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time synchronization error-goautodial vicidial

Asterisk GOautodial

Time synchronization Error , Vicidial ,Goautodial

February 11, 2015January 28, 2016 Satya Prakash 1 Comment 2009 - 40 posts - ‎13 authors hello everyone, 2011 - There a nice build with Asterisk and ViCidial.. but the problem every time is time synchronization server -> agents PC. they are should be at ... Asterisk Call Center Cisco Phones Vicidial.w, 2012 - Time synchronization Error in vicidial goautodial. there is a problem of time synchronization in your system please contact administrator. You've visited this page 2 times. Last visit: 9/2/15 v, 2013 - 7 posts - ‎3 authors I'm new in Goautodial and i install the release 2.1-final. ... 4) Time Sync Error also happens when the agent screen loses contact with the ... vicidial.org • View topic - , goautodial, Goautodial - striker24x7 striker24x7.blogspot.com/2012/.../time-synchronization-error-vicidial.ht... Oct 7, I am trying to test vicidialnow on agent system & it says "there is a problem of time synchronization in your system. please ... time synchronization problem with your ... 12 posts 31 Mar 2011 tim, it will display the current server time. 2. Follow the steps if the server time and timezone is wrong follow the below commands to change it. rm /etc/localtime ln -sf /usr/share/zoneinfo/Asia/Kolkata , or work on the latest goautodial time synchronization error Jobs. FIx GO Autodial 3.0 Error Time synchronization | Asterisk ... https://www.freelancer.in/.../FIx-Autodial-Error-Time-synchronization.ht, PHP time, please tell your ... "http://striker24x7.blogspot.in/2012/10/time-synchronization-error-vicidial.html". There is a time synchronization problem - GOautodial Open ... goautodial.org › Forums › Open dis, please tell your system administrator. How to resovle ... Time synchronization Error, server Time 4.Not updated the New server ip for the vicidial /goautodial How to resolve? 1. Check the server time, there is a problem of time synchronization in your system please contact administrator, These MeetMe Conference Bridge requires a timing resources like Dahdi or Zaptel. if they are not loaded poperly then time sync error occurs 2.wrong System timing . different timing in server and agent, Time synchronization Error, type date in the linux console, vicidial, vicidial-conferece - server ip fields. br Time Synchronization - GOautodial Open Source Call ... goautodial.org › Forums › Help "There is a time synchronization problem with your system, Why This Error?? The major cause for the time sync error are 1.The vicidial based system is using MeetMe conference

Why This Error?? The major cause for the time sync error are 1.The vicidial based system is using MeetMe conference

asterisk manager ami action orginate

Asterisk AMI API Action Orginate: Orginate Asterisk manager Action

February 8, 2015February 8, 2016 Satya Prakash 2 Comments Ami action orginate, AMI API orginate, AMI orginate, asterisk action orginate, Asterisk AMI API Action Orginate: Orginate Asterisk manager Action, Asterisk AMI API orginate, Asterisk Manager API Action Originate Action: Originate Parameters: Channel: Channel on which to originate the call (The same as you specify in the Dial application command) Context: Context to use on, for instance in a Dial command in the definition of the Context. Using Async leads to an OriginateResponse event which contains the failure reason if any. Reason may be one of the following: 0 = no su, the Extension is dialled within the Context to initiate the other end of the call. Also note that the Timeout only applies to the initial connection to the Channel; any timeout for the other end can b, this will cause the outside call not to be placed until the SIP channel has picked up): Action: Originate Channel: Dahdi/g2/8135551212 Context: default Exten: 101 Priority: 1 Timeout: 30000 Callerid: , when that answers

Asterisk Manager API Action Originate Action: Originate Parameters: Channel: Channel on which to originate the call (The same as you

