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Complete newbie need help in understanding how this works

October 19, 2015, 6:02 pm
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start by reading www.asteriskdocs.org. Then go deep with AMI, ami integration with DB, php call backs and possibly webrtc

Statistics : Posted by rhymeguy • on Mon Oct 19, 2015 6:02 pm • Replies 2 • Views 387
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Complete newbie need help in understanding how this works

October 19, 2015, 6:02 pm
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Hi all!
Im complete newbie when it comes to telephone systems, but i actually want to learn something new and hopefully play with it. I know that there are many cloud based services which would be cheaper for setup, but please not that i want to learn something. Money is not important as knowledge.

I will describe what im willing to build.
For start, i will have one analogue line. Next, i have enough knowledge to build web based applications and therefore i want to be able programmatically to make a call to another phone number outside my LAN network. For example i want login to my app where i have phone numbers of my friends, i will click to call button and call should be processed.
Also for example when somebody call back i want to detect that via web app and to be notified which friend calls me. If this is complicated i will be satisfied even with placing simple phone call...

After some researching i came up with some tutorials which are very confusing for me, so i just want to check if im on right track.

Is this what i need?
1) Phone with lan interface connected to FXO gateway
2) FXO gateway connected with IP box (Asterisk)
3) FXO gateway connected with FXS (wall jack)

If yes, when i want to initiate a call, i have to send command to Asterisk. Then Asterisk will connect to FXO gateway and FXO gateway will do all the magic (if it is configured properly).

Now: is this how is it supposed to work or am i mistaken? Can it be simpler?

Thanks!

Statistics : Posted by rhymeguy • on Mon Oct 19, 2015 6:02 pm • Replies 2 • Views 387
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X-Lite (Bria4) - Asterisk / Presence Problem

November 12, 2015, 5:54 am
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if i made the following (sit set debug on) in *CLI I get

.
.
.
ALLOW: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
.
.
.

shouldnt be there another entry like "REFERENCE" !?

Statistics : Posted by schwankit • on Thu Nov 12, 2015 7:54 am • Replies 3 • Views 95
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X-Lite (Bria4) - Asterisk / Presence Problem

November 12, 2015, 5:54 am
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This is the new extensions.conf

Code: exten=6000,hint,SIP/6000,CustomPresence:6000
exten=6000,1,Dial(SIP/6000)
exten=6001,hint,SIP/6001,CustomPresence:6001
exten=6001,1,Dial(SIP/6001)
exten=6002,hint,SIP/6002,CustomPresence:6002
exten=6002,1,Dial(SIP/6002)
exten=6003,hint,SIP/6003,CustomPresence:6003
exten=6003,1,Dial(SIP/6003)
exten=6004,hint,SIP/6004,CustomPresence:6004
exten=6004,1,Dial(SIP/6004)
exten=6005,hint,SIP/6005,CustomPresence:6005
exten=6005,1,Dial(SIP/6005)


Statistics : Posted by schwankit • on Thu Nov 12, 2015 7:54 am • Replies 3 • Views 95
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X-Lite (Bria4) - Asterisk / Presence Problem

November 12, 2015, 5:54 am
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I have something new Image

I can change Presence manual via *CLI: presencestate change CustomPresence:6000 AVAILABLE// (6000 is the caller ID and name is the same) but Xlite doesnt get anything.

In Xlite I only see Presence via peer-to-peer. Doesnt this mean the Asterisk Servers isnt involved in it? So why do I dont get any Presence. Devica state works fine as well!

Statistics : Posted by schwankit • on Thu Nov 12, 2015 7:54 am • Replies 3 • Views 95
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X-Lite (Bria4) - Asterisk / Presence Problem

November 12, 2015, 5:54 am
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Hello everybody,

this is my first post in this forum that means I insttaled asterix with trunks, routing, voicemail, transfer aso. "without" any problems Image ans now 90% of what I need is working fine.

But there are still 2 points on my list and the first i cant't get trough alone will be listet as following.:

I use Xlite as Client on my Windows PC and want to use Messaging and (Custom) Presence (not just the device state what works actually fine.

