At the request of the dynaguy brothers, I have attached a relatively complete experiment to colleagues who are still exploring:
(There are some problems with the 2.0beta experiment, so I will demonstrate it with the most stable 1.2.3)
In this test, we did not discuss the issue of using the SIP Trunk directly to connect other sip servers without board installation to enable the SIP users to make external calls.
This article is only used to discuss questions about using the fxo card to test intern
Http://www.mihua.net/node/279m.htm
Linphone and X-Lite are the most famous ones.Soft Phone soft phones (Open Source)
Source code can be downloaded and modified
Name
Description
Actxphone
An ActiveX-control SIP softphone Based on the Microsoft Real Time Communications (RTC) API. Written by http://www.pernau.at/kd/voip/ActXPhone/. VB
Ekiga
SIP, H.323 audio and video softphone
from domain also contains a display name (Alice) and a sip or sips uri (SIP: alice@atlanta.com) which is used to mark the request's original initiator.This field also contains a tag parameter, which is a random string (1928301774) and a random string added to the URI by softphone. Used for marking purposes.Call_id contains a globally unique identifier used to uniquely identify a call. It is generated by using a random string and softphone's own name
arbitrary, not subject to the initial size of the limit, if set to 0, then empty.
Assign (5,0) changes the vector to 5 sizes and fills with 0
Assign (iax+3,iax+5); From the 4th to 5th fill of the array, note that the left is closed to the right, you can take to iax[3] and iax[4]
9. Use Insert
Insert (it, x), inser
first, iterator last)I. Prototype And ConstructorTypdef list ConstructorList () // nullList (Al) // specify the empty table of AllocatorList (n) // n elements. All elements are T ().List (n, Val) // n elements, all elements are T (VAL)List (n, Val, Al) // same as above, and specify Allocator as AlList (first, last) // copy StructureList (first, last, Al) // specify Allocator ConstructionIi. Operations1. Resize clearUse resize (n) to change the size, and use resize (n, Val) to fill the idle val
Packet Reflector/Forwarder: A tiny tool which forwards or reflects UDP packets. you can also add delay and packet loss. very useful if you want to test RTP applications. homepage: http://www.cs.ucl.ac.uk/staff/s.bhatti/teaching/z02/reflector.html. as I was not able to compile this tool I searched and found a binary somewhere in the web. you can download it local
SIP Phones (SIP User Agents)
X-lite, x-pro: A sip client for Windows; Mac OS and Windows CE, http://www.xten.com/. A really nice S
this tool I searched and found a binary somewhere in the web. you can download it local
SIP phones (SIP user agents)
X-lite, X-Pro: A sip client for Windows; Mac OS and Windows CE, http://www.xten.com/. A really nice SIP UA with a lot of features. The light version is free and reallyRocks, The Pro version not. Supports multiple proxies.
Eyep phone Lite: A sip client for Windows, a fwd version is available for free http://www.eyepmedia.com/eyePPhoneFWD.htm.
Sipps: SIP
callers whose names start with 00 are processed using context as international, while those whose names start with 0 are processed using longdistance. The default value is used for processing by default.
Then you can add the context section after the configuration file, for example:[International][Longdistance][Default]
B) Route stage
In the route stage, the called number matches the Regular Expression in sequence according to the context section. After successful match, the called number mat
http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/www.mundoopensource.com.br/xmpp-asterisk-integration-practical-example-part-2/In past few years integrate XMPP and Asterisk are one of my principal goals. The development of an realtime communication tool with open source software necessarily needs this, technologies and MA Ke It happens is one of my intents.Nine years ago, since I started to work with Asterisk and OpenFire I ' m studying a-to-do it and when I discovered th
Iptables fo Asterisk
Sip on UDP port 5060. Other sip servers may need TCP port 5060 as well
Iptables-A input-p udp-m udp-dport 5004: 5082-J accept
Iax2-The IAX protocol
Iptables-A input-p udp-m udp-dport 4569-J accept
IAX-most have switched to IAX V2, or ought
Iptables-A input-p udp-m udp-dport 5036-J accept
RTP-the media stream
Iptables-A input-p udp-
IptablesAndAsterisk:
Iptables fo asterisk
SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well
Iptables-a input-p udp-m udp-dport 5004: 5082-j ACCEPT
IAX2-the IAX protocol
Iptables-a input-p udp-m udp-dport 4569-j ACCEPT
IAX-most have switched to IAX v2, or ought
Iptables-a input-p udp-m udp-dport 5036-j ACCEPT
RTP-the media stream
Ipta
)
* Trixbox (IP phone server software ).
* Any softphone or hardphone.
Now lets start the process.
1)Download trixbox ce 2.6.2 (stable) from the following link.
Http://master.dl.sourceforge.net/sourceforge/asteriskathome/trixbox-2.6.2.2.iso
After downloading if you are gonna use it on dedicated machine burn the image into CD otherwise you can use ISO with Vmware or any other virtualization software.
2)Here I have assumed that you are using virtual mac
Connect two asterisk servers
There are two asterisk servers. You can call the extension number registered on the other server.
Assume that there are two servers A and B. Extension A starts with extension 3, such as 3000. Extension B starts with extension 8, such as 8000.
Create an IAX trunk on a and name it "interoffice". The configuration is as follows:"Peer detail" itemIP address of host = BQualify = YesType = friendSet a dialing rule, which is
The SIP protocol and some of its applications are worth learning. In this regard, we will explain the configuration of rtp sip today. SIP (Session Initiation Protocol) is usually used for VOIP calls, call establishment, call negotiation, and call termination. it helps two terminals to recognize each other, but it does not process media. When A call is established, it directly transmits media through real-time transmission protocol (RTP) between telephone A and telephone B.
SIP and RTP
SIP is a s
provides comprehensive voice communication with Cisco uniied CallManager and Cisco uniied CallManager Express.
GSM/802.11 IP Phone Fixed-mobile converged IP Solution with Nokia dual-mode commercial telephones and Cisco wired and wireless IP infrastructure. Cisco is working with Nokia to develop mobile phones.
Video IP phone number Cisco uniied IP Phone 7985G is a personal desktop video phone. Cisco uniied IP Phone 7985G uses all the components used to support video calls in a single easy-to-use
the Cisco IP softphone product. cisco IP softphone is a PC based telephone integrated with avvid, and works with the Cisco Call Manager. the primary focus of the winrtp is to ensure that it works well with other products in avvid including desktop IP phones, gateways, etc. it can also be used as an independent component .; it is written in C ++; it is a COM component. (not an ActiveX control ). this makes
Connect! The product is a "terminal service that supports SIP and IAX due to enterprise level requirements", Connect! The service is prepaid, and the cost is to call the U.S. phone number for $11 per month. Broadvoice's BYOD provides various rate schemes for you to choose from (from $9.95 to $29.95 per month), but you can use your own device, of course, Asterisk PBX.
Install AAH
Asterisk @ Home has been bound with CentOS, a Linux-based operating syst
have the option to load modules. Each of the modules you load provides different system functions. For example, some modules can make your asterisk communicate with analog telephone lines, and some modules provide the function of traffic report. Later, we will also discuss the functions and categories of the various modules.Http://www.cnblogs.com/einyboy/archive/2012/11/08/2759969.htmlhttp://blog.csdn.net/yetyongjin/article/details/7520567SIP functionality for AsteriskThe Asterlsk can support t
This article, the original connection: http://blog.csdn.net/freewebsys/article/details/46546205, reprint please indicate the source!1, about FreeSWITCHFreeSWITCH is a soft-switching solution for telephony, including a softphone and soft switch to provide voice and chat product drivers. The FreeSWITCH can be used as a switch engine, PBX, multimedia gateway, and multimedia server.FreeSWITCH supports a variety of communication technology standards, inclu
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