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× Questions on Asterisk with SS7 Chinese variant. (有关Asterisk+中国七号信令的问题)

How to install asterisk, chan_ss7 and zaptel with openvox PRI cards

11 years 4 months ago #3752 by james.zhu
hello:
if you want to install asterisk, chan_ss7 and zaptel, please follow these steps:
1) download asterisk-1.4.20, zaptel-1.4.10 and chan_ss7_1.1
2) unzip asterisk-1.4.20.tar.gz to /usr/src, under asterisk dir, please run: ./configure, make and make install, make samples.
3) unzip zaptel-1.4.10.tar.gz to /usr/src/, under zaptel dir, please run: ./confiugre, make and make install
4) unzip the chan_ss7_1.1 to /usr/src, under chan_ss7_1.1, please do this:
4.1) modify the Makefile, do like this:
=======================
# non-standard places.
INCLUDE+=-I /usr/src/zaptel-1.4.10/kernel ; point to the zaptel source
#INCLUDE+=-I../source/telephony/dahdi/include
INCLUDE+=-I /usr/src/asterisk-1.4.20 ; point to asterisk source
=======================
4.2) save and quit
4.3) run make and make install
4.4) copy the ss7.conf file to /etc/asterisk
4.5) copy the chan_ss7.so to /usr/lib/asterisk/modules
5) edit the zaptel.conf like this:
===========================
# 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: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER)
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-31
# dchan=16

# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2"
span=2,2,0,ccs,hdb3,crc4
# termtype: te
bchan=32-62
#dchan=47

# Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3"
span=3,3,0,ccs,hdb3,crc4
# termtype: te
bchan=63-93
#dchan=78

# Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4"
span=4,4,0,ccs,hdb3,crc4
# termtype: te
bchan=94-124
#dchan=109

# Global data

loadzone = us
defaultzone = us
===========================
6) edit ss7.conf like this:
===========================
[linkset-ls1]
enabled => yes
enable_st => yes
use_connect => no
hunting_policy => even_mru
context => ss7_call
language => en
subservice => auto
variant => CHINA ; 支持中国ss7 号信令
[link-l1]
linkset => ls1
channels => 1-15,17-31
schannel => 16
firstcic => 1
enabled => yes

echocancel => no
echocan_train => 350
echocan_taps => 128

[link-l2]
linkset => ls1
channels => 1-31
schannel =>
firstcic => 33
enabled => yes

[link-l3]
linkset => ls1
channels => 1-31
schannel =>
firstcic => 65
enabled => yes

[link-l4]
linkset => ls1
channels => 1-31
schannel =>
firstcic => 97
enabled => yes

[host-openvox]
enabled => yes
opc => 0x222222 ; get it from your provider
dpc => ls1:0x298922 ; get it from your provider
links => l1:1,l2:2,l3:3,l4:4
===========================
7) edit extensions.conf:
==========================
[ss7_call]
exten => 100,1,Dial(ss7/outgoing number)
exten => 100,2,Hangup
==========================
8) make sure ss7 is up:
==========================
CLI> ss7 link status
linkset ls1, link l1, schannel 16, sls 0, INSERVICE, rx: 5, tx: 1/3, sentseq/lastack: 4/4, total 199328, 199424
CLI> ss7 status
linkset idle busy initiating resetting total incoming total outgoing
ls1 30 0 0 0 0 0
gw1*CLI> ss7 linestat
Linkset: ls1
CIC 1 Idle
CIC 2 Idle
CIC 3 Idle
CIC 4 Idle
CIC 5 Idle
CIC 6 Idle
CIC 7 Idle
CIC 8 Idle
CIC 9 Idle
CIC 10 Idle
CIC 11 Idle
CIC 12 Idle
CIC 13 Idle
CIC 14 Idle
CIC 15 Idle
CIC 17 Idle
CIC 18 Idle
CIC 19 Idle
CIC 20 Idle
CIC 21 Idle
9) use an extension dial 100 to make a call to ss7
Test tools:
asterisk-1.4.20
zaptel-1.4.10
chan_ss7-1.1(支持中国ss7 号信令)
regards!
James.zhu

10 years 10 months ago #4881 by Joe.Yung
如下是来自上海客户关于联通7号信令的一个成功案例供大家分享。如有不对,请大家批评指正!(所有信息均来自客户)
1. 环境:
OS: Centos 5.4
Asterisk Version: asterisk-1.4.21.1
Zaptel Version: zaptel-1-4-11
Chan_ss7 Version: chan_ss7-1.1
上海联通7号信令
OpenVox D410P
普通服务器

2. 安装配置均按照James.zhu.(How to install asterisk, chan_ss7 and zaptel with openvox PRI cards)的标准安装。(所有编译安装步骤均无报错)

3.准备工作:
A:BNC 转接线制作:RJ-48:1245做法.
B:BNC 转接线接法:短线-Rx-12, 长线-Tx-45

4. 几个关键配置文件:
A:/etc/zaptel.conf(这是成功案例的)
===========================================================
# Autogenerated by /usr/sbin/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: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 RECOVERINGClockSource
span=1,1,0,ccs,hdb3 //国内客户CRC4校验一般用不到,建议remove掉
# termtype: te
#bchan=1-15,17-31 //这是默认的PRI信令自动生成的
bchan=1-31 //这是专门为Chan_SS7而设置的
#dchan=16 //如果用Chan_ss7记得将这行注释掉

# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" HDB3/CCS/CRC4 RECOVERING
span=2,2,0,ccs,hdb3
# termtype: te
#bchan=32-46,48-62
bchan=32-62
#dchan=47

# Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3" HDB3/CCS/CRC4 RED
span=3,3,0,ccs,hdb3
# termtype: te
#bchan=63-77,79-93
bchan=63-93
#dchan=78

# Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" HDB3/CCS/CRC4 RED
span=4,4,0,ccs,hdb3
# termtype: te
#bchan=94-108,110-124
bchan=94-124
#dchan=109

# Global data

loadzone = us
defaultzone = us
================================================================

B:/etc/asterisk/ss7.conf(这是成功案例的,只用到link l1和link l2,如果要用link l3,link l4还要做相应的修改)
[linkset-|s1]

; The linkset is enabled
enabled => yes

; The end-of-pulsing (ST) is not used to determine when incoming address is complete
enable_st => yes

; Reply incoming call with CON rather than ACM and ANM
use_connect => yes

; The CIC hunting policy (even_mru, odd_lru, seq_lth, seq_htl) is even CIC numbers, most recently used
hunting_policy => even_mru

; Incoming calls are placed in the ss7 context in the asterisk dialplan
context => ss7

; The language for this context is da
language => en

; The value and action for t35. Value is in msec, action is either st or timeout
; If you use overlapped dialling dial plan, you might choose: t35 => 4000,st
;t35 => 15000,timeout

; The subservice field: national (8), international (0), auto or decimal/hex value
; The auto means that the subservice is obtained from first received SLTM
subservice => auto

; The host running the mtp3 service
; mtp3server => localhost
variant => CHINA

[link-|1]
linkset => |s1
channels => 1-15,17-31
;channels => 1-31
schannel => 16
firstcic => 32 //上海联通7号信令firstcic值由32开始.
enabled => yes

echocancel => no
echocan_train => 350
echocan_taps => 128

[link-|2]
linkset => |s1
channels => 1-31
schannel =>
firstcic => 64
enabled => yes

;[link-l3]
;linkset => ls1
;channels => 1-15,17-31
;schannel => 16
;firstcic => 65
;enabled => yes

;[link-l4]
;linkset => ls1
;channels => 1-15,17-31
;schannel => 16
;firstcic => 97
;enabled => yes

[host-xianxian] //该值为本服务器的主机名,确认正确
enabled => yes
opc => 0xFFFF33 //局端(不同运营商值可能不同)
dpc => |s1:0xFF16E3 //用户端(不同运营商值可能不同)
links => |1:1,|2:2 //对应上面定义的link l1, link l2.


;[jitter]
;
JITTER BUFFER CONFIGURATION
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.

; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".

; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.

; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.

; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.

; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;

5.该客户最后因为BNC转接头上的两根线接反了,后来调换位置之后,则成功配置 .请记住:长线--Tx-45pin, 短线-Rx-12pin

10 years 10 months ago #4882 by james.zhu
:victory:

10 years 9 months ago #5157 by xcoolio
Asterisk1.4+Dahdi该怎么配呢...?
10 years 9 months ago #5158 by Joe.Yung
Hi,
zaptel下的 /etc/zaptel.conf 对应 dahdi的 /etc/dahdi/system.conf
这里有一个类似(dahdi)的链接,你可以参考一下:
http://bbs.voip88.com/redirect.php?tid=10318&goto=lastpost
但是你必须注意:加载openvox板卡对应的驱动.

