similar to: Current possible values for DIALSTATUS?

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

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

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")

Hi, I am using a call file formated like this: Channel: local/12125557891 at outbound/n Callerid: 12125551212 Context: detect Extension: s Priority: 1 This sends the call into the dialplan at the [outbound] context. In [outbound], I have: [outbound] exten => _1.,1,Dial(SIP/${EXTEN}@flowroute,43) If the call is answered, it move on to the [detect] context. When using this method, it appears

Hi I was interested to see how Mongrel uses Lex/Yacc to parse the HTTP requests using a Regal generated parser. I downloaded the source but do not see the lex and yacc files...

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

Hi All, I am relatively new to Asterisk. I have CDR enabled and successfully writing to MS SQL server. In my cdr table I am setting the userfield value with a line in my dialplan. If a call is placed to an invalid number (e.g. 12125551212), I see a cdr record created, however, my userfield value never gets set since the call never made it into the context of my dialplan. I am using AMI with the

Hi, I am attempting to implement Answering Machine Detect and have also played with using BackgroundDetect instead. Does anyone recommend one over the other? Here is the code I am using for the BackgroundDetect method (from voip-info.org). Thanks. [detect] exten => s,1,Set(MACHINE=0) exten => s,2,Answer exten => s,3,BackgroundDetect(silence/5, 1000, 50) exten =>

Hi, I just deployed new Aastra 9480i phones and when I attempt enter digits on other systems, like host pin in a GoToMeeting, the servers on the other end do not get my entries. I am assuming this is a DTMF issue but do not see anything in this phones config other than turning on the display of the digits. I have the DTMF method set to "SIP INFO". I am using AsteriskNow w/FreePBX.

Hi, I am trying to implement call forwarding on the event that my ATA was not reachable to Asterisk, whether due to registration timeout, network interruptions between the ATA and Asterisk, or simply because the network on which the ATA resides in unreachable for any reason. I there a way of implementing such a feature in Asterisk? I have implemented CF unconditional, and CF on busy, CF on

Hi All, My client makes manual sales calls to prospects. He is often sent to voicemail on the prospect's side. If he finds himself having to leave a message, he would like to be able to press a key and let a pre-recorded message play into the prospect's vmail box. This is so he can maintain consistency in his message. Can anyone offer suggestions of how I could accomplish this

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the

I just want to confirm but it seems that if-then-else is not permitted in case structure. It was not really documented but it seems to be the case. Can anyone confirm? switch(${DIALSTATUS}) { case NOANSWER: { // if-then-else not permitted If (${ael-var} = 1) { Playback(beep);

Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken,

Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface

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

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.

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 =>

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