Hi, I have a basic dial plan question; Here is the scenario. Call comes through IAX and my * authenticate, then collect the digits and dials out, simple :). Here is the dial plan; [did-in] ;for did callers exten => 866219xxxx,1,Ringing exten => 866219xxxx,2,Wait,4 exten => 866219xxxx,3,Answer exten => 866219xxxx,4,Authenticate(/etc/asterisk/authenticate.txt|a) exten => 866219xxxx,5,GoTO(s|1) exten => s,1,BackGround(pls-entr-num-uwish2-call) exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten => _9XXXXXXXXXX,1,Playback(pls-wait-connect-call) exten => _9XXXXXXXXXX,2,AbsoluteTimeout(3600) exten => _9XXXXXXXXXX,3,ResetCDR(w) exten => _9XXXXXXXXXX,4,Dial(H323/${EXTEN}@10.10.10.10,90) exten => _9XXXXXXXXXX,5,Congestion exten => _9XXXXXXXXXX,105,Busy 1. Isn't there be a better way to collect digits other than going to S extension ? 2. How do we prevent a salient extension falling into s portion bypassing the authentication ? Appreciate any comments, suggestions. Cheers SW
Hey everyone, Hopefully this will be simple enough to answer. I have a menu setup like below: exten => 850,n,Set(MenuLoop=1) exten => 850,n,Playback(mercury-prompts/welcome) exten => 850,n(MainMenu),Background(mercury-prompts/MainMenu-if-you-know-the-ext) exten => t,1,Gotoif,$[${MenuLoop}=1]?|850|100:|t|2 ;Loops the main menu twice exten => t,n,Goto(Mercury-Sales,852,1) exten => 850,100(FirstLoop),Set(MenuLoop=2) exten => 850,101(SecondLoop),Goto(Mercury-Network,850,MainMenu) Basically we want the caller to be routed through the menu twice which it does very well. I would like to make the code a little easier to update by using 'n' priorities instead of numbers. The part I am having trouble is from the 't' extension. I would like to have to say |850|FirstLoop, but that gives me an error saying it needs a number. I know if i am in the same extension I can just use the name and it works. Is this possible, so far things that I have tried such as using ?context|exten|label hasn't worked. Thanks, Kevin
Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem Vorgang wenden Sie sich bitte an postmaster@teamware-gmbh.de Vorschau: Hi List, this is probably quite straightforward. I need to call a sip extension for 15 seconds, if unanswered I then need to call the same sip extension and an additional sip extension for a further 15 seconds, finally if the calls aren't answered I need it to go to a generic unavailable VM. [...] Inhaltsanalyse im Detail: (5.4 Punkte, 5.0 ben?tigt) Pkte Regelname Beschreibung ---- ---------------------- -------------------------------------------------- 0.1 FORGED_RCVD_HELO "Received"-Kopfzeilen enthalten gef?lschte HELO-Identifikation -2.6 BAYES_00 BODY: Spamwahrscheinlichkeit nach Bayes-Test: 0-1% [score: 0.0000] 0.0 HTML_MESSAGE BODY: Nachricht enth?lt HTML 2.2 DCC_CHECK Gelistet im DCC-System (http://rhyolite.com/anti-spam/dcc/) 1.7 MSGID_DOLLARS Muster in Kopfzeile "Message-ID" typisch f?r Spam 1.9 RATWARE_MS_HASH Bulk email fingerprint (msgid ms hash) found 2.8 RATWARE_OUTLOOK_NONAME Bulk email fingerprint (Outlook no name) found -0.7 AWL AWL: From: address is in the auto white-list Die urspr?ngliche Nachricht enthielt nicht ausschlie?lich Klartext (plain text) und kann eventuell eine Gefahr f?r einige E-Mail-Programme darstellen (falls sie z.B. einen Computervirus enth?lt). M?chten Sie die Nachricht dennoch ansehen, ist es wahrscheinlich sicherer, sie zuerst in einer Datei zu speichern und diese Datei danach mit einem Texteditor zu ?ffnen. -------------- next part -------------- An embedded message was scrubbed... From: "Chris Blunt" <chris.blunt@entropy-it.com> Subject: [asterisk-users] Dial plan question Date: Fri, 14 Jul 2006 11:38:13 +0100 Size: 10517 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20060714/5f14441a/attachment.eml
> I need to call a sip extension for 15 seconds, if > unanswered I then need to > call the same sip extension and an additional sip > extension for a further 15 > seconds, finally if the calls aren't answered I need > it to go to a generic > unavailable VM.> My question is if the first sip extension is busy, > and I don't have the "100 > + x" busy VM defined will it just carry on to the > next priority without > complaining or is there a more elegant way of > achieving this? > > > > Example of my dialplan: > > > > exten => 0870xxx,1,Wait(2) > > exten => 0870xxx,2,Answer() > > exten => 0870xxx,3,Playback(cust-greeting) > > exten => 0870xxx,4,SetCIDName(Tech) > > > exten => 0870xxx,5,Dial(SIP/4902,15,tr) > > exten => 0870xxx,6,Dial(SIP/4902&SIP/4903,15,tr) > > exten => 0870xxx,7,Voicemail(u7003) > > exten => 0870xxx,8,HangupHi Chris Yes that will work and is as you say a simple and fairly straightforward way of doing what you require. Regards Jon Jon Farmer Telford, Shropshire, UK ___________________________________________________________ All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html
Chris: One issue you might find, depending on the SIP phone at 4902, is that it will show a missed call for the first 15 second attempt. If 4902 answers in the second 15 second attempt, it will still show a missed call, when the incoming call was actually answered. If extension 4903 answers the second attempt, 4902 will show two missed calls when it really was just one incoming call. I imagine the CDRs will show something similar, if you are using them. (I was trying to test the DND status of an SNOM phone by hitting it with a 1 second call. The missed call logging on the phone made this approach unacceptable.) I think a better solution is to use dial macros, and an exmaple can be found here: http://www.voip-info.org/wiki-Asterisk+cmd+Dial Look for "Example 3: Dial multiple channels, partially delayed" -Bill On Fri, Jul 14, 2006 at 11:38:13AM +0100, Chris Blunt wrote:> Hi List, this is probably quite straightforward. > > > > I need to call a sip extension for 15 seconds, if unanswered I then need to > call the same sip extension and an additional sip extension for a further 15 > seconds, finally if the calls aren't answered I need it to go to a generic > unavailable VM. > > > > My question is if the first sip extension is busy, and I don't have the "100 > + x" busy VM defined will it just carry on to the next priority without > complaining or is there a more elegant way of achieving this? > > > > Example of my dialplan: > > > > exten => 0870xxx,1,Wait(2) > > exten => 0870xxx,2,Answer() > > exten => 0870xxx,3,Playback(cust-greeting) > > exten => 0870xxx,4,SetCIDName(Tech) > > exten => 0870xxx,5,Dial(SIP/4902,15,tr) > > exten => 0870xxx,6,Dial(SIP/4902&SIP/4903,15,tr) > > exten => 0870xxx,7,Voicemail(u7003) > > exten => 0870xxx,8,Hangup > > > > > > Thanks for your time and advice. > > > > > > -- > > Chris Blunt > > >> _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users