search for: 2000,1

When using the tis time series package (v1.9), I cannot select or alter a subset of a time series when the time series is created from a matrix and the matrix contains NA values. Example: x<-tis(t(c(1:10,NA)), start=c(2000,1), freq=12) x[x>0]<-0 The second line yields "Error in if (any(i > nrow(x))) { : missing value where TRUE/FALSE needed" However, both of the following work fine: x<-tis(t(c(1:10)), start=c(2000,1), freq=12) # Does not contain NA x[x>0]<-0 x<-tis(c(1:10,NA), start=...

Can someone tell me why this is not valid... [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@Local) exten => 2000,1,Macro(DialProxy,115551212) exten => 3000,1,Queue(testq||||45) while this is: [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@start) exten => 2000,1,Macro(DialProxy,115551212) exten => 3000,1,Queue(test...

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA&SIP/deviceB&SIP/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension to dial in order to modify the...

...5-year-intervalls. Every 5 years there's one tick and one label with the respective year. Now I would like to customize the axes in a way that there is a label every year and a tick every quarter. In the previous code I've determined the time series with series1 = ts(x, start=c(2000,1), frequency=4) series2 = ts(y, start=c(2000,1), frequency=4) What I've tried before is deleting the X axes via gpars=list(xaxt="n") in the ts.plot-code. But after that I was not aible to add the customized axes via axis()... In advance, thank you very much for your he...

...000 will be asked for pressing his desired extension then he can press his desired extension then his number is diled exten=>6000,1,Background(enterdesiredexten) exten=>6000,2,Wait(2) exten=>2000,1,Dial(SIP/${EXTEN}) exten=>3000,1,Dial(SIP/${EXTEN}) __________________________________ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

...immediately and as a result I dont hear the audio. -- Started music on hold, class 'default', on channel 'SIP/XXX' -- Stopped music on hold on SIP/XXX NOTICE[380]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! My extensions.conf reads: exten => 2000,1,Answer exten => 2000,2,MusicOnHold(default) I've also tried: exten => 2000,1,Answer exten => 2000,2,MusicOnHold(default) e...

...ording to what I read it is executed everytime something enters the context. Obviously something was misunderstood. The following is in extensions.com: [from-sip] exten=> s,1,Answer exten=> s,2,Background,dir-intro exten=> s,3,DigitTimeout,3 exten=> s,4,ResponseTimeout,10 exten => 2000,1,Dial(SIP/2000,20) exten => 2001,1,Dial(SIP/2000,21) Any ideas are appreciated. Robert

I currently have the following setup: exten => 2000,1,Playback(/var/lib/asterisk/sounds/Greeting) exten => 2000,2,Queue(Qabcdef|t) exten => 2000,3,Playback(/var/lib/asterisk/sounds/EveryonesBusy) exten => 2000,4,Hangup exten => 2000,103,Hangup What happens is, that the greeting in step one is played regardless if anyone is logged into t...

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * a...

...ng exten => _72X.,1,DBput(SIP/${CALLERIDNUM}=${EXTEN:2}) exten => _72X.,2,SayDigits,${CALLERIDNUM} exten => _72X.,3,SayDigits,${EXTEN:2} exten => _72X.,4,Hangup exten => 71,1,DBdel(SIP/${CALLERIDNUM}) exten => 71,2,Hangup ; i'm sure we can do this with macro's exten => 2000,1,DBget(temp=SIP/2000) exten => 2000,2,Goto(${temp}|1) exten => 2000,102,Goto(2000|3) exten => 2000,3,DBget(dnd=DND/2000) exten => 2000,4,Goto(2000|6) exten => 2000,104,Goto(2000|5) exten => 2000,5,Dial(SIP/2000,20) exten => 2000,6,Voicemail2(u2000) exten => 2000,7,Hangup ex...

...even after Asterisk has executed an Answer() and Playback(). The only other problem that I am having with my setup is that the CONSOLE/DSP is not functional... I am not sure if the two problems are related. Any help is appreciated. Please see my two examples below: Unless my incoming DID (2000), is pointed to a SIP station that is registered and functional, I do not receive call progress tones on inbound calls. If I point the DID to an application like: [inbound_pri] ; PRI from the NEAX2400 exten => 2000,1,Wait,3 exten => 2000,2,Answer exten => 2000,3,MeetMe,|Mps exten =...

...made two extentions n both are able to ring each other through soft phone (X-Lite) below is my dialplan ################################################################################################################################### /ETC/EXTENSIONS.CONF [others] [my-phones] exten => 2000,1,Dial(SIP/2000,10) exten => 2000,2,Answer exten => 2000,3,MusicOnHold() exten => 2001,1,Dial(SIP/2001,20) exten => 2001,2,Voicemail(2001,u) exten => 2999,1,VoiceMailMain(${CALLERID(num)},s) #########################################################################################...

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the X100P to call out) 2. When I use my cell p...

...ning Linux ==>SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all ; Allow all codecs allow=ulaw context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=1234 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100...

...s time (1:1/0/0) -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from xxx.xx.xx.xxx Any ideas? # sip.conf [2001] type=friend username=2001 secret=hjksdfg23ASDF context=ario-extensions host=dynamic nat=yes register=yes qualify=yes disallow=all allow=ulaw mailbox=2000@ario [2002] type=friend username=2002 secret=hjksdfg23ASDF context=ario-extensions host=dynamic nat=yes register=yes qualify=yes disallow=all allow=ulaw mailbox=2000@ario # extensions.conf [ario-extensions] exten => 2000,1,GoTo(2001,1) exten => 2001,1,Dial(SIP/2001) exten => 2002,1,Dial(...

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

...68.22.254' failed for '192.168.22.1' Apr 6 14:46:34 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.2' Apr 6 14:46:50 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2000@192.168.22.254' failed for '192.168.22.1' The ips in the 22.1 and 22.2 are 2 Cisco 7940 phones running SIP image 6.3. The host 22.254 is the dhcp/tftp server/asterisk box When I attempt to dial extension 2000 from ext 2001 I get the following... Apr 6 14:51:57 NOTICE[1116957488]: c...

If I have 2 data frames; df1: dim(df1) = (1,10) df2: dim(df2) = (2000,10) Both with column header names, how do I multiply them together please. I.e create a dateframe dim() = (2000,1) Many Thanks in advance Glenn [[alternative HTML version deleted]]

Hi, I have 3 queue set in the table as below. name,autopause 1000,1 2000,1 3000,1 In queue 1000, the autopause works after member failed to answer call. However, other queues don't work for the autopause function. queue 1000: -- Nobody picked up in 25000 ms -- Auto-Pausing Queue Member SIP/1234 in queue 1000 since they failed to answer. queue 2000/3000:...

...ains alot of missing data which has been omitted from the analysis. The only issue is when I make the plot for the TS residual graph you have to state the frequency in the coding i.e. freq= 52. This is a problem as some years only have 40 weeks of data etc. This means instead of showing a x-axis of 2000-2005 it is showing 2000-2004. Does anybody know how you can define different frequencies for the x axis i.e. 37 weeks, 40 weeks, 52 weeks? Please keep in mind that the code has to allow for a smooth line to be presented over the top of the residual plot. The code I am currently using is: plot(...