similar to: manger hangup call

I am running asterisk 1.4.18 trying to use MeetMe and option b. I am getting permissions denied failed to execute conf-background.agi on the CLI lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi -> /home/silentm/bin/conf-background.agi my conf background is a symbolic link - then my permissions are : [root at devcentos5x64 src]# ls -l /home/silentm/bin/conf-background.agi

Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry

All; I wrote a PERL AGI script that prompts a caller to leave a message using print "RECORD FILE $recordfile wav # 60000 BEEP s=3\n"; When the caller is done, they need to press the # key. The message is then delivered. However, the message is not delivered if the caller simply hangs up when finished. If the user hangs up, the script ends right then. How do I keep on processing the

>Are you saying that this worked in earlier versions but you started to >get the delay when you updated to 11.21.0? Or just that you happened to >be using 11.21.0 the first time you tried this scenario? I should have said "first time" trying this. Any thoughts? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am reading comments on the Wiki for meetme http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe from 2004 about how and AGI does work with non zap channels. Is this still valid 3 years later and 1.4.4? How do I bring people into a meetme and play a message to all of them when they are on SIP channels? Jerry

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

Is there anyway a T1/PRI can ignore the ANSWERED signal and just go straight from a dial command to the call was answered? I have a PBX that when calling a certain analog trunk it is not giving me signaling that the call was answered however I hear the PA system come off hook and give dial tone. So is there something in the dial command that can override the signaling and just go straight to

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

Hi All, I'm running an AGI, calling a perl script the does number lookups to a remote server. I would like to put a timeout in the script. The problem I'm running into is if the DNS server is not responding, the script hangs and waits for 30 seconds before returning to the Asterisk dialplan. I would like a timeout of 1 second, then return. Here is my clean script:

I am using the AMI interface to start calls. At one point I have a 10 second delay "Wait(10)" in the dialplan... During this time it "seems" that if I then connect with the manager interface and place a call that nothing happens till the 10 seconds is done... I am requesting Async yes... manager_str Action: Originate[CR ][LF ]Async: Yes[CR ][LF ]Channel: SIP/430[CR ][LF]

Hi, sorry if this is a newbie question, but in fact I am sort of a newbie. Is there a way of connecting two answered and active voice channels together in an AGI script for some time, having the two parties talk to each other, at the same time have asterisk or the AGI script listen for DTMF tones on both channels and react to certain tones, i.e. disconnecting the two channels on reception of

Dear friends, I am having problem with running a sample php and I can't figure out why. I can run the sample.php using CLI but when I run it inside the dialplan it does not work. Can someone please suggest the config problem that I may have made? dommy:/var/lib/asterisk/agi-bin# php sample.php #!/usr/bin/php5 -q VERBOSE "Here we go!" 2 VERBOSE "Call from - Calling

Dear, I am getting this in CLI on release candidate version of Asterisk. Any ideas, or points where to look? -- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi [Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe -- <SIP/916-fc001968>AGI Script rad-auth.agi completed, returning 0 Best regards, Josip

I would like to run an agi script (to calculate the cost of a long distance or international call) right after I execute a Dial app. Can this be configured in extensions.conf? It seems the entries right after a Dial app get executed only if the Dial app was executed unsucessfully. Would I have to execute the dial app from the agi script? Thanks in advance. Dan -------------- next part

Is there any way to see the number of AGI processes that Asterisk is handling? Either console, Asterisk Manager, or from within the AGI? I used to just count the number of running copies of my AGI process (ps aux | grep agi) but once in a blue moon one of my AGI processes will become a zombie or for some other reason not stop when Asterisk disconnects from it. I'd like to know, from

Good afternoon list. I'm trying to use ${AGISTATUS} after the execution of my script in PHP Agi. But after running the script, it just returns me 0 (true). Thus: -- <SIP/213-00000019>AGI Script check.agi completed, returning 0 I tried putting the lines "return false;" or "return 1;" but did not change anything. Does anyone have a clue? Thanks, Rodrigo Lang.

I believe I am running an AGI (to put users in a conf) before the confbridge is built. So the users are not really get in the conf... exten X,1,run agi to put users in conf exten X,n,ConfBridge() How do I have in the dial plan ConfBridge() and someplace run an AGI that brings the users I want into that Conf. I cannot delay in the AGI and wait for the conf because the conf is not built until I

How can extensions.conf be changed to work with both Asterisk 11 and 1.4.X such that 1.4.X calls deadagi and 11 just calls agi as deadagi is no more. Thanks, jerry

Hi everybody, I'm trying to implement a enhanced blacklist system using AGI and Perl, configuration in extension.conf is: exten =>_numbera,1,AGI,blacklist_2_in.agi exten =>_numbera,2,Answer exten =>_numbera,3,AGI,xisco_1.agi exten =>_numbera,4,AGI,blacklist_2_out.agi The problem that I have now, is that blacklist_2_out.agi doesn't execute. I think this is because in