similar to: 2 ring delay before asterisk answer

I need to backup a set of machines that are very similar in nature. I had created an exclude list to backup everything except whats in my list. I then used the --exclude-from=myexcludefile so rsync would not copy unwanted files. eg: /tmp /var/tmp /var/lock .... long list of others /u1 I have 2 drives on each machine and up until now I have no need to be backing up the 2nd drive. Thats what the

I have FreeBSD 7.3 running rsync 3.0.7. I need to backup a remote server to a local drive. This remote server has several filesystems types that dont require backing up, like nfs. If I run the following command: rsync --delete -azHv myremote:/ /machines/myremote/ It will copy ALL filesystems which is not what I want, so I assume I need to do it on a per filesystem basis like: rsync -x --delete

I am running an older version of nut (2.0.2) on a FreeBSD 4.11 machine. This has been running fine for many months until the last few weeks. It seems that several times a week the machine is shutting itself down. Looking at the log files I notice these lines which lead me to think there is something wrong with the serial communication betwen my host and the Ellipse 600 UPS. Jan 5 04:49:02

Hi all, I have a MGE Ellipse 600 UPS connected to com1 on a PC running FreeBSD 4.11 with nut-2.0.2_1 installed. I have everything working and operational. BUT. on rare occasions I reboot my machine and when it comes up I get the following error: Network UPS Tools - UPS driver controller 2.0.2 Network UPS Tools - MGE UPS SYSTEMS/SHUT driver 0.64 (2.0.2) No communication with UPS Driver failed to

Hello all. This is my very first message to the list. I have a TDM400P card, It has 2 FXO channels which are connected to extensions of my PBX (Ericsson BP250), so I can dial from any SIP softphone directly to physical (analog and digital) extensions on my company. My PBX is configured so when I dial 8 on any extension, it will redirect to the first free FXO channel on my TDM400P card.

Hi, I tried using v1.6.0 of the scriptaculous libraries (including latest prototype library) in one of my existing apps that uses the 1.5.1 version, and found that IE 6 was very unstable. Actions get slower and slower over time until the page is basically unusable. This continues until the browser window closes - a page refresh does not solve the problem. I can''t provide the exact code

Hi, I''m new to the prototype and scriptaculous (v1.5.1) libraries. I''m creating a sortable list using the code below, and I''m finding that while a list element is being dragged in IE, the element text is transformed to appear bolder than the original. In Firefox, the text becomes opaque. Is there anyway to stop this from happening, ie don''t change any

Using the latest SVN of zaptel and asterisk, I can no longer receive incoming analog calls. The caller just hears it ringing but Asterisk doesn't pick up. I am seeing these error messages: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]:

hi all, We have a TDM400 card with 4 wfo modules. now the modules load fine and when i start asterisk with on phone line connected it just starts spewing these messages: -- Starting simple switch on 'Zap/4-1' Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)...

Aloha, I'm finishing up the final touches on this install, and have run into an odd problem. I can't seem to get Caller ID on the analog phone lines working. It's a Digium AEX 410 card. I have Verbose set and a line to print the CID: I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf and users.conf [analog] include=>default exten =>

asterisk-users-request@lists.digium.com is believed to have said: >Budgetone and MWI > >The message button can be programmed to dial an extension that checks >voicemail >exten => 160,1,Voicemailmain(${CALLERIDNUM}) > Thanks, this is what I was thinking about. Still, how do you get the BT to dial 160? In my Asterisk setting I have the same mailbox numbers reused for the

I have 1 x100p. Caller id works fine with the beta1 release. Cvshead releases fail with a combination of checksum and ss_thread errors? I'm concerned when beta2 or the 1.2 release comes out it will not work. I have been through the configs I can't find and changes that need to be made to get CVSHEAD to work. Thanks John Hill

Sorry if this is a daft question but when a PSTN call comes in on my X100P the console shows the following; NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)... NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)... NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)...

pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4

Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [s at from-pstn:3] Hangup("Zap/5-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17

So made my first forray into 1.4 and DAHDI and hit a problem. (Not convinced this is a DAHDI issue though...) Testing an analogue line and asterisk sees the caller ID being passed, but then fails to detect ringing. A plain old analogue phone plugged in rings just fine. Console output: == Starting post polarity CID detection on channel 4 -- Starting simple switch on