MATH
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
2016 Aug 23
2
Dial and start music on hold after timeout
How about: exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for the remaining 20 of the 40 seconds, as the Playback will not answer
2014 Nov 13
1
pjsip phoneprov realtime?
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
2020 Feb 13
2
Help with FUNC_MATH
John, >From looking 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
2016 Jun 30
2
how to join 2 channels using AGI/AMI
thanks John yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new
2006 Dec 18
3
Shared Line Appearances (SLA) in 1.4
Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly
2007 May 11
0
The Soup - new rails community resource
Greetings all, I am pleased to announced The Soup - http://the-soup.net/ - a tasty new resource for the Rails Community. About Are you a Talented person that is looking for new and interesting projects to work on? Are you a Project Sponsor that is looking to build quality relationships in anticipation of your next hiring cycle? Inspired by the story of The Stone Soup, The Soup provides a
2013 Apr 10
1
AMI Reload action, returning generated errors?
Howdy, I'm building a webapp to allow my techs to do minor dialplan edits and trigger a reload on my PBX's running 1.8 I have no problem triggering a 'reload pbx_config.so' via manager, The problem is how can I see the results of my reload? For example a missing close parenthesis which would show in /var/log/asterisk/messages [Apr 10 13:46:16] WARNING[23911] pbx_config.c: No
2011 Nov 30
4
[LLVMdev] LLVM Bay-Area Social!
It's a new month already, and with a new month comes a new LLVM social! If you're in the bay area on Wednesday, December 7th, join your peers at St. Stephen's Green in Mountain View starting at 7pm and running until 11pm-ish, usually. This is walking distance from the Mountain View Caltrain and VTA stations, and they'll have great beer, drinks, and tasty food. As is the norm,