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

运行chan_ss7-1.1时出错

14 years 6 months ago #3995 by michael040618
大家好:

在asterisk-1.4.21.2,zaptel1.4.10.1下面装载chan_ss7-1.1是出错:
zby*CLI> module load chan_ss7.so
[Oct 17 06:42:11] NOTICE[215]: config.c:625 load_config_link: Configured link 'l1' on linkset 'ls1', firstcic=1
[Oct 17 06:42:11] WARNING[215]: config.c:877 load_config_host: Missing interface entries for host 'STAMP-ASTFIN'.
[Oct 17 06:42:11] NOTICE[215]: config.c:1045 load_config: Configuring DPC 2722082 for linkset 'ls1'.
[Oct 17 06:42:11] WARNING[215]: transport.c:185 openchannel: Failure in DAHDI_SPECIFY for circuit 1: Device or resource busy.
[Oct 17 06:42:11] ERROR[215]: chan_ss7.c:842 ss7_load_module: Unable to initialize ISUP.

出现以上错误 ,不知道是哪里配置错误了!?多谢!!
14 years 6 months ago #3997 by james.zhu
麻烦你把完整的配置信息发上来。没有那些没有办法判断。
james.zhu

14 years 6 months ago #3998 by michael040618
多谢答复:
配置如下:
ss7.conf:
[linkset-ls1]

; 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 => ss7_call

; 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
variant => CHINA ; 支持中国ss7 号信令

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

[link-l1]

; This link belongs to linkset siuc
linkset => ls1

; 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-zby]
; 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
opc => 0x222222 ; 点码

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

dpc => ls1:0x298922 ; 点码


; 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:
oadzone = cn
defaultzone=cn
span=1,1,0,ccs,hdb3
bchan=1-15,17-31 # set this to 1-15,17-31 for E1
dchan=16 # set this to 16 for E
14 years 6 months ago #4000 by Wayne
麻烦把卡的序列号贴上来。
这样方便我们做支持。

一般来说,我们不方便对非OpenVox客户的系统提供支持。谢谢理解。
14 years 6 months ago #4001 by nobigdeal
把 [host-bzy] 改为 [host-STAMP-ASTFIN]

host 后面必须跟你机器的host name
14 years 6 months ago #4002 by james.zhu
请发你的ssh 帐号到:This email address is being protected from spambots. You need JavaScript enabled to view it.
regards!
James.zhu

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