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 (
, 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 (
, 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".
;