search for: availchan

I sent this earlier under "Editting variable contents" but no-one has responded. So, the subject is now more to the problem, instead of the solution I was trying to implement. ChanIsAvail returns the channel ID plus "-<session>". How can I edit ${AVAILCHAN} to remove this session ID, so I can use its contents in a subsequent Dial statement? Dialing on Zap just gives a warning about unknown option "-", but dialing a SIP channel completely errors out. ------ extensions.conf snippet------------- ; ; Main Home number (8901) ; ; Bedroom1 exten...

...I'm wondering if someone can shed some insight: Asterisk 1.2.9.1 macro newPlaceCallPSTN { s => { TIMEOUT(absolute)=7200; NoOp(Analog Out List: ${ANALOGOUT}); ChanIsAvail(${ANALOGOUT}); NoOp(Available Out List: ${}); // -- theChannel=Cut(AVAILCHAN|-|1); NoOp(Now Calling ${MACRO_EXTEN} on ${theChannel}....); // -- Dial(${theChannel}/${MACRO_EXTEN:${TRUNKMSD}}); }; }; ... I get: -- Executing macro("SIP/162-4c8b", "newPlaceCallPSTN") Oct 13 08:57:22 WARNING[15320]: ast_expr2.fl:183 ast_yyer...

...INUE that the caller exited the queue() and continued on in the dialplan... [incomming] exten => 1,1,Queue(mainqueue,td) exten => 1,2,voicemail(201 at default) exten => 1,3,hangup [screen] exten => _3XX,1,ChanIsAvail(SIP/${EXTEN}&IAX2/${EXTEN}) exten => _3XX,2,GotoIf($["${AVAILCHAN}" = ""]?4) exten => _3XX,3,Dial(${CUT(AVAILCHAN,-,1)},30,mgM(screencallee,s,1)) exten => _3XX,4,Hangup [macro-screencallee] exten => s,1,read(SCREEN_OPT,screenoptions) exten => s,2,GotoIf($["${SCREEN_OPT}" = "" ]?s,1) exten => s,3,GotoIf($["${...

...now is this (I know it's really bad): exten => 16465555555,1,Answer() exten => 16465555555,2,Dial(Local/105@local&Local/106@local&Local/110@local&Local/112@local,30) exten => 16465555555,3,ChanisAvail(Local/105@local&Local/106@local) exten => 16465555555,4,Cut(DESK3=AVAILCHAN||1) exten => 16465555555,5,ChanisAvail(Local/110@local&Local/112@local) exten => 16465555555,6,Cut(DESK4=AVAILCHAN||1) exten => 16465555555,7,Dial(${DESK3}&${DESK4},30,tr) exten => 16465555555,8,Busy (Each local is 1 zap interface) Which is sort of my temporary work around to...

I'm testing ChanIsAvail context and it is not working for me. exten => 55,1,ChanIsAvail(SIP/11&SIP/21) exten => 55,2,Cut(theChannel=AVAILCHAN,,1) exten => 55,3,Dial(${theChannel},r) exten => 55,4,Hangup exten => 55,102,Goto(s,4) It is not dialing SIP/21 when I'm talking on SIP/11, it execute Hangup instruction instruction. According to notes: The channels are checked in the order listed, returning the first available chann...

I am trouble finding a configuration that works for ChanIsAvail and SIP. My two providers are Voxee and Teliax. I have these lines in a macro exten => s,n,ChanIsAvail(SIP/teliax&SIP/voxee) exten => s,n,Cut(CH=AVAILCHAN,-,1) exten => s,n,NoOp(AVAILCHAN= ${CH}) ; Dial the available Channel exten => s,n,Dial(${CH}/${ARG1},60,t) Looking at the execution, I can see what the AVAILCHAN variable returns Executing NoOp("SIP/1001-9906", "AVAILCHAN= SIP/teliax") in new stack However, in order to...

