I’m starting with asterisk and freepbx.
I edit my .conf files for make connections with my telephone exchange without freepbx. I don’t want to use freepbx. I want creat my fast Gui because i want to use it for own needs. So i used asterisk confs (extensions_additional.conf, sip_additional.conf and sip_registration.conf), for creat my trunk connection and it works good. My telephone exchange show register good in status of trunk. Later i did Extensions in sip_additional.conf and then when i have all i edit extensions_additional.conf for use incoming call conf:
[from-trunk-sip-MyTrunk] include => from-trunk-sip-MyTrunk-custom exten => _.,1,Set(GROUP()=OUT_2) exten => _.,n,Goto(from-trunk,${EXTEN},1);--== end of [from-trunk-sip-MyTrunk] ==--;
and then when i did it i have 2 Errors in my asterisk log and i my call can’t be connect with my telephone.
Errors:
WARNING[3848][C-00000001]: pbx.c:6643 __ast_pbx_run: Channel ‘SIP/TestTrunk-00000001’ sent to invalid extension but no invalid handler: context,exten,priority=from-did-direct,1234,1
and
WARNING[3632][C-00000000]: func_callerid.c:905 callerpres_read: CALLERPRES is deprecated. Use CALLERID(name-pres) or CALLERID(num-pres) instead.
For create my confs to creat trunks and extensions i did:
In extensions_additional.conf
OUT_2 = SIP/TestTrunk OUTCID_2 = 48616107xxx OUTMAXCHANS_2 = OUTFAIL_2 = OUTPREFIX_2 = OUTDISABLE_2 = off OUTKEEPCID_2 = off FORCEDOUTCID_2 = PREFIX_TRUNK_2 =
then
[ext-did-0002] include => ext-did-0002-custom exten => 48616107xxx,1,Set(__FROM_DID=${EXTEN}) exten => 48616107xxx,n,Gosub(app-blacklist-check,s,1()) exten => 48616107xxx,n,Set(CDR(did)=${FROM_DID}) exten => 48616107xxx,n,ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) exten => 48616107xxx,n,Set(__CALLINGPRES_SV=${CALLERPRES()}) exten => 48616107544,n,Set(CALLERPRES()=allowed_not_screened) exten => 48616107xxx,n,Goto(from-did-direct,1234,1)
then
in [ext-trunk] include => ext-trunk-custom exten => 2,1,Set(TDIAL_STRING=SIP/TestTrunk) exten => 2,n,Set(DIAL_TRUNK=2) exten => 2,n,Goto(ext-trunk,tdial,1)
next
[from-trunk-sip-TestTrunk] include => from-trunk-sip-TestTrunk-custom exten => _.,1,Set(GROUP()=OUT_2) exten => _.,n,Goto(from-trunk,${EXTEN},1)
next
[outrt-1] ; TestowyRouteOut include => outrt-1-custom exten => _ZXXXXXXXX,1,Macro(user-callerid,LIMIT,) exten => _ZXXXXXXXX,n,Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})}) exten => _ZXXXXXXXX,n,Set(_NODEST=) exten => _ZXXXXXXXX,n,Gosub(sub-record-check,s,1(out,${EXTEN},)) exten => _ZXXXXXXXX,n,Macro(dialout-trunk,2,${EXTEN},) exten => _ZXXXXXXXX,n,Macro(outisbusy,)
In sip_additional.conf
[1234] deny=0.0.0.0/0.0.0.0 secret=mama22 dtmfmode=rfc2833 canreinvite=no context= host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=udp encryption=no callgroup= pickupgroup= dial=SIP/1234 mailbox=1234@device permit=0.0.0.0/0.0.0.0 callerid=Testowy <1234> callcounter=yes faxdetect=no cc_monitor_policy=generic[TestTrunk] username=48616107xxx type=peer secret=mama22 2833compensate=yes relaxdtmf=yes insecure=port,invite host=82.177.59.132 fromuser=48616107xxx dtmfmode=inband defaultuser=48616107xxx allow=ulaw allow=gsm context=from-trunk-sip-TestTrunk
In sip_registration.conf
register=48616107xxx:[email protected]/48616107xxx
Can You tell me what i did wrong? and why when i call to 616107xxx this errors are show and my Extension “1234” no ring?