Asterisk dial command

Asterisk Dial () command

February 8, 2015February 8, 2015 Satya Prakash 0 Comments '15', [URL]]]) Arguments Technology/Resource Technology/Resource - Specification of the device(s) to dial. These must be in the format of Technology/Resource, 1, 10) When the two channels are connected together ("bridged") allowing a conversation to take place between them. When the channel that triggered the Dial command hangs up, 15) exten => s, 20, 20) exten => _908., 3, a call to a SIP device (as defined in sip.conf) might have a destination of SIP/Jane, a forward slash, always set HANGUPCAUSE to 'answered elsewhere' d - Allow the calling user to dial a 1 digit extension while waiting for a call to be answered. Exit to that extension if it exists in the current contex, and a call to an IAX device (defined in iax.conf) might have a destination of IAX2/Fred. If we wanted Asterisk to ring the Zap/1 channel when extension 123 is reached in the dialplan, and bridge the inbound call with whichever destination channel answers the call first. If the Dial () application can’t contact any of the destinations, and continue on with the next priority in the extension. This application sets the following channel variables: DIALEDTIME - This is the time from dialing a channel until when it is disconnected. ANSW, and IAX2. For example, and Resource represents a resource available to that particular channel driver. Technology2/Resource2 - Optional extra devices to dial in parallel If you need more then one enter them as Technology2/R, and the calling DTMF string is sent to the calling party. Both arguments can be used alone. If progress is specified, and the remote endpoint or resource. Common technology types include Zap (for analog and T1/E1/J1 channels), and the resource is 1. Similarly, as shown: Dial(Zap/[gGrR]channel_or_group[/remote_extension]) For example, asterisk dial, Asterisk Dial () command, asterisk dial option, Asterisk takes most of the hard work out of connecting and translating between disparate networks. All you have to do is learn how to use the Dial() application. The syntax of the Dial() application i, Asterisk will set a variable called DIALSTATUS and then continue on with the next priority in the extension. Example: exten => 1265, Asterisk will set a variable called DIALSTATUS with the reason that it couldn’t dial the destinations, asterisl dial with option, but before the call gets bridged. The called DTMF string is sent to the called party, but the switch will not release their line until the destination party (the operator) hangs up. mode - With mode either not specified or set to 1, but when options such as A() and M() are used, by concatenating the destinations with an ampersand (&), caller A might be communicating over the traditional analog telephone network, cancel any dial timeout which has been set for this call. URL - The optional URL will be sent to the called party if the channel driver supports it. As an example, crh) exten => 20, Dial() will attempt to call the destination(s) for that number of seconds before giving up and moving on to the next priority in the extension. If no timeout is specified, Dial() will continue to dial the called channel(s) until someone answers or the caller hangs up. Let’s add a timeout of 10 seconds to our extension: exten => 123, Dial(IAX2/guest@misery.digium.com/s) The full syntax for the Dial() application is slightly different when dealing with Zap channels, Dial(Modem/ttyI0:${EXTEN:1}) exten => 233, Dial(Phone/phone0, Dial(SIP/1234, Dial(SIP/4029&SIP/4027&Zap/4&IAX/jaz, Dial(Zap/1, Dial(Zap/1)\ We can also dial multiple channels at the same time, Dial(Zap/1&Zap/2&SIP/Jane) The Dial() application will ring the specified destinations simultaneously, Dial(Zap/2r2, Dial(Zap/3/5551234), Dial(Zap/4/18005551212) The second argument to the Dial() application is a timeout, do not screen the call. o - If x is not provided, even if the called party isn't actually ringing. Pass no audio to the calling party until the called channel has answered. tone - Indicate progress to calling party. Send audio 'tone' from indications, filename specifies the sound prompt to play as a warning when time x is reached. If not set, filename specifies the sound prompt to play when the call begins. If not set, filename specifies the sound prompt to play when the timeout is reached. If not set, force the CallerID sent on a call-forward or deflection to the dialplan extension of this Dial() using a dialplan hint. For example, force the CallerID sent to x. x F - When the caller hangs up, Gosub to the specified location using the current channel. context exten priority arg1 argN C - Reset the call detail record (CDR) for this call. c - If the Dial() application cancels this call, Gosub to the specified location using the newly created channel. The Gosub will be executed for each destination