I think the best well be i'll post my config files which I think the presence should be configured in, so you all can tell me the mistakes I made Image

sip.conf
Code: [general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
subscribecontext=default
accept_outofcall_message=yes
outofcall_message_context=astsms
allowsubscribe=yes


useres.conf (user 6000 for example)
Code: [general]
fullname=New User
userbase=6000
hasvoicemail=yes
vmsecret=1234
hassip=yes
hasiax=yes
hasmanager=no
callwaiting=yes
threewaycalling=yes
callwaitingcallerid=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callgroup=1
pickupgroup=1
vmexten=99

[6000]
fullname=Christian Schwank
registersip=no
host=dynamic
callgroup=1
mailbox=6000
call-limit=100
type=peer
username=6000
transfer=yes
callcounter=yes
context=DLPN_DialPlan2_9989950
cid_number=6000
hasvoicemail=yes
vmsecret=1234
email=j.braun@samsys.eu
threewaycalling=no
hasdirectory=yes
callwaiting=no
hasmanager=no
hasagent=no
hassip=yes
hasiax=no
secret=samsys11
nat=no
canreinvite=no
dtmfmode=rfc2833
insecure=no
pickupgroup=1
macaddress=6000
autoprov=yes
label=6000
linenumber=1
LINEKEYS=1
disallow=all
allow=alaw,ulaw,gsm,g726,g723


extensions.conf
Code: [general]
static=yes
writeprotect=no
clearglobalvars=no
[default]
exten=6001,1,Dial(SIP/6001)
exten=6002,1,Dial(SIP/6002)
exten=6003,1,Dial(SIP/6003)
exten=6004,1,Dial(SIP/6004)
exten=6005,1,Dial(SIP/6005)
exten=6000,1,Dial(SIP/6000)
exten=_#6XXX,1,Set(MBOX=${EXTEN:1}@default)
exten=_#6XXX,n,VoiceMail(${MBOX})
exten=a,1,VoicemailMain(${MBOX})
exten=99,1,VoiceMailMain(${CALLERID(num)}@default)


Thank you in advice. Image

Statistics : Posted by schwankit • on Thu Nov 12, 2015 7:54 am • Replies 3 • Views 95
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Reparking a call on the same position

November 13, 2015, 8:05 am
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Attempting to use soft keys on Cisco SPA508G phones to implement Parking with BLF. I have the parking lot set up (10000) in Asterisk 13.5, with spaces from 10001-10050 (I'll use the whole range when it goes live). Each button has fnc=blf+sd;sub=10001@$PROXY;ext=10001@$PROXY; (10002, 10003, etc.) which works for the BLF, Park, and ParkedCall. Once the call is parked, the light turns solid red. When the button is pressed again, the call is retrieved, and the light goes green. This is a little misleading. The hint returns to Idle, but the calling extension is still in the application Park(10000), and the callee (parker/retriever) is showing ParkedCall(10000,10001). That parking position is not really available. The callee can then repark the call on a different parking space, but cannot repark on the same space (I need to find a way to repark in the same space). If the callee tries to park on the same space, both caller and callee hear MOH. The caller appears to be in the same place, and the callee is now parking himself.

Ideally, for my scenario, I would like to see the parked call hint show ringing so the light will blink red (or any other device state that will show blinking). When it's picked up, either go solid red (InUse) or join the endpoints, and remove the Park/ParkedCall completely. I've been trying a number of configs for this over the past few days. Besides modifying the source to set the hints does anyone else have an idea to make this work for me? Thanks.

Statistics : Posted by mkozusnik • on Fri Nov 13, 2015 10:05 am • Replies 0 • Views 11
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Asterisk13 Realtime Configuration on Centos7

November 12, 2015, 12:19 pm
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Here's how I have mine set up if it helps.

res_odbc.conf
[asterisk]
enabled => yes
dsn => asterisk-connector
username => asterisk
password => databasepassword
pooling => no
limit => 1
pre-connect=> yes

res_config_odbc.conf
[settings]
connection = asterisk_connector

/etc/odbc.ini
[asterisk-connector]
Description = MySQL connection to 'asterisk' database
Driver = MySQL
Database = asterisk
Server = localhost
Port = 3306
Socket = /var/run/mysqld/mysqld.sock

/etc/odbcinst.ini
[MySQL]
Description= ODBC for MySQL
Driver= /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so
Setup= /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so
FileUsage= 1

extconfig.conf
[settings]
sipusers => odbc,asterisk,sippeers
sippeers => odbc,asterisk,sippeers
iaxusers => odbc,asterisk,iaxpeers
iaxpeers => odbc,asterisk,iaxpeers
voicemail => odbc,asterisk,voicemail_users
queues => odbc,asterisk,queue_table
queue_members => odbc,asterisk,queue_member_table
meetme => odbc,asterisk,meetme

voicemail.conf if you want to store voicemail in the database
[general]
format=wav
odbcstorage=asterisk
odbctable=voicemessages

cdr_adaptive_odbc.conf for storing CDRs in the database
[adaptive_connection]
connection=asterisk
table=cdr

modules.conf
[modules]
autoload=yes

preload => func_periodic_hook.so
noload => res_fax.so
noload => res_config_ldap.so
noload => chan_iax2.so

; Resource modules currently not needed
noload => res_ael_share.so
noload => res_clialiases.so
noload => res_adsi.so
; PBX modules currently not needed
noload => pbx_ael.so
noload => pbx_dundi.so
; Channel modules currently not needed
noload => chan_oss.so
noload => chan_mgcp.so
noload => chan_skinny.so
noload => chan_phone.so
noload => chan_agent.so
noload => chan_unistim.so
noload => chan_alsa.so
; Application modules currently not needed
noload => app_nbscat.so
noload => app_amd.so
noload => app_minivm.so
noload => app_zapateller.so
noload => app_ices.so
noload => app_sendtext.so
noload => app_speech_utils.so
noload => app_mp3.so
noload => app_flash.so
noload => app_getcpeid.so
noload => app_setcallerid.so
noload => app_adsiprog.so
noload => app_forkcdr.so
noload => app_sms.so
noload => app_morsecode.so
noload => app_followme.so
noload => app_url.so
noload => app_alarmreceiver.so
noload => app_disa.so
noload => app_dahdiras.so
noload => app_senddtmf.so
noload => app_sayunixtime.so
noload => app_test.so
noload => app_externalivr.so
noload => app_image.so
noload => app_dictate.so
noload => res_calendar.so
noload => cel_sqlite3_custom.so
noload => cel_odbc.so
noload => cdr_odbc.so
noload => cel_manager.so
noload => cdr_manager.so
noload => res_phoneprov.so
noload => res_stun_monitor.so
noload => res_stasis_playback.so

table asterisk.sippeers
CREATE TABLE IF NOT EXISTS `sippeers` (
`id` int(11) NOT NULL AUTO_INCREMENT,
`name` varchar(80) NOT NULL DEFAULT '',
`context` varchar(80) DEFAULT 'from-internal-sip',
`callingpres` enum('allowed_not_screened','allowed_passed_screen','allowed_failed_screen','allowed','prohib_not_screened','prohib_passed_screen','prohib_failed_screen','prohib','unavailable') DEFAULT 'allowed_not_screened',
`deny` varchar(95) DEFAULT NULL,
`permit` varchar(95) DEFAULT NULL,
`secret` varchar(80) DEFAULT NULL,
`md5secret` varchar(80) DEFAULT NULL,
`remotesecret` varchar(250) DEFAULT NULL,
`transport` enum('tcp','udp','tcp,udp') DEFAULT NULL,
`host` varchar(31) NOT NULL DEFAULT 'dynamic',
`nat` varchar(32) NOT NULL DEFAULT 'force_rport,comedia',
`type` enum('user','peer','friend') NOT NULL DEFAULT 'friend',
`accountcode` varchar(20) DEFAULT NULL,
`amaflags` varchar(13) DEFAULT NULL,
`callgroup` varchar(10) DEFAULT NULL,
`callerid` varchar(80) DEFAULT NULL,
`defaultip` varchar(15) DEFAULT NULL,
`dtmfmode` varchar(7) DEFAULT 'rfc2833',
`fromuser` varchar(80) DEFAULT NULL,
`fromdomain` varchar(80) DEFAULT NULL,
`insecure` varchar(4) DEFAULT NULL,
`language` char(2) DEFAULT NULL,
`mailbox` varchar(50) DEFAULT NULL,
`pickupgroup` varchar(10) DEFAULT NULL,
`qualify` char(3) DEFAULT 'yes',
`regexten` varchar(80) DEFAULT NULL,
`rtptimeout` char(3) DEFAULT NULL,
`rtpholdtimeout` char(3) DEFAULT NULL,
`setvar` varchar(100) DEFAULT NULL,
`disallow` varchar(100) DEFAULT 'all',
`allow` varchar(100) DEFAULT 'ulaw;alaw;gsm',
`fullcontact` varchar(80) NOT NULL DEFAULT '',
`ipaddr` varchar(45) NOT NULL DEFAULT '',
`port` mediumint(5) unsigned NOT NULL DEFAULT '0',
`defaultuser` varchar(80) NOT NULL DEFAULT '',
`subscribecontext` varchar(80) DEFAULT NULL,
`directmedia` enum('yes','no') DEFAULT 'no',
`trustrpid` enum('yes','no') DEFAULT NULL,
`sendrpid` enum('yes','no') DEFAULT NULL,
`progressinband` enum('never','yes','no') DEFAULT NULL,
`promiscredir` enum('yes','no') DEFAULT NULL,
`useclientcode` enum('yes','no') DEFAULT NULL,
`callcounter` enum('yes','no') DEFAULT NULL,
`busylevel` int(10) unsigned DEFAULT NULL,
`allowoverlap` enum('yes','no') DEFAULT 'yes',
`allowsubscribe` enum('yes','no') DEFAULT 'yes',
`allowtransfer` enum('yes','no') DEFAULT 'yes',
`ignoresdpversion` enum('yes','no') DEFAULT 'no',
`videosupport` enum('yes','no','always') DEFAULT 'no',
`maxcallbitrate` int(10) unsigned DEFAULT NULL,
`rfc2833compensate` enum('yes','no') DEFAULT 'yes',
`session-timers` enum('originate','accept','refuse') DEFAULT 'accept',
`session-expires` int(5) unsigned DEFAULT '1800',
`session-minse` int(5) unsigned DEFAULT '90',
`session-refresher` enum('uac','uas') DEFAULT 'uas',
`t38pt_usertpsource` enum('yes','no') DEFAULT NULL,
`outboundproxy` varchar(250) DEFAULT NULL,
`callbackextension` varchar(250) DEFAULT NULL,
`registertrying` enum('yes','no') DEFAULT 'yes',
`timert1` int(5) unsigned DEFAULT '500',
`timerb` int(8) unsigned DEFAULT NULL,
`qualifyfreq` int(5) unsigned DEFAULT '120',
`contactpermit` varchar(250) DEFAULT NULL,
`contactdeny` varchar(250) DEFAULT NULL,
`lastms` int(11) NOT NULL DEFAULT '0',
`regserver` varchar(100) NOT NULL DEFAULT '',
`regseconds` int(11) NOT NULL DEFAULT '0',
`useragent` varchar(50) NOT NULL DEFAULT '',
PRIMARY KEY (`id`),
UNIQUE KEY `name` (`name`),
KEY `name_2` (`name`)
) ENGINE=MyISAM

