VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk

Sumber: http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx

In my previous article we configured Asterisk with some SIP-devices, and created a basic dialplan so that they could dial eachother. We also created two additional extensions for test purposes. This time I will show you how to configure a SIP trunk, and add extensions in the dialplan so that the telephones can dial out through the trunk. And if you also have a telephone number (DID) associated with the trunk, for others to be able to dial your phones, through your Asterisk PBX.

[myphones]

; Call POTS numbers through Foo Provider (any number longer than 5 digits starting with 9)
exten => _9XXXX.,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} through Foo Provider)
exten => _9XXXX.,n,Dial(SIP/fooprovider/${EXTEN:1},60)
exten => _9XXXX.,n,Playtones(congestion)
exten => _9XXXX.,n,Hangup()

There are a couple of things that might need explanation in the above. We use the Dial() application again, to dial the number we entered in our phone, but “${EXTEN:1}” uses the entered number, after the first digit, that is the meaning of “:1″. “60″ is the number of seconds to let it ring, until we give up and let Asterisk play congestion tones to us, increase the time value if you think it is too short. You also exchange “fooprovider” with the name of your real provider that you configured in sip.conf.

Test it

For our configuration to take effect we either have to reload it from Asterisk’s command-line interface, or restart Asterisk. To reload the SIP configuration and the dialplan, connect to the running Asterisk’s command-line:

asterisk -vcr

And run:

sip reload
dialplan reload

Verify that your Asterisk server registers with your provider correctly:

sip show registry

If necessary, troubleshoot the registration, use the following Asterisk CLI commands:

sip set debug on

Now at last, test the configuration. Dial your Asterisk server from your mobile phone, and hopefully your first SIP telephone will ring. Also watch the Asterisk console and see the Log() notice that we added appear and make you smile.

If that works, proceed with dialing out to your mobile phone from any of your configured and registered SIP phones, remember to dial 9 in front of the actual phone number.

While the call is going on, run the following command to see the two channels that are created, and switched together in your Asterisk: One channel to/from your SIP phone, and one through your trunk, to your mobile phone:

core show channels

Full example reference configuration files

Here are the full contents of sip.conf and extensions.conf, from the previous article, with the configuration from this article added, making up a fully working, basic, but yet complete Asterisk configuration.

sip.conf:

[general]
context=incoming

allow=ulaw
allow=alaw
allow=gsm

; Register and get calls from Foo Provider, to our number 1-555-455-1337
register => 15554551337:password123@sip.provider.foo

[fooprovider]
type=friend
secret=password123
username=15554551337
host=sip.provider.foo
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
insecure=port,invite
fromdomain=sip.provider.foo
context=incoming 

; ------------------------------------

[1000]
type=friend
secret=replacethis123
dtmfmode=rfc2833
callerid="First Phone" <1000>
host=dynamic        ; The device must always register
canreinvite=no
; Deny registration from anywhere first
deny=0.0.0.0/0.0.0.0
; Replace the IP address and mask below with the actual IP address and mask
; of the computer running the softphone, or the address of the hardware phone,
; either a host address and full mask, or a network address and correct mask,
; registering will be allowed from that host/network.
permit=192.168.1.0/255.255.255.0
context=myphones

[1001]
type=friend
secret=replacethis321
dtmfmode=rfc2833
callerid="Second Phone" <1001>
host=dynamic        ; The device must always register
canreinvite=no
; Deny registration from anywhere first
deny=0.0.0.0/0.0.0.0
; Replace the IP address and mask below with the actual IP address and mask
; of the computer running the softphone, or the address of the hardware phone,
; either a host address and full mask, or a network address and correct mask,
; registering will be allowed from that host/network.
permit=192.168.1.0/255.255.255.0
context=myphones

Referensi

• http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx

Lebih Dalam

• VoIP: Trunk
• VoIP: Asterisk menerima Anonymous Call
• VoIP: Asterisk route ke arah IP PBX lain dengan kode area
• VoIP: Cara Mengkonfigurasi Trunk di Asterisk
• VoIP: Cara Mengkonfigurasi Dial Out Melalui Trunk di Asterisk
• VoIP: Asterisk forward call ke IP softswitch lain

Asterisk Round Robin

• VoIP: Astersk Dial Round Robin
• VoIP: Asterisk pakai GotoIf

OpenSIPS

• VoIP: OpenSIPS route ke arah Asterisk
• OpenSIPS: Rewrite URI
• OpenSIPS: Rewritehostport

OpenSIPS round robin

• OpenSIPS: dispatcher

Pranala Menarik

• VoIP
• OpenBTS

Latar Belakang

• Menjadikan VoIP dan 4G Legal
• Sekitar VoIP Rakyat
• VoIP: Dasar Hukum Internet Telepon
• VoIP: Beberapa Skenario Topologi
• VoIP: Pilihan Teknologi Internet Telepon
• VoIP: Pengkodean Suara di Jaringan Komputer
• VoIP: Konsep Video Conference

Untuk Pemula

• VoIP: Kebutuhan Peralatan dan Software
• VoIP: Internet Telepon PC ke PC

Untuk Peneliti / Pencoba

• VoIP: Bandwidth Internet Telepon
• VoIP: Softswitch / Server Internet Telepon
• VoIP: Repository Software Internet Telepon
• VoIP: Menghubungkan PSTN dan Selular

Untuk Operator

• VoIP: Server Video Conference
• VoIP: Software dan peralatan client Internet Telepon
• VoIP: Penggunaan DAHDI
• VoIP: Hardware Client VoIP
• VoIP: Hardware Server VoIP
• VoIP: Interkoneksi dan Alokasi Nomor Telepon
• VoIP: Peering Antar Operator VoIP
• VoIP: Menghubungkan PSTN dan Selular
• VoIP: Trunk

Topik Lanjut

• VoIP: ENUM untuk pengenalan nomor telepon di Internet Telepon
• VoIP: Teknik Evaluasi Internet Telepon
• VoIP: Troubleshooting
• VoIP: Video Conference Server

Buku Teknologi VoIP

• Onno W. Purbo, "VoIP Cikal Bakal Telkom Rakyat", Infokomputer, 2007.
• http://125.160.17.21/speedyorari/index.php?dir=ebook-voip
• http://opensource.telkomspeedy.com/speedyorari/index.php?dir=ebook-voip