search for: callednum

...@peer) > > This sets _ORIGCID = CALLERID(number), I think you meant > > Set(_ORIGCID=${EXTEN}) > > ?? > > I will give it a try. > -- Accepting call from '3035551212' to '3035551313' on channel 0/19, span 1 THIS "exten => 3037687402,1,set(CALLEDNUM=${EXTEN})" RESULTS IN THIS "-- Executing Set("Zap/19-1", "CALLEDNUM=3035551313") in new stack" OK SO FAR, VARIABLE SET -- Executing Dial("Zap/19-1", "local/7205551212@alternate1/n") in new stack -- Called 7205551212@alternate1/n THI...

...ink you meant > > > > Set(_ORIGCID=${EXTEN}) > > > > ?? > > > > I will give it a try. > > > > -- Accepting call from '3035551212' to '3035551313' on > channel 0/19, span 1 > > THIS "exten => 3037687402,1,set(CALLEDNUM=${EXTEN})" > RESULTS IN THIS "-- Executing Set("Zap/19-1", > "CALLEDNUM=3035551313") in new stack" > > OK SO FAR, VARIABLE SET > > > -- Executing Dial("Zap/19-1", > "local/7205551212@alternate1/n") in new stack...

...ariable had a value, it was safe to assume that it had already been processed in the call chain somewhere and therefore the system would be looping the call if it continued. Here are some sanitized Perl based AGI excerpts that accomplish this: sub callfwd_loop_check { my %v; ($v{callednum},$v{cfnum}) = @_; $v{num} = $AGI->get_variable($v{cfnum}); if ($v{num}) { debug(" Call Loop Anaylsis for ".$v{callednum}." = LOOPING"); return(1); } else { debug(" Call Loop Anaylsis...

Hi, Yet another question, now that I have callerid working correctly, I'm trying to work out how to utilise the different numbers I have. I have a 100 number range allocated to my E1/PRI/OnRamp service. My incoming calls are handled like this: Advertised/published number is an analogue line terminating on a X101P. If the analog line is busy, it has a call diversion to the PRI on a TE405P

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

hi Today I?ve installed, apache 2.0.52, mysql-4.1.7, asterisk-perl-0.08 and ASTCC prepaid card aplication from CVS, so now I have access to the astcc-admin.cgi from web server http://asterisk/cgi-bin/astcc-admin/astcc-admin.cgi and I?ve been able to create the database from "Configure" menu but I have some doubts to continue: - Do I have to reinstall asterisk with mysql support? -

...everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880", "CAPI/02myisdnnum:347callednum") in new stack -- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0x193 -- Called 02myisdnnum:347callednum -- CAPI[contr1/02myisdnnum]/2 is making progress passing it to SIP/2.7.184.61-08152880 -- CAPI[contr1/02myisdnnum]/2 is ringing > sent FACILITY_REQ (...

I've noticed that with FastAGI (and maybe AGI) that when you sequentially send a sequence of dial commands, if the call is picked up, that after the call ends, the Fast AGI script keeps executing the commands! Is there anyway to stop execution once a call is picked up? I think looking at the result codes after the Dial to determine if the call was picked up or not is not a good idea... if it

Good Day All, I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from