Your connection speed (bandwidth) must be sufficiently
high for the type of vocal codec you are using. Vocal
codecs (or just "codec") are the standard means by which
your IP Phone (UA) encodes your voice and transmits it
across the Internet. The codec that is picked for a
particular phone call is primarily determined by your
UA's configured preferences. Consult your UA's manual to
determine how to change codec preferences.
To use the best quality / highest bandwidth codecs
(G711a/u) we recommend a connection speed of no less
than 90Kb/s in each direction (up and down).
Linksys Codec Selection:
1. G.711 (A-law and mµ-law)
This very low complexity codec supports uncompressed 64
kbps digitized voice transmission at one through ten 5
ms voice frames per packet. This codec provides the
highest voice quality and uses the most bandwidth of any
of the available codecs.
2. G.726
This low complexity codec supports compressed 16, 24, 32
and 40 kbps digitized voice transmission at one through
ten 10 ms voice frames per packet. This codec provides
the highest voice quality.
3. G.729A
The ITU G.729 voice coding algorithm is used to compress
digitized speech. G.729A is a reduced complexity version
of G.729. It requires about half the processing power to
code G.729. The G.729 and G.729A bit streams are
compatible and interoperable, but not identical.
4. G.723.1
The SPA supports the use of ITU G.723.1 audio codec at
6.4 kbps. Up to 2 channels of G.723.1 can be used
simultaneously. For example, Line 1 and Line 2 can be
using G.723.1 simultaneously, or Line 1 or Line 2 can
initiate a 3-way conference with both call legs using
G.723.1.
5. GSM – Uses several codec specifications like: Half
Rate (5.6 kbit/s), Full Rate (13 kbit/s),
Enhanced Full Rate (EFR) (12.2 kbit/s) and a
variable-rate codec called AMR-Narrowband, which is high
quality and robust against interference when used on
full rate channels.
How do I dial an International phone number?
Solution:
To dial any PSTN phone number that is outside of the
NANP (North American Numbering Plan), you must dial:
011 + Country Code + City Code + Phone number
Does the system support IP to PSTN faxing?
Solution:
IP to PSTN faxing is not supported.
Can I use this service with a Satellite Internet
connection?
Solution:
In general, we do not recommend the use of our system
over a Satellite (SAT) Internet connection. However,
some customers do. There are a few technical issues
related to SAT links that can cause problems:
- Round Trip Time (RTT) Delay. Most SAT's used for
Internet connections are geosynchronous. This means that
their orbit period is the same as the length of a day,
so they are stationary in relationship to a given point
on the ground. To achieve such an orbit, the SAT must be
22,235 miles (35,784 km) above the earth's surface.
Radio waves travel at the speed of light, but even the
speed of light takes a noticeable amount of time to
travel that distance 4 times (up and back twice for a
complete round trip from your UA to our servers.) This
delay, when added to other factors on the ground, can
cause delays anywhere from 500 ms to 1000 ms (1/2 second
to 1 second.) It can be very difficult to carry on a
conversation when there is that much delay.
- Network Jitter. Probably more important than RTT Delay
itself is the consistency (or lack of) of the RTT Delay.
Some packets take longer than others to reach their
destination which can cause voice data to arrive out of
order and possibly be dropped. We have found that many
SAT Internet connections have this failing. Many UA's
have adaptive Jitter Buffers which can absorb some small
inconsistencies in RTT delay, but given large
inconsistencies, the result is often an almost unusable
connection.
- Lack of uplink bandwidth. Typically SAT Internet
connections are asymmetrical with their download
bandwidth being much higher than their upload bandwidth.
If your uplink bandwidth is lower than the bandwidth
required by the vocal codec you are using (see the
Knowledge Base article on Codecs), then you will not be
able to use the connection for voice calls reliably.
What does an Error 104 mean?
Solution:
The problem is that the system has detected more than
one IP Phone (UA) registered with a phone number that is
associated with an unlimited calling plan. Calling plans
are restricted for use by only a single UA to prevent
abuse. The simple solution is to make sure that you only
have one UA programmed with the virtual phone number
that is associated with your calling plan.
Occasionally, due to power outages or UA reboots more
than one registration can appear for the same UA,
triggering an error 104. The old registration will
eventually expire (you can check for when on your
control panel.)
