You can download the system easily from the website. Previous post. Next post. Skip to content The sip. Clients must be configured in this file before they can place or receive calls using the Asterisk server. The sip. The first section is for general server options, such as the IP address and port number to bind to. The following sections define client parameters such as the username, password, and default IP address for unregistered clients. Sections are delineated by a name in brackets.
Raw Blame. Open with Desktop View raw View blame. This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters. Learn more about bidirectional Unicode characters Show hidden characters. You signed in with another tab or window. Reload to refresh your session. You signed out in another tab or window. If your Asterisk is installed on a public.
The host or IP address. Asterisk checks the SIP From: address username and matches against. Asterisk checks the From: addres and matches the list of devices.
Devices need a unique. Phone numbers are. Check below. In later releases, it's renamed. To enable callcounters, you use the new. Defaults to 'default'. Default is yes. Uses the Incomplete application to.
The Dial options 't' and 'T' are not. If you set a system name in. Defaults to 'automon'. Works with. Feature must be usable on requesting.
Setting this value to a blank. In cases a and c above, only A records are considered. In case b , only. In case d , when both A. On systems using glibc, AAAA records are given. However, some endpoints either do not include an Allow header. In the former case, Asterisk. Note that. This option may be set in the general section or may. If this option is set both in the general section and. Its use may be expanded in the future.
Since it is new, all of the related configuration options are. If they are changed, the changes will. The order determines the primary default transport. Enable this option to not get error messages. Also fill the. But, after the caller. Asterisk will. It may be specified globally or on. In the. The way legacy.
The SIP. Default is "yes". Also make sure that. If you don't want to expose this, change the. Default: rfc You need to turn this. This assists callfile-derived calls and.
Peerstatus will be "rejected". This reduces. By default this option is disabled. By default, this option is enabled. When this. This option can be defined at both the peer and. That is "demo-alice" is the name you'll have your phone authenticate against when registering. This will be covered more in the next wiki section on registering phones to Asterisk.
Take a minute to look over the pjsip. Then backup your pjsip. The pjsip configuration is a little more complex as the channel driver's architecture allows for more flexibility in configuration, so things tend be more modular and broken out. After the templates, you'll see the definitions of the demo-alice and demo-bob endpoints. In those, all we do is tie them to their own auth and aor sections. In their auth sections we set their unique usernames and passwords.
In their aor sections we don't set anything, because the necessary settings were already set in the template. We have more configuration to do in the next section, so there is no need to load configuration yet.
Therefore it is easiest to restart Asterisk completely here since we have made set transport settings. Evaluate Confluence today.
0コメント