search for: _00

...an stop searching for a fix ? I wrote this dialplan last October when I was really new to Asterisk and just accepted the behaviour until now when I'm wondering if it can be refined - I've tried and failed and read all that I could. My dialplan is for the outgoing SIP call is: exten => _00.,1,AbsoluteTimeout(3600) exten => _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r) exten => _00.,3,Answer exten => _00.,4,Hangup exten => _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r) exten => _00.,104,Answer exten => _00.,105,Hangup (if call can go through on TRUNK1 send it out, if TRUNK1 is o...

...I have outgoing extensions that choose which of two providers to choose (based on cost for different destinations), and I was hoping not having to have two sets of extension rules - one for the 00 and one for the + variety. An example of how I'm having to do this now: [outgoing] exten => _00.,1,Verbose(International call 00 - Vyke) exten => _00.,n,Dial(SIP/vyke/$EXTEN,30,tr) exten => _00.,n,Hangup exten => _+.,1,Verbose(International call + - Vyke)...

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata...

...f($[${GROUP_COUNT(ph4)}>=2]?400:500) exten => s,401,Set(GROUP()=ph4) exten => s,402,Dial(Sip/${ARG1:1}@phonesystems3,40,Tw) exten => s,403,NoOp(PH4) exten => s,500,Playback(all-circuits-busy-now) And my portion of extensions.conf from where we are jumping to that macro exten => _00[123459]XXXXXXXX!,1,Monitor(gsm,${CALLERID(num)}APP-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) exten => _00[123459]XXXXXXXX!,2,GotoIf($[${DB(internet/disponible)}=1]?3:7) exten => _00[123459]XXXXXXXX!,3,GotoIf($[${DB(moyende/telecom)}=0]?4:6) exten => _00[123459]XXXXXXXX!,4,Macro(phones...

...to get an outside line... Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084) or its just another number to dial... I added the following... the playback just advises me which 'route' is being taken.... In 'extentions.conf' I have... ;Cell Phone call exten => _00[78][234].,1,Playback(posix-cellphone) exten => _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}) ;Default catch all - just dial it.... exten => _0.,1,Playback(posix-defaultroute) exten => _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}) No matter what is dialled - I always go out on the '...

...Tyreman" <paul@tyreman.org.uk> To: <Asterisk-Users@lists.digium.com> Subject: Re: [Asterisk-Users] Re: Analogue telephone cards for the UK Date: Sat, 10 Apr 2004 11:55:26 +0100 Reply-To: asterisk-users@lists.digium.com This is a multi-part message in MIME format. ------=_NextPart_000_0005_01C41EF2.B6A5E1E0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable What I want to do is have the asterisk server sat in my house and used = by my family to access the BT landline and to recieve calls made to that = landline. If it is not...

...t me know as most times the SIP phone user will hear half a ring and then the hangup noise generated by the SIP device when a number they call is busy. Many thanks!! Chris PS please Cc: me a copy as well as to the list in case I miss it - Thanks. << extensions.conf >> exten => _00.,1,AbsoluteTimeout(3600) exten => _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r) exten => _00.,3,Answer exten => _00.,4,Hangup exten => _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r) exten => _00.,104,Answer exten => _00.,105,Hangup <<iax.conf>> [iaxy] type=friend account...

Hi, I am registering my Asterisk boxes to a SIP provider for outgoing calls. My "outgoing" dialplan context tries to dial out in sequence, starting with the SIP provider then ISDN lines and finally analog lines. So the idea is that if the SIP trunk fails then all calls are dialed out via ISDN and analog. I noticed however that if I switch my DSL connection off (ie. no internet access

Hi all, if i make a call between one SIP soft-phone to an other soft phone over asterisk, i get a Segmentation fault after take up. The extension is : exten => _00.,1,Dial,OH323/${EXTEN}@<myip>|60|r This means, if a SIP client comes with 00* then dial to <myip> over H323. If the H323 client takes up, a Segmentation fault occures. But, if the extension is exten => _00.,1,Dial,SIP/${EXTEN}@<myip>|60|r it works. And, the other directi...

hi everybody, I try to record a call with *1 - one touch record feature in g723 format. exten => _00[1-9].,1,Set(TOUCH_MONITOR_FORMAT=g723) exten => _00[1-9].,n,Dial(SIP/${EXTEN}@ext-sip-account,,wW) I have chosen g723 format because in my CLI> show translation there is no translation between g723 format and wav (default for *1 feature). After pressing *1 sequence I have two tiles in my...

...es sure that emergency calls can always be placed on a landline. Any ideas would be appreciated! Phil [softoption-zap] exten => _0[123456789].,1,NoOp(${EXTEN}) exten => _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j) exten => _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,) exten => _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,) exten => _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,) ; The below section will allow for 3 digit BT numbers to be called, by prefixing them with 9 ; For example: 154 is BT Business Faults - dial 9154 exten => _9[123456789]XX,1,NoOp(${EXTEN})...

