streamco.org is the official website of Streamco. Company's portfolio includes:
Much more information could be found in our Wiki.
Attention! Our company doesn't relate to our previous domain name streamco.org.ua, which is currently used by other people who doesn't relate to us in any way.
In this release:
Using web access, a customer can check the status of opened tasks for enhancement, tasks for support and tasks for bug fixes. Also client can leave comments and attach files to each task. This will give client chance to closer participate in development process, giving instructions and guidance for better solution while the task is being performed by our engineers.
When the Streamco's employees leave comment for a task or change its status (for example, closing it), the system automatically sends e-mail.
The tasks, opened for a customerm are confidential. Other customers cannot access them.
We are sure, that presentation of a system of such kind will move the customer service to the new level of quality.
Also this system has public access (not for customers). Under public access everyone can explore public information, such as publicly available projects or Wiki.
In this release:
with new mode CDRs are being saved much quicker and don't cause "snow ball" creation on SBC
This allow to integrate existing web-interfaces and applications for Asterisk with the system;
This application allows to automatically block through the firewall system intruders, which try to break in the system by SIP login and password brute force;
This controller detects sudden traffic splashes and informs about the latter system's administrator.
This functionality might by handy for your customers, for preventing the cases when your client has been hacked and used for sending huge amount of call traffic.
Before this it was available for terminators only.
When the company's balance goes below threshold value, the alarm email is sent only once. Furthermore email is sent in case if balance went above the threshold, and then went below again.
This release introduces:
This way we usually install them only once and don't copy once again when doing software upgrades;
So now when creating similar entities (for example, peers) you can copy some entity just supplying new name and don't have to copy all attributes by hand;
Also it's possible to pass through T.38 fax in a call with protocol conversion SIP<->H.323.
This feature is absent in standard Asterisk;
This driver supports H.323 protocol, but unlike the present H323 driver it uses H.323 software stack from ObjectiveSystems. Thus we give to you a choice between 2 stacks - OpenH323 (default) and ObjectiveSystems. This gives advantages in compatibility when interconnecting with different equipment using H.323 protocol.
This enables us to attach to Smartswitch telephony extension cards, for example PRI or SS7 and gain access to corresponding networks.
This functionality works together with standard billing of Smartswitch just as already supported SIP/H.323 channels.
Before this chamge a VoIP core has been polled diectly to get active channels list.
When having significant count of managers, that monitor active calls , the load of VoIP core increased as well. This lead to delays in serving inbound calls.
In new implementation active calls are replicated in dedicated in-memory database once a second, and polling of active calls is performed using this replicated database.
Decreased count of mutual locks when performing different operations (refills, subscribers creation and so on).
This is needed mainly for compatibility with different billing systems, which can round second differently;
We declined to use XML-RPC anymore.
Also ICE is used for API for third party applications.
A list of API interfaces along with examples is available here here.
All previously available XML-RPC interfaces have been transferred to ICE.
MeetMe has been chosen as more functional and stable variant.
Before this change activated SIP subscribers have been loaded in RAM from the database when the configuration has been changed.
When there is a significant amount of subscribers (hundreds of thousands), this leads to RAM starvation.
In new SIP mode subscribers are stored in database on the hard disk drive and are loaded in RAM on necessity (for example, when handling INVITE).
New mode is more CPU-intensive.
Therefore it should be chosen when total SIP subscribers count is more than 1000. In other case it's not effective.
There is a choice between old mode of SIP subscribers handling and a new one per use group or even per single user.
Gateway combines in it a functionality of originator and terminator, therefore it can send and receive calls;
Administrator can use GUI to translate GUI to its own native language;
This way we eliminate codec translation in real-time when playing audio prompts.
Therefore we decrease CPU load and enhance the quality of voice playback;
From now on we use the indicator of opened voice channel together with hangup cause to make a decision of continuation of hunting.
This options control time interval which will be considered by the system when making outbound calls to this terminator;
Previously refill voucher has just been deleted, so there was no mean to see who refilled using this voucher.
This will allow to estimate the effectiveness of a particular agent;
You can re-define the vendor label and sell product as your own;
It allows you to estimate the profit on your destinations for making routing or business decisions;
It allows to authorize whole IP networks or sub-networks as single originator.
It may be used for solving disputes with clients/providers.
