Mostra la font per a Configuracio FreeSwitch

[[Category:VOIP]]
== links interessants ==
https://plone.lucidsolutions.co.nz/linux/freeswitch/install-freeswitch-on-a-centos-v6-vm

https://wiki.freeswitch.org/wiki/Dialplan_XML

https://confluence.freeswitch.org/display/FREESWITCH/Interoperability

http://www.fredshack.com/docs/freeswitch.html

== Instalació ==
<pre>
rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-0.noarch.rpm   # afegim repo
yum install freeswitch                                                   # freeswitch base
yum install sox freeswitch-sounds*                                       # eines per tractar audio + sons per defecte
yum install freeswitch-lang*                                             # config idiomes
yum install freeswitch-freetdm                                           # interficie amb dispositius de telefonia (TDM) (opcional)
yum install freeswitch-spidermonkey                                      # soport per apps javascript (opcional)
yum install freeswitch-codec-*                                           # codecs audio
yum install freeswitch-application-*                                     # apps natives del freeswitch
yum install freeswitch-format*                                           # soport de diferents formats de fitxer (audio, seq tons,....)
yum install freeswitch-xml-*                                             # eines per CDR via xml i comms xml via http
chkconfig --add freeswitch && chkconfig --levels 35 freeswitch on        # activem el servei
service freeswitch start                                                 # inicialitzem el servei
</pre>

== Configuracio ==

Path: /etc/freeswitch

fitxers a tocar:
* vars.xml : definicions usades a la resta de la configuracio important establir el domini
* directory/default/*.xml : important canviar els passwords dels usuaris
* autoload_configs/modules.conf : activar xml_rpc si es vol interf web i mod sofia si es vol SIP


=== vars.xml ===
* canviat default password (linea 16)
* canviat valor variable domain = cadt.com (linea 55)

=== modules.conf.xml ===
* activat mod_xml_rpc  (linia 138)
* activat mod_xml_curl (linia 139)
* activat mod_xml_cdr  (linia 140)

== Proves ==
Instalacio de twinkle a CentOS 6
yum install twinkle <enter>

Configuracio de twinkle 
 Create profile
 . username 1000
 . password cadtboira1
 . proxy 192.168.1.95
 . Transport/NAT: NAT traversal not needed

Modificacio de regles iptables
 -A INPUT -m state --state NEW -m udp -p udp --dport 5060       -j voip-local-IN
 -A INPUT -m state --state NEW -m udp -p udp -m multiport --dports 8000:8005 -j voip-local-IN


Activacio de trace sofia
 Entra a la consola
 * fs_cli <enter>

 Activa trace per SIP
 * sofia global siptrace <on|off>

Configuracio vars.xml

 <X-PRE-PROCESS cmd="set" data="default_password=[PASSWORD]"/>
 <X-PRE-PROCESS cmd="set" data="default_vm_password=[PASSWORD]"/>
 <X-PRE-PROCESS cmd="set" data="ext_prefix=[PREFIX]"/>
 <X-PRE-PROCESS cmd="set" data="domain=[DOMAIN]"/>

 Per exemple

 <X-PRE-PROCESS cmd="set" data="default_password=cadtboira1"/>
 <X-PRE-PROCESS cmd="set" data="default_vm_password=cadtboira1"/>
 <X-PRE-PROCESS cmd="set" data="ext_prefix=10"/>
 <X-PRE-PROCESS cmd="set" data="domain=cadt.com"/>


Configuracio de servidors SIP
 
* sip_profiles/external
 
 . localphone

 <include>
  <gateway name="localphone-gw">
<!--
     <param name="contact-params" value=""/>
     <param name="extension" value=""/>
-->
    <param name="from-domain" value="localphone.com"/>
    <param name="from-user" value="[USERID]"/>
    <param name="realm" value="localphone.com"/>
    <param name="proxy" value="proxy.localphone.com"/>
    <param name="username" value="[USERID]"/>
    <param name="password" value="[PASSWORD]"/>
    <param name="extension-in-contact" value="true"/>
    <param name="retry-seconds" value="30"/>
    <param name="caller-id-in-from" value="false"/>
  </gateway>
 </include>

 . messagenet

 <include>
  <gateway name="messagenet-gw">
<!--
     <param name="contact-params" value=""/>
     <param name="extension" value=""/>
-->
    <param name="from-domain" value="messagenet.it"/>
    <param name="from-user" value="[USERID]"/>
    <param name="realm" value="messagenet.it"/>
    <param name="proxy" value="sip.messagenet.it"/>
    <param name="sip-port" value="5061"/>
    <param name="username" value="[USERID]"/>
    <param name="password" value="[PASSWORD]"/>
    <param name="extension-in-contact" value="true"/>
    <param name="retry-seconds" value="30"/>
    <param name="caller-id-in-from" value="false"/>
  </gateway>
 </include>

== TODO ==

* recepcio trucades entrants
* mode nit trucades entrants
* passar trucada
* trucades en espera
* trunk entre oficines
* musica trucada en espera ( canviar en funcio estacio)
* telefons cisco
* gateway linies analogiques
* usuaris de confianca externs
* conferencia
* trucada a ultim numero
* tornar trucada
* integracio skype
* redireccio e164 / estalvi cost trucada
* videoconferencia/cursos
* documentacio.....

== Abreviacions ==
* DAHDI: Digium/Asterisk Hardware Device Interface
* CDR: Call Detail Records
* CID: Call ID
* DID: Direct Inward Dialing. Numero extern que crida a extensio interna...
* LCR: least cost routing
* POTS: Plain Old Telephony Service
* FXO: Foreign eXchange Office. Port on es pot conectar una linia de telefon del proveidor 
* FXS: Foreign eXchange Service. Port on es pot conectar un telefon  
* snom: control tecles extra softphone
* e164: Base de dades consultable via DNS query que converteix de nº POTS a contacte VoIP
* IVR: Interactive Voice Response. menu de veu de la centraleta
* STUN: Session Traversal Utilities for NAT. Esquiva NAT important per connexio directe
* DND: Do Not Disturb
* LCR: Least Cost Routing
* DTMF: Dual Tone Multiple Frequency. Marcat per tons 
* FSK: Frequency Shift Keying. Parlant de telefonia es refereix al sistema Bell 202 per enviar el CID
* BLF: Busy Lamp Field. Lums d'activitat de les linies en el telef d'operadora
* TR069: estandard de intercanvi de info de configuracio
* e100: extensio del SIP per respostes provisionals