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james.zhu 发表于 2013-11-28 21:48 | 显示全部楼层 |阅读模式
   
国外的文章,不翻译,不误导。不怕被了解,就怕不了解!
History and Propose of the solution
SS7 is suit of protocols and the language between telecom switches. since VOIP is new, still many telecom networks require SS7 for interconnection. There are different solutions and hardware available for interconnection between VOIP and SS7 networks. In this document I will propose a simple way of interconnection if your telecom counterpart is running AXE switch and you are looking for a reliable and cheap and fast interconnection solution.
For 4 E1 link Capacity:
First of all it is suggested to buy E1 hardware in order to physically connect to AXE Switches.
There are variety products from cheap Chinese, to expensive with hardware echo-cancellation based hardware are available, for this interconnection I used Sangoma™ hardware which has a good reputation and fairly stable with echo cancellation.

For this project I used:
  • Asterisk 1.8.22.0
  • Dahdi 2.6.2
  • Chan_SS7 7.2.2
  • Sangome Wanpipe 7.0.5 ( only if you are using Sangoma Hardware)

InstallationTesting and Configuration:If you are using sangoma it is needed to start the wanrouter engine by
  #wanrouter start  // 启动sangoma A104DE 驱动

If everything is fine you will receive the following messages like:

Configuring interfaces: w1g1
done.
Configuring interfaces: w2g1
done.
Configuring interfaces: w3g1
done.
Configuring interfaces: w4g1
done.
Then it is time to configure the dahdi; so we need to modify the /etc/dahdi/system.conf file:
In this example I used channel 1 and 32 for signaling (mtp2=1 mtp2=32)
####################################################loadzone=frdefaultzone=fr#Sangoma A104 port 1 [slot:4 bus:3 span:1] <wanpipe1>span=1,1,0,ccs,hdb3bchan=2-31mtp2=1#Sangoma A104 port 2 [slot:4 bus:3 span:2] <wanpipe2>span=2,2,0,ccs,hdb3bchan=33-62mtp2=32#Sangoma A104 port 3 [slot:4 bus:3 span:3] <wanpipe3>span=3,3,0,ccs,hdb3bchan=63-93#Sangoma A104 port 4 [slot:4 bus:3 span:4] <wanpipe4>span=4,4,0,ccs,hdb3bchan=94-124####################################################

When the modification is done
#dahdi_cfg –v #dahdi_tool
Wait until alarms become OK

edit /etc/asterisk/ss7.conf
####################################################[linkset-siuc]; <--the name of Linkset which is siucenabled => yes enable_st => no use_connect => yes  ; <-- Reply incoming call with CON rather than ACM and ANMhunting_policy => even_mrucontext => ss7-in; <-- this is refered the context in /etc/asterisk/extentions.conf for incoming SS7 traffic language => en t35 => 15000,timeoutsubservice => international[link-1]linkset => siucchannels => 2-31 <-- 2-31 are used for speech/audio/voiceschannel => 1 <-- channel 1 is for signaling firstcic => 1 <-- start the first channel from 1sls=0sltm => noenabled => yesstp => 11111; <-- if you are connected to STP point set the related value here[link-2]linkset => siucchannels => 2-31schannel => 1firstcic => 33sls=1sltm => noenabled => yesstp =>  11111[link-3]linkset => siucchannels => 1-31schannel =>firstcic => 65sltm => noenabled => yesstp =>  11111[link-4]linkset => siucchannels => 1-31schannel =>firstcic => 97sltm => noenabled => yesstp =>  11111[host-ast1]enabled => yesopc => 1223 ; <-- Originating point  code which is our ID in the SS7 networkdpc => siuc:11110; <--  The destination point (peer) code links => 1:1,2:2,3:3,4:4globaltitle => 0x00, 0x03, 0x01,  4546931411ssn => 7route =>  :siuc####################################################

Restart asterisk and issue the command
#asterisk  -rx "ss7 status"
If your installation is correct you have to receive the similar output which shows you have 122 idle channels in SIUC linkset which we defined in /etc/asterisk/ss7.conf

linkset idle busy initiating resetting total incoming total outgoing
siuc    122    0        0                  0                      0                    0
in order to be able to either originate or terminate calls to SS7 peer network we need to modify /etc/asterisk/extensions.conf file
####################################################[ss7-out]exten => _X.,1,Dial(SS7/${EXTEN})exten => _X.,999,Congestion[ss7-in]exten =>  s,1,Answerexten => s, 2,playback(welcome)exten => s, 3,HangUP()####################################################

原文: http://www.voip-info.org/wiki/view/Interconnecting+Asterisk+SS7+with+Ericsson%E2%84%A2+AXE%C2%AE+Switch
cnasterisk 发表于 2013-11-30 09:32 | 显示全部楼层
SS7对接,这个什么玩法啊.高端大气上档次啊.
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