similar to: Interruptable SayUnixTime

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!

Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. In iax.conf, I have something like the following: [General]

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

Hi Gang, Hopefully somebody out there has a "doh" for this one. My dialplan announces the date and time using SayUnixTime. When I run this: exten => 36225,1,Set(ABA=999999999) exten => 36225,n,Background(telbank/${ABA}/${CHANNEL(language)}/thetimeis) exten => 36225,n,sayunixtime(,,Abe 'digits/at' IMP) I get this CLI output -- Executing

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

Yes, After call transfer,I don't want to be media go through Asterisk. Is it possible ? Thanks, Karun. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Dameon D. Welch-Abernathy Sent: Thursday, August 12, 2004 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Blind Call Transfer using

I don't see anything on the Wiki or in the documentation about disabling this feature.

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...

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 =>

Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

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. I wanted to write a "popcorn" app for myself, both to learn how to script in extensions.conf, but also because it was something handy. Along the way, I found myself doing something like: [popcorn] exten => s,1,Set(FUTURETIME=$[${EPOCH} + 10]) ... exten => s,n,While(${EPOCH} < ${FUTURETIME}) exten => s,n,Wait(0.01) exten => s,n,EndWhile() exten => s,n,Play(beep)

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich

Hi All, Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf [general]context = demo ; Default context for incoming callsbindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup = yes ; Enable DNS SRV

Hello, Are you using Asterisk 1.4 ? If positive, are you then successfully using IMAP storage ? Your input would be very valuable to decide if rewite of IMAP storage could be considered as bug fix (non one uses IMAP now) or as a new feature (many use IMAP storage today). So please, take a few seconds to reply as up to now (4 answers), successful IMAP user share = 0% ! Regards PS: If someone

Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten => 125,1,Ringing exten => 125,2,Wait(3) exten => 125,3,Answer exten => 125,4,Wait(2) exten =>

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

To: asterisk-users at lists.digium.com From: webaccounts173 at jgoettgens.de Date: Wed, 29 Jul 2015 16:11:31 +0200 Subject: Re: [asterisk-users] Windows Asterisk Help Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

Hi R developers, On slow file systems with large lists of files, file.exists can take a long time to run. It would be nice if users could interrupt this function. I think it would be simple to add: https://svn.r-project.org/R/trunk/src/main/platform.c, at line 1373, add "R_CheckUserInterrupt();" perhaps every some number of iterations if performance is a concern here. Thanks, Chris