Wednesday, July 18, 2012

Asterisk 10.6.1 Now Available

The Asterisk Development Team has announced the release of Asterisk 10.6.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 10.6.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
  • --- Remove a superfluous and dangerous freeing of an SSL_CTX.
    (Closes issue ASTERISK-20074. Reported by Trevor Helmsley)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.1
Thank you for your continued support of Asterisk!

Asterisk 1.8.14.1 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.14.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.14.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
  • --- Remove a superfluous and dangerous freeing of an SSL_CTX.
    (Closes issue ASTERISK-20074. Reported by Trevor Helmsley)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.1
Thank you for your continued support of Asterisk!

Tuesday, June 26, 2012

Asterisk 1.8.5-rc1 Now Available

The release of Asterisk 1.8.5-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release candidate:
  • Fix Deadlock with attended transfer of SIP call
    (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
    cmaj)
  • Fixes thread blocking issue in the sip TCP/TLS implementation.
    (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
    rossbeer, kowalma, Freddi_Fonet)
  • Be more tolerant of what URI we accept for call completion PUBLISH requests.
    (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
  • Fix a nasty chanspy bug which was causing a channel leak every time a spied on
    channel made a call.
    (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
  • This patch fixes a bug with MeetMe behavior where the 'P' option for always
    prompting for a pin is ignored for the first caller.
    (Closes issue #18070. Reported by mav3rick. Patched by bbryant)
  • Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
    the call that the dialplan started an AGI script for is hungup while the AGI
    script is in the middle of a command then the AGI script is not notified of
    the hangup.
    (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
  • Resolve issue where leaving a voicemail, the MWI message is never sent. The
    same thing happens when checking a voicemail and marking it as read.
    (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
    Mudgett)
  • Resolve issue where wait for leader with Music On Hold allows crosstalk
    between participants. Parenthesis in the wrong position. Regression from issue
    #14365 when expanding conference flags to use 64 bits.
    (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
  • Fix timerfd locking issue.
    (Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1
Thank you for your continued support of Asterisk!

Asterisk 10.5.1 Now Available

The Asterisk Development Team has announced a security release for Asterisk 10.
This security release is released as version 10.5.1.
The release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of Asterisk 10.5.1 resolves the following issue:
  • A remotely exploitable crash vulnerability was found in the Skinny (SCCP)
    Channel driver. When an SCCP client sends an Off Hook message, followed by
    a Key Pad Button Message, a structure that was previously set to NULL is
    dereferenced. This allows remote authenticated connections the ability to
    cause a crash in the server, denying services to legitimate users.

Monday, June 4, 2012

Denphone Asterisk PBX Solution

Denphone Telephone Systems and Solutions : Japan's most flexible PBX solution

We provide the Denphone range of PC/Linux based software PBX systems. Our solutions are based on the Asterisk open source architecture and support a wide variety of configurations and connectivity options.
Hardware choices are made depending on the number of phones to be connected, the desirability of redundant systems compared with budget expectations in order to find the most cost effective solution. The majority of our installations in the 200-300 phone range run on standard rack mount servers from vendors such as Dell or HP. We also normally recommend Dell or Yamaha POE switches although we do install Cisco switches where requested.

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.