similar to: DIALSTATUS for Multiple Dialled Numbers

Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug.

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

I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the call from 3254102 to 3254104. When I try and transfer the call, I get the following on the Asterisk console. Mar 3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '16749440-c28be02e-64b73be7@172.31.16.67'. Both legs must reside on Asterisk box

I've noticed that with FastAGI (and maybe AGI) that when you sequentially send a sequence of dial commands, if the call is picked up, that after the call ends, the Fast AGI script keeps executing the commands! Is there anyway to stop execution once a call is picked up? I think looking at the result codes after the Dial to determine if the call was picked up or not is not a good idea... if it

Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status

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] exten

I'm trying to write a dialplan that will allow me to "stress" test it. I want to be able to dial an extension, or pretend that the extension is busy or out of order (so that I can see what to do) given the dialplan snippet: [outbound] exten => _X.,1,NoOp(${TEST}) exten => _X.,n,Dial(SIP/${EXTEN}) exten => Busy,1,Busy(2) exten => Busy,n,Hangup() exten =>

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -------------- next

I'm trying to find a way in extensions.conf to match ANYTHING dialled, including characters such as *. The following works for numbers... exten => _X.,1,AGI(script) but doesn't catch when someone dialls * first. I tried this: exten => _.,1,AGI(script) which catches when someone dials say, *123 for example, but after the AGI script terminates, Asterisk executes it again with

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}@pstn-1270,60,tr) exten => _7NXXXXXX,n,Hangup() When I try to

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:

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

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-${ARG1},1) exten => s-ANSWER,1,SetGlobalVar(NUM${ARG2} = "ACTIVE")

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:

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

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 defined. I understand

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_01-0000005b", "DIALSTATUS=ANSWER") in new stack

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 not answered by the callee