设为首页 收藏本站
开启辅助访问
注册会员 找回密码

VoIP88

VoIP88 Aster+ 技术文档 查看内容

Asterisk SS7 sangoma A104DE 对接 爱立信 Ericsson™ AXE® Switch

2013-11-28 21:14| 发布者: james.zhu| 查看: 1515| 评论: 0|来自: http://www.voip-info.org/wiki/view/Interconnecting+Asterisk+SS7+with+Ericsson%E2%84%A2+AXE%C2%AE+Switch

摘要: VOIP info 客户发的如何使用asterisk chan_ss7, Sangoma A104DE 对接 爱立信软交换。

   

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)

Installation

  1. Install the Sangoma or any E1 Hardware to your linux box
  2. Install Dahdi, if you don’t know how, consult the following link:
  3. Install Sangoma wanpipe which the installation instruction is available from the below link:
  4. Install Free version of Chan_SS7:
  5. Install Asterisk:

Testing and Configuration:

If you are using sangoma it is needed to start the wanrouter engine by
  #wanrouter start


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=fr
defaultzone=fr

#Sangoma A104 port 1 [slot:4 bus:3 span:1] 
span=1,1,0,ccs,hdb3
bchan=2-31
mtp2=1

#Sangoma A104 port 2 [slot:4 bus:3 span:2] 
span=2,2,0,ccs,hdb3
bchan=33-62
mtp2=32

#Sangoma A104 port 3 [slot:4 bus:3 span:3] 
span=3,3,0,ccs,hdb3
bchan=63-93

#Sangoma A104 port 4 [slot:4 bus:3 span:4] 
span=4,4,0,ccs,hdb3
bchan=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 siuc
enabled => yes 
enable_st => no 
use_connect => yes  ; <-- Reply incoming call with CON rather than ACM and ANM
hunting_policy => even_mru
context => ss7-in; <-- this is refered the context in /etc/asterisk/extentions.conf for incoming SS7 traffic 
language => en 
t35 => 15000,timeout
subservice => international

[link-1]
linkset => siuc
channels => 2-31 <-- 2-31 are used for speech/audio/voice
schannel => 1 <-- channel 1 is for signaling 
firstcic => 1 <-- start the first channel from 1
sls=0
sltm => no
enabled => yes
stp => 11111; <-- if you are connected to STP point set the related value here

[link-2]
linkset => siuc
channels => 2-31
schannel => 1
firstcic => 33
sls=1
sltm => no
enabled => yes
stp =>  11111

[link-3]
linkset => siuc
channels => 1-31
schannel =>
firstcic => 65
sltm => no
enabled => yes
stp =>  11111

[link-4]
linkset => siuc
channels => 1-31
schannel =>
firstcic => 97
sltm => no
enabled => yes
stp =>  11111

[host-ast1]
enabled => yes
opc => 1223 ; <-- Originating point  code which is our ID in the SS7 network
dpc => siuc:11110; <--  The destination point (peer) code 
links => 1:1,2:2,3:3,4:4
globaltitle => 0x00, 0x03, 0x01,  4546931411
ssn => 7
route =>  :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,Answer
exten => s, 2,playback(welcome)
exten => s, 3,HangUP()
####################################################


鲜花

握手

雷人

路过

鸡蛋
关闭

站长推荐上一条 /1 下一条

手机版|VoIP88 ( 粤ICP备11095982号   填写您的邮件地址,订阅我们的精彩内容:

GMT+8, 2017-9-22 17:44 , Processed in 0.271651 second(s), 24 queries .

Powered by VoIP88

© 2001-2017 VoIP88

返回顶部