hello: I install the Digium 2 port E1,asterisk-1.8, dahdi: Version: 2.5.0.2, chan_ss7-2.1 version. the confi files are: ============system.conf====================== # Autogenerated by /usr/sbin/dahdi_genconf on Mon Dec 5 20:12:54 2011 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" HDB3/CCS ClockSource span=1,1,1,ccs,hdb3 # termtype: te bchan=1-31 #dchan=16 # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" (MASTER) HDB3/CCS span=2,2,0,ccs,hdb3 # termtype: te bchan=32-62 #dchan=47 # Global data loadzone = cn defaultzone = cn =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 => 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 ; 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-l1] sltm => no ; 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 => remote,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 ======================= the system always comes out the message: -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20111205/4d0add46/attachment.htm>
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