Test chan_ss7 with two OpenVox D110P cards

zhulizhong

Test chan_ss7 with two OpenVox D110P cards

Written by: James.zhu(zhulizhong@gmail.com)
Date: 29/12/2007

SS7 is a very important protocol in telecommunication. Many users use in their business. We know that SS7 environment is easy to get, if we want to test ss7. Thanks, Knielsen, he has published the reference from voip-info.org. Here we give a more details simple test environment to test ss7 with two OpenVox D110P cards. Some steps have to taken in both of two servers(new-host-3 and new-host-4):

Disclaimer
Linux® is a registered trademark of Linus Torvalds
OpenVox® is a registered trademark of Shenzhen OpenVox communication LTD.
Asterisk® is a registered trademark of Digium


1.        Install chan_ss7(we use chan_ss7, not libss7), Asterisk and Zaptel
1.1   Check the support packages, if not installed, please install that.
rpm -q bison
rpm -q bison-devel
rpm -q ncurses
rpm -q ncurses-devel
rpm -q zlib
rpm -q zlib-devel
rpm -q openssl
rpm -q openssl-devel
rpm -q gnutls-devel
rpm -q gcc
rpm -q gcc-c++
rpm -q kernel-devel

1.2  Download chan_1.0.0, asterisk-1.4.15 and zaptel-1.4.7.1

2.        Modify the Makefile in ss7
         You have to edit the Makefile in chan_ss7. Make sure the “INCLUDE” points to your zaptel and asterisk source files.
         # INCLUDE may be overridden to find asterisk and zaptel includes in
         # non-standard places.
         INCLUDE+=-I../zaptel-1.4.7.1 -I../asterisk-1.4.15/include
         CC=gcc

3.        Compile zaptel, asterisk and chan_ss7
3.1        Compile zaptel->./configure->make->make install
3.2        Compile Asterisk->./configure->make->make install
3.3        Compile chan_ss7->make->make install
3.4        Copy the chan_ss7.so to /usr/lib/asterisk/modules
3.5        Copy ss7.conf to /etc/asterisk

4.        Configure ss7.conf , zaptel.conf and extensions.conf
ss7.conf
[linkset-siuc]
enabled => yes
use_connect => no
enable_st => yes
hunting_policy => even_mru
subservice => auto
context = ss7
language => en
[link-|1]
linkset => siuc
channels => 1-15,17-31
schannel => 16
firstcic => 1
enabled => yes

[link-|2]
linkset => siuc
channels => 1-15,17-31
schannel => 16
firstcic => 1
enabled => yes

[host-new-host-3]
enabled => yes
opc => 0x1
dpc =>siuc:0X2
links =>|1:1

[host-new-host-4]
enabled => yes
opc =>0x2
dpc =>siuc:0X1
links =>|2:1

zaptel.conf
# Autogenerated by ./genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order

# Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1"
span=1,1,0,ccs,hdb3
bchan=1-31
extensions.conf
[from-internal]
exten => 500,1,Dial(ss7/00453377)       ; Call the Asterisk demo
exten => 500,n,hangup           ; Return to the start over message.

5.        Check connection cables and make call to test.
Before making calls, please check the cable connection. It should be RJ48 connector. If you are not sure that, please visit the website to know how to make RJ48 connector (http://www.chebucto.ns.ca/Chebucto/Technical/Manuals/Max/
max6000/gs/cables.htm#17372). It everything is ok. Starting zaptel and asterisk, the LED will in green color. You also can check the ss7 in asterisk console and make sure it is there. If not loaded, please run: load chan_ss7.so to make it be loaded. After dialing 500, the system will forward to ss7. The results are shown in both of host-name-3 and host-name-4.

Reference:
http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+setup
http://www.chebucto.ns.ca/Chebuc ... gs/cables.htm#17372
http://www.sifira.com/chan-ss7/
http://lists.digium.com/pipermail/asterisk-ss7/
www.openvox.com.cn

Test environment:
1        Centos 5.0
2        Zaptel-1.4.7.1
3        Asterisk-1.4.15
4        Chan_ss7-1.0.0
5        Kernel 2.6
6        OpenVox D110P PRI card
Notes: if you have any problems, please report to asterisk-ss7 email list.  want to know more details, download the pdf file.

[ 本帖最后由 zhulizhong 于 2007-12-29 14:42 编辑 ]