similar to: DIALSTATUS not set

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

Just throwing this out here, hopefully someone can tell me why. *CLI> 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

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

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

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

Hi All! Let me explain the problem. When using the Originate? command from the manager api, the dialstatus variable returns results? for whichever phone picks up first, and in this case it is the IAX/2? connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,? or an extension either. What I am ultimately trying to do is get the? dialstatus of the Zap/X/XXXXXXX channel, i.e.,

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 running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without

--> Perhaps someone can share how? First you need to give them the option of turning the feature on and off. I do it with the following: [callback-activate] ; *********************************************** ; Callback activate/deactivate. If this function ; is enabled and there is a call file in the form ; of ${EXTEN}.call, then Asterisk will call the ; phone number contained within the

Hi all, I am from India and has been recently using asterisk for testing and enahncing my telephony knowledge. I am trying to use the originate Command from the Asterisk manager on both SIP and ZAP. The command works successfully but does not return any DIALSTATUS such as BUSY,ANSWER,NOANSWER as in case of command DIAL when used from the dial plan. Can some one guide me how to get the vaue of

I've seen several examples that use extensions such as; s-BUSY s-NOANSWER etc... It's more or less evident what they do, but I've searched for some FORMAL documentation everywhere and have found nothing. Do they work for anything else than "s-"? (I think I've seen other examples, but can't find them now) Are they standard in any way? What are the allowed values

Hello all, I am working on a creating some intelligent failover dial-plan logic and I'm running into something that I'd like some feedback on. Basically, it appears that if you place a call to an IAX2 peer that refuses the connection, or is unavailable, a NOANSWER dialstatus is returned. Example: -- Executing Macro("IAX2/cubix-19",

Phone - TDM430P - home* - IAX2 - office* - PRI - Telco I dial a busy number from the Phone. Home* shows this in the CLI: -- Executing Macro("Zap/1-1", "dial-wu|2922004") in new stack -- Executing Dial("Zap/1-1", "IAX2/andrew@wu-ast/2922004||g") in new stack -- Called andrew@wu-ast/2922004 -- Call accepted by wu-ast (format gsm) --

Hi all, Hopefully, I just need a second set of eyes on this one, but I just can't figure out what I'm doing wrong. I'm using an agi script to dial a number, check the dial result, and act accordingly. The problem is that I'm not getting anything back from DIALSTATUS, or HANGUPCAUSE. Here is the relevant perl code: ===============================================================

Hello Everyone. In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other routes if the chosen route rejects the call. Now, My current scenario is if I get "BUSY" back from the first provider, I send a busy back to my customer. If I get something like CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt the call. This works

Hi, I was trying to use 2 of asterisk servers and interconnected, one of them as a peer to other sever (configured in sip.conf), so all the calls to server 1 will just be passed to server 2 (has PRI Card, TE 412P, only one PRI connected), i was sending calls to server 1 and that would send to server 2 and then dial out using Dahdi, but the problem that i got was the hangup cause codes, i was not

On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit <salah.elharit200 at gmail.com> wrote: > hello list, > > i have asterisk 11.15.0 and i have some trunks sip from my provider > > we have some ip phone astra 6731i > > each Ip-phone is configured with trunk and we call > > no ihave configured another trunk from the same provider in my asterisk > > i can call

Hi List, we're calling a sip phone from our Asterisk Server, and try to add logic depending on the dialstatus Stripped down example; exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =