similar to: API hangup command

I have a few commands I wish to run after a hangup. It looks like only the first 2 commands are run after hangup. I am using 1.4.3 How can I get the entire loop to run 10 times. ( I know my example just has noop's but its an example). exten => h,1,Set(i=1) exten => h,n,While($[${i} < 10]) exten => h,n,Noop(jerry) exten => h,n,Set(i=$[${i} + 1]) exten => h,n,EndWhile exten

How do I stop the "you have been kicked by this conference" message from speaking? I first had MeetMe(conf, l) and I get the kicked message. I tried Meetme(CONF, lq) and I still get he kicked message. and it still says it. Thanks, Jerry

I am running a heartbeat... Asterisk 11.15.0 - same behaviour is noticed on 1.4.43 also I issue a call through the API that does the below. just UserEvent and Hangup -- Executing [s at heartbeat:1] UserEvent("Local/s at heartbeat-0000000f;2", "HeartBeat, Noop") in new stack -- Executing [s at heartbeat:2] Hangup("Local/s at heartbeat-0000000f;2",

Hi all, I am trying to setup a three party conference using the Asterisk Manager API. I am using the Redirect action over an established two party call. The procedure I am using is to try to redirect the two existing channels to a third party. I would expect this to connect both channels to the third party. However, one of the two parties gets disconnected. Is this the expected behavior? Is there

Apparently Verizon is blocking or changing packets on port 5060 so my softphone from my hotspot will not work. How do I set asterisk (11.23.0) to run default 5060 for all other devices I have - BUT for my software run on a different port like 5070? I'm using linphone and is easy to change the ports from 5060 to 5070 ( I think). Thanks, Jerry -------------- next part -------------- An HTML

I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in

>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 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

Hi all, I have setup my Cisco 79XX phone. Did the tftp, put the config files in the right location with the right names. Booted my phone, it does the tftp things, the screen shows my extensions everything seems fine. However, when I come offhook and try to dial 11 which is just a playback of demo-congrats in the dialplan the phone says Calling Out (INV) below is my sip.conf file - I presume it

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]

I have added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]:

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am getting a priveldged command error on the manager API. 16-Feb-09 11:51 am asterisk_command() Action: Login 16-Feb-09 11:51 am asterisk_command() Username: XXX 16-Feb-09 11:51 am asterisk_command() Secret: ZZZZ 16-Feb-09 11:51 am asterisk_command() Events: off 16-Feb-09 11:51 am DEBUG: Response: Success[CR ][LF ]Message: Authentication accepted[CR ][LF ][CR ][LF ] 16-Feb-09 11:51 am

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

I am hearing a you have been kicked from the conference message in asterisk 1.2.23. I dont want to hear that. I am using 1qt for the meetme. How can I disable that message? THanks Jerry

How do you format a command line that needs < > and they are not meant to redirect anything they are part of an email address. command -f "Some Email <someemail at somedomain.com>" -x -y -z I tried putting a backslach in front of the < and > but that didnt do it either. Thanks, Jerry

All, I have noticed that placing a call in the outgoing spool during a reload the call may fail. Try the call again after the reload is done and it will complete. This seems like a bug. During a reload calls should be suspended or something? Thoughts? Jerry

In CentOS 6 gconftool-2 (command line) is used to enable/disable items. In CentOS 7 I found "settings->power-> blank screen" as something I wish to configure by the command line? How is that accomplished? I brought up gconf-editor and searched for power and blank and found nothing. How can I control this setting from the command line? Thanks Jerry

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location