similar to: application call to Gosub affects flow of control, and needs to be re-written using AEL

How to implement some of the function into asterisk like *67 "call number blocking" or *57 "call trace" ? I'm connecting to sipura SPA3K outside line by dialing 9+number. Is there a way to get outside dial tone in SPA3K PSTN-Line by dialing "9"? How to program the extension? -- #Joseph

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

Ladies, Gentlemen We are experiencing an unusual problem in our asterisk 1.4.34.. We are attempting to determine if channels are in use before paging to them. This works correctly, as in it pages the phone.. however, we see the error message below on the console... after googling, we discovered limited information regarding the issue... -- Executing [NPANXX7298 at from-pstn:1]

Hello all. This is my very first message to the list. I have a TDM400P card, It has 2 FXO channels which are connected to extensions of my PBX (Ericsson BP250), so I can dial from any SIP softphone directly to physical (analog and digital) extensions on my company. My PBX is configured so when I dial 8 on any extension, it will redirect to the first free FXO channel on my TDM400P card.

Is there a way to differentiate between a SIP address which hasn't registered (but is within sip.conf) and one that's not there at all (i.e. not in sip.conf) using a straight dialplan. I'd like to differentiate actions depending the state of a SIP device and whether it's in my config or not (if that makes sense, basic automap of dial-in lines to sip phones, but if they've

Hi list, We're trying to set up a phone number that customers can call to get a hold of anyone of a group of sysadmins (and not their voice mails!). We found the findme example ([1]) that makes the callees press 1 to accept the call. It almost works, but it doesn't work correctly when one of the callees, the sysadmins, hangs up after accepting the call. We're using this

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the

I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and then decide to blind transfer them using ## my side of the call is not hung up. Instead it sends me to voicemail. If somebody calls me and then I blind transfer them with ## I am hung up on as expected. I called from 8678 to 28688. I then transferred the call to 8532. Asterisk acts like it wants to hang up, but then

Hello everybody, i have two gsm line (extra channels) and i'd like to schedule the outgoing calls with a round-robin strategy. If all the gsm lines are busy, the call must be sent to the pri lines with a linear strategy. here is the dialplan: exten => gsm,ChanIsAvail(EXTRA/r2&DAHDI/g1) same => n,GotoIf($["${AVAILORIGCHAN}" = ""]?unavail,1) same =>

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??

Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at

Hi list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten

Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered Zap/8-1 Oct 22 11:20:23 VERBOSE[29983]

Hi, I have recently run into the problem with macro implementation in AEL in Asterisk 1.6. I have some older AEL dialplan which runs on 1.4 but it does not on 1.6 and I'm not sure how to solve this correctly. Let me explain... For example, in Asterisk 1.4 I have a macro like this: macro read_digits(digits) { Set(playlist=${SHELL(${PYTHON} ${SCRIPTS}/read_digits.py

How does one go about accessing gosub arguments from Asterisk in extensions.lua? For example, I have the following in extensions.conf: exten => 1000,1,Wait(1) exten => 1000,n,Gosub(functions,mytest,1("123")) exten => 1000,n,Hangup And then the following in extensions.lua: extensions = { functions = { ["mytest"] = function()

It appears that there's no way to get the return value from a GOSUB into an AGI script. Is that correct?

Hello, When reloading a diaplan (asterisk 1.6.1.X), I can see in console : [Apr 2 09:02:00] WARNING[2217]: ael/pval.c:2522 check_pval_item: Warning: file /etc/asterisk/extensions.ael, line 621-621: application call to Gosub affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! What is then the recommended substitution for Gosub() application

Hello, I'm using AEL2 (in Asterisk 1.6.1.6) and I can't find a way to successfully come back into my dialplan. I've tried things like this (in features.conf) : toto => #9,peer,Goto,mylocal2,s,1 But typing #9 (from channel SIP/7275, in example bellow) I've got: -- Feature Found: toto exten: toto -- Started music on hold, class 'default', on SIP/7275-08b7fbe0

Hi - I have a Macro that contains a GoTo. The documentation indicates: If you GoTo out of the Macro context, the Macro will terminate and control will return at the location refered to by the Goto. I thought I might convert the Macro to a GoSub routine, but the documentation doesn't mention what happens if you GoTo out. It does however mention that the return address gets pushed onto the