There are a lot of problems in chan_h323. One of them cause segfaults under heavy load, it was registred in Digium's bug tracker as issue 9299. It is almost 1 year old. In an effort to resolve it, we are putting a patch here. It is aimed to provide detailed log to help us trace this nasty bug. This patch is made against Asterisk-1.4 revision 104125.

Warning! An exploitable bug was discovered recently in the asterisk. This bug affects Asterisk, trixbox, OpenPBX and other Asterisk derived systems. Malefactor can crash a system remotely with a malformed SIP message (INVITE, SUBSCRIBE or REGISTER). Patches are already available.
All users are urged to update asap!

• Asterisk 1.2.16
• Asterisk 1.4.1
• Bug #9182

We are using MySQL to store all Asterisks’ data (Asterisk Realtime). We believe that a lot of user chose MySQL too. It is very important to know how MySQL works to speed it up. Here you can see a very interesting comparison of storage engines: MyISAM, InnoDB and Falcon. We hope this article will help you to choose the best storage engine for your needs.

http://www.mysqlperformanceblog.com/2007/01/08/innodb-vs-myisam-vs-falcon-benchmarks-part-1/

Sometimes strange thing happen: if caller hangup befor callee's answer, callee's phone keep ringing forever. It happens if destination hardware/software uses SIP code 183 to present call progress, in that case asterisk doesn't change channel's state to AST_STATE_RINGING. But Asterisk retransmits CANCEL only to ringing channels, so CANCEL and 183 callprogress doesn't work together for now (revision 48105).

We are glad to provide a quick little patch which solves this issue.
NB: Most likely that this patch would be considered as "quick hack" by asterisk core team, however it does it's job.

There are 3 type of storages for voicemail right now: Standard storage, IMAP storage and ODBC storage. There is also support for Realtime, you can store data about mailboxes in RT database. We believe that voicemail lack of RT storage. It is good idea to store voicemail data in centralised storage, however, you shouldn't use API like ODBC from your moduleю Asterisk provides it's own Database Abstraction layer - Realtime.

Previously, it was impossible to use RT for voicemail storage, because RT API doesn't contain some important methods (INSERT and DELETE). Our previous patch fix that problem, turning RT API into complete DB Abstraction Layer.

We are proudly present a new patch for asterisk-realtime subsystem. This patch extends realtime API with 'store' and 'destroy' method. Previously, you was able only to fetch data from RT engine (*realtime_func) or update data in RT engine (*update_func). Our patch adds 2 new API calls: *store_func - for creating an entry in RT storage and *destroy_func - for removing an entry from RT storage. First patch is based on Asterisk SVN Trunk r44332

To apply this patch, you should execute following commands:


cd /usr/src/asterisk
wget http://voipsolutions.ru/files/rt-store-destroy-r44332.diff
patch -p0 < rt-store-destroy-r44332.diff


We are publishing our patch for IPP. This patch will produce codecs compatable with latest version of asterisk (SVN-TRUNK r41329). Detailed instructions on compiling G723.1/G729a codec can be found here: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/

Use our patch instead of provided there. Please note that this code is available for you to download for education purposes only and if a patent exists in your country for G.729 or G.723.1 then you should contact the owner of that patent and request their p

VoIPSolutions.ru are glad to inform you that we fixed an issue reported last day. Patch can be found in digium bugtracker: Issue #7828 or bellow this text.

To apply it, use the following commands:


$ cd /usr/src/asterisk
$ wget http://voipsolutions.ru/files/segfault_subscribe_mwi_fixed_20060830.diff -O patch.diff
$ patch -p0 <patch.diff


NB: This patch is made against SVN TRUNK Revision 41329, it may be impossible to use this patch with different versions/revisions.

Update: This issue was fi

If you are using realtime in Asterisk, we would recommend you to completly disable SUBSCRIBE MWI support in your sip.conf, to do it - simply add


subscribemwi=no


or


allowsubscribe=no

(to disable SUBSCRIBE completely)

to the [general] section of your sip.conf file. Current version of asterisk (SVN-trunk-r41241) crashes when some SIP phones sending SUBSCRIBE for MWI. We are working on a patch.

There were a lot of activity in asterisk trunk recently. Loader was completly rewritten. Unfortunately current version of asterisk-addons (r283) doesn't work with current SVN-HEAD of asterisk (r40833). To fill up this gap, we are releasing small patch for asterisk-addons (r283). This patch will make asterisk-addons compatable with current version of asterisk (r40833). This patch is released under BSD license.

to apply it:


$ cd /usr/src/asterisk-addons
$ wget http://voipsolutions.ru/files/ast-r40833-aa-r283.diff
$ patch -p0 <ast-r40833-aa-r283.diff


There is a problem with asterisk's CDR subsystem, if you are requesting non-existing CDR-variable, you have a big chance to get some "garbage". We are publishing small patch which will fix this behaviour.

UPDATE: This bug is fixed in current SVN-HEAD branch. Thanks to PCadach.

CIBill is an ultimate VoIP billing for asterisk. CIBill allows you to deploy a lot of VoIP services in a short time. it requires linux operating system with installed asterisk.

Online Demo

VoIP solutions launched the micron project to bring together best-of-breed applications to produce a standardized implementation of Asterisk complete with web-based administrative interface.

VoIP solutions has contributed its administration interface to the project.

"VoIP Solutions" is a young, fast growing company. Our main goal is to create software products which ease your life, making VoIP telephony the ultimate business tool.

Our main interest - large scale solutions.

We specialize in:

• Software Development
• Outsourcing
• VoIP Systems' Installation and Support

• micron - Simple but powerful tool to configure your Asterisk.
• CIBill - Asterisk based VoIP platform.