table asterisk.voicemail_users
CREATE TABLE IF NOT EXISTS `voicemail_users` (
`uniqueid` int(11) NOT NULL AUTO_INCREMENT,
`customer_id` varchar(11) NOT NULL DEFAULT '0',
`context` varchar(50) NOT NULL,
`mailbox` varchar(11) NOT NULL DEFAULT '0',
`password` varchar(32) NOT NULL DEFAULT '0',
`fullname` varchar(150) NOT NULL,
`email` varchar(50) NOT NULL,
`pager` varchar(50) NOT NULL,
`tz` varchar(10) NOT NULL DEFAULT 'eastern',
`attach` varchar(4) NOT NULL DEFAULT 'no',
`saycid` varchar(4) NOT NULL DEFAULT 'no',
`dialout` varchar(10) NOT NULL,
`callback` varchar(10) NOT NULL,
`review` varchar(4) NOT NULL DEFAULT 'no',
`operator` varchar(4) NOT NULL DEFAULT 'no',
`envelope` varchar(4) NOT NULL DEFAULT 'no',
`sayduration` varchar(4) NOT NULL DEFAULT 'no',
`saydurationm` tinyint(4) NOT NULL DEFAULT '1',
`sendvoicemail` varchar(4) NOT NULL DEFAULT 'no',
`delete` varchar(4) NOT NULL DEFAULT 'no',
`nextaftercmd` varchar(4) NOT NULL DEFAULT 'yes',
`forcename` varchar(4) NOT NULL DEFAULT 'no',
`forcegreetings` varchar(4) NOT NULL DEFAULT 'no',
`hidefromdir` varchar(4) NOT NULL DEFAULT 'yes',
`maxmsg` int(11) NOT NULL DEFAULT '100',
`format` varchar(8) NOT NULL DEFAULT 'wav',
`stamp` timestamp NOT NULL DEFAULT CURRENT_TIMESTAMP ON UPDATE CURRENT_TIMESTAMP,
PRIMARY KEY (`uniqueid`),
KEY `mailbox_context` (`mailbox`,`context`)
) ENGINE=MyISAM

table asterisk.voicemessages
CREATE TABLE IF NOT EXISTS `voicemessages` (
`uniqueid` bigint(20) unsigned NOT NULL AUTO_INCREMENT,
`msgnum` int(4) DEFAULT NULL,
`dir` varchar(80) DEFAULT NULL,
`context` varchar(80) DEFAULT NULL,
`macrocontext` varchar(80) DEFAULT NULL,
`callerid` varchar(40) DEFAULT NULL,
`origtime` varchar(40) DEFAULT NULL,
`duration` varchar(20) DEFAULT NULL,
`mailboxuser` varchar(80) DEFAULT NULL,
`mailboxcontext` varchar(80) DEFAULT NULL,
`recording` blob,
`read` tinyint(1) DEFAULT '0',
`flag` varchar(10) DEFAULT NULL,
`msg_id` bigint(20) NOT NULL,
PRIMARY KEY (`uniqueid`),
UNIQUE KEY `uniqueid` (`uniqueid`)
) ENGINE=MyISAM

table asterisk.cdr
CREATE TABLE IF NOT EXISTS `cdr` (
`start` datetime NOT NULL DEFAULT '0000-00-00 00:00:00',
`answer` datetime NOT NULL,
`end` datetime NOT NULL,
`clid` varchar(80) NOT NULL DEFAULT '',
`src` varchar(80) NOT NULL DEFAULT '',
`dst` varchar(80) NOT NULL DEFAULT '',
`dcontext` varchar(80) NOT NULL DEFAULT '',
`channel` varchar(80) NOT NULL DEFAULT '',
`dstchannel` varchar(80) NOT NULL DEFAULT '',
`lastapp` varchar(80) NOT NULL DEFAULT '',
`lastdata` varchar(80) NOT NULL DEFAULT '',
`duration` int(11) NOT NULL DEFAULT '0',
`billsec` int(11) NOT NULL DEFAULT '0',
`disposition` varchar(45) NOT NULL DEFAULT '',
`amaflags` int(11) NOT NULL DEFAULT '0',
`accountcode` varchar(20) NOT NULL DEFAULT '',
`uniqueid` varchar(32) NOT NULL DEFAULT '',
`userfield` varchar(255) NOT NULL DEFAULT '',
`peeraccount` varchar(20) NOT NULL DEFAULT '',
`linkedid` varchar(32) NOT NULL DEFAULT '',
`sequence` int(11) NOT NULL DEFAULT '0',
KEY `start` (`start`),
KEY `dst` (`dst`)
) ENGINE=MyISAM

I have the other tables too if you'd like them.

