similar to: NO ANSWER before playback or background function?

hi, all in my test,it shows Playback will answer the call automaticly, but i don't want to so. i will use answer function to answer the call. could you help me ? -- Best regards, Sucan

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 =>

Hello, I have an ISDN PRI E1. I want to send an audio before answering, I am using noanswer option in playback app but all the audio is muted before the answer. I would like to play this audio. Regards, ia

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

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, 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

Hi I have this escenario: |SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR, A2Billing, etc... The problem is that I can not hear any audio when call from 'sip or H323 phone' and configure something like: exten => _01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ... It works if I remove the 'noanswer' parameter but in this case it connects

Hi! Our telco routes multiple numbers through PRI to our Asterisk. Not all of these numbers are in use. I have noticed recently that someone keeps calling unused phone number from outside world. I called them and asked why do they call dead number. The person on the far end explained that she keeps calling this number because she hears "busy" tone every time... Most telcos these

hi, i try to convert wav file to gsm format.use following commands; sox net263-welcome.wav -r 8000 -g -c 1 net263-welcome.gsm resample -ql the file is located in /var/lib/asterisk/sounds/net263 but cant' play.do you know what's wrong? -- Executing Playback("SIP/1001-00000091", "net263/net263-welcome") -- <SIP/1001-00000091> Playing

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,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, 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,all i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except realtime queue. it seems queue_table works fine, but queue_member_queue not work, the two tables works fine when in 1.4.28. is that something changed related to realtime queue configuration? more detail about two table definition and data stored in , please see: http://pastebin.com/m33f9539e the extconfig.conf file,

hi, all i want to realize more secure communication between asterisk sip end users. so i want to know Does Asterisk 1.6.2.1 Support SIP TLS encryption? if you can tell me same specific example to do encrypt, it's very appreciated. Thanks! -- Best regards, 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 for a long time, i cant understand the difference between ${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)} i know ${CDR(start)} mean when a call is start. and ${CDR(answer)} means when a call was pick up. but what's ${CDR(calldate)} mean? Could you help me ? Thansk a lot! -- Thanks for your supporting, have a nice day. Sucan

hi,all when i upgrade from 1.4.28 to 1.6.2.1, i can't connect to mysql database anymore, error as follow: [Jan 25 15:38:25] WARNING[3003]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk (check res_mysql.conf) the content of res_mysql.conf is: http://www.pastebin.org/81966 i've try command " mysql -uroot -proot" ,i can connect to

Hi, I would to send a message to an incoming call with no answer. My Asterisk server receive the incoming call by a BRI/SIP gateway (Could be a PRI, for instance). I do the command playback with option noanswer, Asterisk send 183 followed by RTP and finish with 603. But the BRI gateway do not allow to pass the RTP without a 200 OK. The question is: are there a SIP command to indicate the gateway