similar to: Can't do make menuselect?

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, 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, 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 , 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

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,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

Hi, Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such a way that I cannot script non-english sound files downloading anymore. The following used to work (unattended) with 1.6.1.9 (for instance): cd /usr/src/asterisk-${ASTERISK_VERSION} ./configure make menuselect.makeopts echo "MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM" >

Hi All, I'm in the process of writing an install script and I would like to change some settings for the install process but I don't want the user to go into menuselect and make the changes manually. Is there a way to make the changes to menuselect from the CLI? As an example, selecting the iLBC codec. menuselect codec ilbc on Regards David. -------------- next part -------------- An

Hello List, Is it possible to select ODBC_STORAGE without entering to 'menuselect'? I'm currently building a package for my distro with a little script, and would like to set this option without entering manually to 'menuselect' I know that I could make the script to change the 'menuselect.makeopts' var from: MENUSELECT_OPTS_app_voicemail= to:

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>

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, I hope this is not too much of a noob question. I am trying to compile Asterisk and I cannot figure out how to get into the Menuselect menu. I do #make clean #./configure #make menuselect, but I cannot figure out how to actually get into the menu select interface. I am running CentOS 5. I have done quite a bit of searching on Google and have not come up with anything. Also, I am reading the

I'd like to be able to save the choices made in menuselect in a way that they can be tracked in a CM system and applied to a later release of Asterisk using an automated tool like Ansible. What's the best way to do that?

Is it possible (and advisable) to copy menuselect options from Asterisk 11 to Asterisk 12? If so, is menuselect.makeopts the only file to copy? -- Mitch

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 , all what's wrong with this command? exten => 222,1,GoToIfTime(11:00-14:00|mon,wed|*|*?1:3,1) as i got the error: -- Executing [222 at 95040:1] GotoIfTime("SIP/1001-00000099", "11:00-14:00|mon|wed|*|*?1:3|1") in new stack [Jan 20 11:21:11] WARNING[16804]: pbx.c:4118 get_range: Invalid day 'wed', assuming none but what should i do. if i want to set

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