yate voip

From the beginning has not yet started to write the server above the code, whether the C/s or b/s are basic contact and achieved, from the beginning of the socket communication, to the transplant boa, realize CGI, groping VoIP server,web server (Php--phalcon), Web server (JAVA-SSH) has been maintained. Contact Yate server time is the longest, but also for the first time a system has a relatively full unders

CentOS-based installation Yate1 PrefaceAfter thinking, a lot of things now understand, will use. It's a long time to bug fix this system. Many know things will forget the same, need to re-spend more time to understand and learn. As the saying goes: good memory is better than bad writing. Or the honest use of words and pictures of the method to record.In order to let the future of their own ease, as long as the present self to do a little more work, re-start to build

The previous article recorded the Yate learning--./yate/tools/log_rotate.sh;Continue reading and Logging section of the script today./yate/packingyate.logrotate, first look at the script source code.# Rotate the log and CDR files before each reaches 2GB in SIZE/VAR/LOG/YATE/VAR/LOG/

Document directory I. Introduction II. Implementation Method Iii. regexroute Iv. Specific Configuration 5. Regular Expressions I. Introduction Based on the research progress, we already know that on a Yate server, we can use the simple registration and Authentication Module: regfile to configure user information. users in the same domain can successfully register on a Yate server, and the call is suc

ArticleDirectory Fork Module No. 1 dual-host The No. 1 dual-host service refers to the caller who calls a called phone number. Two or more calls bound to the called phone call at the same time. Either of them can answer the call first. After the first call, the other calls will stop ringing. Most of the functions implemented by traditional switches are implemented on the intelligent network platform (in), which can be easily implemented on the softswitch platform such as

Please state the source:Namedcounter, Object-named counter:/** * An atomic counter with an associated name * Atomic Counter of association name * @short Atomic counter with Name * @short Atomic counter */class YA Te_api namedcounter:public string{ynocopy (namedcounter);//No automatic copies Pleasepublic:/** * Construct or * constructor * @param name Name of the counter * @ parameter name, counter name */explicit Namedcounter (const string name); /** * Check if the counter is enabled * checks

The VoIP-a reference guide to all things VoIP This wiki covers everything related to VoIP, software, hardware, service providers, reviews, deployments, standards, Tips tricks and everything else related to voice over IP networks, IP telephony and Internet telephony.''Welcome to voip-info.org! Please e let me know at s

Domain: Set the local domain name Outbound Proxy: sets the border server. Registration interval: Internal registered Server Authentication Username: Fill in usrname in Verification Authentication For details, the username in the authorization field is not the actual local user name, A user name that can be changed (becomes the authentication user name) The default port used by the Yate software is 5060, which means that if port 5060 is not occup

News This section is for news, ie news reports, press releases, product release announcements etc. Research: peer-to-peer Internet telephony using SIP PDF Iconv application module for character conversion. Version 0.9.2 of ldapget application module released. bugfix. Over 5 million VoIP subscribers worldwide-dmeurope story Interviews with BKW, twisted and David Mandelstam Interview with drumkilla, the manager of the stable branc

Only the configuration file is pasted as follows, without explanation! Removed many features! Only the SIP Phone is retained! [general]; General settings for the operation of Yate; modload: boolean: Should a module be loaded by default if there is no; reference to it in the [modules] sectionmodload=disable; modpath: string: Overrides the runtime module directory path which is; compiled in or specified with the -m command line option. Note that this

Reprint Description:All base classes based on stream operations in Yate:/** * Base class for encapsulating system dependent stream capable objects * encapsulates the base classes of stream objects that can be based on systems * @short an abstract stream c Lass capable of reading and writing * @short able to read and write abstract classes */class YATE_API stream{public:/** * Enumerate seek start P Osition * Open Search for Loca

Use FireBreath to develop real-time playback interfaces (Yate + SIP + FFMPEG + SDL) and firebreathyate At that time, such a blog post was really needed to guide this function module. Unfortunately, FireBreath has very little information on the Internet and is not very familiar with C ++, so we tried and explored it all the way. Fortunately, we have implemented this module, and now we have recorded it. First of all, our

Please state the source:Messagereceiver, this class is a base class for message acceptance:/** * A multiple message receiver to being invoked by A message relay * Multiple message Receiver call messaging * @short A multiple Message receiver * @short multiple message Receivers */class YATE_API messagereceiver:public genobject{public: /** * This method was called from the Me Ssage relay. * This method is called when the message is forwarded * @param msg the received message * @

instantiated */virtual void* getObject (const string name) const; /** * Check If the object is still referenced and safe to access. * Check to see if the object is still referenced and secure access * Note that you should is not the trust this resUlt unless the object is locked * Note that this result cannot be fully trusted unless the object is locked by another method * by other means. * @return True if the object is referenced and safe to access * @ returns TRUE if the objects are refere

Too many VoIP service providers want to sell you their "full solution", from the phone number on your desk, from different sites to the WAN and public exchange Telephone Network (PSTN). However, as I have seen, Unless users and suppliers have full experience and thoroughly checked every detail, the so-called "full set of VoIP systems" will certainly make some mistakes. Enterprises that have trouble with

In the current network communication, the Email service is no longer the preferred communication method. More instant messaging and voice services are emerging on the network. Now let's talk about the technical principles of VoIP for IP phones.Basic transmission process The traditional VoIP telephone network transmits voice in a circuit exchange mode. The required transmission bandwidth is 64 kbit/s. The so

download tar packages from: http://osip.atosc.org/download/partysip. Mysip: A sip proxy server from Siemens for Windows platforms. homepage: http://www.mysip.ch/ Fomine RTC Server: A sip proxy server for Windows which uses its own SIP stack (does not need the rtc api) homepage: http://www.fomine.com/rtc-server.html. The Unregistered version can be used up to 5 users. Sipxpbx: Part of pingtel's open source releases for VoIP. license: lgpl; homepage

Every organization that is considering deploying a VoIP Phone System has heard the same terrible warning: "routing voice calls over the data network will expose the call content to eavesdroppers ".Although in fact, any phone call is at risk of being eavesdropped to some extent, is the VoIP call system itself at a high risk? In this article, we will explore the ins and outs of

://www.asteriskpbx.com/ Sipd: A Linux SIP proxy from SX-Design written in C (GPL): http://www.sxdesign.com/index.php? Page = developer submnu = sipd Partysip: A Linux SIP proxy based on osip2 (LGPL). Developer homepage is at: http://savannah.nongnu.org/projects/partysip/, you can download tar packages from: http://osip.atosc.org/download/partysip. Mysip: A sip proxy server from Siemens for Windows platforms. Homepage: http://www.mysip.ch/ Fomine RTC server: A sip proxy server for Windows wh

How can I test the VoIP function with an existing PBX or key-press system? There are multiple ways to use the existing PBX system or key-press system to test the VoIP function. How to test the function depends on your purpose. If there are two sites connected with PBX connection lines, but you want to use VoIP so that you can send calls between internal network