search for: 2001,1

hello, i cant find a solution on this (might be) easy problem: i have a time serie by carlandar weeks, so for every carlendar week i have a value. now i would like to use the functions for time series, so i change structur to a time serie with cam <- ts(number,start=c(2001,1),deltat=7/365) or cam <- ts(number,start=c(2001,1),frequency=52) the problem on it is, that 2004 had 53 calendar weeks, which is not recognized there. it follows with using a saisonal structure the weeks are shifting. f.e. first week in 2005 is conected with the second week in 2004. with the...

Kinda confused by this... I have a Cisco 7960 configured with a couple SIP extensions configured on the phone. Just trying to dial one extension from the other on the same phone, but when I do, I get: -- Remote UNIX connection -- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new stack -- Called 2002 -- Got SIP response 486 "Busy here" back from xxx.xx.xx.xxx -- SIP/2002-f29b is busy == Everyone is busy/congested at this time (1:1/0/0) -- Got SIP response 481 "Call Leg/Transaction Does Not Exist&...

...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

...E/NOTIFY mechanism. I can see INVITE, TRYING, RINGING, ACK, BYE but no SUBSBCRIBE in my sip debug traces. I have problem to understand how hint priority works. I follow the instructions from http://www.voip-info.org/wiki/index.php page=Asterisk+standard+extensions but it still doesn't work. [2001] type=friend context=local host=dynamic dtmfmode=rfc2833 incominglimit=1 notifyringing=yes subscribecontext=notify disallow=all allow=alaw allow=ulaw [2002] type=friend context=local host=dynamic dtmfmode=rfc2833 incominglimit=1 notifyringing=yes subscribecontext=notify disallow=all allow=alaw all...

...}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten => _,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => 1001,1,Dial(SIP/one|20|tr) exten => 1001,2,VoiceMail,u1001 exten => 1001,102,VocieMail,b1001 exten => 2001,1,Dial,IAX2/guest@24.202.159.205/2001 exten => 1002,1,Dial(SIP/two|20|mtr) exten => 1002,2,VoiceMail,u1002 exten => 1002,102,VoiceMail,b1002 exten => 6001,1,Ringing exten => 6001,2,Wait(2) exten => 6001,3,VoicemailMain -------------- next part -------------- An HTML attachment...

Can someone enlighten me on why the following doesn't work? setwd('C:/Temp/R') d <- rep(1:53,2) (s <- ts(d, frequency=53, start=c(2000,10))) n <- length(s) k <- n%/%3 for(i in (n-k):n) { st <- c(start(s)[1] + (start(s)[2] + i)%/%frequency(s), (start(s)[2] + i) %% frequency(s)) ed <- c(start(s)[1] +

...ret=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 ; Activate the message waiting light if this ; voicemailbox has messages in it [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=1234 host=dynamic context=from-sip mailbox=101 ==>Extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Di...

...of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' 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 ...

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 dialplan. Here is what I

...s, 1 is yearly (population) and the other is quarterly (bankruptcy statistics). I would like to produce a quarterly time series object that consists of bankruptcy/population. Is there a pre-built function to intelligently divide these time series: br.ts = ts(data=br.df[,-1], frequency = 4, start=c(2001,1), end=c(2008,2)) distPop.ts = ts(data=distPop.df[,-1], frequency=1, start=(2000)) The time series I want is: br.ts[2001Q1]/distPop.ts[2001] , br.ts[2001Q2]/distPop[2001], br.ts[2001Q3]/distPop.ts[2001] , br.ts[2001Q4]/distPop[2001], br.ts[2002Q1]/distPop.ts[2002] , br.ts[2002Q2]/distPop[2002],...

If I try to execute this dialplan, and nobody picks up at any of the three extensions (7780 7781 and 7782), it's supposed to go to voice mail; instead, it hangs up and gives me a busy signal: exten => 2001,1,Dial(sip/7780,20) exten => 2001,2,Goto(2001,102) exten => 2001,102,Dial(sip/7781,20) exten => 2001,103,Goto(2001,203) exten => 2001,203,Dial(sip/7782,20) exten => 2001,204,Goto(2001,304) exten => 2001,304,VoiceMail2(u7782) exten => 2001,305,Hangup However,...

