search for: timeslotting

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 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

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:

Dear all, In order to reduce clutter, may I suggest that cancelled timeslots be removed from the "Past Office Hours" section of the "OfficeHours" wiki page? Also, I noticed that the URL for 20th Jan 2014 event is missing. Regards, Timothy Lee

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, 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 List, The following code returns an "Error in as.POSIXlt.character(x, tz, ...) : character string is not in a standard unambiguous format" require(raster) require(rts) require(stringi) r <- raster(ncol=100, nrow=100) values(r) <- runif(ncell(r)) list(ID=seq(1:24),month=rep(str_pad(1:12, pad = 0,width = 2 , "left"),2),year=sort(rep(2016:2017,12)))->dt

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.

Hi Team, i have been facing issues with sangoma card with 16 E1. used LibSS7 asterisk 1.6 with 8 E1 the links are stable , but moment i add another card of 8 E1 for to support 16 E1. link keeps fluctuating any idea why ? Please help Thanks Vinod Dharashive -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Hi, I am experiencing one-way audio from: SIP Device ----> Asterisk -----> Cisco 7200 The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass audio from SIP Device to Asterisk through the Cisco 7200 to the other end, but the Cisco 7200 does not return any audio back to the SIP Device or Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no

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

The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and I will be submitting future changes via patches to Digium for additional sounds. Ideas welcome for more text; I may

Hi everyone, This is just a reminder of our upcoming monthly meeting, on the 7th of September at 18:00 GMT - we've changed to a new timeslot for this month, so I thought it'd be worth reminding people a couple of days in advance. Hopefully the new time will be a little easier for most of our contributors and other interested parties to make it to. We've got a preliminary agenda at

Hi everyone, This is just a reminder of our upcoming monthly meeting, on the 7th of September at 18:00 GMT - we've changed to a new timeslot for this month, so I thought it'd be worth reminding people a couple of days in advance. Hopefully the new time will be a little easier for most of our contributors and other interested parties to make it to. We've got a preliminary agenda at

Hi All, These are the code that i used to plot the kmeans clustering. DataSetS01022<-rbind(matrix(rnorm(50),ncol=2), + matrix(rnorm(50),ncol=2)) > colnames(DataSetS01022) <-c("timeslot","em") > (cl <-kmeans(DataSetS01022,2)) >plot (DataSetS01022, col=cl$cluster) > points(cl$centers, col = 1:2, pch = 8, cex=2) Now, i would like to do *the same process

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

Hi everyone, This is just a reminder of our upcoming monthly meeting, on the 7th of September at 18:00 GMT - we've changed to a new timeslot for this month, so I thought it'd be worth reminding people a couple of days in advance. Hopefully the new time will be a little easier for most of our contributors and other interested parties to make it to. We've got a preliminary agenda at