Absolute_Codec_String Not Working In Dialplan

The solution is to make use of another necessary variable known as inherit_codec. This means, you presumably can enable the outbound name to barter a codec then pass that decided value again to the inbound leg earlier than media was established. This will then pressure the inbound leg to request the same codec because the outbound leg and remove transcoding. When calling the instance extension, the decision hits the XML dialplan and executes the directions contained within the action tags. First, the variable inherit_codec is ready to true after which the call is bridged to over SIP. Because we enabled the inbound-late-negotiation parameter, the codec has not yet been chosen for the inbound leg. The outbound leg then proceeds to connect to cluecon.com where a codec will be chosen when the far-end answers or establishes media. At this level the FreeSWITCH call origination engine will take the codec from the outbound leg and set it as the absolute_ codec_string on the inbound leg. Next, the media indication is passed across, which can prompt the inbound leg to barter media and offer the same codec because the outbound leg. This , nevertheless, does not work properly under NAT situations.

Introduction Routing calls is on the core of any FreeSWITCH server. There are many methods for carrying out the surprisingly complex task of connecting one phone to another. However, you will want to make certain that you have the basic instruments necessary to complete this task. The most basic part of routing calls is the dialplan, which is basically an inventory of actions to perform relying upon what digits had been dialed . The dialplan is damaged up into a number of contexts. Each context is a group of one or more extensions. Finally, every extension contains specific actions that can be performed on the decision. The dialplan processor makes use of regular expressions, which is a pattern-matching system, to find out which extensions and actions to execute. To make the most effective use of the recipes in this chapter, it's especially important to grasp tips on how to use regular expressions and the three contexts in the default configuration. Getting prepared You will want a text editor and at least one SIP telephone for this recipe. It can also be recommended that you just evaluation the phrase file on your language. For English that is found within the FreeSWITCH source directory in docs/phrase/phrase_en.xml. The phrase_en.xml file incorporates both the file name of each pre-recorded immediate as properly as the actual spoken textual content. Prompts are divided into sections similar to voicemail, IVR, foreign money, digits, and time. By far the most important collection of sound prompts is in the IVR section. In this recipe we'll create a easy dialplan extension that may learn again to the caller his extension number. We will use a phrase macro to handle the work of stitching collectively sound prompts and using the say application to read back the caller's extension quantity. Keep in thoughts that the ESL is only an abstraction library—you can connect with the event socket with any socket-capable utility, together with telnet!

The tips in this chapter will focus most of their attention on utilizing the event socket for some frequent use instances. Getting ready You will want a text editor and a telephone connected to FreeSWITCH as properly as access to the fs_cli for your system. You may even must have the ESL compiled and working for your scripting language of alternative (see Setting up the occasion socket Library earlier in this chapter). The language used in this is example is Perl, however the ideas apply to all ESL-enabled languages. When we are via we may have a simple script that can pay attention for a socket connection from FreeSWITCH, answer the decision, play a file, wait for a DTMF digit, and then exit. The dialplan socket application sends management of the decision to the process listening on the specified TCP port. This recipe will information you thru the steps necessary to get a simple call control script up and running. You may discover it simpler to know the information introduced right here if you're a minimal of somewhat conversant in the concept of TCP sockets. Make a test call from 1007 to a different cellphone, answer, then hangup. You will now have a file named sql.csv in your cdr-csv/ listing. This technique can be utilized to customise the sorts of data which may be saved. Using accountcode and a CDR CSV template permits you to tailor the habits as needed. Now we can restart the FS service or we will reload "mod_dialplan_xml" from the FScli. Once the modifications are into effect, we are able to test whether or not the call is getting routed by way of Plivo.

