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...l/0036701234567 at CustomCallOut-1 WaitTime: 45 MaxRetries: 0 RetryTime: 0 ; 2nd CID SetVar: callid2=0036204313763 SetVar: azon2=masodik hivas azonosito { V1pr: ehehhe } Context: CustomCallOut-2 ; 2nd phone num Extension: 003617654321 The contexts: [CustomCallOut-1] ; set custom CDR exten => _0X.,1,Set(CDR(azonosito)=${azon1}) exten => _0X.,n,Set(CALLERPRES()=allowed) exten => _0X.,n,Set(CALLERID(number)=<${callid1}>) exten => _0X.,n,Set(KEEPCID=TRUE) ; pass the call to internal routing include => from-internal [CustomCallOut-2] exten => _0X.,1,Wait(1) ; set custom CD...

...ber)=0317998975) exten => 977,n,Dial(SIP/0317998977) [Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272 answer_call: Dropping call because extensions '977', 's' and 'i' doesn't exists in context [inputinterior.se] Rewriting of outgoing is working... snip exten => _0X!/0317998975,1,Set(CALLERID(number)=317998975) exten => _0X!/0317998977,1,Set(CALLERID(number)=317998977) exten => _0X!/0317998978,1,Set(CALLERID(number)=317998978) exten => _0X!/0317998985,1,Set(CALLERID(number)=317998985) exten => _0X!/0317998987,1,Set(CALLERID(number)=317998987) exten...

How to make outgoing calls from DID and what is theway to get incoming calls from DID. -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110526/3f19091d/attachment.htm>

HI, I'm using the usual Set(Callerid(num) function to change the incoming from skype callerid but it's not working. Asterisk 1.4.31 and last release of skype channels This is the dialplan exten => _0X.,1,NoOP(${CALLERID(num)} - ${CALLERID(name)}) exten => _0X.,n,Set(STRINGA="Skype") exten => _0X.,n,NoOP(${STRINGA}) exten => _0X.,n,Set(CALLERID(num) = ${STRINGA}) exten => _0X.,n,NoOP(${CALLERID(num)} - ${CALLERID(name)}) and is the output NoOp("Skype/lab.trentinonetwor...

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 signalling to my SIP-Phones. In this case "capi debug&...

I have my peers registered to SER.asterisk seems to be sending mwi for the peers seen in the sip show peers CLI command. i have my ser server registered with asterisk as a type=friend and all clients register to ser.how do i get mwi to work for these clients registered to SER. Thank you, -AA

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/${EXTEN} I cant send them out. I am getting the following error: Mar 26 23:37:07 WARNING[5189]: pbx.c:1291 pbx_extension_helper: No application 'Zap/1/${EXTEN:1}' for extension (default, 00290785472, 1) == Spawn extension (default, 00290785472, 1) exited non-zero on '...

...o(emergency,s,1) exten => 0000,1,SetVar(CALLFILENAME=/mnt/asterisk/EMERGENCY_CALL-${CALLERID}-${TI MESTAMP}) exten => 0000,2,Monitor(gsm,${CALLFILENAME},m) exten => 0000,3,Goto(emergency,s,1) exten => _00011X.,1,AGI(blockintl.agi|${EXTEN:1}) exten => _01902X.,1,Hangup exten => _0X.,1,SetVar(CALLFILENAME=/mnt/asterisk/${CALLERID}-${EXTEN:1}-${TIMEST AMP}) exten => _0X.,2,Monitor(gsm,${CALLFILENAME},m) exten => _0X.,3,Dial(Zap/g1/${EXTEN:1}) exten => _0X.,4,Congestion exten => _0X.,5,Hangup include => phatphingers [transfer-record] exten => _52XX,1,SetVar(...

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} -- Arnaud Pignard (apignard@frontier.fr) Frontier Online - Op?r...

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} -- Arnaud Pignard (apignard@frontier.fr) Frontier Online - Op?r...

