similar to: AMI Reload action, returning generated errors?

Hi All, I have one asterisks server and 3 client (i'm using voip sip client for my handphone). I've configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too. what i want to ask is, i was try to use 1 user (ex:1001) in 2 different client. example: client 1 (1001) make a call to client 2 (1002) --> ok then in client

Hi, I need to do some analysis on historic daily radiosonde data I download from the Wyoming Univ. web page ( http://weather.uwyo.edu/upperair/sounding.html). I am trying to use the RadioSonde package (V 1.3), but the format of the files from Wyoming don't match what RadioSonde is expecting. Has anyone used the Radiosonde package on the Wyoming data? Here is a sample of the Wyoming file

I hope I have this wrong, but when I have a bunch of priority 1 NoOp's created from regexten in sip.conf, and I do an 'extensions reload', I lose all the priority 1 NoOps! This can't be right... this means that in a production environment, if you make a change to your dialplan and do an 'extensions reload', you lose your ability to terminate calls to phones on this system.

Whenever I issue the reload command, asterisk crashes. Below is the output I get from (gdb) bt Any help is appreciated. *************************************************************** *CLI> reload == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/rtp.conf': Not found (No such file or

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > JR Richardson > Sent: Tuesday, December 05, 2006 3:49 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] RE: regcontext,NoOp extension > vanishes when extension reload > > > > > Let me guess: The

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New

> > Let me guess: The context in which you have the 2 thru n priorities is > the same one as you're using for regcontext right? > > Don't do that, bad things will happen (as you've noticed). > > I'd have to review the code again, but I think what you're seeing is as > a result of this. > > Regards, > - Brad > No, not exactly, I have a

OK this was an easy one to fix. All I had to do is RTFM. Note on the wiki: ATTENTION: Make sure you take a look at bug report 7144 Just do what Kevin said, include the regcontext in whatever static context you have the priority 2 extension and don't make a static regcontext in extension.conf. Let sip module do the rest. Works great. Thanks Guys. JR On 12/5/06, JR Richardson

I've not needed to do a dialplan reload for a while, so I don't know exactly which version is stopped working, but on 15.5, I'm not seeing ANY debug info at any debug level. So I'm not really sure how to find mistakes in the dialplan. This is all I get... how do I enable this debug mode to see the previous behaviour? Thanks asterisk -rvvvvvddddd (enters console) dialplan reload

Hello, Have a look at the plm package https://cran.r-project.org/web/packages/plm/index.html It has a convenient way to structure your data into panel according to some id. Best regards, Jeremie On Wed, Jan 10, 2018 at 5:41 PM, deva d <devazresearch at gmail.com> wrote: > dear all, > > i need some help in structuring my data file for a hierarchical time series > analysis.

Good morning, We recently fell back to the most recent build of asterisk 1.8 down from 11.3 and I believe we've crossed some sort of limit for 1.8. Our dialplan is 515723 entries long with 6263 distinct contexts. Both are loaded realtime via odbc (mysql). Previously at the end of a dialplan reload we would get a summary of how long it took to reload everything. Now it just shows the last line

In the interest of making things cleaner, I'd like to know if I can just reload one single conf file. Let's say I have two files, extensions.conf which includes small_file.conf. I only want "small_file.conf" reloaded, not the main file. Is this at all possible? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

I upgraded to 0.7.1 from a cvs version from a few weeks before 0.7.1 was relesed. now I am having troubles with my dialing plan and voice mail. As part of the upgrade I re-built the machine so there was a blank slate however after installing 0.7.1 I had no mail box creation script and could not figure out how to go about creating a mailbox, any suggestions would be usefull. I have looked at

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

So, incoming calls on zap work just as I expect them - an intro is played, the caller hits 1 for sale 2 for support or dials an extension. I'm using the privacy option for all extensions. When calls come in from zap, they caller is played the priv-recordintro recording, they say their name, and everything happens normally from there on out. However, when the call comes in from sip and

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]