similar to: Remove cancelled timeslots from "Past Office Hours" section on "OfficeHours" page

Hi, is there a dovecot feature I did not found yet, which can limit the access to the server to special timeslots like working hours? Or is that a serverside / sssd / auth / pam / account feature? Thanks for hints to some helpfull documentation and sugesstions. Regards . G?tz -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:

Hi. Can anyone help me with the following: [globals] OFFICEHOURS .................................... [internal] exten => *80,2,SetGlobalVar(OFFICEHOURS=100) exten => *80,2,SetGlobalVar(OFFICEHOURS=200) .................................... [incoming] exten => s,1,GotoIf($[${OFFICEHOURS} = 100}]?incoming-officehours:incoming-officehours-off 1. Am I using the right sytanx when

Hi Guys, For a few weeks we ran the CentOS officehours, that were fairly well recieved. And now that CentOS-7 is out of the door, it would be good to restart those again, and perhaps focus on conversations wider than just about the Project and the SIG's process. We should do those too, but we should perhaps also consider including technical content and howto's etc. Would there be

Hi, I created a calendar class and i now have a method which gives me a 1 day view with timeslots etc. now i want to re-use that code for generating 2 and 5 day views. how can i re-use my code so i can make a show_two_days class which uses the logic of the show_one_day class? am i correct it can''t be done with helpers? regards, Remco -- Posted via http://www.ruby-forum.com/.

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hello List, I have few installations out there based on 1.6.1 or above. I'm trying to play different voice mail messages based on certain criteria's. For example, I want during office hours to play (in short): "we are not available to take your call, please leave a message", during off-hours and weekends I would play: "we are closed, our opening hours xx:xx-yy:yy, please

G'afternoon everybody, I'm having a problem with consistently being able to ring our extensions from an outside line. I don't have a problem reaching the number, but during our calls to Background(msg) that I am having a problem. It seems to be an issue with timing. If I press the extension towards the end of the Background(msg) the it often works. However, in the middle of the

Has anyone got a Premisys Slimline channel bank working with a Cisco AS5400 or similar? I'm not sure if my unit is bad, or what. I'm using FXS Loop Start. Calling the port connects immediately without ringing the attached phone. If I pick up the phone, it's connected and I can talk to the caller. Hanging up has no effect. I can see the bit transitions (0101 to 1111 when I go

Hi We interfaced our ASCOTEL PBX to Asterisk. by EuroISDN PRI , DSS1 It works fine on channels 1- 15, but on 17-31 the miststood each other. Asterisk speaks in Timeslots, the PBX in B-channels The signalling is ok, but the bridging is shifted. The first incoming connection is bridged to "nirwana" also no indication is hearable, calling a second internal subcribes bridges them to the

Hi All, I wish to connect * to a Cisco using a E100P board. When I load the driver I got this error message: -bash-2.05b# modprobe wct1xxp ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows Cisco configuration: isdn switch-type primary-qsig

[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups. [Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net' changed from 64.2.142.17 to 64.2.142.29 [Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed Jitterbuf max 600 timeslots And the calls are dropped. I fixed this by turning off enable in dnsmgr.conf My question is: Do you attempt to

I'm experimenting (using for the first time) with using a cisco3620 to connect to the PSTN via a channelised E1 interface, with * handling all of the SIP calls. If anyone has any installation tips / help / documentation I would be most appreciative :) However, my first question is this: when I am in the setup, I see the following: Current interface summary Controller Timeslots

Hello, I have an E100P, and in the zaptel.conf I have: span=1,1,0,ccs,hdb4,crc4,yellow fxsks=1-10 the light on the card is green( BTW what do all those states of the card that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or for the card?) in the asterisks` zapata.conf I have: [channels] context=default switchtype=euroisdn signalling=fxs_ks usecallerid=yes hidecallerid=no

Heh guys, I just received a T400 card, I've been using a T100 for a little while, and it works fine when using a raw channelized T1. I'm relocating my asterisk machine, and PRI's will only be available, haven't found any good config info for PRI's, can someone point me to PRI config info, or let me know what changes I need to make in order to bring them up, I imagine,

Hi all, I need to setup call using particular timeslot on one of E1's. I've looked into http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and it says that: exten => TestTrakt,1,Dial(ZAP/1-2/517255333) exten => TestTrakt,2,hangup should work and force call setup via span 1 (port 1) but when I try setup call rasterisk says: -- Executing [TestTrakt at

Hello, I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and Nortel TX-1. I had problems with name transfer and with the help of Cisco support I've fixed it. Enclosed here are the definitions needed for it. BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM. Here I am using SIP so the router must decode/encode the Q.sig. The Nortel should be defined

Hi i test a new equipment on my backbone: a Cisco AS5300 with voice dsp ressource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec

Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get handed off to the Asterisk server via SIP, at least not in any format that Asterisk

>> Generate a test signal (10+x sine waves per frame), where x increases by >> one for each iteration, and wraps around at 100. > > Testing with sine waves is usually not a good idea. If you intend on > cancelling speech, then test with speech. Ok, I tested more extensively with both music and two-way speech. More on this below. >> However, when peeking at the

I would think I could do this but for some reason I am stymied. I have a PRI from RCN connected to a cisco 3640 (in my day "cisco" is all lower case :-)). My config looks something like this on the cisco... --------------------------------------------------------- voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode