Hi there.
This its like a real nightmare.
4 hours ago this should be working and its impossible.
Outbound and internal call works perfect.
But anytime someone try use our telephone and call us, freepbx send a “message to the caller” telling them that the number he has call doesnt exist. I saw on log of it plays the wav files of the numbers so, the call get in.
This is what i get:
[2013-12-12 19:00:15] VERBOSE[741][C-00000025] netsock2.c: == Using SIP RTP TOS bits 184
[2013-12-12 19:00:15] VERBOSE[741][C-00000025] netsock2.c: == Using SIP RTP CoS mark 5
[2013-12-12 19:00:15] VERBOSE[23030][C-00000025] pbx.c: -- Executing [00493027890330@from-trunk:1] Set("SIP/VoIP_DE-00000013", "__FROM_DID=00493027890330") in new stack
[2013-12-12 19:00:15] VERBOSE[23030][C-00000025] pbx.c: -- Executing [00493027890330@from-trunk:2] NoOp("SIP/VoIP_DE-00000013", "Received an unknown call with DID set to 00493027890330") in new stack
[2013-12-12 19:00:15] VERBOSE[23030][C-00000025] pbx.c: -- Executing [00493027890330@from-trunk:3] Goto("SIP/VoIP_DE-00000013", "s,a2") in new stack
[2013-12-12 19:00:15] VERBOSE[23030][C-00000025] pbx.c: -- Goto (from-trunk,s,2)
[2013-12-12 19:00:15] VERBOSE[23030][C-00000025] pbx.c: -- Executing [s@from-trunk:2] Answer("SIP/VoIP_DE-00000013", "") in new stack
[2013-12-12 19:00:16] VERBOSE[23030][C-00000025] pbx.c: -- Executing [s@from-trunk:3] Wait("SIP/VoIP_DE-00000013", "2") in new stack
[2013-12-12 19:00:18] VERBOSE[23030][C-00000025] pbx.c: -- Executing [s@from-trunk:4] Playback("SIP/VoIP_DE-00000013", "ss-noservice") in new stack
[2013-12-12 19:00:18] VERBOSE[23030][C-00000025] file.c: -- Playing 'ss-noservice.gsm' (language 'en')
[2013-12-12 19:00:21] VERBOSE[23030][C-00000025] pbx.c: == Spawn extension (from-trunk, s, 4) exited non-zero on 'SIP/VoIP_DE-00000013'
[2013-12-12 19:00:21] VERBOSE[23030][C-00000025] pbx.c: -- Executing [h@from-trunk:1] Macro("SIP/VoIP_DE-00000013", "hangupcall,") in new stack
[2013-12-12 19:00:21] VERBOSE[23030][C-00000025] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/VoIP_DE-00000013", "1?theend") in new stack
[2013-12-12 19:00:21] VERBOSE[23030][C-00000025] pbx.c: -- Goto (macro-hangupcall,s,3)
[2013-12-12 19:00:21] VERBOSE[23030][C-00000025] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf("SIP/VoIP_DE-00000013", "0?Set(CDR(recordingfile)=)") in new stack
[2013-12-12 19:00:21] VERBOSE[23030][C-00000025] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup("SIP/VoIP_DE-00000013", "") in new stack
[2013-12-12 19:00:21] VERBOSE[23030][C-00000025] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/VoIP_DE-00000013' in macro 'hangupcall'
[2013-12-12 19:00:21] VERBOSE[23030][C-00000025] pbx.c: == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/VoIP_DE-00000013'
trunk data
The sip account:
Trunk Name :VoIP_DE
OUTBOUNDCALLERID: TELEFON
PEER Details
username=SIPGATEUSERNAME
type=peer
secret=SIPGATEPASSWORD
qualify=yes
nat=yes
insecure=invite
host=sipconnect.sipgate.de
fromuser=SIPGATEUSERNAME
dtmfmode=rfc2833
context=from-trunk
authuser=SIPGATEUSERNAME
allow=gsm,ulaw,alaw
USER context: SIPGATEUSERNAME
USER details
context=from-trunk
fromuser=SIPGATEUSERNAME
secret=SIPGATEPASSWORD
type=user
Register
SIPGATEUSERNAME:[email protected]/TELEFON
inbound route data
name: office
DID number: TELEFON
CallerID: TELEFON
Asterisk 11
Debian 7
FreePBX 2.11
Thanks folks, im really desperate.