...=yes canreinvite=no incominglimit=1 context=recordings Here's the dial string I'm trying to use: exten => 775,1,SetMusicOnHold(default) exten => 775,2,ChanisAvail(Sip/frontdesk1&Sip/frontdesk2&Sip/frontdesk3&Sip/frontdesk 4&Sip/frontdesk5) exten => 775,3,Cut(DESK1=AVAILCHAN||1) exten => 775,4,ChanisAvail(Sip/frontdesk6&Sip/frontdesk7&Sip/frontdesk8&Sip/frontdesk 9&Sip/frontdesk10) exten => 775,5,Cut(DESK2=AVAILCHAN||1) exten => 775,6,Dial(${DESK1}&${DESK2},15,tr) Any tips, tricks or ideas would be greatly appreciated. Thank you, Christi...

ChanIsAvail returns the channel ID plus "-<session>". How can I edit ${AVAILCHAN} to remove this session ID, so I can use its contents in a subsequent Dial statement? Dialing on Zap just gives a warning, but dialing a SIP channel completely errors out. ------ extensions.conf snippet------------- ; ; Main Home number (8901) ; ; Bedroom1 exten => 8901,1,Macro(twoline,Zap/1,Z...

...after the function like, zap channel hangup. Here is the copy of my extensions.conf file and messages display on consol while making the call. Please help me to fingure out this issue. Thanks Deepak Extension.conf : exten => _9NXXXXXX,1,ChanIsAvail(${TRUNK}) exten => _9NXXXXXX,2,NoOP,${AVAILCHAN} exten => _9NXXXXXX,3,Cut(TheChannel=AVAILCHAN,,1) exten => _9NXXXXXX,4,NoOP,${TheChannel} exten => _9NXXXXXX,5,Dial(${TheChannel}/${EXTEN:${TRUNKMSD}}) exten => _9NXXXXXX,6,Hangup Log File: -- Executing ChanIsAvail("SIP/201-57f5", "Zap/g1") in new stack --...

...07. Anyone who has sipredirect working? Any other ideas, how to transfer a call from Callmanager extension 10 to Callmanager extension 20? Here is my extensions.conf: [general] static=yes writeprotect=yes [macro-dialout-callmanager] exten => s,1,ChanIsAvail(SIP/callman01) exten => s,2,Cut(AVAILCHAN=AVAILCHAN,,1) exten => s,3,Dial(${AVAILCHAN}/${ARG1}) exten => s,4,Hangup exten => s,102,Congestion [default] exten => 2672,1,Dial(SIP/2672,30,tr) exten => 2672,2,Hangup exten => 3333,1,SIPRedirect(6672@cm1:5060) exten => _98XXXX,1,Macro(dialout-callmanager,${EXTEN:2}) Kind...

...t. [globals] TELIAX=username1@teliax VOXEE=username2@voxee [macro-longdistance] ; ; Standard extension macro: ; ${ARG1} - Number to dial ; exten => s,1,SetCallerID("NetConcept"<1234567890>|a) exten => s,2,ChanIsAvail(IAX2/${TELIAX}&IAX2/${VOXEE}) exten => s,2,Read(${AVAILCHAN}) exten => s,3,Cut(C=AVAILCHAN,,1) exten => s,4,NoOp(AVAILCHAN= ${C}) exten => s,5,Dial(${C}/${ARG1},60,tr) -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

...going on? -- Executing ChanIsAvail("SIP/501-304d", "IAX2/teliax&IAX2/voxee") in new stack -- Hungup 'IAX2/teliax-3' -- Executing Set("SIP/501-304d", "CH=IAX2/teliax") in new stack -- Executing NoOp("SIP/501-304d", "AVAILCHAN= IAX2/teliax") in new stack -- Executing Dial("SIP/501-304d", "IAX2/teliax/12222222222|60|tr") in new stack -- Called teliax/19032303776 -- Call accepted by 207.174.202.4 (format g729) -- Format for call is g729 -- IAX2/teliax-7 is making progress passi...

...f exten => _91XX.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _91XX.,n,Dial(SIP/${EXTEN}@192.168.53.197,,tToR) exten => _91XX.,n,hangup() Server_B[192.168.53.197] for call forwarding extension.conf exten => _911X.,1,ChanisAvail(${TRUNK_GRP3}) exten => _911X.,2,gotoif($["${AVAILCHAN}" = ""]?lbl_busy:) exten => _911X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)}) exten => _911X.,n,Gotoif($["${RECORDING_ENABLED}"="Y"]?lbl_dbc:lbl_dial) exten => _911X.,n(lbl_dbc),Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d...

