similar to: sched_settime: Request to schedule in the past?!?!

Dear List, I have made an update to asterisk RC1 - all works well :) but I am getting all the time an error message: Aug 2 13:34:57 NOTICE[15375]: sched.c:221 sched_settime: Request to schedule in the past?!?! Aug 2 13:34:57 NOTICE[18446]: sched.c:221 sched_settime: Request to schedule in the past?!?! I cannot figure out if this error message is important or not and if there is a real

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 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, 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 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,

Some people on this group may have understood from messages posted here that the Sipura SPA-2000 is not currently available for shipping. That is not the case. Voxilla.com has the Sipura SPA-2000 available for immediate shipping, and has had them since late November. The price is $109.95, and it comes with a month of free VoicePulse service with activation fees waived (a $65 value). In return

We have extra equipment that was over-ordered or unused. All of the equipment is brand new. The equipment has been highly discounted to move quickly - the last set of equipment sold in 48 hours. If this equipment is of interest to you, call or e-mail quickly. Buy on VOXILLA and SAVE $300 each (Cisco routers & switches): http://store.voxilla.com/customer/home.php?cat=259 For Sale (all new):

Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't

Anyone have one of these yet? http://www.voxilla.com/shop/index.php?action=item&id=38 -Dan

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

Going over the needs of any PBX that replaces our current system (working toward Asterisk) and have VM functionality question. Currently when someone leaves a voice mail for a sales person (Who is in the field) the system takes the VM and then in turn dials over a POTS line and pages the sales person notifying them of a VM (Does not deliver the message-just notifies) Is this possible with

How to Connect SPA-3000 to Asterisk so * will answer? After setting up Asterisk on Gentoo the extension.conf contains [demo] context; but my asterisk is not answering? In SPA-3000 in PSTN Line - tab under Dial Plan 1 I have: S0<:demo@10.0.0.101> Default dial plan is set to 1. My box's IP where Asterisk is running has IP: 10.0.0.101 Line 1 - tab has: SIP Settings Port: 5060 Nat is

Hi, I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP). The ISDN-BRI is in PTP-Mode (Point to Point "german: Anlagenanschluss") which is enabled within I4L with "hisaxctrl fcpcipnp0 7 1". Everything works fine except that I can not see the called number/MSN of incoming calls within Asterisk and because of this I can not route incoming calls

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab -> Dial Plans -> Dial Plan 8 (<S0:66610>) Should I put extension [66610] in sip.conf with a context in extensions.conf that

Does anybody out there have any evidence that Sipura support is still in operation? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.

Hi all, I need a fully developed web based remote provisioning system. I cant find anything reliable on the internet. Have already checked ataconfig.com and voxilla-ays.com. have tried to contact them but got no response. So if anybody knows a good provisioning system then plz tell me about it. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part

Asterisk-users, Our two-part tutorial explaining how to use VoIP and Asterisk in Amazon?s Elastic Compute Cloud (EC2) has garnered quite a bit of attention. But due to the time required to complete the many steps needed to get up and running, some of you have asked if it is possible to create a much simpler to install ?pre-built? Asterisk EC2 ?instance.? In short, yes it is. And we?ve