Configuration Samples

Voxbone provides call forwarding via SIP to an IP address or a SIP URI.

After you have been provisioned a DID from us you need to route the calls to your SIP equipment.

You also need to configure your equipment to accept calls coming from the Voxbone IP addresses.

The current list of IP addresses is:

You will find below typical configuration sample for famous SIP equipments compatible with Voxbone.

1. Asterisk
2. Brekeke OnDO PBX
3. Cisco
4. CommuniGate Pro
5. Dialexia
6. IVR Technologies, Inc.
7. Portaswitch
8. SER
9. SipX
10. Voipswitch

Add the following to /etc/asterisk/sip.conf:

[81.201.82.20]
host = 81.201.82.20
type = friend
insecure = very
context = your-context
canreinvite=no

[81.201.82.21]
host = 81.201.82.21
type = friend
insecure = very
context = your-context
canreinvite=no
...
( Do it for all voxbone IPs address listed above )

Replacing 'your-context' with an existing context in /etc/asterisk/extensions.conf.

The 'insecure = very' allows all calls from this IP address.

In extensions.conf, then add a line to route the DIDs as you wish. For example, to route 1234567890 to SIP phone 100:

exten => 1234567890, 1, Dial(SIP/100)

Once you have edited sip.conf and extensions.conf, load the Asterisk console, and type the following commands:

sip reload
extensions reload

Users of Asterisk behind a NAT must add the 'externip' line in their "sip.conf" file :

externip=<External IP Address> ; e.g. externip=11.22.33.44 where 11.22.33.44 is the external IP address of the router.

If the external IP address is dynamic (e.g. a DSL provider handing out a different IP address every day) you will need to use a Dynamic DNS client.

Asterisk servers with both internal and external SIP clients will need to add the 'localnet' line in their "sip.conf" file :

localnet=<internal range with mask> ; e.g. localnet=192.168.1.0/255.255.255.0 where 192.168.1.0 is the local internal range

1. Register a SIP phone with OnDO SIP Server by specifying the sip proxy and
   outbound proxy address and sip user name e.g. user name=100
2. Login to your Voxbone account and add the Endpoint URI in the following format:

   sip:100@1.2.3.4

   
   Where:
   
   a) 100 = registered SIP Phone number
   b) 1.2.3.4 = OnDO IP address if it is public or your router address if OnDO SIP server is behind a router.
   
   
3. If your OnDO SIP Server is behind a NAT router you need to setup port forwarding on your router.
   The following ports will need to be forwarded:
   
   • SIP exchanger -- Local Port [UDP]
   • Default value: 5060
   • RTP exchanger -- From Minimum Port to Maximum Port [UDP]
   • Default value: 10000-10999
   
   
   You also need to set router's IP address on the OnDO SIP Server configuration page:

   Interface Address # -- Your router's global IP Address

You need to create an incoming dialpeer as follows :

dial-peer voice 100 voip
description ## incoming calls from Voxbone ##
incoming called-number 123456789
voice-class codec 1
session protocol sipv2
session transport udp
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs5 signaling
no vad

where '12345678' is your DID, formatted as you wish from the Voxbone web config panel.

Then you need to route the calls towards the PSTN with an outbound dial peer, and possibly changing the DNIS or ANI/CLI with translation rules.

If you have an access-list on the cisco gateway, do not forget to allow the traffic coming from the Voxbone IP's.

Connect to your CommuniGate Pro web administration interface.

If your subscribers will exist on your Local Area Network, you can define the local area network range of IPs under:

you can then proceed to:

and enable "Process LAN IP Addresses as Clients".

If your subscribers are on a separate network, enter the range of IP addresses under "Client IP Addresses" that you wish to treat as local subscribers for your CommuniGate Pro voice network.

You can then control relaying for SIP under:

The default setting of "Relay to Non-Clients for: clients" will allow the Voxbone DIDs to ring your local subscribers.

Create an SIP endpoint URI on Voxbone that routes your DID to the CommuniGate Pro account or PBX attendant you wish to receive the call.

Forward your Voxbone DID to the CommuniGate Pro you have created with "unmodified" as the URI Formatting option.

First open an ITSP carrier account.
Setup the Voxbone IP's in the "gateway" menu.
Go to the "DID pool" menu and generate the blocks of DIDs assigned by Voxbone.
Now you can assign the list of DIDs to an agent or reseller, to be assigned later to a SIP subscriber.
Or if you want to resell the DIDs to another ITSP go to the "ITSP" menu, create an ITSP client account and assign the desired DIDs with the right rate.

Talking SIP is a fully integrated application, media, real-time billing, location, and registration server provided in a single, cohesive, all-in-one turnkey solution.
Talking SIP is fully interoperable with Voxbone's services and provides revenue generating enhanced services and real-time billing for next-generation networks.