10 years 8 months ago #5282 by xin.liu
如下是来自北京客户关于SS7号信令的一个成功案例供大家分享
1. 环境:
OS: Centos 5.4
Asterisk Version: asterisk-1.4.22
Zaptel Version: zaptel-1.4.12.1
Chan_ss7 Version: chan_ss7-1.1
Openvox D110P
2. 所有编译安装步骤均无报错
3.几个关键配置文件:
A:/etc/zaptel.conf(这是成功案例的)

span=1,1,0,ccs,hdb3
bchan=15
bchan=16
bchan-17-31
loadzone=cn
defaultzone=cn

B:/etc/asterisk/ss7.conf
[linkset-siuc]

; The linkset is enabled
enabled => yes

; The end-of-pulsing (ST) is not used to determine when incoming address is complete
;;;;;;enable_st => no
enable_st => yes

; Reply incoming call with CON rather than ACM and ANM
;;;;;use_connect => yes
use_connect => no

; The CIC hunting policy (even_mru, odd_lru, seq_lth, seq_htl) is even CIC numbers, most recently used
hunting_policy => even_mru

; Incoming calls are placed in the ss7 context in the asterisk dialplan
context => ss7_call

; The language for this context is da
language => da

; The value and action for t35. Value is in msec, action is either st or timeout
; If you use overlapped dialling dial plan, you might choose: t35 => 4000,st
t35 => 15000,timeout

; The subservice field: national (8), international (0), auto or decimal/hex value
; The auto means that the subservice is obtained from first received SLTM
subservice => auto

; The host running the mtp3 service
; mtp3server => localhost
variant => CHINA

[link-l1]

; This link belongs to linkset siuc
linkset => siuc


[linkset-siuc]

; The linkset is enabled
enabled => yes

; The end-of-pulsing (ST) is not used to determine when incoming address is complete
;;;;;;enable_st => no
enable_st => yes

; Reply incoming call with CON rather than ACM and ANM
;;;;;use_connect => yes
use_connect => no

; The CIC hunting policy (even_mru, odd_lru, seq_lth, seq_htl) is even CIC numbers, most recently used
hunting_policy => even_mru

; Incoming calls are placed in the ss7 context in the asterisk dialplan
context => ss7_call

; The language for this context is da
language => da

; The value and action for t35. Value is in msec, action is either st or timeout
; If you use overlapped dialling dial plan, you might choose: t35 => 4000,st
t35 => 15000,timeout

; The subservice field: national (8), international (0), auto or decimal/hex value
; The auto means that the subservice is obtained from first received SLTM
subservice => auto

; The host running the mtp3 service
; mtp3server => localhost
variant => CHINA

[link-l1]

; This link belongs to linkset siuc
linkset => siuc

; The speech/audio circuit channels on this link
channels => 1-15,17-31

; The signalling channel
schannel => 16
; To use the remote mtp3 service, use 'schannel => remote,16'

; The first CIC
firstcic => 1

; The link is enabled
enabled => yes

; Echo cancellation
; echocancel can be one of: no, 31speech (enable only when transmission medium is 3.1Khz speech), allways
echocancel =>no
; echocan_train specifies training period, between 10 to 100 msec
echocan_train => 350
; echocan_taps specifies number of taps, 32, 64, 128 or 256
echocan_taps => 128


[host-DIANXIN]
; chan_ss7 auto-configures by matching the machines host name with the host-<name>
; section in the configuration file, in this case 'gentoo1'. The same
; configuration file can thus be used on several hosts.

; The host is enabled
enabled => yes

; The point code for this SS7 signalling point is 0x8e0
opc => 0xXXXXXXX

; The destination point (peer) code is 0x3fff for linkset siuc
dpc => siuc:0xXXXXXXXX

; Syntax: links => link-name:digium-connector-no
; The links on the host is 'l1', connected to span/connector #1
links => l1:1

; The SCCP global title: translation-type, nature-of-address, numbering-plan, address
;globaltitle => 0x00, 0x04, 0x01, 4546931411
;ssn => 7
;route => 919820405471:ra_geb, 919820367598:ra_geb, 919820706441:ra_geb, :ra_geb




[jitter]
;
JITTER BUFFER CONFIGURATION
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.

; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".

; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.

; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.

; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.

; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;

111,1 Bot
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.

; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".

; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.

; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.

; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.

; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;

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