Help in solve errors and help with .conf

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 [email protected] 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?

Yea. But i have to build my gui for my customers. They should have creat a private exchange and use it but without freepbx… We have our voip solution and we want to great gui for customers for use it like queue, ring group, announcments etc.
If i install asterisk without freepbx and edit conf it will works?

You don’t want to use FreePBX but you want to use the FreePBX dialplans.
This is essentially where you have gone wrong.
You should chose one or the other.
The FreePBX dial plans are very involved and rely on things you may not be supplying. Unless you understand the dialplans 100% to know how to make them do what you want it is best to not use them. If you wish to not use FreePBX the best route is to build a base Linux box and add only the components you need. Then build your own dial plans so there is no mystery.

Realize FreePBX and the resulting dial plans are the result of years of work.

It is not a set of base configurations it is a intricate framework.

All the features in FreePBX can be done without FreePBX sure. With that said it is not going to be done with 5 minutes of pho development in your garage.

If you wish to seriously use Asterisk as a platform I would recommend googling “asterisk the future of telephony” you can download it free and read it a few times