The following are just some of the many services offered by Talking SIP:

• Voice over Broadband Services
• Hosted Voice Mail and PBX
• Prepaid/Postpaid Calling Card
• DID, Web, GPRS, smartphone, and SMS Callback
• One Number Locator (Find Me/Follow Me Services)
• Reminder and Wake Up Services

1. Add one or more Devices (System Management|Devices followed by Edit|Add or just right click on the grid and select Add from the popup menu) for the Voxbone servers, specifying an IP address, CIDR string (used for a range of IP addresses) or an FQDN for each entry.
2. Create a Device Module Map entry for each Device created in step 1. (System Management|Device Module Map followed by Edit|Add or just right click on the grid and select Add from the popup menu) and check 'By DNIS' if you wish to provision each access number individually and/or specify a catch-all default module to service all calls with a particular module if a matching DNIS entry cannot be found in step 3.
3. If you have checked the 'By DNIS' option in step 2. then add a DNIS Module Map entry for each provisioned Voxbone DID (System Management|DNIS Module Map followed by Edit|Add or just right click on the grid and select Add from the popup menu) and specify the desired service module that you wish to override from the default that may have been created in step 2.

The sequence of actions to configure your DIDs with PortaSwitch is the following:

1. VoxBone gateways need to be configured in PortaSwitch to be authorized by IP address. To achieve this, type the following command at Master server command prompt:

   shell> mysql -u root porta-sip
   After that, type at MySQL prompt:
   
   a) for MR10:
   insert into check_ua values ('^Voxbone Callcontrol.*', 5);
   b) for MR11:
   insert into ip_auth values ('remote_ip', '^213\.246\.57\.33$', NULL);
   insert into ip_auth values ('remote_ip', '^213\.246\.216\.81$', NULL);
   insert into ip_auth values ('remote_ip', '^213\.246\.216\.82$', NULL);
   insert into ip_auth values ('remote_ip', '^213\.246\.216\.90$', NULL);
   insert into ip_auth values ('remote_ip', '^213\.246\.216\.91$', NULL);
   insert into ip_auth values ('remote_ip', '^213\.246\.216\.92$', NULL);
2. Create a tariff that describes the termination cost of your incoming calls.
3. Create a vendor associated with VoxBone DIDs.
4. Under that vendor, create five accounts with logins:
   213.246.216.81
   213.246.216.82
   213.246.216.90
   213.246.216.91
   213.246.216.92
   and password 'cisco'. Choose any convenient name.
   
5. Go to the newly created vendor's connections page and add three connections of 'VoIP from Vendor' type, with the following parameters:
   
   Node = your PortaSIP node;
   Transl. rule - leave empty
   Account = one of the accounts created at the previous step
   Tariff = tariff created at step 2
   Description - chose a descriptive name for a connection
   Capacity = maximum number of simultaneous calls for the DID (e.g. 2).
6. Press Save.

After that, your system is ready to accept incoming VoIP DID calls from VoxBone.

If you use SIP Express Router (SER), you should proxy all calls from the Voxbone IP addresses without asking for authentication. The exact method you use depends on your ser.cfg and how you wish to route calls. For example, if you wish to proxy calls to a fixed IP address, such as Asterisk:

Since sipX is a SIP proxy no specific configuration is required for sipX to receive SIP calls. The only requirement is that the user name that is called in the SIP URI has to exist in sipX. E.g. if you forward calls from Voxbone to operator@yourdomain.com, then the user "operator" has to exist in the sipX system.

1. Forward calls to a user line.
   A user is configured using the Web interface of the sipX Configuration Server (sipXconfig). The user ID can be an alpha-numeric string or an extension number. That user is then assigned to one or several devices again using the Web interface. Upon calling the user all devices to which this user is assigned will ring. User specific forwarding can be configured if necessary (parallel or serial forking). More information is available here: http://sipx-wiki.calivia.com/index.php/How_to_configure_Users.

2. Forward calls to an auto-attendant.
   Calls that originate with Voxbone can also be directly sent to an auto-attendant defined in the sipX system.   Either the default auto-attendant ("operator") or a custom auto-attendant can be used. More information is available here: http://sipx-wiki.calivia.com/index.php/HowTo_configure_the_sipX_Auto_Attendant

3. Forward calls to a hunt group.
   Similar to forwarding to an auto-attendant, calls can be handled by a hunt group. More information is available here: http://sipx-wiki.calivia.com/index.php/Call_Groups_and_Hunt_Groups_Configuration

Firewall / NAT Traversal.

We were able to test calls from Voxbone to sipX with the following firewall / NAT solutions:
a) Intertex IX-67 (http://sipx-wiki.calivia.com/index.php/HowTo_use_Intertex_IX67_for_Firewall/NAT_Traversal)
b) Ingate SIParator (http://sipx-wiki.calivia.com/index.php/HowTo_configure_InGate_SIParator_to_work_with_sipX)

If no SIP aware firewall / NAT is available, you can open the necessary ports by defining the sipX host as an exposed host. Port 5060 has to be opened for SIP signaling traffic to the sipX host. The RTP port range depends on the phone you use as media traffic does not go through the sipX host.
E.g. the default RTP port range for Polycom phones starts at 2222. If you prefer using the "exposed host" feature then the phone needs to be exposed for RTP traffic. Please consult the manual of your phone as well as the manual of your firewall / NAT to find out more about how it handles SIP.

In 'GW clients' menu you need to create a new client, with the following parameters�:
In 'login' and 'password' you put whatever you want.
In 'IP numbers' you need to add all the Voxbone IP addresses.
Check the box 'PIN source' if you want to use the voipswitch IVR with our DIDs
Check the box 'active'
Uncheck the box 'Authorize by login/password'
In 'tariff' choose a special tariff such that you do not bill calls coming from Voxbone.
Add some funds to the account