|
Who Can Use this Service? |
| |
|
|
|
 Any
ITSP or enterprise with a SIP compatible IP PBX that wants to buy
quality VoIP termination services to all worldwide destinations. Our
service works best for users with a static public IP address, and
therefore is usually not ideal for home users with a NAT router. |
|
|
|
What codecs do you support? |
|
|
|
|
|
Our carrier selection has been optimized for use with the g.729
codec. If you set this as your first priority codec, you will
maximize your chances for completing calls. That said, we use
many underlying carriers for termination, and each carrier has
different capabilities. Many support g.711, but not all. The
same goes for g.723.1 and iLBC. As a result, if you set any of
these codecs as your preferred codec, a call may or may not
complete using it, depending on which of our underlying carriers
receives the call. Our softswitch will allow your endpoint and
the carriers to negotiate the mutually preferred codec. |
|
|
|
|
How can I configure my Asterisk server? |
|
|
|
|
|
Here is an Asterisk configuration example. Be sure to replace
the example domain and prefix with the domain and prefix that
was assigned to you.
in sip.conf
[PSTN]
allow=all
canreinvite=no
context=from-trunk
fromdomain=example.domain.com
host=example.domain.com
outboundproxy=sbc.domain.com
insecure=very
type=peer
in extensions.conf
[from-trunk]
exten => _1XXXXXXXXXX,1,Dial(SIP/example.domain.com/11332400${EXTEN})
Notes:
1. The example above will route US calls to our service and
prepend the prefix 11332400 to the call.
2. The resulting SIP INVITE message ’sip:1133240012127773456@example.domain.com’
should be sent to our outbound proxy ’sbc.domain.com’
3. If you would like for all destinations to be sent to us,
then modify the configuration to this: exten => _Z.,1,Dial(SIP/example.domain.com/11332400${EXTEN})
4. Remember to replace the sample values for domain and prefix
with the actual ones that were assigned to you.
5. Outboundproxy configuration syntax is: outboundproxy=domain/host[:port] |
|
|
|
|
Why do you require a dialing prefix? |
|
|
|
|
|
We offer several different routing choices: Standard, Premium,
Special. A different prefix is used for each route so that you
can easily choose among the various routing options. We find
that this level of choice is very popular among our customers,
as it allows them to mix and match the best routes for their
needs, without making route change requests to our support
staff.
for standard route : no prefix is required
for premium route prefix is : 98461350
for special route prefix is : 98462450
However, if you are unable to use a route prefix, we would be
happy to modify your account as needed. For more information,
please send an email to
support@10telecom.biz – When you apply for the first time
standard routes are by default applied to your account to change
type of routes please
see the
instruction that you can find in your web platform. |
|
|
|
|
Do I Need a SIP Username and Password? |
|
|
|
|
|
If your equipment is situated behind NAT or if you are using a
SIP domain name for authentication, you will be required to
configure a SIP username and password in your equipment.
(a) The SIP username and password are generated automatically
and sent by mail to your email address when you apply our
services |
|
|
|
|
What should I do if my payment did not get credited to my balance? |
|
|
|
|
|
Please send an email to support@10telecom.biz with details about
your payment. We will get back to you right away. |
|
|
|
|
How many concurrent call sessions can you support? |
|
|
|
|
|
There is no set number of sessions. You access to the system
will depend on usage by other customers, which varies from time
to time. However, we monitor traffic closely, and if it appears
that your traffic may exceed the available capacity, we will
contact you and ask you to utilize dedicated facilities.
As a general rule of thumb, you should consult with tech support
if you intend to send more than 150 concurrent calls. |
|
|
|
|
Do you have basic configuration guidelines? |
|
|
|
|
|
When sending traffic to TENTEL, your gateways should be
configured as follows:
Your originating gateways should send traffic to our session
border controller at 188.165.195.217
Our session border controller will accept both SIP and H.323
traffic and is a signaling proxy only. We use the standard
ports - 5060 for SIP and 1720 for H.323.
All interconnected gateways should be configured to use RFC2833
for DTMF relay.
For H.323 Gateways, FASTSTART must be enabled.
Our routing is based on e164 addresses with a unique prefix.
Therefore, please send the destination number in the format of:
PREFIX+COUNTRY CODE+CITY(or AREA) CODE+NUMBER |
|
|
|
|
How often are the rates updated? |
|
|
|
|
|
Rates are usually updated twise per month. However, there is
not set schedule and the frequency of updates may vary due to
market conditions. |
|
|
|
|
What can I do when the Special Routing fails? |
|
|
|
|
|
Special routes are offered on a best efforts basis only, and may
have limited capacity. Unlike the Standard and Premium routes,
Special routes have only one termination partner. If that
partner is unable to complete your call, the call will fail.
It is highly recommended that you create a failover route using
the Standard or Premium prefix (or your own secondary route) to
maximize the value that special routing offers.
As with all of our routes, you will receive an appropriate error
code to allow you to forward route in cases where special
routing is not available |
|
|
|
|
|
