[default]
exten => s,1,noop()
exten => _*[1-5]xx,1,Macro(stdexten,${EXTEN:1},SIP/${EXTEN:1})
exten => *900,1,VoicemailMain(${CALLERID(num)})
exten => _1831.,1,Goto(${EXTEN:4},1)
exten => _*0x,1,MeetMe(${EXTEN:1}|MsI)
[dialplan]
include => default
exten => _NXXXXXXX,1,Macro(trunkdial,${faktortel}/${EXTEN})
exten => _0[2356789]XXXXXXXX,1,Macro(trunkdial,${faktortel}/${EXTEN})
exten => _04XXXXXXXX,1,Macro(trunkdial,${faktortel}/${EXTEN})
exten => _1[38]00XXXXXX,1,Macro(trunkdial,${faktortel}/${EXTEN})
exten => _000,1,Macro(trunkdial,${faktortel}/${EXTEN})
exten => _12[23]X,1,Macro(trunkdial,${faktortel}/${EXTEN})
exten => _124XX,1,Macro(trunkdial,${faktortel}/${EXTEN})
exten => _13XXXX,1,Macro(trunkdial,${faktortel}/${EXTEN})
exten => _180XXXX,1,Macro(trunkdial,${faktortel}/${EXTEN})
exten => _001[145689]XXXXXX!,1,Macro(trunkdial,${faktortel}/${EXTEN})
<DEBUG FROM CALL>
Connected to Asterisk 1.4.14 currently running on asterisk (pid = 28571)
Verbosity is at least 8
Sending to 10.0.1.97 : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.0.1.97:2250
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.1.97:2250
Looking for 0405060165 in default (domain 10.0.0.145)
<--- Reliably Transmitting (no NAT) to 10.0.1.97:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.1.97;branch=z9hG4bKcd3fcea43B706C37;received=210.18.217.1
From: "Angie Cheung" <sip:102@10.0.0.145>;tag=78F5233B-29A2BE3A
To: <sip:0405060165@10.0.0.145;user=phone>;tag=as6e803872
Call-ID: 90871d66-86be28e1-b9552d18@10.0.1.97
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Nov 23 10:59:48] NOTICE[28585]: chan_sip.c:13753 handle_request_invite: Call from '' to extension '0405060165' rejected because extension not found.
Scheduling destruction of SIP dialog '90871d66-86be28e1-b9552d18@10.0.1.97' in 32000 ms (Method: INVITE)
Really destroying SIP dialog '90871d66-86be28e1-b9552d18@10.0.1.97' Method: ACK