Hello all, With some help from people in #asterisk on freenode, I've managed to get incoming SIP calls working. Outgoing calls however are however a different matter. My whole working (incoming calls only) SIPgate configuration can be found here. [1] When I uncommon what's in there, nothing works. There doesn't appear to be any useful error being logged , even when debug is

...e = very port = 5060 provider = registeriax = no registersip = yes secret = xxxxxxxx trunkname = Custom - iinet trunkstyle = customvoip username = 028012xxxx The dialplan, Just dial 0, then number, then strip the first 0 and dial [numberplan-custom-2] include = default plancomment = home exten = _0X!,1,Macro(trunkdial,${trunk_1}/? ${EXTEN:1}) comment = _0X!,1,All Numbers,standard The trunks context, Wich is all incoming calls go to exten 400 (office) [DID_trunk_1] include = default exten = _X.,1,Goto(default|400|1) exten = s,1,Goto(default|400|1) sip show registry asdev*CLI> sip show...

...xten => _381659XX,1,SetMusicOnHold(record) exten => _381659XX,2,Dial(Zap/g4/2${EXTEN:-2},6000,t) exten => _381659XX,3,Dial(Zap/g4/211,600,t) exten => _381659XX,4,Hangup [te405p-outtelstra] exten => _00011X.,1,AGI(blockintl.agi|${EXTEN:1}) exten => _01902X.,1,Hangup exten => _0X.,1,Dial(Zap/g1/${EXTEN:1}) exten => _0X.,2,Congestion exten => _0X.,3,Hangup include => dialstring [to-sip] exten => 098,1,WaitMusicOnHold(45) exten => 099,1,Echo ;simple echo test when you dial 099 on your phone exten => 511,1,VoiceMailMain(s${CALLERIDNUM}) exten => 5...

...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 => _0X.,1,Answer() > exten => _0X.,n,MixMonitor(${CALLERID(num)}-${STRFTIME($ > {EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav) > exten => _0X.,n,Dial(IAX2/pabx-canall/${EXTEN},60,tT) > > exten => _2XX,1,Answer() exten => _2XX,n,MixMonitor(${CALLERID(num)}-$ >...

Thank you Melissa. I love the phone but the dial keypad is a little bouncy. I was hoping for a more solid feel like on the analog PT390's or my quality standard, the Nortel 9417CW. Other than the MWI problem, I'd like more documentation on the configuration paramters. I have found little online configuration documentation other than very basic stuff on the Sayson website. I'd

...yback(tt-weasels) exten => s,n,Hangup() [incoming_calls] ;exten => 7977529,1,NoOp() ;exten => 7977529,n,Dial(SIP/241|SIP/243,30,Tt) exten => 7977529,1,Dial(SIP/241&SIP/243,30,Tt) ;exten => 7977529,n,Dial(SIP/243,30,Tt) exten => 7977529,n,Hangup() [outgoing_calls] exten => _0X.,1,NoOp() exten => _0X.,n,Dial(Sip/${EXTEN}@tellfree,30,Tt) exten => _0X.,n,Hangup exten => _7X.,1,NoOp() exten => _7X.,n,Dial(Sip/${EXTEN}@tellfree,30,Tt) exten => _7X.,n,Hangup [internal] exten => _24[1-9],1,Verbose(1|Estension ${EXTEN}) exten => _24[1-9],n,SayDigits(${EXTEN...

What is the proper method to install/configure an X100P FXO card? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031009/ce8a7803/attachment.htm

...lcapi 46112 4 c4,b1,capidrv,capi 'capi info' from * console returns correctly : Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. (chan_capi-0.3.5) I'm doing outgoing calls in this way : (extensions.conf file) exten => _0X.,1,Dial(CAPI/0227006796:b${EXTEN:1}) exten => _0X.,2,Congestion I can use only the first controller, on the third call out the line is occupied, asterisk console says : Connected to Asterisk 1.0.7 -- Executing Dial("SIP/annap-5827", "CAPI/0227006796:b3282045") in ne...

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/${EXTEN}@PSTN-GW,60) exten => _0X.,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() [transfer] exten => 123,1,Goto(common,${EXTEN},1) Scenario A: SIP Phone dials 123 and hangs up after 10 seconds. CDR is recorded just fine. Scenario B: SIP...

Hello; I am looking for SIM adaptor to be connected with Asterisk to be able to send and receive calls from the mobile operator and if possible the same adapter to be used for SMS "sending and receiving". But what if anyone called this SIM card that is connected to this adapter and no one relied his call, how this miss call can reach for the use at the asterisk PBX? Regards Bilal