asterisk

问题 I am using asterisk 11.9.0 and i want to generate an outgoing call.I found that for outgoing i have to make a .call file and place it in a var/spool/asterisk/outgoing.I am following the link below http://the-asterisk-book.com/1.6/call-file.html#call-file-parameter my code is same as given in the above link,the above example uses only single fixed number to call. My problem is that i have to generate an outgoing to a number fetched from database(outgoing to new number everytime),so how to

问题 I have a server with Debian wheezy x64, I have problem with asterisk server "Try increasing max file descriptors with ulimit -n", I try to change the file descriptor limit as follows: # su - asterisk $ ulimit -Hn 4096 $ ulimit -Sn 1024 $ exit # vi /etc/security/limits.conf I added in the end of the file: .... asterisk soft nofile 65535 asterisk hard nofile 65535 # End of file And when I try to test: # su - asterisk $ ulimit -Hn 4096 $ ulimit -Sn 1024 $ am I miss somethings? (I rebooted the

问题 I have installed asterisk latest package and using mysql DB for realtime simulation. But in the CLI i am getting the error: config.c:2444 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available and while executing command MODULE LOAD RES_CONFIG_MYSQL i am getting the following message. loader.c:918 load_resource: Module 'RES_CONFIG_MYSQL' could not be loaded. My machine is with linux mint 16. 64 bit. 回答1: If you have version <1.4 you have download

问题 This was working fine in Asterisk 1.8.32.3 - I'm testing with Asterisk 13.22.0 on Centos 7 running as root (already - as you'll see below): same=>n,System(/usr/src/bash/setData.sh ${CHANNEL(accountcode)}) The script's permissions: [root@localhost bash]# ls -l -rwxr-xr-x 1 root root 1493 Jan 20 2016 setData.sh [root@localhost bash]# pwd /usr/src/bash [root@localhost bash]# I've upgraded to Asterisk 13, and Asterisk is now incapable of executing the setData.sh script: [Jul 25 10:33:21] WARNING

问题 I have two sip extensions: 200 and 300 and a queue, let's call it my_queue. Extension 200 is talking to extension 300 and decides to transfer extension 300 to my_queue. So extension 200 puts extension 300 on hold and dials the queue number in the dialplan that eventually will call ... same => n,answer() same => n,queue(my_queue,,,,${TIMEOUT},,,) ... Now, at this point extension 200 is listening to the queue's MOH and 300 is listening to the call in progress (as expected). When 200 hangs up

问题 I am writing an IVR menu and I need to allow my users to press 0 anytime during the communication to exit. The following is how I do it: exten => 0,1,Playback(good-bye) exten => 0,2,Playback(beep) exten => 0,3,Hangup However, by doing so, when the user presses zero while some file is being played back or some other operation is taking place, he/she cannot exit, it is like if he/she didn't press zero. I hope I am clear enough and that you can help me out with this. cheers 回答1: For an IVR, you

问题 Sorry in advance if my question makes no sense to you. I am newbie in asterisk, and what I am trying to do is writing a dial plan which can connects 2 soft phone end point (VoIP client end points) and then try to detect silence in ongoing call. I am able to make through call by using following dial plan exten = 100, 1, Answer() same = 100, n, Monitor() same = 100, n, Dial(SIP/client1,15) when I dialed 100, it makes call to client1, which I received gracefully and now call is on going, now I

问题 I am working project which written in java & handle asterisk. By java program i select station numbers like max no 90 or 900 or 9000. For adding in asterisk i handle 3 files sip.conf queues.conf extensions.conf In extensions.conf i done following setting specifically for adding stations quick. extenpatternmatchnew=yes By this i can enter 90 stations within 20-30sec in extensions.conf 900 stations within 8-9 min & 9000 withing 5 hours But i got info that we can include as many conf files as we

问题 This is not a programming question per se. I am trying to build a system which consists of the following: User calls system using regular land line Some processing is done by asterisk Call is forwarded to an external number (another landline/mobile phone) Now I would like atleast 2 simultaneous lines on which the user can call. I would like to know the following: Will the TDM410 work for what I am trying to achieve? Since I want call forwarding, do I need an extra line for that? Or can I do

问题 I have followed the instructions in this thread: Asterisk AMI - pickup call. However, I am still unable to answer calls via AMI. I can make the call to the extension, but corresponding phone for that extension doesn't ring. I can then run the AMI command to answer that call, it does answer, but obviously there isn't any actual response. Dialplan (testing with extension 116): exten => 116,1,AGI(agi:async) Any ideas what I am doing wrong here? 回答1: Use exten => 116,1,Answer exten => 116,2,AGI