similar to: Unable to set callerid for incoming skype calls

Hi, I need help for cleaning this: "[2440810] / www.tinyurl.com/hgaco4fha3" My desired output is: "[2440810] / tinyurl". My attemps: stringa <- "[2440810] / www.tinyurl.com/hgaco4fha3" b <- sub('^www.', '', stringa) #wanted to get rid of "www." part. Until first dot. b <- sub('[.].*', '', b) #clean from

hi, i'm trying to perform a clustering on a big dataframe the code is this: print("load required R packages") require(spgrass6) require(cluster) gmeta6 <- gmeta6() print("read in our 7 raster files from GRASS") x <- readFLOAT6sp(c ("er","crosc","longc","slope","profc","minic","maxic"))

hi to all, frustated for this error, to day i buy a 1 GB memory slot for my laptop now it have 1,28GB instead the old 512, but i've the same error :-( damn!damn!....how can i do? repeat for a little area (about 20X20 km and res=20m) it work fine! have you any suggestion? is ther a method for look if this error depend from my ram or other....? thanks foe any suggestion! i need your help.

Hi! Does anybody know a string function that would calculate how many characters two strings share? I.e. ("Hello World","Hello Peter") would be 7. Thanks. Laetitia

Hi All, I'm having some troubles with using call files. I'm trying to establish the following: - want to use call files to connect two (outside) extensions - want to use the outbound routes set in FreePBX - want to set the outgoing callerid for both calls - want to set a custom CDR field in MySQL ( field name 'azonosito' ) Asterisk is version 1.8.2.3 with freepbx 2.8.1. What

Hi! Trying to figure out how to rewrite calling number of an incoming call... A cell phone (0733025975) dials a X-Lite (977). X-Lite "shows" 733025975 at the display, but I want it to be 0317998975. I thought i could do something like: exten => 977/733025975,1,Set(CALLERID(number)=0317998975) exten => 977,n,Dial(SIP/0317998977) [Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272

Hi, Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram. For some odd reason now that I have the asterisk box almost to the stage I want it, I hit a problem. I have a te405p in the system, Zap/g1 is connected to the telco as an ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250 phone system. When calls come in on g1 they go straight through instantaneously to the

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor() application like this is my extensions.conf: > > exten =>

I have an asterisk installation connected to 2 isdn lines via an AVM C2 card. modules seems to load well, lsmod gives : c4 19588 4 b1 24192 1 c4 capidrv 28468 2 isdn 134604 9 capidrv slhc 7552 1 isdn capi 18112 4 capifs 6024 2 capi kernelcapi 46112 4

Hi, I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer. My extensions.conf: [globals] __TRANSFER_CONTEXT = transfer [common] exten => 123,1,Playback(demo-congrats) exten => 123,n,Hangup() exten => _0X.,1,Dial(SIP/${EXTEN}@PSTN-GW,60) exten => _0X.,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() [transfer] exten

Ciao Simone, why you want each backend to recalc quota only for its managed users and not run "doveadm quota recalc -A" only one time from a backend tha recalc quota for all users? Il 10/03/20 11:18, Simone Lazzaris ha scritto: > Hello dovecot! > > I administer a dovecot installation with 30k users. I've got 4 dovecot > directors as frontend and 10 backends. >

Hi, I tested the capi_chan with latest cvs of * and I have problems with Early-B3. The following dialstring works for me (without Early B3): exten => _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30) But if I add the 'b' for using Early-B3 exten => _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30) nothing changes (no dialtone). If in this example the called party discards the call, there is no

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Hi everybody, I have some problems making calls from a sip user (HT286) to the pstn trough Digium Wildcard TE110P, i allways have an error : SIP 403 INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79 To: <sip:0170708959@192.168.1.4;user=phone>

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

hi all , please need help for an asterisk version 1.4.21.2 i created a write func odbc list records files in sql table: [R] dsn=connector write=INSERT INTO ast_records (filename,caller,callee,dtime) VALUES ('${ARG1}','${ARG2}','${ARG3}','${ARG4}') prefix=M and set it in dialplan : exten => _0X.,n,Set( M_R(${MIXMONITOR_FILENAME}\,${CUSER}\,${EXTEN}\,${DTIME})= )

I allow our internal extensions to transfer calls, so I have the appropriate "t/T" in the Dial() command. When I do the transfer, though, I don't know what context the user entry is interpreted in... The one which calls the macro which does the dial? This is an issue because my internal phones are in the 201-208 range. In my initial context, though, I only match on the

Hi and thank you. I am a beginer trying to install my first TDM04B. I am able to receive call with the card using: [incoming] exten => s,1,Dial(SIP/robgol,20,tr) on my extensions but, with [outgoing] exten => _0X.,1,Zap/1/${EXTEN} I cant send them out. I am getting the following error: Mar 26 23:37:07 WARNING[5189]: pbx.c:1291 pbx_extension_helper: No application