Configure a gentle phone with a default FS person and make an outbound call which matches the regex that we now have talked about for routing to Plivo. Now if all works we should get a call on the destination quantity. We can check the FS logs at "/usr/local/freeswitch/log/freeswitch.log". Other choices There are a couple of different choices in the settings part of cdr_csv.conf.xml. Setting this to true will simply trigger each call to perform an data dump when the decision hangs up. Note that this will dump both to the fs_cli and to the FreeSWITCH log file, so concentrate on disk area. This will determine which call leg or legs get a CDR. You can set this parameter to "b" to log solely the B leg or you can set it to "ab" so that you receive a CDR for each leg. Handling A and B legs is mentioned later in this chapter. Introduction Call element records are an necessary a part of the accounting process on any telephone system. They are additionally a useful useful resource for troubleshooting. FreeSWITCH supplies several totally different strategies for generating CDRs. The most typical method is to create plaintext, comma-separated worth information. Each line in the CSV file represents one phone name . There are other choices for processing CDRs, most notably using mod_xml_cdr to store extra detailed information about calls as well as using the event socket to process CDR info. What this guide covers Chapter 1, Routing Calls; getting calls from one endpoint to another is the first operate of FreeSWITCH. This chapter discusses strategies for effectively routing calls between phones and repair suppliers. Chapter 2, Connecting Telephones and Service Providers; telephones and service suppliers have specific requirements for connecting to FreeSWITCH.

This chapter will help in shortly getting your FreeSWITCH server linked to other VoIP devices. Chapter 3, Processing Call Detail Records; Call Detail Records, or CDRs, are essential for companies. This chapter discusses a variety of methods to extract CDR data out of your FreeSWITCH server. Chapter 4, External Control; FreeSWITCH may be managed externally by the highly effective and versatile event socket interface. This chapter presents numerous real-world examples of controlling FreeSWITCH from an exterior course of. Tagged dial plan, freeswitch, freeswitch channel variables, freeswitch core, freeswitch listing, Media server, Sip, sip modules. There are lots of different parameters which we will add it right here, like caller id and so forth. Replace the username and password with the Plivo endpoint credentials. Once the gateway file is created, we will either restart the service or we are in a position to run "reload mod_sofia" on the FScli. If the FreeSwitch service si began in foreground, we'll get the FScli, so we are ready to run the reload command directly on it. Once the FreeSwitch is working fantastic, we can start configuring the SIP trunking via Plivo. So first we have to create an external gateway to join to Plivo. I'm going to use the SIP endpoint created on the Plivo Cloud to initiate the connection. Go to "/usr/local/freeswitch/conf/sip_profiles/external/", here we can create an XML gateway config file. Getting prepared Be certain that you've got got configured ESL on your system and that you have followed the steps in Establishing an inbound occasion socket connection earlier on this chapter. The examples listed right here are written in Perl but the ideas apply to any ESL-enabled language. Of course, you will need a textual content editor and a SIP telephone registered to your FreeSWITCH server so as to test this instance. Launching a name with an inbound event socket connection Using an inbound occasion socket connection to launch a name is a standard requirement for some purposes, similar to outbound IVRs.

In a case like this it's advantageous to handle the generating of the calls in a non-blocking manner using the ESL connection object's bgapi() technique. This recipe discusses the way to use the bgapi() technique with the corresponding "Background-Job UUID". Establishing an inbound event socket connection An "inbound" occasion socket connection signifies that an external script or program is connecting to a FreeSWITCH server. The connection is inbound from the server's perspective. In truth, every time you run the fs_cli utility you are making an inbound occasion socket connection. This program comes with FreeSWITCH, as part of the default set up, and works in Linux/Unix, Mac OS X, and Windows. What is less well-known about fs_cli is that it is a wonderful example of an ESL program. Beyond that, anything that you are capable of do with fs_cli, you can do with ESL and the occasion socket. Indeed, it is probably certainly one of the most important instruments for interacting with your FreeSWITCH server. If you're acquainted with C programming then you might recognize the supply code for fs_cli. It is found in libs/esl/fs_cli.c under the FreeSWITCH source directory. It is considered a finest apply not to add a broad range of dialable numbers within the Local_ Extension without having the corresponding users in the directory. Doing so could make troubleshooting dialplan points tougher. As a reminder, be sure to execute the reloadxml command each time you modify the regular expression . Now mod_xml_cdr will load automatically when FreeSWITCH begins. However, if FreeSWITCH is already running then we have to load it manually. Simply issue the command load mod_xml_cdr at the fs_cli and the module might be loaded. XML CDR data will now be stored within the $FS_INSTALL/log/xml-cdr/ listing. To evaluation them, open the file conf/autoload_ configs/xml_cdr.conf.xml. We shall be discussing some of these options later on this recipe.

Introduction As its name implies, FreeSWITCH will "swap" or "connect" varied endpoints together. Part of that switching includes making semi-permanent connections to individual telephones or telephone service suppliers. Service providers are usually telephone companies or ITSPs . Read on to learn in regards to the many ways that FreeSWITCH can join your phone to the world. The recipes on this chapter will delve into the varied methods to connect FreeSWITCH to telephones and service providers. FreeSWITCH also can utilize a locally installed sound card by the use of the PortAudio library. The final recipe is for advanced users and discusses the topic of codec negotiation. The key to this operation is the highlighted traces. In the primary one, we set continue_on_ fail to true. This channel variable tells FreeSWITCH to maintain processing the actions within the extension even when a bridge attempt fails. After each bridge try, you presumably can then do some processing. Note, too, that we set hangup_after_bridge to true.

This is done in order that the dialplan doesn't keep processing after a successful bridge attempt. (For instance, if the decision to userA was successful, we would not need to call userB after userA hung up.) You could add as many further bridge endpoints as needed. Now we have to the setup the Dial Plan to route the outgoing calls via Plivo. Go to "/usr/local/freeswitch/conf/dialplan/" folder and add an extension on the "public.xml" file. The enter worth is matched in opposition to the regular expression within the pattern option. In most instances $ incorporates solely digits, so our pattern will capture these into the special variable $1. At this level we now have a little bit of logic to assist decide what to do. If the input matches the sample, then the actions within the node might be executed. You have most likely found out by now that the actions contained inside the match are executed sequentially. You can even see that phrase macros are not merely restricted to playing particular person sound files. You can name functions like sleep and say to customize the way the immediate is played to the consumer. You may even call a text-to-speech software when you have one put in. Uncomment this parameter to enable late-negotiation for all calls. At the fs_cli press F6 or issue the reloadxml command after which concern the command sofia profile inside restart. You are now able to experiment with codec negotiation. This sends the decision back through the dialplan on the lookout for a destination_number of 5001_ open or 5001_closed. We have outlined each of those locations with the extensions "office is open" and "office is closed," respectively. Now we are able to play completely different greetings to the caller—one for when the workplace is open and a unique one for when the office is closed. As a pleasant contact, for all calls, we play a sound file that says, "Good morning", "Good afternoon", or "Good night", depending on what value is within the channel variable day_part.

The execute_extension and transfer dialplan applications These two purposes both inform FreeSWITCH to execute one other a part of the dialplan. The primary distinction is that execute_extension will return after executing one other portion of the dialplan, whereas a switch sends management to the target extension. In programming parlance, execute_extension is like a gosub command and switch is type of a goto command. Once the decision hits the common public context, we attempt to match the destination_number area. The destination_number is generally the DID quantity (see the There's extra part under for some caveats). The make command issued above installs the MOH files in 8 kHz, sixteen kHz, 32 kHz, and forty eight kHz sampling rates . The default dialplan extension quantity will play the default music on hold files to the caller. The music is supplied by the module mod_file_stream. It is feasible to customise the MOH in your system by adding other streams. Press F6 or issue the reloadxml command from fs_cli. Dial 1411 from your phone and hearken to the options. For this check, dial the primary three letters of the final name (568 for "L-O-V") and take heed to the outcomes. This script runs constantly—a daemon in Unix parlance—and waits for socket connections from FreeSWITCH on port 8040. As soon as a socket connection is established, the script forks a toddler process. This baby course of then creates the ESL connection object $con. Once the $con object is created we are saying a greeting to the caller after which enter the outer while loop. This loop causes the script to exit if the caller hangs up.

The inside whereas loop uses the play_and_get_digits software to truly play the immediate and acquire the digits from the caller. We then learn again to the caller the digit she or he pressed using the say software. Finally, if the caller dialed the digit 9 then we are saying "goodbye", after which hold up. The youngster process then exits however the father or mother continues to be working. You can have multiple simultaneous calls in existence and each one will get its own course of. You can use this script as a template for creating your own interactive dialogs. All of the caller interactions take place throughout the inner whereas loop, so focus your attention there. Also, should you plan to play various sound prompts to the caller make certain to evaluate the recipe Use phrase macros to build sound prompts in Chapter 5. Getting prepared This recipe relies on the occasion socket interface to FreeSWITCH. However, there are lots of different ways of connecting to the occasion socket.