channel. context exten priority arg1 argN B - Before initiating the outgoing call(s), here is how you would dial 1-800-555-1212 on Zap channel number 4. exten => 501, if it exists. D - Send the specified DTMF strings after the called party has answered, in milliseconds m - Provide hold music to the calling party until a requested channel answers. A specific music on hold class (as defined in musiconhold.conf) can be specified. class M - Execute the s, in milliseconds y - Warning time, in milliseconds z - Repeat time, is busy, it will be ignored. When the destination answers (presumably an operator services station), it will ring their phone back. p - This option enables screening mode. This is basically Privacy mode without memory. P - Enable privacy mode. Use x as the family/key in the AstDB database if it is pr, its DTMF is sent immediately after receiving a PROGRESS message. called calling progress e - Execute the h extension for peer after the call ends f - If x is not provided, let’s assume that we want to call a Zap endpoint identified by Zap/1, like this: exten => 123, options, or is otherwise unavailable, or the context defined in the EXITCONTEXT variable, sip, some PSTNs do not allow CallerID to be set to anything other than the numbers assigned to you. If x is provided, specified in seconds. If a timeout is given, specify that the CallerID that was present on the calling channel be stored as the CallerID on the called channel. This was the behavior of Asterisk 1.0 and earlier. If x is provided, specify the CallerID stored on the called channel. Note that o(${CALLERID(all)}) is similar to option o without the parameter. x O - Enables operator services mode. This option only works when bridgin, the calling channel is answered when the called channel answers, the calling channel is not answered until all actions on the called channel (such as playing an announcement) are completed. This option can be used to answer the calling channel before doing anything, the channels are bridged and the dialplan is done. If the destination simply does not answer, the default behavior is adequate in most cases. b - Before initiating an outgoing call, the Dial command exits. If the call is answered before the timeout, the introduction will always be deleted. N - This option is a modifier for the call screening/privacy mode. It specifies that if Caller*ID is present, the originator hanging up will cause the phone to ring back immediately. With mode set to 2, the originator no longer has control of their line. They may hang up, the recorded introduction will not be deleted if the caller hangs up while the remote party has not yet answered. With delete set to 1, the time remaining will be announced. FILENAME LIMIT_CONNECT_FILE - If specified, the time remaining will be announced. FILENAME LIMIT_WARNING_FILE - If specified, the time remaining will be announced. FILENAME x - Maximum call time, this action will be taken after the macro finished executing. ABORT - Hangup both legs of the call CONGESTION - Behave as if line congestion was encountered BUSY - Behave as if a busy signal was encou, this defaults to 136 years. options A - Play an announcement to the called party, this variable causes Asterisk to play the prompts to the callee. YES NO default: (true) LIMIT_TIMEOUT_FILE - If specified, this variable causes Asterisk to play the prompts to the caller. YES default: (true) NO LIMIT_PLAYAUDIO_CALLEE - If set, timeout, transfer the called party to the next priority of the current extension and start execution at that location. g - Proceed with dialplan execution at the next priority in the current extension if the d, transfer the called party to the specified destination and start execution at that location. context exten priority F - When the caller hangs up, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten priority h - Allow the called party to hang up by sending the DTMF sequence , tTr) exten => 500, we’d add the following extension: exten => 123, What is command and how it works? One of Asterisk’s most valuable features is its ability to connect different callers to each other. This is especially useful when callers are using different methods, when the operator flashes the trunk, where Technologyrepresents a particular channel driver, where x is the prompt to be played x - The file to play to the called party a - Immediately answer the calling channel when the called channel answers in all cases. Normally, which (in its simplest form) is made up of a technology (or transport) across which to make the call, which is an FXS channel with an analog phone plugged into it. The technology is Zap, while user B might be sitting in a café halfway around the world and speaking on an IP telephone. Luckily, you can dial into a demonstration server at Digium using the IAX2 protocol by using the following extension: exten => 500

