search for: 2000,2

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...

...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...

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...

...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 =...

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...

...ace for demo > > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) > > > > [sports] > > exten => s,1,ResponseTimeout,5 > > exten => s,2,Answer > > exten => s,3,Wait(1) > > exten => s,4,Playback(sports/gafanaSports) > > exten => s,5,Goto(2000,2) > > exten => 2000,1,wait(1) > > exten => 2000,2,Background(sports/teams/theLosAngelesLakersU) > > exten => t,1,Playback(goodbye) > > exten => t,2,Hangup > > > > ******** sip.conf *********** > > [general] > > context=default > > p...

I have a very simple query with regard to summarizing the number of factors present in a certain snippet of a data frame. Given the following data frame: foo <- data.frame(yr = c(rep(1998,4), rep(1999,4), rep(2000,2)), div = factor(c(rep(NA,4),"A","B","C","D","A","C")), org = factor(c(1:4,1:4,1,2))) I want to get two new variables. Object ndiv would give the number of divisions by year: 1998 0 1999 3 2000 2 Object...

...; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls disallow=all allow=ulaw allow=alaw tos=lowdelay ;tos=184 maxexpirey=18000 ; Max length of incoming registration we allow defaultexpirey=12000 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY [opoffice] type=friend secret=opoffice host=dynamic dtmfmode=rfc2833 mailbox=1000 context=local callerid="Operator Office" <1000> [opfield] type...

...have come up with: in extensions.conf: [beepvoicemail] exten => 1000,1,answer() exten => 1000,2,NoCDR() exten => 1000,3,wait(2) exten => 1000,4,Set(TIMEOUT(absolute)=5) exten => 1000,5,playback(voicemail/beeps) exten => 1000,7,SendDTMF(9) exten => 1000,8,hangup() exten => 2000,1,Set(TIMEOUT(absolute)=5) exten => 2000,2,NoCDR() exten => 2000,3,extenspy(,g(${mailbox})WqX) exten => 2000,4,hangup() Here is what I run: Action: Originate Channel: Local/2000 at beepvoicemail MaxRetries: 0 RetryTime: 15 Context: beepvoicemail...

...configuration of asterisk 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) allow=all ; Allow all codecs context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=soft1 ; This device takes and makes calls username=2000 ; Username on device secret=friend ; 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 ; Acti...

Hi, I have two softphones connected to an Asterisk "stable". I have two extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on extension 2000 will ring, but as soon as the call is picked up, extension 2000 will hang up the call. The softphone on 1000 (SIP, SJphone, e.g.) will give a "403 Forbidden" resul...

...shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below. *CLI> sip show peers Name/username Host Mask Port Status 2001/2001 (Unspecified) (D) 255.255.255.255 0 UNKNOWN 2000/2000 (Unspecified) (D) 255.255.255.255 0 UNKNOWN I am pasting sip.conf & extension.conf sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = INVALID ;Autocreatepeer= yes [2000] type=friend username=2000 secret=2000 host=dynamic context=from-sip mailbox=100 canrein...

...command is called ("[.POISXct"). This results in dates being printed in the locale specific format after a subscript operation is applied to a POSIXct object which has cause several problems for me in the past. Here is an example of this problem under R 1.9.1: > x <- ISOdate(rep(2000,2),rep(3,2),rep(26,2),hour=0) > x [1] "2000-03-26 GMT" "2000-03-26 GMT" > x[1] [1] "2000-03-25 19:00:00 Eastern Standard Time" > > unclass(x) [1] 954028800 954028800 attr(,"tzone") [1] "GMT" > unclass(x[1]) [1] 954028800 > Here i...

...: min not meaningful for factors I tried making columns by year: ys=with(dists,by(V2,V1,function(x) x)) or even making that into a matrix: ysm<-as.matrix(ys,ncols=11) But still vioplot isn't happy. After a bit of faff, I end up with: with(dists,vioplot(dists[V1==1999,2],dists[V1==2000,2],dists[V1==2001,2], dists[V1==2002,2],dists[V1==2003,2],dists[V1==2004,2],dists[V1==2005,2], dists[V1==2006,2],dists[V1==2007,2],dists[V1==2008,2],dists[V1==2009,2], names=1999:2009)) ...but this is clearly awful. Surely there's a better way of achieving this? Thanks, Matthew -- Matthew...

...c reinvite=no canreinvite=no dtmfmode=inband callerid="Fred F"<200> dissallow=all Extensions.conf [default] exten => 1000,1,Dial,Zap/1|20 exten => 1000,2,Voicemail,u1000 exten => 1000,3,Hangup exten => 1000,102,Voicemail,b1000 exten => 1000,103,Hangup exten => 2000,1,Dial,Zap/2|20 exten => 2000,2,Voicemail,u2000 exten => 2000,3,Hangup exten => 2000,102,Voicemail,b2000 exten => 2000,103,Hangup exten => _NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) ; Dial Broadvoice for 30 seconds exten => _NXXNXXXXXX, 2, congestion() ; No answer,...

...odify lattice graphics but do not know if these are applicable. I have unreasonable faith that lattice can do anything. Since my eventual goal is to make use of a grouping variable as with dat2 below, lattice will be preferable to complex layouts. Thanks, Andy dat <- data.frame(Year = 1751:2000, Stuff = rnorm(250), Samples = floor(seq(5,30,length.out=250) +rnorm(250,5)), Grp = rep('SiteOne',250)) par(mar=c(5,4,4,4) + 0.1) plot(Stuff~Year, data=dat, type='l') par(new=T) plot(...

I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better

...stered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at this time My config files are below: sip.conf [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw allow=G723.1 context=from-sip [2000] type=friend username=2000 secret=xxxxxxxx host=dynamic mailbox=2000 extensions.conf [general] static=yes writeprotect=yes [globals] PSTN-1=Zap/1 [from-sip] exten => 2000,1,Dial(SIP/2000,20) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hang...