Statistics : Posted by hmtaurus • on Thu Nov 12, 2015 2:19 pm • Replies 4 • Views 214
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Asterisk13 Realtime Configuration on Centos7

November 12, 2015, 12:19 pm
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One thing I noticed...

When using mysql, the second part in extconfig.conf should be the name of the database unless that's been changed.
sippers=>mysql,asteriskDB,sip_buddies
extensions => mysql,asteriskDB,extensions
For odbc, it would be general in this case.

Do you have any logging when the phone tries to register? Can you hard-code the endpoints in sip.conf and get it to work (just for testing purposes)?

Statistics : Posted by hmtaurus • on Thu Nov 12, 2015 2:19 pm • Replies 4 • Views 214
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Asterisk13 Realtime Configuration on Centos7

November 12, 2015, 12:19 pm
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Thanks for your replay, actually the point is that endpoints can not be registered :/ Image

Statistics : Posted by hmtaurus • on Thu Nov 12, 2015 2:19 pm • Replies 4 • Views 214
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Asterisk13 Realtime Configuration on Centos7

November 12, 2015, 12:19 pm
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If your endpoints are able to register, you could add rtcachefriends=yes to the [general] section of sip.conf. Once they make a call, they will show in 'sip show peers'.

mark*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime
1000/1000 192.168.1.66 D No No 5060 OK (3 ms) Cached RT
1001/1001 (Unspecified) D No No 0 UNREACHABLE Cached RT
2 sip peers [Monitored: 1 online, 1 offline Unmonitored: 0 online, 0 offline]

I used to use the MySQL setup, but would not be surprised if that became deprecated in the future in favor of odbc.

Statistics : Posted by hmtaurus • on Thu Nov 12, 2015 2:19 pm • Replies 4 • Views 214
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Asterisk13 Realtime Configuration on Centos7

November 12, 2015, 12:19 pm
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Hello Dear Fellows
I Installed Asterisk 13 on Cent OS 7 in which MariaDB Database is installed on it.

I made proper configurations on:Res_mysql.conf / extconfig.conf / sip.conf /extensions.conf /modules.conf and finally I created suitable database and tables and I inserted some values in them .

how ever apparently asterisk cannot load my database information so that I can register my users. I checked "core show peers" and RealTime Architecture was disabled.

I Highly appreciate any hint that what the problem would be ? I attached my configured files in the following. Best Regards


Res_Mysql.conf
[general]
dbhost=127.0.0.1
dbname=asteriskDB
dbuser=root
dbpass=123a123A
dbport=3306
----------------------
Extconfig.conf
[setting]
Sippers=>mysql , general , sip_buddies
Extentions=> mysql,general , extensions

----------------------

Sip.conf
[general]

context= from-sip ;
bindport=5060;
bindaddr=0.0.0.0;
disallow=all;
allow=gsm
allow=ulaw
registerattempts=10
registertimeout=20

dbhost=127.0.0.1
dbname=asteriskDB
dbuser=root
dbpass=123a123A
dbport=3306
-----------------------------

Extensions.conf

[general]
dbhost=127.0.0.1
dbname=asteriskDB
dbuser=root
dbpass=123a123A
dbport=3306

[globals];
[from-sip]
Switch =>Realtime
-----------------------------

Modules.conf

Autoload=yes
Load =>res_config_mysql.so
load => func_realtime.so
Load =>app_realtime.so
Load =>pbx_realtime.so
------------------------------------------------------------------------------------------------------------------------------------------

create database if not exists `asteriskDB`;
use asteriskDB;

-----------------------------------------------------------------------------

1.Create Tables (sip_buddies & Extensions)

