YATE: Install di Ubuntu

Sumber: http://yate.null.ro/pmwiki/index.php?n=Main.GettingStartedOnUbuntu

Instalasi

Instalasi menggunakan perintah

apt-get install yate php5-cli

php atau python dibutuhkan untuk dialplan (menjalankan urutan step dan menerima input dari user) tidak menggunakan extensions.conf seperti di asterisk.

Running & Configuring

Cara paling gampang untuk menjalankan & debug, saat setup adalah

yate -vvvv

When you make changes to the configuration files (located in /etc/yate, you'll need to hit CTRL-C and re-run.

Once you get things configured, you need to edit /etc/default/yate and set NO_START=0. Then you can start with /etc/init.d/yate start and stop with /etc/init.d/yate stop. When in this mode, output is logged to /var/log/yate.

Connecting Softphones

For Yate to be setup as a basic PBX, it must be setup as a registrar and we must make accounts for our phones. If you want to have the phones in a database, you'll need the ubuntu package yate-pgsql or yate-mysql, but if you just want to get started with a simple file based setup, you can set them up in /etc/yate/regfile.conf. Just scroll to the bottom and add:

[101] password=welcome

[102] password=welcome For ekiga, goto Edit | Accounts | Add and fillout the following fields Account Name: yate (or whatever) Protocol: SIP Registrar: IP address or DNS of the server running Yate User: 101 (or 102 - just match bracketed username in regfile.conf) Password: welcome (or whatever you set)

Once you create the account you can click default so that way you can make calls by typing 101 or 102 rather than 101@server.

If you're using Ekiga on the same machine running the server you need to change the port it listens to. You can do this by running gconf-editor and navigating to /apps/ekiga/protocols/sip/listen_port. I set mine to 5061 and it works fine with Yate (and asterisk though you need to add port=5061 in sip.conf).

As you change things and restart yate, Ekiga may think it's registered when it is not, so it's a good idea to keep that accounts dialog open and uncheck and recheck the account to get it to re-register.

At this point, you should be able to start yate, have the phones register and make a test call.

Routing Calls

Routing calls is done in /etc/yate/regexroute.conf. It's similar to asterisk's extensions except there's no concept of sequencing steps other than calling a php or python script and having that do the work. You may have already noticed that the calls to the phones work with out configuring anything. The registered users actually have a priority associated with them and they may or may not take precedent over this routing configuration.

There are already some examples, and I suggest you try them. 99991003 will play ringing. When you look in the file, you'll see the corresponding entry.

^99991003$=tone/ring

All the routing is done with Regular Expressions. Each extension can map to a module just like tone is mapped to the Tone Generator Module.

Here are some examples:

^1\?\([0123456789]\{10\}\)$=sip/1\1;line=callwithus;caller=15555555555 ; Call out to the PSTN. See Section below.

this defines a meeting room called sales and immediately puts everyone in here.

Note there is no voice prompt or password functionality included.

You'll need php or python for that.

^701$ = conf/sales

^1234$=fork sip/101@localhost | external/nodata/leavemail.php ; voicemail The PSTN

See ConnectingToFWD

Voicemail

FIXME

Music on Hold

FIXME

Transferring calls

FIXME

Conferencing

Say you want people to be able to call to extension 701 to join a conference. With

^701$ = conf/sales

Referensi

• http://yate.null.ro/pmwiki/index.php?n=Main.GettingStartedOnUbuntu

Lebih Detail

• YATE
• YATE Softswitch
• OpenBTS: Yate Softswitch
• Instal YATE
• YATE: Compile
• YATE: Install di Ubuntu
• YATE: Menjalankan
• YATE: Konfigurasi Dial Plan
• YATE: Konfigurasi User