similar to: Perform Action after X invalid tries

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

Using an example provided by "The Hitchhiker's Guide to Asterisk", I made the following addition to my extensions.conf file: [inbound-analog] exten => s,1,Wait(1) exten => s,2,SetVar(counter=0) exten => s,3,Answer() exten => s,4,Wait(1) exten => s,5,DigitTimeout(15) exten => s,6,ResponseTimeout(10)

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Hi all, Did anyone manage to make the GotoIf command work with regular expression ? I would like to make the following thing: ${DTMSeq} contains a menu choice. Only 1, 2 and 3 are allowed menu choices. If DTMFSeq contains 1 or 2 or 3 => OK, Goto 4 else Goto 2 I've tried the follwoing syntax but it is alway going to 2 whatever the value of DTMFSeq: exten =>

Hello All, I am trying to set up faxing using Asterisk@home 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing

Quick question...Im authenticate all exten except this one(2006). If I call from ext 2006 I still have to authenticate. If I call form any other ext I have to authenticate. Any suggestions? Thanks extex => s,1,GotoIf($[${EXTEN} = "2006"]?3) exten => s,2,Authenticate(731) exten => s,3,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) exten =>

This is a followup to the posting earlier about Hunt Groups with Call Screening. I have implemented the following macro and for some reason the Macro does not exit and continue the context it was called from when the called party rejects the call - It always drops through to the NoOp at the end and connects the call. Below are two examples of the dial commands I am using to call the macro.

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these extensions.conf lines: exten => s,1,SetVar(SET_EMERG_FLAG=0) exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK}) exten => s,n,SetGlobalVar(EMERGENCY=1) exten => s,n,SetVar(SET_EMERG_FLAG=1) exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM}) exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY} =

I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink T1 ---- Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

When I make a call out on the Faktortel number I am then able to transfer to call to my asterisk meetme room of 801 by hitting 'transfer' then '801' then 'send' on my grandstream phone. This connects my faktortel trunk (and who ever is on the other end) to my conference room I can then make another call using my local pstn service and set up a 3 way (or whatever number

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, Dave sip.conf -------- [general] port =

Greetings, Below is part of the contents of my extensions.conf file. exten => s,1,Wait,1 ; Wait a second before answering. exten => s,2,Answer exten => s,3,ResponseTimeout,10 ; Set the amount of time the user ; has to make a selection. exten => s,4,DigitTimeout,5

Hi, I am having difficulty getting my record phone call dial-plan script working. I have tried the example record call scripts but they start recording before they are actually connected to an end point, e.g. you get ringing or announcements being recorded. It seems to me that these are bugs with the Dial() command: 1) Using M(x) in a dial command does not allow argument to be passed. Using

Hello, i've recently switched over from 1.0.9 to 1.2.3. I've experienced some (to me) weird behaviour. This is the config for an example queue.conf: [654] wrapuptime=30 timeout=20 strategy=ringall retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=