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Discussion on SIP server problems (1)

Some friends may be familiar with the SIP protocol. In this regard, the most prominent thing is the VoIP service. In VoIP services, the SIP protocol and the SIP server are often involved. Next, let's take a look at the traversal problem on the SIP server. 1. Description of SIP

MAC EI Capitan Update system comes with SVN version number (Turn off SIP side sudo rm)

Following last night. decided to update the SVN that comes with the system. The SVN version number that comes with it is 1.7. Crossing Network Svn:http://www.wandisco.com/subversion/download#osx The latest version number is 1.9.13, decided to upgrade under.Unexpectedly due to EI Capitan sip problem Toss a good freshman meeting. Would not have wanted to record. But because sip this egg-ache thing decides sti

Configure a SIP gateway to dial an external phone

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 create an XML file in conf/sip_profiles/external/, with a name that can be randomly used, s

Linux Firewall NAT-SIP network topology, natsip

Linux Firewall NAT-SIP network topology, natsip Environment: Firewall: Ubuntu Server 17.10. SIP-Yate client, Asterisk Server Network Topology: 1) configure the IP address of the firewall and enable forwarding: ifconfig enp2s0 192.168.1.131 ifconfig enp3s0 192.168.100.1 echo 1 > /proc/sys/net/ipv4/ip_forward 2) load the nf_nat_sip module to create expectations: modprobe nf_nat_sip 3) configure th

SIP Protocol Response Code

Answer code The answer code is included, and the http/1.1 answer code is extended. Not all http/1.1 answer codes are properly applied, and only the appropriate ones are indicated in the fold. Other http/1.1 answer codes should not be used. Also, SIP defines a new answer code series, 6xx. 1 Temporary answer 1xx A temporary response, a message-nature response, flags that the other server is processing the request and has not decided on the final answe

FreeSWITCH connecting the external SIP Provider

1, added SIP Provider, add the configuration file in Conf/sip_profiles/external This SIP Provider requires REGISTER, and since FreeSWITCH is accessed through NAT, it is set to send pings for 30 seconds. 2, the did mapped by this SIP Provider to the corresponding extension SIP Profile External.xml Sets the context d

WebLogic SIP Server and conference Application

At BEA, my role is to help build and support ISVs of applications on WebLogic Communications Platform to build an ecosystem. It is easy to use WebLogic SIP Server to compile an aggregate J2EE/HTTP/SIP application. This article details the architecture behind a complete conference application that uses Cantata's Media Server to stream audio and video. The two smart developers took less than a month to comp

When SIP is in progress

SIP, which has always been known as "simple", is not so simple, but it is difficult to grasp anything.This document is designed to keep track of the various doubts and problems encountered during SIP usage.First, Response 422 Session Interval Too SmallThe invite messages sent are as follows:INVITE SIP:806@192.168.8.11sip/2.0Via:sip/2.0/ws 9srpbdc87v1s.invalid;bra

Application of Stun/turn/ice protocol in peer-to sip (II.)

Reprint: http://www.cnblogs.com/ishangs/p/3816689.htmlApplication of Stun/turn/ice protocol in peer-to sip (II.)1 description2 dozen holes and the concept of crossing ... 13 hitting holes and crossing ... 24 using the STUN Series Protocol traversal features ... 25 Stun/turn/ice The relationship of the agreement ... 36 Stun protocol (RFC 5389) 36.1 Why the STUN protocol is used ... 3How the 6.2 stun protocol works ... 47 Turn protocol ... 47.1 Why the

FreeSWITCH Docking Other SIP devices

These days use to FreeSWITCH docking other equipment knowledge, here to tidy up, also convenient I later check. Operating system: debian8.5_x64 FreeSWITCH version: 1.6.8 First, FreeSWITCH as the called deviceFreeSWITCH as a device and other devices docking the situation is relatively simple, you can directly through the 5080 port inbound.FreeSWITCH default configuration turns on port 5080 docking (for public in conf/dialplan/public.xml):extensionname= "Public_extensions"> co

Basic settings of SIP Trunk in trixbox

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 the trunk number to the extension. Create a new SIP Trunk, provided that you have obtain

Implement dual-stream through the SIP protocol

This article original from the http://blog.csdn.net/voipmaker reprint indicate the source. Dual-stream is the concept in video conferencing. It generally means that the client can display two video streams at the same time, one is the main video (main), and the other is usually the content sharing (slides ), it is also called share content. Content is usually screen, PPT, document, and other content. The SIP protocol implements dual-stream. The SDP c

Sip rfc 4538 authorization request through dialog

Background: Generally, when a UA receives a request that creates a dialog (invite/Subscribe/refer), it must decide whether to authorize the request, in some cases, UA determines whether to authenticate the request by determining whether the request is in a created dialog, For example, after invite creates a dialog, UA does not need to authenticate other requests (prack, Act, etc.) sent in this dialog, but the problem is that for refer, message, and SUBSCRIBE requests, A new dialog will be crea

The next day, learn about SIP (FreeSWITCH add recording function) (2)

Learn some of the FreeSWITCH core commands, and then learn more about FS in detail.To see if it was not previously suspected, two times when programming changes the configuration file, or Java injects some parameters into the configuration file, learn more about the following configuration file.This should be difficult, not clear.Ask Mr. Baidu.Learn a new knowledge of how FS adds recording functions to configureThe general telephone system can record voice calls in the system, and voice recordin

Differences between update and re-invite methods in SIP

In sip, both re-invite and update are used to modify the session parameter. The difference is that update does not affect the status of the dialog, and re-invite changes the status of the dialog. Therefore, update can be sent before the first invite is responded (that is, before 200ok of invite is received ). That is to say, update can be used to control early media. The re-invite can only be sent after the first invite cup responds (that is, after th

Conflict between MessageBox and sip on PDA Platform

InProgramWhen the MessageBox pop-up error message is applied, when you click the OK button, it is found that the keyboard icon on the mainmenu has disappeared. You must click it once to appear. Although it does not affect the use of normal functions, it may be confusing for PDA cainiao, and many methods to refresh the interface cannot be solved. Baidu cannot find the answer to this strange problem. You can directly query the problem on Google and find that foreigners also encounter this proble

Design and Implementation of the SIP protocol stack-Transaction Layer

A transaction is a request transaction sent by the customer (through the communication layer) to a server transaction, together with all the responses to the request of the server transaction, sent back to the client transaction. The transaction layer processes the re-sending of the application service layer, matches the response of the request, and the timeout of the application service layer. All tasks completed by a user agent client UAC are composed of a group of transactions. Generally, a

Comprehensive application of SIP and RTP 3

Based on the practices in the past few days, we have found an Optimal Configuration: 1. The SIP server uses trixbox. If you are familiar with Linux, we recommend that you use asterisk directly. 2 If the client is used directly, it is recommended that ekiga. By the way, how do I feel when using several clients: 1 Linphone: It seems to be well known. However, the latest version 3.1.2 crashes after being installed. I installed a general XP-SP3, compu

Python graphical interface Development programming: WxPython (SIP)

, has been completely attracted by WxPython , can not wait to practice, hereby record the individual learning process (Windows system):First, Wxpython environment installationThe most realistic and practical way to install Pip, and must be used in a manner that will synchronize the installation of dependent packages: pip install-u wxPythonsecond, Wxpython tastedAs a beginner, must not blindly directly into the subject, it is necessary to go through this stage, can be very good to help understand

Python/c++ Interface Library Comparison (Swig,boost.python, Pycxx, py++, sip, Weave, Pyrex)

"Python/c++ Interface Library comparison" (Swig,boost.python, Pycxx, py++, sip, Weave, Pyrex) http://blog.csdn.net/lainegates/article/details /19565823There are many open-source python/c++ binding tools, search a lot of 岾, here a little summary.SWIG Supports Python 2 and 3 Properly configured, the package can be fully automated (*.i files need to be written by themselves) When it is not fully automatic, it will mostly repeat your. h f

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