custom parking lot

1. exten=>_2XXX,1,Answer()

   same=>n,Set(PARKINGEXTEN=${EXTEN:2})

   same=>n,park()

   same=>n,Set(CALLERID(name)=Failed Parking Lot)

   same=>n,Noop( checking var ${BLINDTRANSFER:0:7})

   same=>n,Set(ALERTINFOA=\;info=Ring2)

   same=>n,SipAddHeader(${ALERTINFOA}))

   same=>n,Dial(${BLINDTRANSFER:0:7})

   

IF you blind transfer to  the  specific parking lot it  will park the customer on that specific lot, if lot is busy, it will call   the extension who transferred the call

Note this will park spaces  parkpos => 701-720 , using the default Parking Lot created on res_parking.conf, if you want use spefic slot you will need create you parking lot on res_parking.conf and then include the contextt

Autoanswer and Server side auto answer.

Zoiper 3 Biz supports both Autoanswer and Server side auto answer.

Autoanswer is a BIZ feature, It is not available for the free version of Zoiper.

Enabling  Auto answer in Zoiper 3:

Open the Settings menu and select Automation

A new window will open. Configure it by your own preference.

What is BLF?

Busy Lamp Field is a feature that will allow you to see if your SIP contact is online.

This tutorial will show you how to make the BLF work with Asterisk-v.11.4.0 and above.

For the showcase purposes we have used extensions: SIP/1000 and SIP/2000.

How to enable BLF in Asterisk?

To enable BLF you need to specify the hints in the extensions.conf file so the subscribed
extensions would appear with their statuses in the Zoiper softphone contact list.
Please login to your Asterisk server via SSH, make sure Asterisk is not running and open
the extensions.conf file for editing using your favorite editor.

type peer vs friend

Howdy!

This is always a confusing part of the chan_sip SIP channel driver.

Rather than try to dig into any history, here is the current
documentation (from sip.conf.sample in the Asterisk source of
1.8,11,12) that you should base your decision to use a particular
"type" on:

"; SIP entities have a 'type' which determines their roles within Asterisk.
; * For entities with 'type=peer':
;   Peers handle both inbound and outbound calls and are matched by
ip/port, so for
;   The case of incoming calls from the peer, the IP address must
match in order for
;   The invitation to work. This means calls made from either
direction won't work if
;   The peer is unregistered while host=dynamic or if the host is
otherise not set to
;   the correct IP of the sender.
; * For entities with 'type=user':
;   Asterisk users handle inbound calls only (meaning they call
Asterisk, Asterisk can't
;   call them) and are matched by their authorization information
(authname and secret).
;   Asterisk doesn't rely on their IP and will accept calls regardless
of the host setting
;   as long as the incoming SIP invite authorizes successfully.
; * For entities with 'type=friend':
;   Asterisk will create the entity as both a friend and a peer.
Asterisk will accept
;   calls from friends like it would for users, requiring only that
the authorization
;   matches rather than the IP address. Since it is also a peer, a
friend entity can
;   be called as long as its IP is known to Asterisk. In the case of
host=dynamic,
;   this means it is necessary for the entity to register before
Asterisk can call it."

Most new work for SIP support in Asterisk is happening around
res_pjsip[1][2]. I don't know that there is any plans to deprecate
type=user going forward in chan_sip.

> Is that still correct?  Will type=user be phased out, and should even new
> installs of older asterisk versions (eg: 1.6) use type=peer only?

New installs of older Asterisk versions? That doesn't sound wise,
seeing as the 1.6 branch doesn't have any support, even for security
issues... A new install of Asterisk should be on a version of Asterisk
supported by the developers.[3] Right now, that would be the latest of
the 1.8,11, or 12 branches. That being said, 12 is rather new and has
many significant changes that should be considered.[3]

[1]: https://wiki.asterisk.org/wiki/display/AST/New+in+12
[2]: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
[3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

Hope that helps, thanks!

-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

Asterisk survey

[survey1]
exten=>_991,1,Answer()
same=>n,Playback(/var/lib/asterisk/agi-bin/survey/welcomesurvey)
same=>n(q1),read(q1,/var/lib/asterisk/agi-bin/survey/pregunta1,1)
same => n,GotoIf($[${q1}>5]?q1)
same=>n(q2),read(q2,/var/lib/asterisk/agi-bin/survey/pregunta2,1)
same => n,GotoIf($[${q2}>5]?q2)
same=>n(q3),read(q3,/var/lib/asterisk/agi-bin/survey/pregunta3,1)
same => n,GotoIf($[${q3}>5]?q3)
same=>n,Playback(/var/lib/asterisk/agi-bin/survey/goodbyesurvey)
same=>n,system(mysql --user=root --password='1932' survey -e "INSERT INTO survey.results (q1,q2,q3,did) VALUES ('${q1}','${q2}','${q3}','${EXTEN}')")



database;

create database survey;

use survey;

CREATE TABLE IF NOT EXISTS `results` (
  `res_id` int(11) NOT NULL AUTO_INCREMENT,
  `q1` varchar(1) DEFAULT NULL,
  `q2` varchar(1) DEFAULT NULL,
  `q3` varchar(1) DEFAULT NULL,
  `did` varchar(20) DEFAULT NULL,
  `res_time` timestamp NOT NULL DEFAULT CURRENT_TIMESTAMP,
  PRIMARY KEY (`res_id`)
) ENGINE=InnoDB  DEFAULT CHARSET=latin1 AUTO_INCREMENT=8 ;

FreePBX activation not working

The issue is we do not support OpenVZ as we only support Activation and Commercial Modules with our Distro and to install in OpenVZ means it would not be our Distro anymore.

http://community.freepbx.org/t/solved-cant-activate-freepbx-13/35540/5

playing sound file every 30 seconds using local channels

exten=>201,1,Dial(SIP/201&Local/s@alert)

[alert]
exten=>s,1(start),Noop()
same=>n,wait(30)
same=>n,System(/usr/bin/php /home/spy.php 201)
same=>n,Goto(alert,s,1)

Asterisk Dial() command example

Dial()

Synopsis

Attempt to connect to another device or endpoint and bridge the call.

Description

This application will place calls to one or more specified channels. As soon as one of the requested channels answers, the originating channel will be answered, if it has not already been answered. These two channels will then be active in a bridged call. All other channels that were requested will then be hung up.

Unless there is a timeout specified, the Dial application will wait indefinitely until one of the called channels answers, the user hangs up, or if all of the called channels are busy or unavailable. Dialplan execution will continue if no requested channels can be called, or if the timeout expires. This application will report normal termination if the originating channel hangs up, or if the call is bridged and either of the parties in the bridge ends the call.

If the OUTBOUND_GROUP variable is set, all peer channels created by this application will be put into that group (as in Set(GROUP()=...). If the OUTBOUND_GROUP_ONCE variable is set, all peer channels created by this application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP, however, the variable will be unset after use.

Asterisk logrotate

logrotate

'logger rotate' and logrotate

Asterisk logs files (typically under /var/log/asterisk) need some attention because they can get quite large.

You can do "logger rotate" from the asterisk command line or

/usr/sbin/asterisk -rx 'logger rotate'

from a bash prompt .. and you can add it to a cron job so it does it automatically for you.