The Asterisk pbx

Our services (making and receiving calls) are compatible with your Asterisk server. We can only provide you with a trunk sip and some configuration advice. If you need a specialist to configure your Asterisk, you must visit our category dedicated to VoIP integration. You will find voip specialists available for any type of voip integration...

Yes, you can forward calls from one or more numbers to one or more IP addresses of your choice. In this case, you must place an order and once the numbers are active, ask us for the activation of the IP forwarding and we will do the necessary setting. Please note that, if you wish to change this configuration in the future, you must notify us, otherwise, the line will not work...

Asterisk PBX is undoubtedly one of the best solutions to answer the needs of many companies that want to migrate to IP telephony for its attractive costs and innovative features. Easy to install, configure, manage, consuming very few memory resources, it is easy to integrate with other systems, it is compatible with several protocols and codecs available in the market. Choosing Asterisk is really betting on the evolution of your telephone system.

The only drawback is that you need to have à server, install Asterisk, and have the technical knowledge to do the job. On our website you will find independent technicians, who offer their services for the configuration of Asterisk. But, if you are looking for a complete virtual PBX without a server or installation, choose the Hivoox virtual switchboard...

Configure your Asterisk PBX server through Digest authentication. This scenario is used when one would like to use digest authentication with Asterisk.
You will have to use regular SIP account like User=878xxxxxxx and password=xxxxxxxx.

In this scenarios Asterisk configuration should be a bit more complex.

[general]

register => 878xxxxx:passwordxxxx@sip_proxy/my_incoming_extension

useragent => PBX
[sip_proxy]
type=friend
authuser=878xxxxxxx
username=878xxxxxxx
fromuser=878xxxxxxx
secret=passwordxxxx
host=87.238.224.117
qualify=yes
insecure=very
canreinvite=yes

Note, that it is important to set useragent parameter in [general] section as shown above, because otherwise our system will try to apply ip authentication instead
of digest. ‘register’ parameter in [general] section configures what to do with the incoming calls, in the example above such calls will be sent to context my_incoming_extension...

Configure your Asterisk PBX server through IP authentication. This type of authentication is used when you have static ip address and this is default scheme.
You need to give us your static ip address.

sip.conf:
[sip_proxy]
type=peer
host=87.238.224.117
disallow=all
allow=g729
nat=no
context=ringring
usereqphone=yes
outboundproxy=87.238.224.117

Then you may use this configuration in your dialplan anyway you want, for example to send all calls in the context you may do the following:

extensions.conf:

[ringring]
exten => _X., 1, Ringing()
exten => _X., 2, Wait(1)
exten => _X., 3, dial(SIP/${EXTEN}@sip_proxy)...

By using our SIP trunk, you can freely manage the CLI (number displayed when you make a call). By default, we configure the CLI with the number subscribed with us or another number of your choice. However, you can administer the CLI directly from your server.

If you use a sip trunk and want to configure the CLI, just let us know so that your account is configured accordingly. In this case, the configuration of outgoing calls will depend exclusively on your server. We do not provide support to configure your server...