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For the registration function, Asterisk SIP protocol stack provides two services,
1. Asterisk is used as the SIP client and registered with other
Tags: blog http os ar using file data on artOriginal: Asterisk real-time add SIP number--sqliteAsterisk real-time add SIP number--sqliteToday, I tried to use Asterisk's real-time mode to add a SIP account to SQLite without restartingAsterisk, no need to reload, you can successfully register a
FreePBX SIP TrunkDockingbackground: PBX1 is a virtual machine running FreePBX, whichnow needs to be connected via SIP TRUNK docking , PBX2, using PBX2 E1 The line calls out the phone. PBX1 192.168.100.1PBX2 192.168.100.2PBX1on the configurationOneConfigurationTrunkNew SIP TRUNK650) this.width=650; "src=" http://s3.51
In asterisk, there are three types of peer: Peer, user, and friend.Let's take a look at the three types of VoIP-info.
Peer: a sip entity to which asterisk sends CALS (a sip provider for example ). if you want a user (Extension) to have multiple phones, define an extension that CILS two
Basic settings of SIP Trunk in trixbox
Http://www.voclub.net/zone? Action-viewthread-tid-1065The basic settings of the SIP Trunk in trixbox are as follows: the extension can call the phone number through a SIP Trunk, and then ring
exists, asterisk considers this request to be re-invite (! P-> owner). Otherwise, it is considered as a new invite. There are many stories about re-invite, involving whether Asterisk is b2bua or proxy. Next we will discuss non-re-invite requests.
See from printed information
Ast_verbose ("using invite request as basis request-% s/n", p-> callid );
Using invite request as basis request-zjriyjzkyzyzzdnjndr
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View hardware configuration # Dahdi_hardware==============================================View Dahdi Service ConfigurationMore/etc/dahdi/system.confShow the following content, obviously less my PSTN card configuration# Global DataLoadzone = usDefaultzone = usRebuilding the Dahdi service configuration#dahdi_genconfView Dahdi Service Configuration again# more/etc/dahdi/system.confShow# autogenerated by/usr/sbin/dahdi_genconf on Wed Aug 15 22:09:20 201
Label:Recently studied asterisk configuration, before the SIP account is configured in the sip.conf file, manual writing dead, the current demand, is the dynamic new SIP account, saved in the database.After adding data to the database, use the command SIP show users to not load the
Is the call flowchart of Asterisk:
We use the call process of SIP as an example to describe the call process of other channels.
The call process (incoming) is as follows:
Do_monitor-> sipsock_read-> handle_request-> handle_request_invite-> sip_new/ast_pbx_start-> pbx_thread->__ ast_pbx_run
-> Ast_spawn_extension-> pbx_extension_helper-> pbx_exec-> execute dialplan
When the chan_sip module is loaded, an inde
The network structure is as follows:Asterisk (192.168.1.99) That is to say, both Asterisk and SIP terminals are behind Nat.
The solution is as follows:1. Modify the SIP Extension settings in the SIP _. conf file.Nat = YesQualify = yes; it seems this item is not requiredExternip = 55.66.77.88; change to match our exter
Release date: 2011-12-08Updated on: 2011-12-09
Affected Systems:Asterisk 1.xDescription:--------------------------------------------------------------------------------Bugtraq id: 50990
Asterisk is a free and open-source software that enables the Telephone User Switch (PBX) function.
Asterisk has a security vulnerability. Attackers can exploit this vulnerability to obtain valid user names.
When the regu
Release date: 2011-12-08Updated on: 2011-12-09
Affected Systems:Asterisk 1.xDescription:--------------------------------------------------------------------------------Bugtraq id: 50989
Asterisk is a free and open-source software that enables the Telephone User Switch (PBX) function.
Asterisk has a security vulnerability in implementation. Attackers can exploit this vulnerability to cause invalid memory
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