[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

; 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 => seq_lth

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

; 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 => 8

noa => 0x1

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

[link-l1]
;sls => 0
linkset => siuc				; This link belongs to linkset siuc
channels => 1-15,17-31		; The speech/audio circuit channels on this link
schannel => 16				; The signalling channel, To use the remote mtp3 service, use 'schannel => remote,16'
firstcic => 1				; The first CIC
enabled => yes				; The link is enabled
; ptype => tup
; b_nature => 0
; trans_ind => 0
;echocancel => no			; Echo cancellation, echocancel can be one of: no, 31speech (enable only when transmission medium is 3.1Khz speech), allways
;echocan_train => 350		; echocan_train specifies training period, between 10 to 100 msec
;echocan_taps => 128			; echocan_taps specifies number of taps, 32, 64, 128 or 256
sltm => yes
;stp => 0x850
;stp => 0x0x7ef

[link-l2]
linkset => siuc
channels => 1-31
schannel => 
firstcic => 33
enabled => yes
; ptype => tup
; b_nature => 0
; trans_ind => 0
;echocancel => no
;echocan_train => 350
;echocan_taps => 128

[link-l3]
linkset => siuc
channels => 1-31
schannel => 
firstcic => 65
enabled => yes
; ptype => tup
; b_nature => 0
; trans_ind => 0
;echocancel => no
;echocan_train => 350
;echocan_taps => 128

[link-l4]
linkset => siuc
channels => 1-31
schannel => 
firstcic => 97
enabled => yes
; ptype => tup
; b_nature => 0
; trans_ind => 0
;echocancel => no
;echocan_train => 350
;echocan_taps => 128

[link-l5]
linkset => siuc
channels => 1-31
schannel => 
firstcic => 129
enabled => yes
; ptype => tup
; b_nature => 0
; trans_ind => 0
;echocancel => no
;echocan_train => 350
;echocan_taps => 128

[link-l6]
linkset => siuc				; This link belongs to linkset siuc
channels => 1-15,17-31		; The speech/audio circuit channels on this link
schannel => 16				; The signalling channel, To use the remote mtp3 service, use 'schannel => remote,16'
firstcic => 161
enabled => yes
; ptype => tup
; b_nature => 0
; trans_ind => 0
;echocancel => no
;echocan_train => 350
;echocan_taps => 128

[link-l7]
linkset => siuc
channels => 1-31
schannel => 
firstcic => 193
enabled => yes
; ptype => tup
; b_nature => 0
; trans_ind => 0
;echocancel => no
;echocan_train => 350
;echocan_taps => 128

[link-l8]
linkset => siuc
channels => 1-31
schannel => 
firstcic => 225
enabled => yes
; ptype => tup
; b_nature => 0
; trans_ind => 0
;echocancel => no
;echocan_train => 350
;echocan_taps => 128

[host-MediaSvr1]
; 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
default_linkset =>siuc 
; The point code for this SS7 signalling point is 0x8e0
; opc => 0x8e0

; The destination point (peer) code is 0x3fff for linkset siuc
; dpc => siuc1:0x3fff
opc => 0x1467
dpc => siuc:0x7ef

; Syntax: links => link-name:digium-connector-no
; The links on the host is 'l1', connected to span/connector #1
links => l1:1,l2:2,l3:3,l4:4,l5:5,l6:6,l7:7,l8:8

; 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".
;-----------------------------------------------------------------------------------