The call will jump to the s extension. While a Macro is being executed, it becomes the context, so you must be able to handle the h, i , and t extensions if required within that context.
Let's build our small macro dialplan. We have a variable defined in the globals section of the extensions. We have seen that we can pass one dial string, but let's now pass other variables to the Dial application, such as a backup route for outgoing calls, and the caller ID we want to use for the call. Now it's time to bring some. Using the setvar facility in the sip. Basically, anything that will help you handle calls more efficiently.
One problem using. This isn't too much of an issue but the need can be removed by using the AstDB for storing commonly changed settings, such as caller ID and recordings. You may think that all this variable use is over-complicated, but consider a system that supports multiple tenants.
Using these techniques, you will only need one dialplan for multiple tenants instead of one per tenant. Simply set the tenantID in the relevant. The concept is the same for other scenarios, such as departments that require cross charging of telephone costs. Setting and retrieving variables in the AstDB is very simple and achieved through the use of the Set application.
Variables can exist in splendid isolation or be grouped into families. The syntax for setting a variable is:. As we can see, we have been able to cut down the amount of code and make it universal for different types of users and systems. Using a macro lets us pass an ordered list of arguments. It is easiest to think of macro arguments as a list of variables since they are handled the same way.
Due to the way macro is implemented, it executes the priorities contained within it via a sub-engine, and a fixed per-thread memory stack allowance. Macros are limited to seven levels of nesting. It can be possible that stack-intensive applications in deeply-nested macros could cause Asterisk to crash. Take this into account and be very careful when nesting macros.
We will see how we can check and change the "status" of devices with the DEVSTATE function and use the system application to cause scripts on the server to be run. This is very useful in many applications. We will cover a few of them here but you will be able to find many more. This can now be found at:. One very common use is to activate phone lamps, showing users if they have set a feature such as DND or call forwarding. In the following examples, we will look at both setting and checking methods:.
The application can be used here to check that an outgoing peer is "available" and not "down", before you send a call to it. This is useful if you have peers or remote systems that are on variable quality connections. This example is an expansion of our previously used macro and has a couple of extra arguments passed to it. This makes it very flexible, as the backup peer can be different for each dialed number. The application lets you see if extensions are busy or "out of service" before calling them.
This can be useful for handsets that support call waiting, but you don't want to fully disable it for all calls. Before calling the extension, you can check to see if the extension has call waiting enabled and then, depending on the result, check the device status as follows:.
In the previous example, we have used the internal database to set the flag to say if call waiting is enabled or not. As we will see next, we can expand this to light a BLF Busy Lamp Field key as well, to give a visual indication to users of the device status.
This is particularly helpful if you are using the dialplan for setting call forwards or DND. It can also show if a call center agent is logged in or not, on their phone. To illustrate this functionality, we have a very simple example showing how to turn the light on and off. It uses one number to toggle the light status and is not specific for the particular phone all phones dial the same number and it is the CHANNEL variable, which is used to set it for a specific phone.
In this example, we have two hints and , and these are linked to extensions and Using this code and adding additional code to set the database key for call waiting as we have already covered would give the phone user a visual indication as to whether call waiting is set or not.
By using userevent , you can also send out manager events to update the Flash Operator Panel. The following would set the CW flag for the Flash Operator Panel for our extension and change the icon to reflect the status. This means you can write dialplan logic based on the state of an extension in use, ringing, on hold, and so on.
The extension just needs to have a hint so we can determine which devices to check. A call-centre customer wished to temporarily increase their outgoing call capacity, in this case by 20 concurrent calls, to cater for a particular project. However, in their location, with their budget and given the temporary need for extra capacity, the only effective means of boosting bandwidth is to utilize multiple ADSL circuits. In other words, SDSL and leased line circuits were too costly for consideration.
Therefore, there was a need to bond multiple ADSL circuits together within Asterisk, in order to provide a single high-bandwidth circuit for outbound calls.
It doesn't matter if you have a superfast 20 MB DSL circuit, chances are that you only have an uplink speed of kbps or less. This gives you a theoretical limit of up to 10 uncompressed calls per circuit if you're really lucky. Of course, with the GSM codec you can get a lot more, but at the cost of audio quality, which your customer is unlikely to accept. They expect PSTN quality and nothing less. If bandwidth utilization is on the borderline with an uncompressed codec, it can be advantageous to use a commercial non-free codec such as G, which is obtainable from Digium.
Using the criteria already discussed and assuming a kbps uplink speed, it was determined that four broadband circuits were needed.
As we'll see later in the sales appendix, a major benefit of VoIP is that the customer is paying much less for line rental. This solution only reinforces that benefit. We are going to describe a solution that used four broadband circuits, but another advantage of this approach is that it is very scalable. To illustrate, a system has been set up for a charity in the UK that had 75 agents placing thousands of calls a day on just eight broadband circuits. This example was tested and proven using Asterisk 1.
Once the circuits are delivered, you will end up with four routers connected to the broadband service of your choice. Each router will have a unique IP address. In our case, we shall assume they are as follows:. We will also assume that the VoIP ITSP has multiple IP addresses that you can connect to, though if not, you can probably do some clever address translation in the routers. Within Linux, you can easily set up different gateway addresses for a given destination.
To configure the gateways, we append the following to the ifip-routes file:. Taking the last entry, what we're saying is that for all traffic to If you reboot and run the command route n in a terminal session, you'll see these routes in place. We now turn our attention to the Asterisk configuration. When we make a call, we're going to keep count of how many calls we have on a broadband line, so that when the circuit is "full", we can move on to the next available one.
Firstly, in the extensions. We've previously set up four entries in the iax. Here we assign the appropriate external IP address, that is, As expected, we use a macro to manage the call routing. The example below only shows two lines for the sake of brevity. At "4", we look to see if we've exceeded max calls on this line. If we haven't, we'll place a call on the first router, otherwise go to the next.
If in the meantime a call is dropped from router 1, the next call placed will go back to router 1, even if other calls are ongoing on router 2.
The above technique is scalable. You can add as many broadband lines as you need. The end result is that you can say to your customer, "want more outgoing capacity? Just add another DSL line". However, it must not be forgotten that there may be more stable solutions such as SDSL and leased lines, depending on location.
The above example works really well for outbound calling, but not so well for inbound. If you own the server the customer is connecting to, then you can reverse the logic at your end.
Check the voip-info. There are some files and configuration options that you will have to change, even though we made every effort possible to maintain backwards compatibility.
To discover the major new features of Asterisk 1. In addition, your system needs to have the C library headers available, and the headers and libraries for OpenSSL, ncurses and zlib. On many distributions, these files are installed by packages with names like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
You may also want to check the configuration files that contain examples and reference guides. Normally, this is automatically done with the presence of the file.
If this is your first time working with Asterisk, you may wish to install the sample PBX, with demonstration extensions, etc. If so, run: 6 "make samples" Doing so will overwrite any existing config files you have. Finally, you can launch Asterisk in the foreground mode not a daemon with: asterisk -vvvc You'll see a bunch of verbose messages fly by your screen as Asterisk initializes that's the "very very verbose" mode. Then you can use "answer", "hangup", and "dial" to simulate the actions of a telephone.
Comments are delimited by ';' since ' ' of course, being a DTMF digit, may occur in many places. A configuration file is divided into sections whose names appear in []'s. In general, the parameter will apply to instantiations which occur below its specification. Manually changing the system time using date 1 or other similar commands may cause SIP registrations and other internal processes to fail.
NTP is designed to keep the system clock synchronized by speeding up or slowing down the system clock until it is synchronized to "real time" rather than by jumping the time and causing discontinuities.
Most Linux distributions include precompiled versions of NTP. Any ideas? Finally I got Asterisk compiled with H, but I have one question How can I configure Asterisk to accept calls from OpenPhone h and then send it to an external endpoint. Thank you very much! Automatically restarting Asterisk. Let me know. Hi I am the problem: [root asterisk1 asterisk And zaptel???? I not see the compiler from zaptel Thanks OP.
Solution: 1.
0コメント