Elastix sip.conf file




















Change-Id: I2a6c6cd7c2f5f30d1dfe3ea 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. Improve this question. What is question? This question is like "why the car is not go and how to fix".

Becuase you have to understend what are you doing maybe? Can You share with some help? However you should understand what is nat for, how network structure affect sip etc etc. There are no info and not think anyone will test your config instead of you just to find where is error.

You can check sip debugging guide or this command list. Show 1 more comment. Active Oldest Votes. Previous post. Next post. Skip to content In this article I will try to explain that how to Edit sip. The sip. Sipura 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. The first section is called general which cannot be used as a client name. The following sections begin with the client name in brackets, followed by the client options. The default is , in keeping with standards.



0コメント

  • 1000 / 1000