CREATE TABLE `sip_buddies` (
`id` int(11) NOT NULL auto_increment,
`name` varchar(80) NOT NULL,
`callerid` varchar(80) default NULL,
`defaultuser` varchar(80) NOT NULL,
`regexten` varchar(80) NOT NULL,
`secret` varchar(80) default NULL,
`mailbox` varchar(50) default NULL,
`accountcode` varchar(20) default NULL,
`context` varchar(80) default NULL,
`amaflags` varchar(7) default NULL,
`callgroup` varchar(10) default NULL,
`canreinvite` char(3) default 'yes',
`defaultip` varchar(15) default NULL,
`dtmfmode` varchar(7) default NULL,
`fromuser` varchar(80) default NULL,
`fromdomain` varchar(80) default NULL,
`fullcontact` varchar(80) default NULL,
`host` varchar(31) NOT NULL,
`insecure` varchar(4) default NULL,
`language` char(2) default NULL,
`md5secret` varchar(80) default NULL,
`nat` varchar(5) NOT NULL default 'no',
`deny` varchar(95) default NULL,
`permit` varchar(95) default NULL,
`mask` varchar(95) default NULL,
`pickupgroup` varchar(10) default NULL,
`port` varchar(5) NOT NULL,
`qualify` char(3) default NULL,
`restrictcid` char(1) default NULL,
`rtptimeout` char(3) default NULL,
`rtpholdtimeout` char(3) default NULL,
`type` varchar(6) NOT NULL default 'friend',
`disallow` varchar(100) default 'all',
`allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw',
`musiconhold` varchar(100) default NULL,
`regseconds` int(11) NOT NULL default '0',
`ipaddr` varchar(15) NOT NULL,
`cancallforward` char(3) default 'yes',
`lastms` int(11) NOT NULL,
`useragent` char(255) default NULL,
`regserver` varchar(100) default NULL,
PRIMARY KEY (`id`),
UNIQUE KEY `name` (`name`),
KEY `name_2` (`name`)
) ENGINE=MyISAM AUTO_INCREMENT=893 DEFAULT CHARSET=latin1;
CREATE TABLE `extensions` (
`id` int(11) NOT NULL auto_increment,
`context` varchar(20) NOT NULL default '',
`exten` varchar(20) NOT NULL default '',
`priority` tinyint(4) NOT NULL default '0',
`app` varchar(20) NOT NULL default '',
`appdata` varchar(128) NOT NULL default '',
PRIMARY KEY (`context`,`exten`,`priority`),
KEY `id` (`id`)
) ENGINE=MyISAM AUTO_INCREMENT=257 DEFAULT CHARSET=latin1;

-----------------------------------------------------------------------------

2.Insert into Tables (sip_buddies & Extensions)



INSERT INTO asteriskDB.sip_buddies (
NAME, defaultuser, secret, context, HOST, nat, qualify, TYPE)
VALUES (
'1000', '1000', '1234', 'from-sip', 'dynamic', 'yes', 'no', 'friend');
;

INSERT INTO asteriskDB.sip_buddies (
NAME, defaultuser, secret, context, HOST, nat, qualify, TYPE)
VALUES (
'2000', '2000', '1234', 'from-sip', 'dynamic', 'yes', 'no', 'friend');




INSERT INTO `extensions`(`context`,`exten`,`priority`,`app`,`appdata`)
values


('from-sip','12121111111',1,'Dial','SIP/1000|60'),
('from-sip','12122222222',1,'Dial','SIP/2000|60');

Statistics : Posted by hmtaurus • on Thu Nov 12, 2015 2:19 pm • Replies 4 • Views 214
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Reparking a call on the same position

November 13, 2015, 8:05 am
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I just found EXTENSION_STATE. I may see how that works since I'm using the parking space as an extension rather than a device.

Statistics : Posted by mkozusnik • on Fri Nov 13, 2015 10:05 am • Replies 2 • Views 126
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Reparking a call on the same position

November 13, 2015, 8:05 am
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Here's what the code looks like right now. I've made so many changes to try and get it all to work, there may be some pieces missing. At this point, it's working, but can't repark, and when the parked call is answered, the hint changes to Idle.

extensions.conf
[subInternalDialer]
...
same=> n,GotoIf($["${devicetype}" = "parking"]?parking,s,1)
...


