similar to: SayUnixTime quandry

Can anyone tell me what I missed? I'm trying to setup a simple extension (400) that reports the time when it is dialed. I searched the threads and it seems like this should work... Here's what's in my extensions.conf: exten => 400,1,Answer() exten => 400,n,Wait,1 exten => 400,n,SayUnixTime(,EST5EDT,) exten => 400,n,Playback(tt-weasels) [BTW, tt-weasels is hillarious!

If I use SayUnixTime to an extension, it will tell me the CORRECT time: extension =>698,1,SayUnixTime If I use the same in the wakeup-agi, it tells me the time 14 hours in the future: $agi->exec('SayUnixTime', sprintf("%s||IMp", UnixDate($time, "%s"))); Where is the difference? I am in Time zone Asia/Taipei (GMT+8:00) Any ideas? bye Ronald

I'd like to announce the time when people call and hit my voice-menu context, but the complaint is that the time announcement isn't interruptable. Is there any way to make SayUnixTime interruptable? -- PhoneBoy

I'm trying to get SayUnixTime to parse the date correctly and say 'May Twenty Fifth', however it refuses to do so and keeps using the phrase 'May Twenty Five'. I've read all the syntax, tried using both 'd' and 'e', and in the doc it says: d or enumeric day of monthfirst, second, ..., thirty-first However, it doesn't read my h-#.gsm files...

i know with icecast2/ices , karl heyes has some example config files for mixing live/static content ( if i recall correctly ) . however, what i had done in the past is add the static file to the .pls prior to the live mount . if you generate the .pls files on demand , this would work . ALSO , i'm quite sure that both icecast1/2 support intro/outtro files , etc .. this may achieve the same

I've done this and it works great. But there are drawbacks. If the player client is in random track mode, the player may play the live stream first, and never the static content. I've always wanted an option to have a static file(s) presented to the client before the live stream. Having it done on the server would make sure that whatever the static content was it couldn't be

I have a case where I am abstracting the creation of models. After a fair bit of indirection, I eventually get around to making a .new call. I need to be able to pass params to that .new call. I know I can''t use def initialize() in an AR model (rather frustrating). So, I have found some references to after_initialize, but that doesn''t work either. I see several threads on

Hello all: I have a Samba PDC (Samba 2.2.3) on RedHat 7.3 currently serving about 40 to 50 clients with the intent of adding another 60 to 70 in the coming weeks. Everything is working decently well. I do have one problem that I am concerned about though. I have had to make many users Admin users so that a certain application can run. This does not concern me so much except for where data

Hi list, I have a set of 300 or so WAV files I was combining and playing using playback/background in 1.4.X. Now that I have moved on to the 10.0 set, I understand that I can replace my 8 Khz mono files with virtually unlimited Khz mono files (still no stereo, but a quantum leap forward). I've played with this and get good throughputs using SLIN44 formats on SIP. The 2 questions

I've been running classic icecast + liveice for some time now (several years) without problems. A couple days ago, the station manager (NPR affiliate) came to me and asked if we could 'inject' a static file into the stream. Basically what he wants is for the client to first receive a static file (requesting donations of course) and then on to the live stream (*not* intermittently).

There was mention of this [r-sig-fedora at r-project.org] mailing list on one of the other R lists overnight. I thought the list needed a bit of posting, as I could not recall seeing content recently on it. I cross post to the Red Hat hosted list as well, it raises issues relevant there as well I have been packaging in support of many of the financial packages at CRAN and in R-Forge [

On a CentOS 6 64bit system, I added a couple prototype SAS SSDs on a HP P411 raid controller (I believe this is a rebranded LSI megaraid with HP firmware) and am trying to format them for best random IO performance with something like postgresql. so, I used the raid command tool to build a raid0 with 2 SAS SSDs # hpacucli ctrl slot=1 logicaldrive 3 show detail Smart Array P410 in Slot 1

Just checking the transcoding on our Asterisk boxes and I get the following results. I have the g726, ilbc and lpc10 formats and codecs enabled in 'make menuselect' so I dont understand why its showing as no translation path. Any ideas? I am running certified-asterisk-11.2-cert2 Thanks Gareth > core show translation paths alaw --- Translation paths SRC Codec "alaw"

I had a compile problem with the CVS I downloaded on 21 June. I have a Debian box with 2.4.18 kernel (version needed for support of Conexant ADSL). There is a difficulty with Zaptel build regarding HDLC detection. It tries to build it in and then results in unresolved kernel symbols and fails to load. I have had to comment out the entire HDLC defines in zconfig.h to get a driver to install at

I have a 'day' and a 'night' mode. In the day mode, I play a 'background' message which is interruptable by the pushing of a DTMF key - ie - all is normal. In night mode - I decided to get smarter... I play two backgrounds with a 'sayunixtime' in between and now DTMF does nothing - the menu times out to my 'lets get the operator then'... If I change the

Hello All, I was wondering if somebody could elaborate the change in translation of codecs specifically the amount of time increased in Asterisk 11. For example *Asterisk 11* * **alaw **speex * *gsm **15000 **15000 * *ulaw 9150 15000* * * *Asterisk 1.6.x* * **alaw **speex * *gsm **2 12002 * *ulaw 1 12002* I did recalculate the

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

Hi again I'd like to configured my Asterisk to use german sounds for the "Say"-commands... I installed the sounds-files and I tried them with "Playback(de/demo-echodone)" and it works. Now I tried to add an extension to say the current time: exten => 24,1,Verbose(2,Time asked by ${CALLERID(num)}) Exten => 24,n,Set(CHANNEL(language)=de) Exten =>

I have no speex translation ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000

Here is the ufw.log after enabling logging medium and trying to connect to the windows net. Unfortunately the web Microsoft page is in German. I think it says window uses smb1 syntax. Jan 27 15:11:09 martin-RB042AV-ABA-a1410y kernel: [ 887.241685] [UFW BLOCK] IN=enp2s5 OUT= MAC=00:19:21:a2:11:5e:74:27:ea:ab:1e:e0:08:00 SRC=192.168.254.15 DST=192.168.254.39 LEN=90 TOS=0x00 PREC=0x00 TTL=128