What is command and how it works? One of Asterisk’s most valuable features is its ability to connect different callers

disk alert in linux

Email alert on disk space outage in linux

February 2, 2015February 24, 2016 Satya Prakash 26 Comments disk alert in linux, disk outage alert, disk usages alert, Email alert on disk space outage in linux, You need to be installed sendmail ( yum install sendmail) and mailx (yum install mailx) Create a shell script in linux. vi diskalert.sh #!/bin/sh support="itsupport@xyz.com" # set alert level 90% is d

For Send mail Tutorial Click Here Need to be installed sendmail ( yum install sendmail) and mailx (yum install mailx)

Asterisk CDR configuration file

January 31, 2015January 31, 2015 eduguru 2 Comments ; ; Asterisk Call Detail Record engine configuration ; ; CDR is Call Detail Record, ; and the [general] section is defined, ; asterisk stores the time in terms of microseconds and seconds. By setting ; initiatedseconds to 'yes', ; by looking in the "2. Call Detail Recording" option in the main menu. If your ; backend is marked with XXX, ; etc. If you are not getting output as you would expect, ; it might be acceptable to just use the scheduler thread, ; which is usually /var/log/asterisk. Of course, ; you will get 3 CDR's, a new thread is recommended, all formats are commented ; out except for the cdr-csv format. ; ; Here are all the possible back ends: ; ; csv, also, and ; has a [global] section with the proper variables defined. ; ; For logging to radius databases, and ; has a [global] section with the proper variables defined. ; ; For pgsql, and check what modules are available, and contains the proper [mappings] section. The advantage to ; using this backend, and has the ; proper definitions in it. If there are any problems, and in ; what order. By default, and in that section, and more. ; [general] ; Define whether or not to use CDR logging. Setting this to "no" will override ; any loading of backend CDR modules. Default is "yes". ;enable=yes ; Define whether or not to log , and no destination ; channel.) ;unanswered = no ; Define whether or not to log congested calls. Setting this to "yes" will ; report each call that fails to complete due to congestion conditions. Defau, and that cdr_tds.conf exists and is properly set up with a [global] category. ; ; Also, and that is the intended behaviour. (It also results in some B side CDRs being ; output, and the [radius] ; category is defined in this file, and the cdr_odbc.conf file exists, and the cdr_pgsql.conf file exists, and the originating ; channel on the other, and then one CDR for each channel attempted. This may seem ; redundant, and this option is "yes", and you get no output. ; ; Also, are still ; logged, as they have the B side channel as their source channel, Asterisk CDR configuration file, but are unanswered, but cannot be helped. ; ; In brief, but expending more CPU cycles to do so! ; ; To get manager events generated, by creating ; a new backend module in the source cdr/ directory!) ; ; Some of the modules required to provide these backends will not build or install ; unless some dependency requirements are met. Ex, CDR's are not closed out until after all extensions are finished ; executing. By enabling this option, compliance with ; Sarbanes-Oxley aka The Enron Act, custom, define the [csv] category in this file. No database necessary. The example ; config files are set up to provide this kind of output by default. ; ; To get custom csv CDR records, due to licensing ; requirements) ; (please note, etc. ; WARNING WARNING WARNING ; ;batch=no ; Define the maximum number of CDRs to accumulate in the buffer before posting ; them to the backend engines. 'batch' must be set to 'yes'. Default is 100. ;, etc. Useful for billing, files, fraud prevention, if you try to dial 3 extensions, in "yes" ; mode, is that you can define which fields to output, kill -9, make sure all the proper libs are installed, make sure the 'cdr.db' file exists in the log directory, make sure the 'radiuscfg' ; variable is properly pointing to an existing radiusclient.conf file. ; ; For logging to sqlite databases, make sure the cdr_custom.conf file ; is present, make sure the cdr_manager.conf file exists, make sure the proper libraries are available during the 'configure' ; phase, manager, mysql is available via the asterisk-addons, odbc, one for each phone that was rung. Default is "no". Some ; find this information horribly useless. Others find it very valuable. Note, or a new ; thread can be spawned for the submission of every batch. For small batches, or that the appropriate config file exists, pgsql, please note that you can generate CDR records in as many formats as you ; wish. If you configure 5 different CDR formats, power failure, QOS evaluations, radius, regardless of the value defined for 'size'. 'batch' must be set to ; 'yes'. Note that time is in seconds. Default is 300 (5 minutes). ;time=300 ; The CDR engine uses the internal asterisk scheduler to, remember, say anything over size=10, set this to "yes". Default ; is "yes". ;safeshutdown=yes ; ; ; CHOOSING A CDR "BACKEND" (what kind of output to generate) ; ; To choose a backend, so ; set this to "no". Default is "no". ;scheduleronly=no ; When shutting down asterisk, so set this to "yes". ; For large batches, sqlite, tds ; (also, that "make menuselect" ; shows that the modules are available, that if you wish to log CDR info to a database, that other backends can be created, the 'billsec' field logged to the backends (text files or databases) ; is simply the end time (hangup time) minus the answer time in seconds. Internally, the CDR will be ended before executing ; the "h" extension and hangup handlers so that CDR values such as "end" and ; "billsec" may be retrieved inside of of this extension. ; The default value is "no, the data will be stored in a buffer to help alleviate load on the ; asterisk server. Default is "no". ; ; WARNING WARNING WARNING ; Use of batch mode may result in data loss after unsafe asterisk term, the entry will ; silently ignored, the example configs are set up to mimic the cdr-csv ; output. If you don't make any changes to the mappings, the first thing to do ; is to run the command "make menuselect", the proper libraries should be available ; during the 'configure' operation. ; ; For tds logging, then data will likely be lost. You can always check the size of ; the CDR batch buffer with the CLI "cdr status" command. To enable blocking on ; submission of CDR data during asterisk shutdown, then each event will be logged ; in 5 different places! In the example config files, then it will post the ; records, then the billsec time is incremented one second. ; The default value is "no". ;initiatedseconds=no ; Define the CDR batch mode, this is ; when the microsecond part of the end time is greater than the microsecond ; part of the answer time, this option controls the reporting of unanswered calls which only have an A ; party. Calls which get offered to an outgoing line, usually, when it was not ; answered. For example, where instead of posting the CDR at the end of ; every call, which provides logging services via a variety of ; pluggable backend modules. Detailed call information can be recorded to ; databases, with one of the call targets on one side, with the single variable 'enabled = yes'. ; ; For odbc, you are basically generating ; the same thing as cdr-csv, you can block until the CDRs are submitted. If ; you don't, you can force asterisk to report any seconds ; that were initiated (a sort of round up method). Technically, you have to make sure either the right category is ; defined in this file, you know that the "configure" command could not find ; the required libraries for that option. ; ; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv ; file, you will have to define ; a specific table in that databse to make things work! See the doc directory for more details ; on how to create this table in each database. ; [csv] usegmtime=yes ; log date/, you will see one CDR

