What is Voice over Internet Protocol (VoIP)?

Voice over Internet Protocol (VoIP) is a technology used to transmit voice communications through IP packets via the Internet.

What Gateway vendors do you support?

We support most Gateway vendors and protocols.  Our network has the ability to meet most any of your connection requirements.

 

Yes, we do support Asterisk open source PBXs!.

 

Do you have a sample configuration for Trixbox?

Trixbox GUI Trunk Setup Entry - Inbound Service

1. Create an inbound route for your DID pointed to the function you are trying to accomplish.
2. Create a SIP trunk. SIP trunk details below. No registration is used.

Trunk Name: ipcomms
USER Context: from-pstn
USER Details:
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
host=64.154.41.100
insecure=very
nat=yes
type=friend

3. Test DID.

TalkinIP GUI Trunk Setup Entry - Outbound Service

Trunk Settings

[***enter username***]

username=***enter username***
type=peer
secret=***enter secret***
nat=yes
insecure=very
host=sip.talkinip.net
fromuser=***enter username***
dtmfmode=rfc2833
canreinvite=no
allow=ulaw

Registration Statement
register=username:password@sip.talkinip.net/username

Do you have configuration samples for Asterisk?

Below are some configuration examples for the open source PBX Asterisk ^TM . These examples may vary depending on your software versions, implementation type, and more.  These are examples taken from most common customer configurations.

 

SIP Trunk Sample Entry - Inbound Service - g711ulaw
(Listed below is an example of what you would put in your SIP Trunk entry.)

Create a SIP trunk for g711ulaw.

Trunk Name: ipcomms

PEER Details:

allow=ulaw
context=from-pstn
dtmfmode=rfc2833
host=64.154.41.100
insecure=very
nat=yes
type=friend

USER Context: from-pstn

USER Details:

allow=ulaw
context=from-pstn
dtmfmode=rfc2833
host=64.154.41.100
insecure=very
nat=yes
type=friend
____________________________________________________________________________________________________________________________

SIP Trunk Sample Entry - Inbound Service - g729
(Listed below is an example of what you would put in your SIP Trunk entry.)

Create a SIP trunk for g729.

Trunk Name: ipcomms

PEER Details:

allow=g729
context=from-pstn
dtmfmode=rfc2833
host=64.154.41.100
insecure=very
nat=yes
type=friend

USER Context: from-pstn

USER Details:

allow=g729
context=from-pstn
dtmfmode=rfc2833
host=64.154.41.100
insecure=very
nat=yes
type=friend

What do I do if my asterisk server has a private IP Address?

Please enter the following two settings (externip and localnet) at the top of your sip.conf file in the general section.  If public, nothing needs to be added to the general section. Depending on your setup, you might need to forward all ports to your Asterisk server's private IP address or put the private IP address of the Asterisk server on the DMZ.

externip=WAN IP address

localnet=192.168.1.0/255.255.255.0 (or whatever your private LAN is configured for)

Do you have configuration samples for Cisco Gateways?

Below are some configuration examples for Cisco gateways. These examples may vary depending on your software versions, implementation type, and more.  These are examples taken from most common customer configuration. Depending on your gateway version some commands may differ.

Sample POTs Entry - Inbound Service

 

!

dial-peer voice 6781475 pots

 description IPC

 huntstop

 destination-pattern 6784601475

 no digit-strip

 port 0:D

!

Sample VoIP Entry - H323 - Outbound Service

!

dial-peer voice 1234 voip

 description IPC

 huntstop

 destination-pattern . or .T

 progress_ind setup enable 3

 session target ipv4:xxx.xxx.xxx.xxx

 dtmf-relay rtp-nte h245-signal h245-alphanumeric

 codec g729r8

 tech-prefix yyyyyyy

!

xxx.xxx.xxx.xxx is the IP address of IPC's destination gateway.  yyyyyyy is the

prefix assigned to your account.

 

Sample VoIP Entry - SIP - Outbound Service

!

dial-peer voice 1234 voip

 description IPC

 huntstop

 destination-pattern yyyyyyy. or yyyyyyy.T

 progress_ind setup enable 3

 session protocol sipv2

 session target ipv4:xxx.xxx.xxx.xxx

 dtmf-relay rtp-nte h245-signal h245-alphanumeric

 codec g729r8

!

xxx.xxx.xxx.xxx is the IP address of IPC's destination gateway.

yyyyyyy is the prefix assigned to your account.

What Protocols and CODECs do you support?

Depending on your specific requirements, we support several delivery methods (VoIP protocols).  Below is a list of our supported delivery methods, and the associated Codec each supports.

 

H323  (g729,   g711ulaw,   g711alaw)

SIP     (g729,   g711ulaw,   g711alaw)

IAX    (g711ulaw,   g711alaw)

 

Can all of my phone numbers be sent to the same IP Address?

Yes.  With  most services we can point your phone numbers to one IP address.