Wednesday, May 23, 2012

Helping Your Company do better Business With Voip





VoIP is replacing conventional telephone services; there are 30 million or more users by now. To penetrate the market there are few hindrances which are to be overdone, since VoIP is a hot industry in current days, the market is saturated and hindrances do come across. The question here is how you can help your Company do better business? And what setup should be there in order to capture the market chunk? Breezecom is among all the big exiting players and fulfills all the benchmarks and is heading to success.

The world has witnessed many revolutionary changes in the current century with information technology sector leading from the front. Many ground breaking methods of communication evolved due to this advancement. People started to use these new found ways of keeping in touch with their friends and family enabling more users for a specific product or service. VoIP is one such technology commonly used nowadays.

The primary two things which anyone in the VOIP industry seeks are the Price and Quality. The lowest price is not always the best service but it shouldn’t be too high with respect to market at the same time. And when you buy keep in mind to choose a provider that has a price that is reasonable with the competition, but not a low cost provider. Price change should be there in accordance with the market. When you buy VOIP, Price is only one of many factors that should be evaluated.

When buying or selling, get the idea of the second party as it’s very important in terms that it’s a long term relationship and not few days. Get the idea of business counterpart with the outlay they have in the Website, the domains they have, the legal system, the payment mode diversity. Like a car, a test drive gives clear idea what the car is like! Likewise these factors help evaluate the counterpart well.

Setting up a place for customers to call and allow important clients access both to your business and your employees is also important and holds vital position. Meeting customers enhances the trust and confidence level in the long run.

Your VOIP network should be strong, and make sure that it does not have problem. Even simple networks can have problems that will affect the quality of VoIP calls.  Make sure that your switches are in good shape and that your computers are running efficiently.

Provision of user friendly CRM is another factor which holds quite a big importance. It should give access to the customer if he wants to see CDRs and many other options for the customers. Breezecom provides provision of real time CDRs in the Customer Relationship Panel. Also 24/7 efficient customer support and technical service should be available.

VOIP is spreading day by day and the competition is getting tougher day by day. It has been proven that consumers will save a bundle of money by transitioning away from the traditional telephone companies and switching to a VoIP provider.


Sunday, May 6, 2012

Digium IP Phones

Digium manufactures a line of SIP phones that are built specifically for use with Asterisk and Asterisk-based phone systems. The phones support all of the standard SIP features and include a number of enhancements including easy provisioning, direct integration with Asterisk features and an open API that allows end user and integrators to write custom applications that run on the phones.

Digium D40 -  A 2-line SIP phone with HD voice and PoE support.


Digium D50 -  A 4-line SIP phone with 10 rapid dial buttons with busy lamp field indicators for your most important contacts.


Digium D70 -  A 6-line SIP phone with 10 digital rapid dial buttons with real-time status information and busy lamp field indicators for 100 of your most important contacts.

Wednesday, April 4, 2012

Asterisk 1.4.44, 16.2.23, 1.8.10.1, 10.2.1 Now Available (Security Release)

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.4.44 and 1.6.2.23 resolve an issue wherein app_milliwatt
can potentially overrun a buffer on the stack, causing Asterisk to crash. This
does not have the potential for remote code execution.

The release of Asterisk 1.8.10.1 and 10.2.1 resolve two issues. First, they
resolve the issue in app_milliwatt, wherein a buffer can potentially be overrun
on the stack, but no remote code execution is possible. Second, they resolve
an issue in HTTP AMI where digest authentication information can be used to
overrun a buffer on the stack, allowing for code injection and execution.

These issues and their resolution are described in the security advisory.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2012-002 and AST-2012-003, which were released at the same
time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-...

The security advisories are available at:

http://downloads.asterisk.org/pub/security/AST-2012-002.pdf
http://downloads.asterisk.org/pub/security/AST-2012-003.pdf

Asterisk 1.8.11.0 Now Available

The following are the issues resolved in this release:

--- Fix potential buffer overrun and memory leak when executing "sip
show peers"
(Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)
--- Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389.)
--- Remove possible segfaults from res_odbc by adding locks around
usage of odbc handle
(Closes issue ASTERISK-19011. Reported by Walter Doekes)
--- Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
(Closes issue ASTERISK-19322. Reported by aragon)
--- Copy CDR variables when set during a bridge
(Closes issue ASTERISK-16990.)
--- push 'outgoing' flag from sig_XXX up to chan_dahdi
(Closes issue ASTERISK-19316. Reported by Jeremy Pepper)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.11.0

Asterisk 10.3.0 Now Available

The following are the issues resolved in this release:

--- Fix potential buffer overrun and memory leak when executing "sip
show peers"
(Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)
--- Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389.)
--- Remove possible segfaults from res_odbc by adding locks around
usage of odbc handle
(Closes issue ASTERISK-19011. Reported by Walter Doekes)
--- Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
(Closes issue ASTERISK-19322. Reported by aragon)
--- Copy CDR variables when set during a bridge
(Closes issue ASTERISK-16990.)
--- push 'outgoing' flag from sig_XXX up to chan_dahdi
(Closes issue ASTERISK-19316. Reported by Jeremy Pepper)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.3.0

Sunday, December 11, 2011

PRI for Asterisk

PRI is an abbreviation for Primary Rate Interface and is a telecommunication standard for carrying multiple data or voice DS0 transmissions between 2 physical locations.

All data and voice channels are (ISDN) and operate at 64 kbit/s.

North America and Japan use a T1 of 23 B channels and one D channel. Europe, Australia and most of the rest of the world use a slightly higher capacity E1 of 30 B channels and one D channel.

A B-channel is used to transmit the voice or data and is sometimes also called a user channel, a D channel is used for control messages and signalling.

In some T1 configurations, no D-channels might be used, instead the signalling will be sent inband over the B channels, this is called in-band signalling or bit robbing, resulting in lower transmission rates than the E-carrier system. This resulted in many US ISDN installations only having an effective data rate of 56 kbit/s over a nominal 64 kbit/s channel. See also A&B. This depends on the framing format used.

In the states its common to order a fractional T1, with less than 23 B channels.


1.3. signalling

On both E1 and T1, one timeslot is usually reserved for a D-channel for call setup and call teardown. (called signalling).

- CAS: Channel associated Signalling: with this kind of signalling, a set of bits is used to replicate opening and closing the circuit (as if picking up the telephone receiver and pulsing digits on a rotary phone), or using tone signalling which is passed through on the voice circuits themselves.

- CCS: Common Channel signalling: A more recent kind of signalling, (ISDN signalling and ss7 signalling are a subgroup of CCS signalling.) In this kind of signalling, short messages are sent over the signalling channel, with more information about the call, including caller ID, type of transmission required, etc. etc.


1.4. Framing

- HDB3
- AMI: Aternate Mark Inversion
- B8ZS:


1.5. Timing or clock sources

A PRI connection needs a timing device on one of both ends.
A PRI line can be clocked internally or can be clocked by the telco.

Sunday, December 4, 2011

USB 2.0 port support heavy telephony traffic

The USB 2.0 interface provides a theoretical speed of 480,000,000 bits per second. A typical uncompressed phone conversation uses about 64,000 bits per second per direction, plus some overhead; in total less than 200,000 bits per second per phone call is used. Thus, the theoretical concurrent number of conversations that the USB 2.0 interface can handle is 480,000,000 divided by 200,000: roughly 2,400 calls for a single USB 2.0 port.

This is a theoretical number, of course, and the typical host processor will not be able to handle this amount of concurrent calls, but it gives a clear answer to the question: "Can the USB port support heavy telephony traffic with multiple Astribanks?"

Connecting multiple Astribank units to a single USB 2.0 port is easy: simply use a USB 2.0 Hub to connect many Astribank units.