...1002 member => Agent/1003 in extension_addtional.conf exten=> 2020,1,Answer exten=> 2020,2,Ringing exten=> 2020,3,Wait(2) exten=> 2020,4,Queue(MyQueue) exten=> 2020,5,Hangup my agent 1001 and 1002 has been login (in file extensions.conf) [ext-agents] ;Agent Login ;exten=> 2001,1,AgentCallbackLogin(|${CALLERIDNUM}@ext-agents) exten=> 2001,1,AgentCallbackLogin(|${CALLERIDNUM}@from-internal) ;Agent Logout exten=> 2002,1,AgentCallbackLogin(|l) my dial option as DIAL_OPTIONS = twWrm in features.conf [featuremap] automon=1 when from telco, i dial to asterisk, then...

...(IAX2/jmproductions:xxxxx@iaxtel.com/${EXTEN}@iaxtel) [sip] include => iaxtel exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM}) exten => s,1,Dial(SIP/one|20|tr) exten => 1001,1,Dial(SIP/one|20|tr) exten => 1001,2,VoiceMail,u1001 exten => 1001,102,VocieMail,b1001 exten => 2001,1,Dial,IAX2/guest@24.202.159.205/2001 exten => 1002,1,Dial(SIP/two|20|mtr) exten => 1002,2,VoiceMail,u1002 exten => 1002,102,VoiceMail,b1002 exten => 6001,1,Ringing exten => 6001,2,Wait(2) exten => 6001,3,VoicemailMain ________________________________________ From: asterisk-user...

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan. ? sip.conf ? [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=outside secret=1234 host=dynamic [2001] type=friend context=outside secret=1234 host=dynamic [2002] type=friend context=outside secret=1234 host=dynamic [2003] type=friend contex=outside secret=1234 host=dynamic ? [2004] type=friend contex=outside secret=1234 host=dynamic [2005] type=friend contex=outside secret=1234 host=dynamic ? [...

...treams". If I enable SIP debugging on asterisk, then I find the following output "-- Got SIP response 500 "Internal server error (cannot align media streams)" back from 197.7.75.129" followed by the following debug message (no NAT) to 197.7.75.129:5060 -- SIP/2001-a513 is circuit-busy == Everyone is busy at this time We're at 197.7.75.85 port 16816 Answering with preferred capability 2147483647 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Below is the configuration of asterisk SIP.CONF [general] p...

Hi All Has anyone managed to get the hint priority with Swissvoice IP10S phones working? I have 2 phones: a Snom 360, setup as the reception phone on extension 11, and a Swissvoice IP10S on extension 12. When calling each other (tested both ways) I can only ever see the Snom 360 in the Active State from 'show hints'. The Swissvoice stubbornly remains in the Idle State when on a call!

...manager 8 server. I can call from the call manager to a phone on asterisk, but I can't call from a phone on asterisk to call manager. Any help would be greatly appreciated. sip.conf [2000] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2001] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2002] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes ;******************************* ; BEGIN: CUCM(s) Added Below ;************************...

...4af", "Zap/1") in new stack Jul 28 15:51:25 NOTICE[1910]: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Hangup("SIP/2002-a4af", "") in new stack == Spawn extension (default, 2001, 2) exited non-zero on 'SIP/2002-a4af' -------------------------- zapata.conf signalling= pri_cpe switchtype=national pridialplan=unknown group=1 channel=>1-15 channel=>17-31 group = 2 channel => 32-46 channel => 48-62 group = 3 channel => 63-77 channel => 79-93...

I'm wondering if the page application is broken in 1.2.5 The following: exten => 2001,1,Page(SIP/3254105) does strange stuff. The caller's phone immediately drops into the call, while the callee's phone is still ringing. I'd think it was a SIP messaging issue, except that the Dial() command is working fine, which makes me wonder if it's a bug in the Page appplicati...

...internal announce-frequency = 30 announce-holdtime = yes member => Agent/1401 member => Agent/1402 member => Agent/1702 extensions.conf -------------------------------------------------------------------------------------------------------------------------------------------- exten => 2001,1,Answer() exten=> 2001,2,Wait(1) exten => 2001,3,AgentCallbackLogin(||${CALLERIDNUM}@internal) the console message: -- Executing Answer("SIP/1601-aeb3d650", "") in new stack -- Executing Wait("SIP/1601-aeb3d650", "1") in new stack -- Exe...