; ; Asterisk Call Detail Record engine configuration ; ; CDR is Call Detail Record, which provides logging services via

Autodestruct on dialog….: Asterisk issue

January 28, 2015 eduguru 11 Comments Autodestruct on dialog, Autodestruct on dialog in Asterisk issue, Autodestruct on dialog....: Asterisk issue, i.e. someone hung up, it still executes the rest of the script. I now check multiple places to see if the channel has been closed before executing certain actions., WARNING[1812] chan_sip.c: Autodestruct on dialog '38617eb73184c21225d3133b398b0dc6@10.0.8.18:5060' with owner SIP/1000-00000533 in place (Method: BYE). Rescheduling destruction for 10000 ms Solution:

WARNING[1812] chan_sip.c: Autodestruct on dialog ‘38617eb73184c21225d3133b398b0dc6@10.0.8.18:5060’ with owner SIP/1000-00000533 in place (Method: BYE). Rescheduling destruction for 10000 ms Solution: We

start asterisk autometically when system got restarted: Start Asterisk service on reboot

December 28, 2014 Satya Prakash 8 Comments change to asterisk source directory and do it: make config chkconfig asterisk on after that you can reboot.

change to asterisk source directory and do it: make config chkconfig asterisk on after that you can reboot.

asterisk subroutine

Asterisk Subroutine Call : Pre-Dial Handlers

December 25, 2014December 25, 2014 Satya Prakash 22 Comments [default] exten => pre_dial_handler, 1, Asterisk Subroutine Call : Pre-Dial Handlers, B(default^caller_handler^1)) same => n, b(default^pre_dial_handler^1), but before any actions have been taken to actually dial the callee channels. You can execute a dialplan subroutine on the caller channel and on each callee channel dialled. There are two ways in which, Channel is ${CHANNEL}) same => n, Dial(SIP/alice, Dial(SIP/bar, Hangup() exten => caller_handler, In caller pre-dial handler!) same => n, NoOp() same => n, Pre dial handler, Pre-dial handlers allow you to execute a dialplan subroutine on a channel before a call is placed but after the application performing a dial action is invoked. This means that the handlers are execut, Return(), Return() exten => dial_example, Verbose(0, Verbose(1

Pre-dial handlers allow you to execute a dialplan subroutine on a channel before a call is placed but after the

Asterisk start on server reboot

December 25, 2014 Satya Prakash 22 Comments Asterisk start on server reboot

Asterisk Manager API Action Redirect : Asterisk Call transfer

December 25, 2014 Satya Prakash 50 Comments Action: Redirect Synopsis: Redirect (transfer) a call Privilege: call, all Description: Redirect (transfer) a call. Variables: (Names marked with * are required) Channel: Channel to redirect ExtraChannel: Second call leg to transfer (optional) Exten: Extension to transfe, asterisk call transfer, Asterisk Manager API Action Redirect : Asterisk Call transfer

Action: Redirect Synopsis: Redirect (transfer) a call Privilege: call,all Description: Redirect (transfer) a call. Variables: (Names marked with * are