asterisk sip gateway

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

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

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

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

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

Introduction 1. About the SIP protocol Currently, there are three basic communication protocols based on VOIP: H 323, SIP, and MGCP. The following describes the comparison between the H 323 protocol and the SIP protocol. H.323 was proposed by ITU. It attempts to regard IP phones as well-known traditional phones, but the transmission mode is changed from circuit s

If you have a SIP account from a carrier, you can configure the SIP to dial an external phone. The SIP account (or the device providing the account) is referred to as the SIP gateway in FreeSWITCH. Adding a gateway only needs to c

This article from csdn ucser, http://blog.csdn.net/perfectpdl reprinted to indicate the source, thank you! I have created a freeswitch learning and communication group, 45211986. welcome to join. Freeswtich can be used as the rtmp and SIP gateway of the Streaming Media Protocol. It can communicate with the SIP video phone through flash in a web browser. Th