similar to: ast_yyerror - 'space' in Caller-ID - string comparison

Hey all, We're starting to see "all circuits are busy" and a few dropped calls. When these happen, in the messages log, I see the following error. May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 0?7: What causes this?

If I want to send outgoing DTMF over a BRI interface, can I do it with 'isdn4linux' or must I use the 'capi' library? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ mje@posix.co.za - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 -------------- next part -------------- A non-text attachment was

I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys.... Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has

I have a Grandstream GSX-2000 with .. Software Version: Program-- 1.0.0.3 Bootloader-- 1.0.0.3 I tried to do an HTTP update from the Grand Stream web site... After half an hour, I recycled power and now its dead... LED's come on and stay on, screen and buttons are dead. Connectivity to Grandstream.com was always good - whenever I checked (I downloaded the "User Manual" in a

Playing with time ranges - using the examples found in one of the asterisk cook books... (pdf - page 17) ; After Hours include => night_menu|00:00-08:00|Tue-Fri|*|* include => night_menu|17:00-24:00|Mon-Thu|*|* this gives... ... pbx.c:2962 get_timerange: 24:00 isn't a valid end time.... -- Including context 'night_menu|17:00-24:00|Mon-Thu|*|*' in context 'default'

I'm sure I saw a posting about someone updating the CVS with a more richly featured voicemail system. What happened? Am I wrong? Can't seem to find anything on this... -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ mje@posix.co.za - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 -------------- next part

I have a 'day' and a 'night' mode. In the day mode, I play a 'background' message which is interruptable by the pushing of a DTMF key - ie - all is normal. In night mode - I decided to get smarter... I play two backgrounds with a 'sayunixtime' in between and now DTMF does nothing - the menu times out to my 'lets get the operator then'... If I change the

I'm using a SNOM 360 with Ver 4.3 software. Asterisk is.... Asterisk CVS-D2005.05.02.22.00.00-05/04/05 (BRI Stuff + Head) I've used the wiki info to set up some lines to monitor some internal extensions. When the extension is rung - the lamp comes on, when the call is answered, the lamp goes off.. I was expecting something a little more exciting - like the lamp to flash when the

Howdy, I recently obtained a Dell Axim X30, running Windows Mobile 2003 SE. It totally refuses to connect to a Samba share due to authentication errors.The share is set up with clear text passwords and functions perfectly with a Win98 SE box. I have tried encrypted passwords and nt status support = no in the global section of my smb.conf. I am running Debian unstable, Samba 3.0.7-1.

What I've got... Software: Linux: Slackware 9.1 Asterisk: out of CVS - so its new. isdn4k-utils: to test the ISDN Card Hardware: PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM 1 x ISDN BRI Card - DIVA EICON (Installed + working) 2 x Grandstream (Barbie?) BT100 SIP Phones. What Works.. I can call from one phone to the other... get read voicemail... I can

I've got asterisk able to make and receive calls via the Internet via E164 lookups. If I get such a call - I'd like to display the original phone number on my phone. In the log is the following - which displayed '601' on my phone. The caller was +886288097680 - am I getting the wrong ClID because of my end or the caller end?

I've got two patterns I want to match on making an outgoing call... (one day - to do Least Cost Routing for Cell/Mobile calls) Firstly - I prefer '0' rather than '9' to get an outside line... Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084) or its just another number to dial... I added the following... the playback just advises me which 'route' is

OK, I'm going nuts here trying to correctly identify null values, specifically when callerID info is not available. FYI, I'm running Asterisk CVS-HEAD-08/17/04-13:08:53, and Bison 1.875a (debian Sid). A snippit of my dialplan looks like this: exten => s,1,SetCIDNum(${CALLERIDNUM}) exten => s,2,NoOp,${CALLERID} exten => s,3,DBGet(temp=idiot/${CALLERIDNUM}) ; Is the person calling

Hi All, I got this error: [Jan 18 09:56:58] WARNING[9617]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: 1 > ^ exten => s,3,GotoIf($[${GROUP_COUNT(${ARG1})} > ${LINELIMIT}]?101) exten => s,3,GotoIf($[${GROUP_COUNT(${TRUNKGROUP})} >

Hello people: First, excuse any misspelling, english is not my mother tongue I'm triying a simple gpo, put a fixed image as wallpaper on all the pc connected to the dc but is not working. Using the remote administration tool had has defined an Organizational Unit with one user and defined a policy with two restrictions 1) use a shared image [ \\dc\common\backgroud.jpg ] as wallpaper 2)

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 want to authenticate a user before he is able to use the phone. I also want to set his privilege as to where he is allowed to call to... Preferably, the password should be their VoiceMail password, (every extension (or is that user?) can have voicemail defined - even if its not in use?) ...one should be able to enter the password (variable length) as part of the dial sequence - eg the number

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

-- Executing DateTime("SIP/phone1-07ff", "") in new stack -- Playing '/var/lib/asterisk/sounds/digits/day-1' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/mon-3' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/h-19' (language 'en') This works - the pathname is complete - Joy.

I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a call with '*8' - the call will drop after about 20 or so seconds. Is this a general problem with Asterisk 1.0.2? As this is the latest release that it appears Klaus-Peter Junghanns has for public consumption - is there anything I can patch for just this problem - or has Klaus-Peter Junghanns (or anyone else) been