similar to: Dial plan errors

I'm having the same problem here. Any solution to this problem? -Manuel (sorry for top-posting, I'm having a stupid mail client here) -----Messaggio originale----- Da: Simon Brown [mailto:Simon.Brown@otterson.com.au] Inviato: giovedì, 1. luglio 2004 02:05 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Dial plan errors I am attempting to implement the new features added

I am attempting to implement the new features added recently where you can have "Goto(s-DIALSTATUS)" in the dial plan. My extensions.conf looks like this: exten => s,1,Dial(${ARG2},20,r) exten => s,2,Goto(s-${DIALSTATUS}) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-CHANUNAVAIL,1,Voicemail(b${ARG1}) exten => s-BUSY,1,Voicemail(b${ARG1}) exten =>

My dial plan seems to work great - in that when I call extensions 1234 it connects to 1234. Strangely, after the call terminates (the other side hangs up first), Asterisk continues in the same context and then matches to extensions _. which causes an invalid extension error! Why does asterisk not leave the context (called internalmenu) after the remote hangup? Instead, it continues to the

Hi, I have an odd problem when doing a blind transfer. The transfer is intiated and the transferred caller hears nothing until the timeout. I have tried setting the 'r' and the 'm' variables in the dial command. Nothing happens when I use the 'r' variable when I use the 'm' variable I briefly hear music on hold and then it stops until the timeout for no answer

I am using the sample (slightly modified) macro for dialing phones. My extensions are in the 2000 range. Each extension has it's own external DID. Is there any way to have the macro look up the DID to be used for the extension and set the DID as the callerid? Maybe from a flat file somewhere? Or is there a better way to do this??? I know I can use callerid in sip.conf, but I only want the

Hi, This problem may be related to a configuration problem but I believe it is a bug in the FOP since restarting the FOP server clears the problem. Here is the scenario: Using AgentCallBackLogin and have four agents logged in a call is made to one of the agents directly from an internal phone. Okay so far. Call is hung up and the same extension is used to call another agent okay again, no

In the following macro, a call is dialed and control branches according to DIALSTATUS, much like the default std-exten macro. What I'm trying to figure out is how to regain control when the call is answered. ; Standard extension logic [macro-stdexten] ; ${ARG1}=Extension ${ARG2}=Device(s) to ring exten => s,1,NoOp(stdexten ${EXTEN}) exten =>

I'm trying to set up my extensions.conf file using some of the existing macros like stdexten, etc. while at the same time having two logically separate virtual PBX's (with no "default" context) and two trunks coming into separate contexts, i.e. one for residence and one for my at-home business. I noticed, however, that macro-stdexten depends on the "default" context:

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5 exten =>

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",

Dear All, I have a problem with call transfer. When I dial a number and then I want to transfer current call to an extension, I'm getting disconnected. Transfering incoming call works fine. I'm using macro for dialing. extensions.conf: [from-internal] ignorepat => 9 exten => 200,1,Macro(stdexten,200,SIP/dzalewski) [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})})

I've been banging my head on a brick wall for about an hour now trying to understand why the following doesn't work (which is even provided as an example in the distribution!). The goal is to create a voicemail-only extension not associated with a phone. I'd rather not have an extension dedicated to VoicemailMain(), so I would like the user to be able to hit '*' during

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

Dear All I want to integrate sugarcrm and asterisk , so when customer call the call center the agent or extension which answers the call , before pickup the phone and talk to customer , view his/her information if it is available. I do this as described below : 1-Setup login username for sugarcrm for each extension 2-Extension Users will login to the sugarcrm. 3-Develop php script to handle new

I have a pretty standard setup with Asterisk acting as a PABX for a bunch of SIP handsets (in this case, SwissVoice IP10S). My users are complaining that when they forward their phones to their cellphones on unavailable (i.e. forward when no-answer), their cellphone only rings once or twice, and then Asterisk sends the call through to Voicemail. I'm using the standard extension Macro

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hi, Has anyone managed to create a setup with a Sipura SPA-1001 as a client? Right now I can connect to the device by dialing the extension number but when I try to connect from the phone handset to make an outbound call it gives an unavailable tone. I'm using Line 2 on the SPA-1001 to connect to the local asterisk server, line 1 is used to connect to my VOIP provider until I can get the