asterisk sip conf

This article is from csdn lidp and reposted the famous source. Thank you. For information and technology trends in the VoIP industry, see www.voip123.cn. For the registration function, Asterisk SIP protocol stack provides two services, 1. Asterisk is used as the SIP client and registered with other

to call through your asterisk. friend is both. But what is the actual situation? Try the following definition:[1001]Username = 1001Secret = XXXXXHost = dynamicPort = 5060Type = peerContext = from-extensions Then, use a SIP client to register it with your Asterisk. What happened?You will find that the client can also be registered! In fact, the true significance

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

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

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 requiredExte

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

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

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

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

============================================== 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