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VoIP Singapore SIP Trunk with Asterisk

VoIP Singapore SIP account with Asterisk
voip singapore asterisk freepbx sip trunk**Click above image to view clearer
Asterisk or FreePBX is an open source server based PBX system which is widely used due to its flexibility and lower rate. AlienVoIP is running on SIP protocol, so it can be connected to any Asterisk based systems.

The Asterisk/FreePBX can make calls to PSTN network, which includes all mobile operators and fixed lines by just using AlienVoIP SIP account.

By using AlienVoIP Asterisk SIP trunk, the whole office will beneficial from low rate communication while voice quality is maintained. Meanwhile, Asterisk PBX can be configured with dial plan to meet the environment’s requirements. For example, some calls are directed to VoIP Singapore (especially oversea calls) and some are still using local providers.

Asterisk Installation & Configuration
Before configuring Asterisk SIP Trunk server, make sure that you have registered AlienVoip and your account is active with credits and at least one SIP username.

Kindly click HERE for a full and comprehensive guide for creating an account with AlienVoIP.

The guides below show you step-by-step and assuming that you already have a running Asterisk server and have also done the necessary steps mentioned above as well as agreed to our terms & conditions.

Step 1

Configure your sip.conf file.By default, the file is located in /etc/asterisk/sip.conf

[general]                                         ; General setup
bindport=5060
bindaddr=0.0.0.0
context=default
defaultexpirey=3600
maxexpirey=3600
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=1
dtmfmode=rfc2833
nat=yes
disallow=all
allow=g729
qualify=yes
registertimeout=60

; Register your AlienVoip SIP username and password to AlienVoip's SIP server
register => 605600110:password:605600110@sip1.alienvoip.com:5060/605600110

; Setup an account
[alienvoip]                                  ; Give your gateway a meaningful name
port=5060                                 ; AlienVoip uses port 5060 for voip authentication
type=friend                                ; Set this to friend
secret=password                     ; This is your password for your SIP username
username=605600110        ; This is your SIP username
fromuser=605600110          ; This is your SIP username
host=sip1.alienvoip.com     ; This is where AlienVoip's sip registration server is located
canreinvite=no
qualify=yes
nat=yes                                        ; Depending on your network condition, if your Asterisk is located behind a router,
; set it to yes, if your asterisk server has a public IP of its own, set it to no
context=INCOMING               ; If your asterisk server allows incoming AlienVoip to AlienVoip calls, give
; this a meaningfull context
disallow=all                                ; First disallow all codecs
allow=ilbc                                  ; Allow codecs in order of preference
                                   ; see doc/rtp-packetization for framing options

[101]                                              ; Register an account for your SIP client
username=101
callerid=testing <101>
secret=101
regexten=101
host=dynamic
nat=yes
canreinvite=no
type=friend
qualify=yes
context=alienvoip
disallow=all
allow=ilbc
allow=gsm
dtmfmode=rfc2833
canreinvite=no

Step 2

Then, configure your extension.conf file to use the SIP user you have just defined in sip.conf. By default, it is located in /etc/asterisk/extensions.conf

[alienvoip] exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/alienvoip/${EXTEN},60)
exten => _X.,n,Hangup()

The example above is just to give you a general picture of a basic dial plan in Asterisk, you can also configure your dial plan to suit your production server’s requirements.

Once you have done the above configurations, your server is now ready to make calls to AlienVoIP.

Kindly contact us if you need any support in configuring Asterisk SIP trunk at your PBX.

VoIP Singapore offers a wide range of VoIP Hardware and Packages. Do check out our Cheap VoIP Singapore Call Rates for your desired countries. If you have further inquiry about VoIP Singapore, please do not hesitate to Contact Us.

VoIP Singapore SIP Trunk with Asterisk September 10, 2014

Web ASP Sdn. Bhd. (832475-W)

Unit 1-3-31, i-Avenue,
No. 1, Jalan Tun Dr Awang,
11900 Bayan Lepas,
Penang, Malaysia.

No.B-2-7, Block B,
Kuchai Exchange,
Jalan Kuchai Maju 13,
Kuchai Lama 58200,
Kuala Lumpur, Malaysia.

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Tel 1: +603 2780 3880
Tel 2: +603 7980 1388
Fax: +603 7980 2388

Penang: +604 642 0621
Fax: +604 6115620

Singapore Sales Representative

KM Ho
Tel: +6012 687 2201
sales@mobiweb.com.my