similar to: gotoif syntax error

Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) Which should allow me to listen to conversations that hit the first (Set

does asterisk support imaps for voicemail storage? Ide really like to use gmail as my imap server... -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080421/00ed120c/attachment.htm

Hello. I'm trying to use gmail's imap feature w/ asterisk imap voicemail. I compiled c-client with the following settings: make lr5 IP6=4 and asterisk with: ./configure --with-imap=/usr/src/imap-2007a/ However if i enable any if the imap settings in voicemail.conf, asterisk starts acting funny and dosent allow any calls imapserver=imap.gmail.com imapport=993 mapfolder=Voicemail Where

Here is a section of my dialplan (macro) exten => s,200,Wait(1) exten => s,201,read(holdopt|screen-onhold|1) exten => s,202,GotoIf($[${holdopt} = 1 ]?4) exten => s,203,GoTo(200) it's simple really it loops telling you the caller is on hold until you press 1 and then it sends you off to another area. The problem right now is that if the read() times out i get these warnings...

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Hi Gang, I'm testing 1.8.0 on one of my machines and this snippet "chokes" on line 7 (works fine with 1.4.30) [tb-account-balance] exten => s,1,Set(BALCOUNT=0) exten => s,n,NoOp(Verbose(acct ${digitacc} pwd ${digitpwd} )) exten => s,n(runagi),Set(TEST_RETURN="NONE") exten =>

Hi, I have the shown (end of this post) net work configuration. In a "few" words: My Debian Sarge server is connected to a D-Link ADSL Router (DSL-562T). DMZ is enabled for the Debian Sarge IP on the Router. My Linux server has two NIC''s. ethlan = internal Net ethdsl = external -> D-Link My Linux server is configured to make NAT via iptables. Current state -

Hello I've tried a bunch of things, but still get errors/warnings when using the IF() function: ============== TEST #1 exten => h,n,Set(WAV_FILE=${IF($[ ${STAT(e,/tmp/${CALLTIME}.wav)} ]?${CALLTIME}.wav)}) [Nov 26 21:52:34] WARNING[5074]: func_logic.c:107 acf_if: Syntax IF(<expr>?[<true>][:<false>]) ============== TEST #2 exten =>

I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears in quotes before the SIP URI. I want to extract the SIP URI from MESSAGE(from) in order to (conditionally) route a failure message back to the source peer. My test dialplan

Hi! I use the following condition: if (${FOOBAR}=YES) { ... } The problem is, that if FOOBAR is not defined at all Asterisk generates a warning: WARNING[11982]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: =YES Of course I could use the following code, but this bloats up the code: if (${EXISTS(${FOOBAR})}) {

I get an error when I execute this code exten => rejected,n,Hangup($[-1*${Z}]) May 2 13:42:09] WARNING[23128]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: -1* The variable "Z" has a negative number, which is the code that I need to use in the

Folks, I'm trying to implement a simple loop in a dialplan. The object is to set a counter, run through some IVR options, increment the counter, return to the start, then finally fall through to an operator or voicemail. Am using 1.4.10 and have reviewed doc/ exten => s,1,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=20) exten => s,n,Set(loop = 0) exten =>

Hello! I use tincd to interconnect 3 LANs: A, B and C. So long, it works fine: everybody reaches everybody. But I want a different behavior: A and B should be allowed to talk, as should B and C. I tried to simply delete the host-files on the nodes that should not be allowed to talk to eachother: A has a hostfile from B B has a hostfile from A and C C has a hostfile from B But this is no

Hi all, Maybe I'm being thick here, but I've had a look through the mailing list and the Wiki, and I can't seem to see details of anybody else with this problem.... With the following line: exten => s,1,HasNewVoicemail(201) I am getting the following error: -- Executing HasNewVoicemail("SIP/201-2f1e", "201") in new stack Sep 2 12:41:09

Hi, all. I'm making my first foray into AEL. I'm starting with a simple macro, but I've already encountered an odd behaviour. I'm wondering if someone can shed some insight: Asterisk 1.2.9.1 macro newPlaceCallPSTN { s => { TIMEOUT(absolute)=7200; NoOp(Analog Out List: ${ANALOGOUT}); ChanIsAvail(${ANALOGOUT}); NoOp(Available Out List:

Hi list, on asterisk 13 I use same => n,Set(__myCpt=$[${myCpt} + 1]) which is working well. On an Asterisk 16 I get, for this same command [2019-10-01 16:15:01] WARNING[28197][C-00000008]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '+', expecting $end; Input: + 1 ^ What changes in 16 version creates this behavior and how to

Hello list, getting warning : *syntax error, unexpected '<token>'* dialplan : exten => pbx,n,Macro(CheckNetworkProblems,${custID}) exten => pbx,n,NoOp(status = ${STATUS}) exten => pbx,n,GoToIf($["${STATUS}"="congestion"]?backup:nocongestion) CLI : [Sep 9 12:27:07] -- Executing [pbx at cust:15] NoOp("SIP/test13-0000002a",

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

I've some code to manipulate incoming Caller-ID - so its suitable for replying to... [sipdef] exten => s,1,NoOp(FWD SIP: "${CALLERIDNAME}" <${CALLERIDNUM}>) ; Alter incoming calles from pulver - add a '87' exten => s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4) exten => s,3,SetCIDName(87${CALLERIDNUM}) exten => s,4,SetCIDNum(87${CALLERIDNUM}) exten

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway