![]() ![]() xml extension - check the existing examples in conf/directory for the layout. Defining usersĬreate an XML definition in conf/directory with a. Check the SIP Provider Examples for the format of the file - it lists example configurations for most of the common providers. ![]() xml extension in either conf/sip_profiles/internal or conf/sip_profiles/external. Defining gatewaysįor the default internal and external profiles, drop a gateway configuration file with a. xml extension - check the existing examples in conf/sip_profiles for the layout. Defining new profilesĬreate an XML definition in conf/sip_profiles with a. By default, FreeSWITCH includes all profiles defined in conf/sip_profiles, and each profile runs independently on it's own IP:port, and can have it's own settings, gateways, etc. Multiple SIP profiles can be configured - in fact, it's encouraged. via the event socket - using global_setvar via an API commandĪ single SIP profile is configured via sip.conf, which stores basic SIP configuration parameters, registrations to gateways, and credentials and configuration for SIP endpoints registering to the server.In a script called by the dialplan - for example, in a Lua script you would use a global_setvar call via the freeswitch.API() object. ![]() ![]() In any other XML file included in the XML diaplan - nice if you have some need to group the variable definitions.In conf/vars.xml - the default config has examples of how to do it.More focus is put on clarifying the most common methods. Not all implementation methods are necessarily discussed in each section.The time spent reviewing these pointers will almost assuredly pay off in time saved from beating your head off a wall, saying "I know how to do this in Asterisk, but." Caveats It is far from comprehensive, but instead a place to hold the various tricks and lessons that people learn in the process. This document is an effort to help with the roadbumps that come up in the conversion. FreeSWITCH works differently, and it can be somewhat frustrating to rebuild a telephony system in FreeSWITCH that was originally built in Asterisk. In some ways, this is actually a handicap compared to discovering FreeSWITCH first, because you have to, in a sense, unlearn 'the Asterisk way'. If you've been involved in open source VoIP and are now looking into FreeSWITCH, chances are you came from a background in Asterisk. ![]()
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