similar to: RC1 - error message : Request to schedule in the past

Does anybody know what this message means? [2009-10-17 08:11:59] DEBUG[9598]: sched.c:204 sched_settime: Request to schedule in the past?!?! I've tried all I can find on google with no change It seems to happen when call is at a READ() or Background() - What the caller hear is small delays after a press during a message. Still using asterisk 1.4.22.1 because it seems to have the best DTMF

Dear All, Here the version 2.2 a new version of your dear CallingCard Software !!! http://www.areski.net/areskicc-doc-v2/ Many new features have been added and several enhancements made! Newest features : - A new re-build rate-engine - LCR & LCD management (OOOOHHH YESSSSS) - Billing Increment - Progressive Rate - Scheduled Rates (days of the weeks) - Expiration rates - Buy rates

Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as

Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since.... -- Chris ----- Original Message ----- From: "Chris Earle

Hi, Whenever someone leaves a Voicemail in our system we get this message on the console: NOTICE[18447]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Does anybody know what it means? Ricardo.

Hi, I'm trying to install Areskicc but with no much sucess. After installing the UI I login as root/mypass, but in the left menu I only have 3 options: Main Template, disconnect and logout. When I click on Mail Template> Show mail template it shows an error page. First, is there a way to increase the debugging level so I can know what's going on behind the scenes? Second, I've

Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has

Folks! Let me clarify something to the Asterisk community about the CDR tool. 1) This is *not* my code to start with. I picked the original code from this forum here... http://www.voip-info.org/tiki-index.php?page=Asterisk%20CDR%20Areski%20GUI 2) The original code was not working (for most part, as the MySql portion has bugs) and I fixed this and added a few bells and whistles. 3) The

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

Hi all, I'm looking for a reporting solution for Asterisk CDRs. I have a small Asterisk server that will eventually have 4 - 6 trunks. the system is up and the CDRs are being written to a MySQL DB. I tried installing Areski, but had no success .. I assume it's no longer supported... the last update was in March 2005 according to this page..

Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd

Hi all, Just wondering if someone have already done something like that : SIP Client_A ---> 1)call ---> ASTERISK ---> 2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <----------------------------' with user A Make 2 user in conference (point 1 and 2), it's definitely easy, but call

hi everyone! Two days ago we installed asterisk in our labs to do some testing and try the product with a couple of ITGs. Overall, we really loved it! We found it easy to configure and manage, and with good debugging options. There are a couple of questions I would like to ask: a) We had some authentication issues trying to register a Planet ITG with asterisk. Apparently, asterisk ignored

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Hello, I am posting this after spending hours digging through the list archives. Problem : When asteirsk plays a voice prompt, the voice clip is really choppy. I figure that this is something to with the sound card, the timing of playback etc. But cannot seems to find an answer. Here is the Notice which appear when voice prompt is played. NOTICE[1217602880]: File sched.c, Line 209

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is anyboby using the areski CDR reporting tool? I have installed asterisk-stats v1.2 three days ago, but I found a possible bug in it. My calls compare graphic shows most of the on the calls at first hours past midnight, and it never logs anything after lunch time. This is wrong, my calls are made on business hours. The call log lists those calls at the right time. Is there

Last update before I head to Korea for an IP phone deployment New diffs and PHP web interfaces at www.fitawi.com/Asterisk -New in the interface: New fields for conference conference owner and title Delete conferences Changed date/time to listbox to reduce user input error -This will likely have to become a java or other client-side code, since I don't see a way to tie the number of

Hi guys, I have a customer with asterisk registering 100 lines from my Voip Provider. In some times during a day we receive this messages: [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear all. I'm new to this so please forgive my ignorance if I missed something obvious. I've set-up asterisk 0.9_1 on a FreeBSD 4.10 server (I know it's not linux but that's all we have available at that stage). After some struggle to understand how everything works, I set up some SIP accounts for test purposes. I can log in,