similar to: GoToIfTime issue

hi, All one thing confused me a long time. when i change the extensions.conf file. why not take effects after restart the asterisk? details as follow: my dailplan is : [95040] exten => _95040XXXXX,1,Set(CALLINNUM=${CALLERID(dnid)}) exten => _95040XXXXX,n(start),Answer exten => _95040XXXXX,n(welcome),Background(${welcomefile},,123) ... exten => i,1,Playback(invalid) exten =>

hi, all i want to use g729 codec for set up a call. so i donwloaded the so file from web site: http://asterisk.hosting.lv/#bin and install it properly. *CLI> *CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin

Hello list, why is it that GoToIfTime thinks a date of **|*|29-*|jun *is not valid ?? [Jun 29 14:06:34] -- Executing [s at macro-vac:10] *GotoIfTime*("SIP/testcorp-00000036", "**|*|29-*|jun*?onvac") in new stack [Jun 29 14:06:34] WARNING[3076]: pbx.c:4127 get_range: Invalid end day '*', assuming none [Jun 29 14:06:34] -- Executing [s at macro-vac:11]

hi, all recently, i face a GotoIfTime problem GotoIfTime("08:00:00-07:00:00,mon-sun,*,*?95040263008,start") as you can see the section is 08:00:00-07:00:00 , which is the begin time is later than the end time what's this refers then? in my test , my system time is 10:57:00, but this check will pass, although i guess i will not. is begin time later than the end time means

Hi, I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the ampersand (&) to combine multiple days / day ranges but this gives me an error. For example mon&wed gives the error (in the asterisk console): [Nov 23 18:04:27]

hi, guys, when i create a manager account used for freepbx, the follow info produce all the time? do you know that's the reason? == Manager 'bitzsk' logged off from 127.0.0.1 == Manager 'bitzsk' logged on from 127.0.0.1 == Manager 'bitzsk' logged off from 127.0.0.1 == Manager 'bitzsk' logged on from 127.0.0.1 == Manager 'bitzsk' logged

hi, list i want to know what is the best OS for install Asterisk 1.6.2.9, which should work properly on working system. i want to use CentOS5.2 or CentOS 5.4. Which is better and stable? Thanks for your help. -- Thanks for your supporting, have a nice day. Sucan

hi , all when i do the command "module load cdr_radius.so" ,error happens. i have installed radiusclient-ng , what's wrong with it? thanks! error message as follow: ZHANGSHUKUN*CLI> module load cdr_radius.so Unable to load module cdr_radius.so Command 'module load cdr_radius.so' failed. [Dec 23 17:55:41] WARNING[31072]: loader.c:380 load_dynamic_module: Error

hello,all there is a AudioCodes Mediant 2000 out there. i want to realise ip to PSTN and PSTN to ip connection. after some configuration on AudioCodes Mediant 2000, PSTN to ip connecttion works. next ,i want to dial from asterisk to PSTN now. i have see the sample in the extensions.conf relevent to PSTN as follow: ; If you are freely delivering calls to the PSTN, list them here ; ;exten =>

hi,list Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ? after i make and make install. i cant find the .so file. is this mean it can't install on 64bit Cent-OS. ps: it works fine on the 32 bit Cent-OS Thanks very much! -- Thanks for your supporting, have a nice day. Sucan

Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several times and ring another? ; A strategy may be specified. Valid strategies include: ; ; ringall - ring all available channels until one answers (default) ; roundrobin - take turns ringing each available interface ;

hi, all after my installation of asterisk and adds-on . when start astrisk, error accours as follow: [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available what's wrong with me ? Thanks. -- Best regards, Sucan

hi, i have installed a2billing , when i open /admin web pages. errors as follow: Fatal error: Call to undefined function bindtextdomain() in /usr/local/src/a2billing/common/lib/languageSettings.php on line 130 do you know what's wrong? -- Thanks, Sucan

hi, Does A2Billing has mial list? -- Thanks, Sucan

Hi,all i want to hear the voicemail recorded, but when hear "if you want to play message , press 3", after i press 3 i only hear that that's the time the file recorded. not the content. do you know how to hear content of voicemail fle? debug message: == Parsing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001.txt': Found -- <SIP/1003-00000058>

hi, group there are two module can used for meeting. MeetMe and Conference(which is a plugin) My question is : which is better for large conference(maybe above 100 people in a meeting)? -- Thanks & Regards Sucan

Hey All, I need to forward an extension to an other depending on the current time but I could not get it done with GotoIfTime. What I'm trying to do is ring on the extension 1 if time is between 8:00AM and 2:00PM and on extension 2 if is between 2:01PM 11:00PM. exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333) exten => 111,2,Dial(${Person1}) exten => 111,3,Dial(Hangup) exten

Does anyone know if gotoiftime can take any subset of 7 days for the days of the week or only a contiguous range? I want to use gotoiftime to change dialplan behavior on Monday, wedneday and Friday -- Executing GotoIfTime("Zap/8-1", "09:00-20:00|MON WED FRI|?21") in new stack Aug 27 19:27:25 WARNING[2676]: pbx.c:3729 get_dow: Invalid day 'MON WED FRI',

Hello there, I have successfully installed and configured asterisk for use as an office PBX using SIP trucks and Voip handsets (using g.729 codec) which works great. Now I wish to try and configure asterisk to do a HTTP request and submit callerID to an external website when a call is missed. eg Someone calls PBX and rings extension 100 -> Call is not answered -> HTTP request is initiated

I want to play a sound file using GotoifTime function. 1) What should be the appropriate format of this type of sound file? 2) Is there any method to copy this file into the destination directory using the browser of a PC other than the asterisk PC (currently i am using cp to copy the file in /var/lib/asterisk/sounds on asterisk PC)??? Waiting for ur kind reply !!