Asterisk Realtime conference

For asterisk 1.6 and above

Create a new database and table in your mysql database. For adding the table use the below query

CREATE TABLE meetme (
confno char(80) NOT NULL default ‘0’,
starttime datetime NOT NULL default ‘0000-00-00 00:00:00’,
endtime datetime NOT NULL default ‘2099-12-31 23:59:59’ ,
pin char(20) default NULL,
opts char(100) default NULL,
adminpin char(20) default NULL,
adminopts char(100) default NULL,
members int(11) NOT NULL default ‘0’,
maxusers int(11) NOT NULL default ‘0’,
PRIMARY KEY (confno,starttime)
);

The above query will create a table meetme, with the needed fields in it.

Now add the following database connection details to /etc/asterisk/res_config_mysql.conf

[context]

dbhost = 127.0.0.1

dbname =dbname

dbuser = user

dbpass = password

dbport = 3306

dbsock = /var/lib/mysql/mysql.sock

dbcharset = latin1

requirements=warn ; or createclose or createchar

 

Now edit your extconfig.conf file inside /etc/asterisk and add the following line at the bottom

meetme => mysql,dbname,meetme

 

Also make sure you have the following parameters in your meetme.conf file inside /etc/asterisk

schedule=yes (starttime and endtime options for meetme conference will work only if this parameter is set to yes. You can schedule a conference with this and the conference will be active only between the startime and endtime)

logmembercount=yes (This parameter helps keep track of members in an active conference. maxusers option in a conference will work only if this parameter is set to yes)

 

Now reload asterisk and start using asterisk realtime conference

Sip Trunking Basics

Don’t know much about SIP Trunking? No worries!

SIP Trunking is popular medium to provision calls for many who use VoIP. Along with flexibility and scalability SIP Trunking gives huge cost benefits.

First, you need to ask yourself some questions.

What is the use my phone system?

  • Are you simply receiving calls from clients?
  • How valuable are conference calls to your business?
  • Do you rely on voicemails while you are out of the office?
  • What about faxing?

You need to understand how important the phone system is to you or your business. How often you receive calls? When you receive them, is someone there to answer the call at all times? How is the state of your phone system going to reflect your business model to the clients?

These answers will dictate what services you are looking for in a SIP Trunk provider. While many providers are out there, they do not all offer the same services.

Will my needs for the phone system grow over time?

We are a very digital oriented human race today. The SIP trunk phone system simply will expand on that. With offerings like voicemail to email, texting, and video chats, you may not need all these features. However voicemail to email is usually a pretty standard feature and one that can keep employees connected even when they are out of the office. The ability to forward a desk phone to a cell phone is another feature that is generally included, and another excellent feature for those who are rarely in the office.

SIP Trunk gives ability to accept calls and voicemails from anywhere. It gives flexibility to do business anytime, and anywhere.

How much bandwidth do I need from my ISP? Do I need an additional access point?

While searching for a provider, you should be looking at the information they provide on bandwidth usage over a specific number of calls. There are providers who will supply dedicated connections for your voice service. This is sometimes far more effective when and if you have issues with your trunks. Simply contacting the provider will be enough to sort out issues. However if you choose to utilize an existing network connection, or even if you plan to have your provider bring in a second connect specifically for voice, it is wise to keep in the front of your mind that you now have multiple points of contact for phone issues. Will it be the trunk provider, will it be the ISP, could it simply be an internal issue? The complexity of issues that arise are not frequent, but as with any other type of support, it can be easy to point at someone else. Be sure to check the reliability of not only Sip trunk providers, but of the ISP as well.

The complexity of the system you should be dictated by the needs of your staff and your customers, not the price of the provider. SIP trunks are a powerful tool for your business. Having a clear understanding of your own needs, as well as those of your customers will give you the knowledge and flexibility to select a provider that offers the services you require with the room to grow.

SIP vs. PRI

SIP vs. PRIYou can continue to utilize a PRI trunk while running a VOIP server and utilizing the SIP protocol internally.   However there are several issues that come up due to this configuration.

First, you will need a PRI box, that basically takes the PRI stream and converts it to digital for the server to understand. The hardware aside, this can become difficult to troubleshoot issues with calls from external sources as well as calls originating inside the system to external numbers. While the system will work internal extension to internal extension, external out bound/inbound calls may not, and can result in a line that is simply silent, presents a busy tone, or even rings and then disconnects.

The end result is that your server is working, but we need to know why calls into the system from outside are not, and why the system can not call out of the network.

  • Are we actually connected to the internet? We can login to the server via an ssh session or even the web GUI and see if we can hit an external web page, or ping tests, as well as running trace routes. So, if we are successful, this still does not answer why we are not able to make or receive external calls
  • We can check our Asterisk CLI and see if we can see the invite on an inbound call. If we do not, and we can successfully reach the internet, then we have an issue along the PRI.
  • At this point we would need to login to the PRI box, verify it can see our server and that all it’s settings are correct, and reboot the device

Rebooting of the PRI box will generally solve several connectivity issues, but the question is why? The simple answer is generally the provider of the PRI handoff has made some configuration changes and usually do not notify end users that there will be changes, and this requires the PRI box to reboot to connect back to the PRI trunk.

Another common issue along the PRI line, is the handoffs card will go out, and a resulting call to the PRI provider generally results in the provider on the phone telling you they do not see an issue in the circuit, and will require a few hours of prodding to have a provider technician come to the site and change the faulty hand off card. At that point you will now need to login to the PRI box, and reboot it to bring the phone system back online and functional once again.

When we utilize a SIP trunk, this is a direct internet connection and is already being transported digitally and is ready for the server to accept, with no other hardware conversions required. Your only issues that can arise in this situation would be a failed internet connection, or an issue with the SIP trunk provider directly. However the Sip Trunk providers use several fail over connections so this will be unlikely, and a configuration change from your ISP will not affect your SIP service.

While the overall call quality will not show a difference, the troubles that can arise as well as the added hardware costs and extra configurations of a PRI box that are needed make a direct SIP trunk more economical in the end.

Is your Business Ready to implement SIP TRUNKING

Business Sip Trunking 

Are you ready to join the VoIP SIP Trunk trend & take your business ahead?

Switching to SIP Trunk for your telephone needs is the best decision you have made. So how do you know your business is ready to implement SIP Trunking?

Today implementing Sip trunk is not as hard as most think. There is very little hardware required to implement the systems, and if you already have an internet connection, your most likely there.

The major component you will add are the handsets, and when utilizing a hosted solution, you may find a provider that can simply rent you the handsets as well, although that is an additional cost. However they should also be able to provide updated and new handsets over a given time line as you retain a contract with them as well.

Even if you are choosing to host your own VOIP server, the start up is relatively easy as it simply requires a server, a switch, and the handsets. Equipment you would own, and would be reflected in the lower cost of your yearly sip trunking costs.

With a reliable internet connection, sip trunking will be very easy and cost effective and given the added flexibility of utilizing IVR menus for your callers, specialized ring groups, time conditions, as well as conference rooms and video chat.

The added features to sip trunking and the low cost of running it over time compared to initial start up costs, or the switch from the old standby PBX will far better suit business today than to not make the switch.

If you are already running your own domain within your business, the ability to do a self hosted solution will be just about as easy as a hosted solution. Your typical VOIP server will sit outside your network, or if you have one, it can sit in your DMZ, either way your infrastructure will most likely be in place already and this is simply an addition of a server and switch.

If you are not hosting your own servers within the confines of your business, this will still be a very easy switch. Simply adding a server and possibly a switch will do the trick.   Location of those devices would typically be near the internet providers hand off, or in the MDF Closet. With the current platforms available to handle the sip trunks you can easily pick up server with a small enough foot print to fit directly into a patch panel as opposed to a dedicated rack. Typically these types of servers come bare bones, meaning there is no operating system installed, but are more than capable of handling medium size business systems in excess of 50 handsets.

While cell service could be considered as an alternative, business will not reap the benefits of conference rooms, complex IVR trees, time conditions, and a host of other valuable assets provided with sip trunking. Whether the solution is hosted off site, or hosted by the business itself, sip trunking is a definite asset to any business that requires phone service.

How to choose right SIP Trunk Provider?

how to choose sip trunk providerWhat is the right SIP trunk provider?

It really all depends on your needs and your understanding of what SIP trunking is?

Several internet providers can provide sip trunking, but not all sip trunk providers can provide internet.

One of the biggest worries you will have with sip trunking is latency, and most carriers can provide that data within their QoS (Quality of Service) reports. Generally (as always) we like to see as small of an amount of packet loss in the transmissions as we can. For voice calls 1% or less packet loss would be an excellent quality call.

You will see codecs mentioned by providers, and if your using an open source VOIP server, it will support all codecs, and this should not be an issue as most carriers conform to a standard that is now well supported throughout the VOIP industry.

If you are not hosting your own solution, you obviously should be concerned with a usable control panel, as well as Call detail records interface so you can see what calls where sent/received and when.This is something you will want to track closely as the number of calls made from your system can affect billing on a monthly basis. The times calls are placed can also indicate a potential intrusion into the system as well.

Packaging can vary greatly from provider to provider as well, the number of channels supported to the cost per call. Careful investigation into pricing packages based on your specific needs of the system your running will make a big difference in your pocket book down the road. Also with VOIP, you are now utilizing E911 which will incur fees when a 911 call is placed, so you also want to be aware that even an accidental call placed can incur the fee, as well as alarm systems that may end up tied into the phone system.

Your SIP trunk may connect back to multiple servers in different locations based on the providers network, and this ensures maximum uptime for your system. It is a simply redundant system set up to make sure you are always connected, so it is a good idea to ask, how that will work based on a given package you may be looking at. After all, if your SIP trunk provider is not connected, you are not connected, so call quality and price may be great, but a provider without any type of fail over plan, will not be as reliable as the provider with a viable plan in place.

Does the provider also have a block of sequential DID numbers available for you? This may not be an issue for everyone, but if this is a part of how you choose to set the system up, it can become extremely important, and I have seen from experience that sequential DID’s can be extremely helpful to business.

If you have designed your own system, or are simply looking to have your system hosted, understanding how you plan on using that system, what the needs of your business, and employees are in regards to that system will supply more than enough information to choose a quality provider for the best price. The first step is to know your needs and how you will use the system.