similar to: Setting up calls through the manager interface

I have data in the form of a column such as 1234. 2345. 3435. 4343. I want to have the data in this form ..i.e to remove the "dot" at the end of each number above. 1234 2345 3435 4343 I am trying to use split but it is not working. Any suggestions? [[alternative HTML version deleted]]

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

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 =>

Hi all, How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command : Channel = SIP/0041435215301 Context = default Exten = 00982166501553 Priority = 1 CallerID = 0041435215301 this works with out any problems I initiate a call from one of my

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

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:

No Jeremy, excuse me, the error was in my email. The correct command is /bin/echo "Channel: Local/$1@chiamamezzi-dialout";\ /bin/echo "Variable: callid=123456|number=$1|url=pippo|menuid=FOP|redirectnum=0554202880";\ /bin/echo "Context: chiamamezzi-Wave";\ /bin/echo "Exten: s";\ /bin/echo "Priority: 1";\ /bin/echo "Callerid: Asterisk Automatic

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

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

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

Here's a segment of my dialplan, I'm working on the freenum/ISN functionality: same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) same => n,GotoIf($["${isnresult}" != ""]?:fn-CONGESTION,1) ; set up our outgoing call state same => n,Set(SIPFROMUSER=${CALLERID(num)}) same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" ==

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

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 =>

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

I have been trying to get a DIALSTATUS output from a call started with originate. I searched a fair bit and have found several references to using local channels to do this. However, I could not find enough of the specifics to get it working myself. What I need to do is dial a zap channel and run various scripts if the channel is answered, busy, no-answer,etc. Here is the dial plan I am

I have recently noticed that the "Action: Originate" options in asterisk 1.0 CVS has changed sometime between 2/23 and 3/18. I have a 2/23/04 CVS installation (cvs checkout -r v1-0_stable asterisk ) that allows me to make calls like this using the Manager Interface on port 5038. action: login login: admin secret: mypass action: originate exten: 200

Hi, I'm currently trying to originate a call with 2 variables set. I tried doing it via manager API and call File and both failed, because the vars were not separated. I'm using Asterisk 1.2_beta1 on this machine Can anyone here verify wether this is a bug or just a stupid error on my part? This is the callfile I tried to use, after the manager way failed: Channel:

I have the following in my dialplan, counts the number of loops and when it hits greater then 5, exit. It works, but errors initially with, "syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or tolken; Input: +1". Could somebody tell me why? Thanks: ; **************************************** ; Setup a varriable to count the number of ; times the message has been

Hi All, I just noticed something interesting. When a sip device registers and regcontext is setup in sip.conf, a NoOp priority 1 extension is dynamically created in the dialplan within the specified regcontext. Works great. If for some reason, modification is made to the extension.conf and a >reload extension is performed, those dynamically created extensions in the regcontext vanish. Now