...esilient 'dialout' macro. The logic is simple; try Zap/g1 (a group of two E1s), and if that fails, try locating a channel via DUNDi. Here's a massively cut down version to illustrate the problem I'm having. macro dialout ( dest ) { ChanIsAvail(Zap/g1); noop(Value of AVAILCHAN is ${AVAILCHAN}); Dial(Zap/G1/${dest}||H); }; Here's what happens when only the second bearer is connected: -- Executing Macro("SIP/1210-082a9768", "dialout|0800789456") in new stack -- Executing Set("SIP/1210-082a9768", "dest=0800789456&quot...

...g read urb, error -22" and then obviusly my command does not work. however, again - If I go back to the command line and my command it runs just fine. Any ideas on what might be happening here? Jerry extensions.conf exten => s,1,ChanIsAvail(Console/Dsp) exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1) exten => s,n,System(/home/silentm/bin/usbserial -start) exten => s,n,Playback(beep) exten => s,n,Dial(Console/dsp) exten => s,n,Hangup exten => h,1,System(/home/silentm/bin/usbserial -stop)

...direct Bridge: <none> -- PBX -- Context: XXXXXXXXXXX Extension: XXXXX Priority: XXXXXX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=XXXXXXXXX SIPURI=sip:303 at XXXXXXXXXXXXXXXXX CDR Variables: level 1: clid="Ext. 303" <303> level 1: src=303 level 1: dst=XXXXXXXXXX level 1: dcontext=X...

In 1.4.43 I would see things from "core show channels" like DAHDI/18/xxxxx for line 18 in Asterisk 11 its DAHDI/i4/xxxx How do I get the line number back? Jerry

...; PAGE4441,n,Set(__SIP_URI_OPTIONS=intercom=true) exten => PAGE4441,n,Set(TIMEOUT(absolute)=60) exten => PAGE4441,n,Dial(${DB(DEVICE/4441/dial)},5, A(beep)) exten => PAGE4441,n(skipself),Noop(Not paging originator) exten => PAGE4441,n,Hangup exten => PAGE4441,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available (state = ${AVAILSTATUS})) exten => PAGE4442,1,GotoIf($[ ${CALLERID(number)} = 4442 ]?skipself) exten => PAGE4442,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL) exten => PAGE4442,n,Set(AVAILSTATUS=not checked) exten => PAGE4442,n,Goto(SKIPCHECK) exten => PAGE4442,n(AVAIL),...

...the second call. So both ChanIsAvail() or Dial() both hang on the second call in. So only 1 call in will work. Below is the CLI report of the call that works. This is my context [smvoice-mediacontroller-public-address] exten => s,1,ChanIsAvail(Console/default) exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1) exten => s,n,Playback(beep) exten => s,n,Dial(Console/default) exten => s,n,Hangup Now what ??? Jerry onnected to Asterisk 18.18.0 currently running on nuc11cdev2 (pid = 282195) == Using SIP RTP CoS mark 5 > 0x7feeec0086b0 -- Strict...

...up=1 channel=3; singalling=fxs_ks group=4 callgroup=1 pickupgroup=1 channel=4 and for outbound calls I have this context on my extensions.conf [out-callb] exten => 44,1,Set(LANGUAGE()=es) exten => 44,n,ChanIsAvail(Zap/g1&Zap/g2&Zap/g3&Zap/g4) exten => 44,n,GotoIf($["${AVAILCHAN}" = ""]?4:6) exten => 44,n,Congestion exten => 44,n,Hangup exten => 44,n,Playback,ggestion/varios/moment exten => 44,n,SetMusicOnhold(dialtone) exten => 44,n,Set(TIMEOUT(response)=10) exten => 44,n,Set(TIMEOUT(digit)=5) exten => 44,n,WaitExten(25|m(dialtone))...