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Always a step ahead
We offer you specialized services and products tailored to your needs at very low prices, satisfying all your telecommunications habits, putting you at the heart of broadband telecommunications. To always be one step ahead.
Our services include cutting-edge features
Such as: Unlimited Outgoing Calls, International Numbering, Multiple Incoming Calls with one DID Number, HD Voice & Video, Unified Communications and more..
Cloud vPBX
Get off your communications literally with a call center in the Cloud. Virtual PBX solutions from 6 to 25 Extensions with added value Features such as Call Recording, IVR, Call Queue, Call Center, Standby Music, Smartphone Client, FAX2mail, Transfer (Announced and Blind), Bridges, CDR, Voicemail, Follow with, Call forwarding, Time base routing, Call Block, Conferences and many more
Cloud vPBX
Get off your communications literally with a call center in the Cloud. Virtual PBX solutions from 6 to 25 Extensions with added value Features such as Call Recording, IVR, Call Queue, Call Center, Standby Music, Smartphone Client, FAX2mail, Transfer (Announced and Blind), Bridges, CDR, Voicemail, Follow with, Call forwarding, Time base routing, Call Block, Conferences and many more
Stay Connected
Extend your organization’s Communications enable a BYOD solution by using your famous sip client app, or save money using our hosted voice service – vPBX puts business-class communications tools in your pocket so no matter where you are in the world, you stay connected.

Setting instructions

TrixBox / Elastix / AsteriskNOW / FreePBX/Issabel

Sip Trunk settings for TrixBox/Elastix/AsteriskNOW/FreePBX


SIP Server: Outgoing

Outbound Caller Id

XXXXXXXXXX

Never Override CallerID

Checked

Trunk Name

Company Name

PEER Details

username=your sip This email address is being protected from spambots. You need JavaScript enabled to view it..4

fromuser=your sip This email address is being protected from spambots. You need JavaScript enabled to view it..4

type=peer

secret=sip device password

host=46.4.60.4

port=5060 or 5070

disallow=all

context=from-trunk

allow=alaw

qualify=yes

canreinvite=no

sendrpid=yes

dtmfmode=auto

USER Context

(Leave this field blank)

USER Details

(Leave this field blank)

Register String

your sip device:sip device This email address is being protected from spambots. You need JavaScript enabled to view it..4/your sip device



Press Submit Changes, Apply Configuration Changes, the asterisk server must be registered.


Troubleshooting - Known Issues

You can see the registration status for asterisk by running the command line:

    /usr/sbin/asterisk -rx "sip show registry"

If the registration has not been successful:

  1. Try setting the port=5070

 

 

SIP Server: Incoming

PEER Details

username=your sip This email address is being protected from spambots. You need JavaScript enabled to view it..4

type=user

secret= sip device password

host=46.4.60.4

context=from-trunk

allow=all

insecure=port,invite

USER Context

(Leave this field blank)

USER Details

(Leave this field blank)

Register String

your sip device:sip device This email address is being protected from spambots. You need JavaScript enabled to view it..4/your sip device


Products and services

  • Our Softphones with Tapi Support

    Our Software phones are compatible with every VoIP telephone system.

  • Web Meeting

    Web Meeting enables you to share your audio, slides, chat, video, and desktop with Participants.

  • Cloud vPBX

    Virtual PBX Solutions from 2 to 100 Extensions with Value added Features

  • SIP Trunks

    SIP Trunks from 2 up to unlimited Channel per sip Trunk depending on your internet connection bandwidth

DID numbers in more than 50 Countries