search for: izol

...$pocpart6F)){ l<-subset(k,k$pocpart6F==i12) for(i13 in levels(l$styk6F)){ m<-subset(l,l$styk6F6==i13) for(i14 in levels(k$kond6F)){ n<-subset(m,m$kond6F==i14) for(i15 in levels(k$styd)){ o<-subset(n,n$styd==i15) for(i16 in levels(k$zasl)){ p<-subset(o,o$zasl==i16) for(i17 in levels(p$izol)){ r<-subset(p,p$izol==i17) i<-i+1 n<-length(r$pohl) data2[i,]<-c(n,i1,i2,i3,i4,i5,i6,i7,i8,i9,i10,i11,i12,i13,i14,i15,i16,i17) }}}}}}}}}}}}}}}}} I believe that there exist smart solution, thanks -- View this message in context: http://r.789695.n4.nabble.com/counting-across-leves-...

hi all! Is it possible to trigger some external device trough parallel port or. serial on some caller id ? or if it's possible tu run some external (linux) program on specific caller id ? OR if anyone have idea how to open the doors with asterisk :) by, Thomas

hello, i have a problem starting asterisk with one x100p digium and two hfc chipset isdn cards with bri-stuff.0.0.2. "ztcfg -vv" shows me a this info: Zaptel Configuration ====================== SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03:

Is any way to limit music on hold (or ringtones) to specified time ? I need it to play it ~ for 7 seconds . How to do this ? in dial plan i have: exten => _021XXXXXX,4,Dial,Zap/1/BYEXTENSION||r when go to this extension it rings once! and then asterisk say : -- Zap/1-1 answered Modem[i4l]/ttyI0 and it stop ringing ;) becouse mean that other end is ringing :) .. BUT when the other

hi! when i have busy signal on analog line (zap card) it doesn't detect that line is busy ? is it possible to change detected sequence (frequency) of busy tones on line (zapata.conf ??) tnx, Thomas my zapata.conf [channels] language=en context=lin1 signalling=fxs_ks channel => 1 group=1 echocancel=yes echocancelwhenbridged=yes rxgain=3.0 txgain=3.0 busydetect=yes

what can be wrong ? -- Executing MusicOnHold("CAPI[contr1/8]", "default") in new stack WARNING[13326]: File res_musiconhold.c, Line 259 (moh0_exec): Unable to start music on hold (class 'default') on channel CAPI[contr1/8] == Spawn extension (remote, s, 6) exited non-zero on 'CAPI[contr1/8]' thomas

hi, I have a problem with passing caller id information from telco (isdn) to sip client (grandstream). i see callerid in asterisk verbose console but on grandstream (sip) phone is just internal (own-gs) 101 number. Isdn line is connected with hfc card and p2p , asterisk is latest CVS in extensions.conf i have: exten => 2442242,1,Dial,SIP/101,r|T and in console is this: -- Executing

hi, how to set caller ID for internal SIP users when dialing out on telco ISDN p2p (hfc card) line? I need to setup numbers from 0 - > 9 (10 sip users).internal caller id is working correctly .. from 0 to 9 ,but when I dial on isdn telco line -> gsm show only our prefix number ( xxxxxxY ) , only 6 (x) numbers of 7 (Y). here is extension 1 dialing on isdn line but on gsm is only 6

hi! Anyone on list has working asterisk box with hfc based card (bristuff) and a x100p adapter? Becouse together in box I can't get it working in any way .. thank you, Tomaz

hi, when i call asterisk (on x100p) i got this : CLI> -- Starting simple switch on 'Zap/7-1' Jul 13 15:03:34 ERROR[311316]: callerid.c:192 callerid_feed: fsk_serie made mylen < 0 (-9) Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4735 ss_thread: CallerID feed failed: Success Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4777 ss_thread: CallerID returned with error on channel

hi, I have some problem with musiconhold or playtones (background,...) in this context someone dial out thru sipura 3000: Executing Dial("Zap/1-1", "SIP/sipura3000/054419949|20|m") in new stack -- Called sipura3000/054419949 -- Started music on hold, class 'default', on Zap/1-1 -- SIP/sipura3000-61fe is ringing -- SIP/sipura3000-61fe answered Zap/1-1

hello, i need some suggestion how to indicate caller that calling number is unavailable if some iax user is not registered: this is what I got in asterisk console: app_dial.c:727 dial_exec: Unable to create channel of type 'IAX2' == Everyone is busy/congested at this time why is send to busy congested but user is just unavailable !? any way to play unavailable message if user is

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making

hi! anyone has a idea how to send caller id (incoming number) to shell script like: exten => s,1,wait,2 exten => s,2,System(/bin/script callerid-number) exten =>s,3,answer i need to run script in shell with command line like "/bin/script -a 123456789 " this 123456789 must be callerid (incoming call) number. tnx, Thomas

is it possible to do with asterisk something like this how ? maybe some copy of extensions and some other file ..pls. call to asterisk server -> user 100 is busy -> recorded msg say "all lines are busy , pls wait" ,"you are second caller in 'queue', pls wait" -> caller is on hold till user 100 busy then ring user 100 (ext100) another call to asterisk

hello, anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode? what kernel ? and second question mISDN driver .. anyone has working solution with mISDN and maybe fritz card? what you suggest for DDI -> point2point mode (card,kernel,chan_..., ...) ? thank you, Tomaz

"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas