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lunes, 25 de julio de 2016

freepbx 13 centos 6

[root@FreePBX-13 freepbx]# ./install -n --dbpass 123456

Feature Code Call Transfers

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Overview of Feature Code Call Transfers

A call transfer is when one party of a call directs Asterisk to connect the other party to a new location on the system.
Transfer types supported by the Asterisk core:
  • Blind transfer
  • Attended transfer
    • Variations on attended transfer behavior
Transfer features provided by the Asterisk core are configured in features.conf and accessed with feature codes.
Channel driver technologies such as chan_sip and chan_pjsip have native capability for various transfer types. That native transfer functionality is independent of this core transfer functionality. The core feature code transfer functionality is channel agnostic.

viernes, 15 de julio de 2016

how to provisioning polycom

Getting Started

This is a publicly accessible server intended only for use of provisioning, upgrading, or downgrading the software of Polycom IP phones with stock Polycom software and configuration files.

Multiple 'stock' directories exist with different versions of SIP, UC, and BootROM software, and are named according to the SIP or UC software version contained within. These directories are completely stock; no changes should be made to any configuration files within.

Each directory contains an ABOUT.TXT to indicate which SIP and BootROM software versions exist within that directory.
To Provision Your Polycom VoIP Phone:
1.       From the phone's local menu interface, press Menu > Settings > Advanced (default password: 456) > Administration Settings > Network Configuration > Server Menu.
2.       Configure your DHCP menu:
On 3.x Phones: Main Menu > Settings > Advanced > Admin Settings > Network Configuration > DHCP Menu
On 4.x/5.x phones: Main Menu > Settings > Advanced > Admin Settings > Network Configuration > Provisioning Server > DHCP Menu set the following:
Boot Server: Static
BootSrv Type: IP Address������ (Once phone is upgraded, please remove the IP address from the phone!!!)
3.       From the Provisioning Server menu set:�� Server Type: HTTP
4.      Next input Server Address as or IP�� Example to load the latest SIP 4.06 =
a.       Note can be entered in IP form, enter:� Example for convenience.

martes, 5 de julio de 2016

freepbx queuegosub

nano /etc/asterisk/globals_custom.conf



nano /etc/asterisk/extensions_custom.conf

exten=>s,1,Noop(var values are ${CALLERID(num)} ${EXTEN} ${CDR(src)} ${CDR(dst)} ${CDR(clid)} ** ${mycid}** //${AMPUSERCID}// )

sábado, 2 de julio de 2016

orginate syntax

asterisk-dominicana*CLI>originate SIP/trunk/13052361010 extension 700@pstn

llama al numero 13052361010  lo transfiere al contexto pstn en la extension  700