[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?
Default FreePBX context,
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = ext-did-post-custom
include = from-did-direct; MODIFICATOIN (PL) for findmefollow if
enabled, should be bofore ext-local
include = ext-did-catchall; THIS MUST COME AFTER ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)
The call seems to be going here
[ext-did-catchall]
include = ext-did-catchall-custom
exten = s,1,Noop(No DID or CID Match)
exten = s,n(a2),Answer
exten = s,n,Wait(2)
exten = s,n,Playback(ss-noservice)
exten = s,n,SayAlpha(${FROM_DID})
exten = s,n,Hangup
exten = _.,1,Set(__FROM_DID=${EXTEN})
exten = _.,n,Noop(Received an unknown call with DID set to ${EXTEN})
exten = _.,n,Goto(s,a2)
exten = h,1,Hangup
; end of [ext-did-catchall]
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[asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?
Hi all, I have a connect between a siemens hipath Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting The number you have dialed is not in service In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk) then the number 1905 (Freefone number in Ireland) Please help I cant figure this one out. Thanks, Joe CLI - [Feb 11 17:45:25] VERBOSE[4526] logger.c: -- Accepting overlap call from '0339' to 'unspecified' on channel 0/31, span 1 [Feb 11 17:45:25] VERBOSE[5764] logger.c: -- Starting simple switch on 'Zap/31-1' [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:1] Set(Zap/31-1, __FROM_DID=91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:2] NoOp(Zap/31-1, Received an unknown call with DID set to 91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Goto (from-pstn,s,2) [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:2] Answer(Zap/31-1, ) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:3] Wait(Zap/31-1, 2) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:4] Playback(Zap/31-1, ss-noservice) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'ss-noservice' (language 'en') [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:5] SayAlpha(Zap/31-1, 91905) in new stack [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'digits/9' (language 'en') [Feb 11 17:45:39] VERBOSE[4526] logger.c: -- Channel 0/31, span 1 got hangup request, cause 16 [Feb 11 17:45:39] WARNING[5764] file.c: Failed to write frame [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, s, 5) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:1] Hangup(Zap/31-1, ) in new stack [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on Zap/31-1 [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on Zap/31-1 [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Hungup 'Zap/31-1' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