...eriskA *********iax.conf: [asteriskA] type=user host=voip.xxx.xx username=asteriskA auth=rsa inkeys=name context=default accountcode=asteriskA 2. asteriskB *********iax.conf: [asteriskA] type=peer host=xxx.xxx.xxx.xxx auth=rsa outkey=name username=asteriskA *********extensions.conf: exten => _00[34][01].,3,Dial(IAX/asteriskA:[name]@xxx.xxx.xxx.xxx/${EXTEN}@default,30) or exten => _00[34][01].,3,Dial(IAX/asteriskA/${EXTEN}@default,30) but nogo. Thanks in advance, regards, Rob.

...g to dial PART1${Predigits} PART2${EXTEN}) exten => _X.,2,Goto(nationalcalls,${Predigits}${EXTEN},1) exten => t,1,Goto(#,1) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again" [nationalcalls] exten => _00.,1,Dial(${OUTBOUND}/${EXTEN}) exten => _00.,2,Congestion exten => _01.,1,Dial(${OUTBOUND}/${EXTEN}) exten => _01.,2,Congestion exten => _02.,1,Dial(${OUTBOUND}/${EXTEN}) exten => _02.,2,Congestion exten => _07.,1,Dial(${OUTBOUND}/${EXTEN}) exten => _07.,2,Congestion Unfortunat...

...fused so I am asking to the list. I want to make with * a gateway from PSTN to H323, and to send all incomings call to a predefined IP, which will treat the h323 calls. let's assume that all my incoming numbers starts with 00 here is my extensions [incoming] exten => s,1,Answer exten => _00.,1,Answer exten => _00,2,Dial(OH323/${EXTEN}@XXX.XXX.XXX.XXX,20,r) But nothing happens. I can see on asterisk console: -- Starting simple switch on 'Zap/1-1' -- Executing Answer("Zap/1-1", "") in new stack -- Executing Dial("Zap/1-1", "OH...

...t first, thanks for helping! In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: exten => _00., 1, NoOp() same => n, Set(FAXOPT(gateway)=yes) same => n, Dial(SIP/${EXTEN}@27XgY8YwfI2S9NAg) same => n, Hangup() The problem vanished when I changed that to: exten => _00., 1, NoOp() same => n, Answer() same => n, Progress() same => n, Set(FAXOPT(gateway)=yes)...

...llation period is >> short in both cases), and as soon as one of these works like expected >> I'll cancel the contract with the other one). So I don't need to have >> a general solution which works with every provider around the world. >> >>>> exten => _00., 1, NoOp() >>>> same => n, Answer() >>>> same => n, Progress() >>>> same => n, Set(FAXOPT(gateway)=yes) >>>> same => n, Dial(SIP/${EXTEN}@27XgY8YwfI2S9NAg) >>>> same => n, Hangup() >>> >>> One may assume t...

...you would like will consume a large amount of your time, you will also find yourself doing a lot of research. What you should have found out by now (or perhaps deduced) is that the T.38 is a standard that is varied thus one cannot be assured a T.38 solution will always work. > exten => _00., 1, NoOp() > same => n, Set(FAXOPT(gateway)=yes) > same => n, Dial(SIP/${EXTEN}@27XgY8YwfI2S9NAg) > same => n, Hangup() > > The problem vanished when I changed that to: > > exten => _00., 1, NoOp() > same => n, Answer() > same => n,...

...exten => _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@voipms) > exten => _1NXXNXXXXXX,n,Hangup() > exten => _NXXNXXXXXX,1,Dial(PJSIP/1${EXTEN}@voipms) > exten => _NXXNXXXXXX,n,Hangup() > exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms) > exten => _011.,n,Hangup() > exten => _00.,1,Dial(PJSIP/${EXTEN}@voipms) > exten => _00.,n,Hangup() > > ; inbound context example for your DID numbers, do not add the number > 1 in front > > [voipms-inbound] > exten => {redacted},1,Goto(hello,200,1) ; My  DID > > [phones] > exten => 101,1,Dial(PJSIP/...

...send the DTMF only if the phone is answered. [voip] exten=>i,1,NoCDR() exten=>i,2,Hangup() exten=>s,1,Wait(2) exten=>s,2,Background(beep||) exten=>s,3,DigitTimeout(6) exten=>s,4,ResponseTimeout(10) exten=>s,5,SendDTMF(c) exten=>t,1,NoCDR() exten=>t,2,Hangup() exten=>_009[13456789].,1,Dial(SIP/operador/${EXTEN},60,tr) exten=>_009[2].,1,Dial(SIP/operador/${EXTEN},60,tr) exten=>_00[12345678].,1,Dial(SIP/operador/${EXTEN},60,tr) exten=>_6[0123456789].,1,Dial(SIP/operador/${EXTEN},60,tr) exten=>_9[123456789].,1,Dial(SIP/operador/${EXTEN},60,tr) thanks ___...

...AX2 overhead bandwidth requirement for every call? I don?t know though - The next steps are to 1) hear back for you ;) and 2) put in mrtg bandwidth logging to get an idea of my usage. I have VOIPJET setup as a peer in iax2.conf and then my extensions.conf dial command just looks like: exten => _00.,2,Dial,IAX2/xx@voipjet/011${EXTEN:2} ; Make Call - (In Ireland we use 00 as the International prefix but US people (and VOIPJET) like 011 as the international dial prefix) I'd be very keen to hear from anyone who has knowledge on using IAX2 to route multiple outgoing calls. Thanks very much,...