[parking]
exten=> s,1,NoOp(${ARG1} - ${CHANNEL} - ${parkedchan} - ${CHANNEL(parkinglot)} - ${BLINDTRANSFER})
same=> n,Set(lot=10000)
same=> n,Set(pos=${ARG1})
same=> n,GotoIf($["${DEVICE_STATE(park:${pos}@parkedcalls)}"="NOT_INUSE"]?park:pickup)

same=> n(park),Set(PARKINGEXTEN=${pos})
same=> n,Park(${lot})

same=> n(pickup),NoOp(${ARG1})
same=> n,ParkedCall(${lot},${pos})

exten=> i,1,Hangup()
exten=> t,1,Hangup()
exten=> h,1,Hangup()

[hints]
exten => _10XXX,hint,park:${EXTEN}@parkedcalls
exten => _ZXXXX,hint,SIP/${EXTEN}


res_parking.conf
[general]
parkeddynamic = yes
...
[10000]
findslot => first
parkinghints => yes
parkingtime => 60
comebacktoorigin => yes
parkedcalltransfers => both
parkedcallreparking => both

context => parkedcalls
parkext => 10000
parkpos => 10001-10020

Statistics : Posted by mkozusnik • on Fri Nov 13, 2015 10:05 am • Replies 2 • Views 126
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Both inbound and outbound call dial plan (Anal /Ditl)Cards

November 14, 2015, 8:47 am
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Hello, in my asterisk (Debian) server both diguim Analogue and Digital cards are installed, i will have to configure these two card to different group in : /etc/asterisk/chan_dahdi_groups.conf>>>to make them work, already know.

Could someone provide a great link to set dial plan for both inbound and outbound calls comes from these two trunk to make them go through a specific / extension / ring groups....
I know how to set up dial plan for extension,but i am new for calls comes from trunk
I look forward to your reply Image .

Thanks !

Statistics : Posted by tella • on Sat Nov 14, 2015 10:47 am • Replies 0 • Views 12
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Both inbound and outbound call dial plan (Anal /Ditl)Cards

November 14, 2015, 8:47 am
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You want the Biz and Jobs forum. Also what do you mean by a ring group. That is a term used by FreePBX, but not by Asterisk.

Statistics : Posted by tella • on Sat Nov 14, 2015 10:47 am • Replies 1 • Views 88
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incomming call from extention , not from asterisk !!

November 11, 2015, 9:52 am
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The address that is wrong is10.23.10.2。 However this is only a technicality, as Asterisk will ignore it in this context.

The trace should have continued with a second INVITE containing authentication data based on the password. As it doesn't, it means that the device has not been configured with a password to use.

Statistics : Posted by dr.x • on Wed Nov 11, 2015 11:52 am • Replies 7 • Views 345
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incomming call from extention , not from asterisk !!

November 11, 2015, 9:52 am
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As I understand this, the remote entity is a phone, not an ITSP.

Statistics : Posted by dr.x • on Wed Nov 11, 2015 11:52 am • Replies 7 • Views 345
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incomming call from extention , not from asterisk !!

November 11, 2015, 9:52 am
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Quote:SIP/2.0 401 Unauthorized

I think he need to add the remotesecret parameter or set the old hack insecure=invite in his trunk configuration

Statistics : Posted by dr.x • on Wed Nov 11, 2015 11:52 am • Replies 7 • Views 345
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incomming call from extention , not from asterisk !!

November 11, 2015, 9:52 am
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hi david this is the full trace

agian

which ip is wrong in the header and how to correct it ?

is it 196.41.228.32 (my peer trunk ) ??
or
196.41.228.244
????

agian , if i replace the asterisk with zoiper softphone with the same network settings , im able to send and receive calls !!!

any explanation for that ?

thanks agian Mr david

cheers

Statistics : Posted by dr.x • on Wed Nov 11, 2015 11:52 am • Replies 7 • Views 345
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