Does the system support IP to IP faxing?
Solution:
This FAX service is not guaranteed.
IP to IP faxing, both the sender and receiver need to be
AleesVoip members and have their fax machines attached
to IP analog telephone adapters (ATAs).
One other important factor when faxing over VoIP is the
quality and speed of your Internet connection. Since fax
data cannot be compressed, the G711u/a CODEC must be
used. This codec requires a minimum of 64Kb/s in both
directions to be reliable, but we recommend more than
90Kb/s.
What does an error 205 mean?
Solution:
The system has detected a PSTN call from an IP Phone
(UA) that is not registered correctly. To prevent abuse,
all UA's must register with our proxy server before they
can place a PSTN call. Common reasons for your UA not to
be registered are:
PROBLEM: Incorrectly set USER-ID (phone number) or
PASSWORD in the UA
SOLUTION: Make sure your USER-ID and PASSWORD are set
correctly. Re-enter them if necessary.
PROBLEM: "Registration" is set to "Never" or "No" in the
UA's configuration
SOLUTION: Modify your UA's configuration to force it to
register before placing a call.
PROBLEM: Call placed too soon after registration
SOLUTION: Wait at least 5 seconds after successful
registration before making a PSTN call.
What value should I set my Registration Interval to?
Solution:
Registration is the event where your IP Phone (UA)
contacts our system and tells it where you can be
reached for incoming calls. The registration interval is
the time (measured in minutes or seconds) between when
your UA re-registers. Registration is required before
you can make PSTN calls (see the Knowledge Base article
on Error 205.)
Our sip proxy forces a 60 minute registration regardless
of the interval specified in your VoIP adapter.
What does "PSTN" mean?
Solution:
PSTN is an acronym for Public Switched Telephone
Network. It is the traditional telephone system that the
world has been using for decades to make phone calls. In
order to access the PSTN from your IP Phone (UA), you
need to use a "bridge" between VoIP and PSTN.
If you are a Premium Member, this is accomplished
seamlessly by dialing a 1 + areacode + phone number for
a North American call or 011 + country code + city code
+ phone number for an International call.
How do I configure my Asterisk server with AleesVoip?
AleesVoip does not provide technical support for
Asterisk
When my number is called, the system immediately replies
with "the number is not available." Why?
Solution:
Check the account CP to verify the device connected to
the number is registering with the system. Next to the
number you will see a line containing: Registration
expires @ ... UA Type: type of SIP device.
-If the registration line is missing, the device is not
connecting to the system and therefore it will not
receive calls. Thus the system will immediately detect
the lack of registration and reply with the "person is
not available to take your call at this time". In such
case, the device is either offline because it is turned
it off or there is a problem with it. If the device is
not off, the connection to the network should be checked
as well as the device configuration settings should be
reviewed. A network may be in order.
-If the registration line is present, the following
needs to be verified:
a. check call forwarding-CF for that number. If CF is
set to "Send immediately to my voice mail" the system
will immediately send the call to VM. The send to VM
call forwarding option neds to be removed in order for
the device to receive calls normally.
b. the device is having connection issues where the
registration interval is so long that the device is
offline and the control panel registration is still
present. The device registration interval needs to be
changed to 300 seconds or 5 minutes.
c. the device has problems with the router, NAT issues:
If the router lost connection to the Internet, most
routers will flash for an instant and have to reconnect
to the attached devices. As a connection is
re-established the router will block outside connections
to an internal device unless the internal device makes
contact first.
Please consider forwarding the device's ports on the
router so it will not cut off your SIP device from
outside connections.
If you reboot the SIP device, it establishes contact
with the SIP server and then the router opens up that
link and allows two way communication between the
devices. If the router looses contact with the SIP
device, then when the SIP server tries to make contact
with the device, the router blocks the connection, hence
busy signal.
As with all NAT sessions unless you have port forwarding
setup the sessions need to be established from the
inside. The problem is if the router resets or has an
issue then any existing sessions normally get forgotten
about and thus dropped. The SIP protocol deals with this
by having a reregistration period.
At InPhonex this reregistration period is 5 minutes or
300 seconds. This normally means that if the SIP device
is working correctly and the router does lose its
information then the SIP device should fix that issue
within 5 minutes.
Putting the SIP in the DMZ on your router should fix
this problem as well.
