Download > Installation > Configuration

Download:

Download the LTP Channel here

create a ltp user with www.spokn.com

download the LTP channel to /usr/src/ on your linux system

Note: You should have the asterisk software installed before you can install the LTP channel, for instructions to install asterisk please check www.asterisk.org or www.voip-info.org, you will also need a ltp userid, regsiter a ltp userid with the following service providers

Installation:

extract the file using tar
tar -zxf chanltp.tar.gz

change directory to /usr/src/ltp
cd ltp

run make to compile the ltp channel
make

run make install to copy the ltp channel to the asterisk modules folder /usr/lib/asterisk/modules
make install

if you get a libgsm incompatible error copy the libgsm from your asterisk src /usr/src/asterisk-1.2/codecs/gsm/lib/libgsm.a to the ltp folder
cp /usr/src/asterisk-1.2/codecs/gsm/lib/libgsm.a /usr/src/ltp

Configuration:

create a ltp.conf file in /etc/asterisk in any favorite editor most common is vi

the following parameters are to be included in the conf file
vi /etc/asterisk/ltp.conf

;add a general context to the file at the start
[general]

;your ltp userid as registered with LTP server
userid=ltpuserid

; your ltp password
password=ltppassword

; context in extensions.conf
context=ltpcontext

;extension in the context where the ltp incoming call should hit, this can be any valid extension or your ltpuserid itself
extension=ltpextension

; the LTP server IP for the the ltp server you are using for spokn.com its 212.187.211.5
server=0.0.0.0

;your local IP on the system for the ltp channel to bind to, the channel listens on port 4060
localip=0.0.0.0

If you are using a firewall open port UDP port 4060 for ltp channel


Edit extensions.conf

now we need to edit the extensions.conf file and add a context for the ltp channel
vi /etc/asterisk/extensions.conf

here you can do something with the incoming call on ltp like sending it to your phone over SIP or LTP or IAX or let a AGI script handle the call, in the below example we are sending the incoming ltp call to ring on a PSTN network through SIP

[ltpchannel]
_X.,1,Answer ; answer the incoming call
_X.,2,Dial(SIP/18472016170@sipsever) ; bridge the call with SIP to call a pstn number
_X.,3,Hangup() ; finally hangup

note: the sipserver is defined in the sip.conf file

restart your asterisk server to load the ltp channel

note: if you make changes to the ltp.conf file you will need to restart asterisk , a reload wont work