search for: dialstatus

...|90|HL(" . ($maxtime * 60 * 1000) . ":60000:30000)"; $res = $AGI->exec("DIAL $dialstring"); $answeredtime = $AGI->get_variable("ANSWEREDTIME"); $dialstatus = $AGI->get_variable("DIALSTATUS"); print LOGFILE "Dialstring: $dialstr DIALSTATUS: $dialstatus\n"; $callstart = time(); if ($d...

In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts in the s extension. Goto() is used in examples. Is the prefix "s-" mandatory? Is it related to the original extension "s"? (Apparently Goto(${DIALSTATUS}) won't work for me.) Yuan Liu

What am I doing wrong? Goif dialstatus: busy CONGESTION not working. exten => _7NXXXXXX,1,Dial(SIP/7780${EXTEN:1}@pstn-5665,60,tr) exten => _7NXXXXXX,n,GotoIf($[$["${DIALSTATUS}" = "BUSY"] | $["${DIALSTATUS}" = "CONGESTION"]]?line2) exten => _7NXXXXXX,n(line2),Dial(SIP/9780${EXTEN:1}@pst...

Hello list, Hope all doing well! I've been checking some cases when a Dial fails and dialplan execution continues to handle this. I am finding it a little confusing how we should handle the DIALSTATUS and the HANGUPCAUSE in this situation.... More specifically, I am facing a case in version 13.6.0 where I am getting a DIALSTATUS=BUSY and HANGUPCAUSE=19 after receiving a 480 SIP error. Seems wrong to me, since 480 should be converted to HANGUPCAUSE=19 and DIALSTATUS = NOANSWER (https://wiki.aster...

...sure why. To me they look identical and it has me stumped. This works: [to-test] exten => _X., 1, SetCallerPres(allowed) exten => _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb) exten => _X., 3, Ringing exten => _X., 4, Dial(SIP/9330 at a-test,20,ro) exten => _X., 5, GotoIf($["${DIALSTATUS}" = "ANSWER"]?9) exten => _X., 6, GotoIf($["${DIALSTATUS}" = "NOANSWER"]?7) exten => _X., 7, Dial(SIP/9330 at a-test2,20,ro) exten => _X., 8, GotoIf($["${DIALSTATUS}" = "ANSWER"]?9) exten => _X., 9, Hangup This does NOT work: [to-t...

Thanks Dovid! Indeed looks a bug but regardless of this, this problem made me think that the HANGUPCAUSE could be used for this purpose with benefits. I couldn't find an explanation about when DIALSTATUS would actually be better. The HANGUPCAUSE was reworked in version 11 ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find someone actually stating it is a better alternative or replacement to the DIALSTATUS or something similar. Cheers, Patrick Wakano On 14 March 2018...

Folks, When I have a dial string like this: Dial(SIP/3254101&SIP/3254102,20,tr) and I want to check the ${DIALSTATUS} variable after the dial, how do I know which number I am getting the variable for? And, what about this? Dial(SIP/3254101&SIP/3254102@proxy1,20,tr) What happens in that case? How can I get the ${DIALSTATUS} variable for EACH NUMBER dialled? Thanks, Doug.

...imeout before continuing with dialplan execution. I was under the impression that app_dial would timeout on the signalling prior to the timeout parameter specified in the dial parameter. For example, consider the following dialplan: exten => 111,1),Dial(SIP/phone1,30,tg) exten => 111,n,NoOp(DialStatus=${DIALSTATUS}) exten => 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail) exten => 111,n,GotoIf($[${DIALSTATUS} = NOANSWER]?unavail) exten => 111,n,GotoIf($[${DIALSTATUS} = BUSY]?busy) exten => 111,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy) exten => 111,n(unavail), Goto(voice-ma...

Hi, As of v 1.6.1.1, can anyone tell me what the current possible values for DIALSTATUS could be? I found http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS but believe it is outdated since there is no FAIL or FAILED in this list. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/a...

...Op(${TEST}) exten => _X.,n,Dial(SIP/${EXTEN}) exten => Busy,1,Busy(2) exten => Busy,n,Hangup() exten => Congestion,1,Congestion(2) exten => Congestion,n,Hangup() exten => NoAnswer,1,Wait(10) exten => NoAnswer,n,Hangup() exten => h,1,NoOp(XXXXXXXXX) exten => h,n,NoOp(${DIALSTATUS}:${HANGUPCAUSE}) why if I call the Busy or Congestion extensions, the DIALSTATUS and HANGUPCAUSE variables are not set ? If I call the NoAnswer extension, DIALSTATUS is blank and hangupcause is 16. I presume that this is correct ? If I call (say) extension 1234 all things are set ok. Julian.

...show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) Why, if 5551234 is busy, is DIALSTATUS set to CHANUNAVAIL? Should it not be BUSY? And now with IAX2 (I am using a user:pass that is purposely invalid): exten => 1234,1,Dial(IAX2/user:WRONGpassword@peer/${EXTEN}) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DI...

Hi, I am trying to pass DIALSTATUS to a Macro so that i can set a variable when a call is placed (call is placed via a call file to another extension first). Basically i don't want to dial a number where a call is already bridged and thats why i am setting a variable. [macro-afterdial]; exten => s,1,Goto(s-${ARG1},1) exten =...

I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing. Below is a dialplan snippet and the resulting CLI output. This is running in an 'h' extension. Noop(DIALSTATUS=${DIALSTATUS}) Noop(CDR(disposition)=${CDR(disposition)}) -- Executing [h at pbxmax-dial-simple:1] NoOp("SIP/msx_0...

...LEDTIME=$[${EPOCH} - ${DIALSTART}]); } else { Set(DIALEDTIME=0); } ChanIsAvail(SIP/${EXTEN}); // NoOp(DIALSTATUS EXT ${EXTEN}:); // NoOp(DIALSTATUS=${DIALSTATUS}); // Necessary for Phoenix NoOp(DIALEDTIME=${DIALEDTIME}); // NoOp(HANGUPCAUSE=${HANGUPCAUSE}); // NoOp...

Hello, We have a strange situation (asterisk 1.6.2.14), where we get a result for DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL. This is the (relevant) test dialplan: -------------------------------- [incoming-private] exten => _X., n, Dial(SIP/1001,30) exten => _X., n, NoOp(${DIALSTATUS}) exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1) [incoming-status] ex...

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I...

According to http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS when a caller hangs up before the callee has time to pick the phone up then DIALSTATUS should be CANCEL. And it is. However, the disposition field in the CDR table is "NO ANSWER". So if I analyze the CDR data I won't be able to discriminate calls cancelled by the caller and calls...

Hello, Is it possible to retrieve the DIALSTATUS variable when placing call through a call file. This variable is set when using the Dial() application from the dialplan, but I am using a call file for my current application and need to get the dialstatus. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL...

...ocal and std and the other for mobile calls, all though they can be backups for each other. and then the pstn line originally i had (and still till i get around to changing them) macros to dial each interface and macro's that handle trying each voip and then pstn in order, this is based around dialstatus (hackled from the website). My problem is when I have had long adsl problems (pain in the back side, back to base alarm system) eventually asterisk seems to not want to talk to the voip phone - nor does it allow any calls to be placed. my guess is this srvlookup=yes which I had until recently -...

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia