sip speakerphone

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

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 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

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

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

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

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

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

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 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

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

"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

novice?The discovery directory has a lot of configuration files, is it necessary to change these configuration files two times? Let's take one to learn.FreeSWITCH is configured by default to 1000 to 1019 (ext.) A total of 20 users. Let's not start by downloading a SIP phone client on our own phone and try to talk.First Ipconfig/all know your LAN address. The password default appears to be 1234.My own IP is 192.168.0.113, and then I correspond with th