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Questions on Asterisk with SS7 Chinese variant. (有关Asterisk+中国七号信令的问题)
ss7安装问题
15 years 5 months ago #2128
by zy7884
ss7安装后，asterisk不能启动，日志如下，请问是什么问题
[root@data modules]# asterisk -vvvvvvvvvvcg
Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <This email address is being protected from spambots. You need JavaScript enabled to view it.>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
[2008-11-05 16:36] ERROR[5341]: asterisk.c:2982 main: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. You have options:
1. You only have to compile Zaptel support into Asterisk if you need it. One option is to recompile without Zaptel support.
2. You only have to load Zaptel drivers if you want to take advantage of Zaptel services. One option is to unload zaptel modules if you don't need them.
3. If you need Zaptel services, you must correctly configure Zaptel.
ss7.conf 文件内容
[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 => yes
; Reply incoming call with CON rather than ACM and ANM
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 => incoming_1
; 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 (
, 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
;addr_type => ansi
;a_cat => 0x18
;trans_ind => 0x00
[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-gentoo1]
; 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 => 0x8e0
; The destination point (peer) code is 0x3fff for linkset siuc
dpc => siuc:0x3fff
; 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
;[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".
;
zaptel.conf内容
# 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: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0"
span=1,1,0,ccs,hdb3
# termtype: te
bchan=1-31
loadzone = cn
defaultzone = cn
[root@data modules]# asterisk -vvvvvvvvvvcg
Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <This email address is being protected from spambots. You need JavaScript enabled to view it.>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
[2008-11-05 16:36] ERROR[5341]: asterisk.c:2982 main: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. You have options:
1. You only have to compile Zaptel support into Asterisk if you need it. One option is to recompile without Zaptel support.
2. You only have to load Zaptel drivers if you want to take advantage of Zaptel services. One option is to unload zaptel modules if you don't need them.
3. If you need Zaptel services, you must correctly configure Zaptel.
ss7.conf 文件内容
[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 => yes
; Reply incoming call with CON rather than ACM and ANM
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 => incoming_1
; 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 (
, 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
;addr_type => ansi
;a_cat => 0x18
;trans_ind => 0x00
[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-gentoo1]
; 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 => 0x8e0
; The destination point (peer) code is 0x3fff for linkset siuc
dpc => siuc:0x3fff
; 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
;[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".
;
zaptel.conf内容
# 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: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0"
span=1,1,0,ccs,hdb3
# termtype: te
bchan=1-31
loadzone = cn
defaultzone = cn
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