similar to: Crazy stuff in latest CVS HEAD

i get lots of the below from friday 15.7.05 cvs as well ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ... ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

So it turns out that -pre13 was quite buggy after all. Once I started running the list of tests that I described in my email, a bunch of those bugs came to heed. Now I think I've fixed them, and so have released -pre14. I have also formally decided to call the final release 3.00 when it's ready. Between code base rewrite + EXTLINUX + libcom32 + simple menu, there is plenty of

I got some weird mail back, claiming that my e-mail was blocked, w/o indicating what server was blocked, after the mail appeared in the mailing list. Is this some BS shakedown? I use postfix on a DSL line forwarding to earthlink, this is hardly something that should be rare in the BSD world... the mail appeared to come from a web mail server and the first link has the following lovely text:

Hello, I have just set up an asterisk box (Debian unstable) and I would like to test it with a H.323 application (gnomemeeting). When I call the demo voice menu, I can't hear any sound. asterisk says that the soundfile is played: -- Executing BackGround("H323/ip$212.9.189.172:30005/29597", "demo-congrats") in new stack -- Playing 'demo-congrats' (language

hello, I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. Anybody could help me to configure Asterisk in order to set instant message and presence ? I've tried with Ondo sip server it's ok ! Regards D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail

Hi, What can it be when I can establish a connection between two Softphones but no voice is transfered ? thnx Hugo,

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

Okay, now that I've got NUT up and running with the newhidups driver, I can give it a quick shakedown. Most of the values look good, but are few are off. Here's what I get: elrond@foxstar:~$ upsc foxstarups@localhost battery.charge: 100 battery.charge.low: 10 battery.charge.warning: 50 battery.date: 3150/08/01 battery.mfr.date: 2005/03/22 battery.runtime: 1935 battery.runtime.low: 120

Hi - > I am not sure what I did but blind transfers do not work. The Polycom does > not allow me to dial the extension of the person I want to transfer to after > I hit: > > transfer -> blind I would strongly suggest getting the latest firmware, and using the sample configuration files with that firmware to set up your phone. This SHOULD work. If it still does not work

What is the reason for llvm-gcc? Will regular gcc work (except for the releases that are broken, of course)? Does llvm in any way depend on features of llvm-gcc instead of gcc? Or is it optional? Was it conceived to give llvm itself a through shakedown? Is it there just in case you really want llvm code in some form from a C program? Is such llvm code necessary for some kind of use --

Hi - We have a couple of large spaces that we'd like to cover with dedicated conference units like the Polycom Soundstation IP3000. We're concerned about adequately covering the spaces, though, one of which is very long and narrow. I wanted to get external add-on microphones for the IP3000, but I've found that unlike some of their other conference products, it does not have this

I think Polycom has added another feature that nobody wants. With MWI configured, and a phonexxx.cfg that has this: <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="XXX" msg.mwi.2.subscribe=...> </msg> Under 1.3.4 and earlier, the phone would immediately

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

I've been getting these really weird errors that scroll almost nonstop across the screen when someone attempts to place a call and it starts ringing. Jul 19 17:55:01 ERROR[23246]: utils.c:509 tvfix: warning negative timestamp -536835.-70000 Does anyone know what this is all about and what I can do to fix it? Thanks for any help in advance, this is causing huge problems!

Hi All - I've got a load of Polycom phones, and for the most part, I think they're great, but one thing that is bugging the heck out of me (and my users) is the "on-hold" feature. When you're on a call, and another one comes in, it doesn't ring the second line appearance on the phone, even though I have it registered separately, and I've tried to make my

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

I've been evaluating asterisk for quite some time now and am attempting to create services on it. The system is simple right now. asterisk seems to look up atleast every week if not more. I am running asterisk 1.0.9 and would like to find similiar experiences of long term stability. I attempted to debug it, but my asterisk isn't compiled with all the possible debugging flags, which

Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for