similar to: Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'

Hello there I have been struggling with queues, because i think this is the right module for our business. My main goal, is when we receive external calls, the receptionist should be able to transfer the call to us Technicians, and I am trying to add 2 extensions to a queue name [teknisk] Extension 301 and 302. I have a test setup now which I thought should look like this: When a external call

Is it possible to build a dialplan in which shifting from daytime to nightime is not hour based but phone driven ???

I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu --------Dial 1 for support | Dial 2 for special | Dial 3 sales

Hello dear list. I am currently working on a Automatic attendant, and the core things work, but I think the loop function isn't working as expected. I am testing this environment: a sip internal call from 301 to 501. The setup here is when 301 calls 501, and 301 doesn't enter an extension, it will go in loop, 3 times, and then hangup...Can't get that working. Could someone please

Hello, How would you implement free sitting ? The idea is to offer teachers the ability to share the same desk and hardphone : for instance, Mr Foo is teaching mechanics on mondays while Mr Bar is teaching english on wednesdays. Each has his own extension but use the same hardphone. 1. Does a program check a calendar or database somewhere to allocate a phone to a user (as teachers schedules are

Hello dear list. I am having issues on parkedcalls. I am using a Cisco SPA525G as a test phone, and I have the transfer button there when I am in a call, But when I want to transfer the current call I am in, I push the transfer button, and onscreen I se "Enter Number", and if I enter ex sip 200, I have to wait Almost 10 seconds, before the transfer to sip 200 is made, can I reduce

Hi, I posted the following message to the RedHat linux list to see if anyone had any thoughts about this problem. I have only received a few replies, all of which state that they have experinced poor performance problems with ssh on Solaris. They do not, however, state if it is openssh, which version, what hardware, or what version of Solaris. (All I can say is that we have had no problems with

Hello, I'm having problems working with date values in POSIXct format. Here is what I got (eg.lig attached): x <- read.table("eg.txt", sep = ',', col.names=c("ok","time","secs","lig")) # it gives time as factor z <- cbind(x,colsplit(x$time, split="\\s", names=c("date", "clock")))

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

Hello, We are using CentOS 6.6 and keepalived 1.2.13 on two servers for failover, no load-balancing. Failover is governed by the NIC being present, and the Apache and Tomcat processes being present. Both servers are configured as 'EQUAL' (not master/backup). An initial priority of 100 is set, and if a process or NIC fails, then this is reduced by 60 - causing a lower priority to be seen

Hi, I am trying to use the attended transfer. So I put this in my feature.conf: [general] [featuremap] atxfer => *0 blindxfer => #0 I completly restart asterik, and not just make a RELOAD. But during a call, when I press # it runs a blind transfer and if I press * I am disconnected. I am using the CVS version of * get as explain here

Hi, Asterisk 1.4.31 here. I have a queue with members defined, and those member have member names "member 1", "member 2", etc. They are in a realtime DB. When I modify those member names (column membername) the changes aren't reflected in the queue status ("show queues" from cli. They aren't reflected when a new call comes in, or when I reload the

Hi, One of the big features of 1.6 was described as multi-tenant parking. Basically, parking people in different "lots" so the sales dept. could only pick up their calls, and tech support theirs and no mix up was possible. I can only find the original announcement and others asking the same question. Is there some sort of sample conf file of how I would get this functionnal on

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

Mates, I am at the point that I will have to update my trusty samba 2.0.9 to 3.0.6 or 2.2.11. (damn SP2) Problem is that my production machine is still running Mandrake 7.2 and, of course, there aren't any RPMs for the old distro. So, I guess my option is to install from source. Do I need to rpm -e the 2.0.9 before attempting to compile and install 2.2.11 or 3.0.6? My server is a simple

Spring is back in the Stockholm area. After a few day's worth winter re-runs, the sun is back and night-time temperature is at least 5 degrees celsius. Time to move out all my annual flowers and prepare the garden for summer. Sweden is famous for our annual five week holidays - by law. From june to late august, it's almost impossible to make any business decision, since there's always

Hello Everyone, I need to setup a dialplan where if a incoming call is rec'd to a number, Asterisk needs to dial several SIP extensions at the same time. The SIP extensions are for Cisco 7960s and each have multiple line appearnces. For example, exten => 9043442342,1,DIAL(SIP/102&SIP/103&SIP/104&SIP/105,,20) exten => 9043442342,1,Voicemail(u102) The issue I have is

Hello dear list. A couple of years ago, I worked with a Alcatel IP pbx and Alcatel Sip phones, and we had the opportunity to unregister user by typing *-a number and -* again, ex * 99 *, and then the phone number/sip extension was unavailable, and all of the calls to that extension was redirected to the receptionist. When the user came back and wanted to register her sip account/extension, the

K-means recluster data with given cluster centers Dear R user, I have several large data sets. Over time additional new data sets will be created. I want to cluster all the data in a similar/ identical way with the k-means algorithm. With the first data set I will find my cluster centers and save the cluster centers to a file [1]. This first data set is huge, it is guarantied that cluster