FreePBX und Asterisk keine Abgehenden Gespräch

Monotron

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Also ich bin hier schon ewig am herumdoktorn. Aber ich kann nicht raus telefonieren.

Das Telefon gibt immer ein "Fehler 404" wieder.
Ich hab keine Ahnung. Würde mich freuen wenn mir da einer helfen könnte.
Wie gesagt, es geht um FreePBX und Asterisk.

Intern und ankommend ist alles sauber!
Habe einmal Arcor für abgehend und ankommmend und nochmal Sipgate für ankommende Calls.

Hier wie ich die Arcor Trunk eingerichtet habe:

Code:
[B]General Settings[/B]
Outbound Caller ID: 
Maximum Channels:

[B]Outgoing Dial Rules[/B]
Dial Rules:
Outbound Dial Prefix:

[B]Outgoing Settings[/B]
Trunk Name: arcor
Peer Details: canreinvite=no
                  fromuser=02115*****
                  host=0211.sip.arcor.de
                  nat=no
                  secret=********
                  username=02115******

[B]Incoming Settings[/B]
User Context: 02115******
User Details: allow=ulaw
                  context=ankommend
                  diallow=all
                  fromdomain=0211.sip.arcor.de
                  host=0211.sip.arcor.de

[B]Registration[/B]
02115*****:******@0211.sip.arcor.de
Und hier dann wie ich die Outbound Route eingerichtet habe:

Code:
RouteName: ArcorOut
Route Password:
Emergency Callking:
Dial Patterns: 0.
Trunk Sequence: 0 SIP/arcor
Hier mal das Ergebnis einen Abgehenden Calls im Sip Debug Modus:

Code:
Peer audio RTP is at port 192.168.1.2:49406
Found description format GSM
Found description format iLBC
Found description format iLBC
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 02119****** in from-internal (domain 192.168.1.5)
list_route: hop: <sip:[email protected]:5060>
Transmitting (no NAT) to 192.168.1.2:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a80102000008d3454077a10000712400002c33;received=192.168.1.2
From: "unknown"<sip:[email protected]>;tag=9498910925858
To: <sip:02119******@192.168.1.5>
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02119******@192.168.1.5>
Content-Length: 0


---
    -- Executing Macro("SIP/100-081d0570", "dialout-trunk|2|02119******||") in new stack
    -- Executing Set("SIP/100-081d0570", "DIAL_TRUNK=2") in new stack
    -- Executing Set("SIP/100-081d0570", "DIAL_NUMBER=02119******") in new stack
    -- Executing Set("SIP/100-081d0570", "ROUTE_PASSWD=") in new stack
    -- Executing GotoIf("SIP/100-081d0570", "1?noauth") in new stack
    -- Goto (macro-dialout-trunk,s,6)
    -- Executing Set("SIP/100-081d0570", "GROUP()=OUT_2") in new stack
    -- Executing Macro("SIP/100-081d0570", "user-callerid") in new stack
    -- Executing GotoIf("SIP/100-081d0570", "0?report") in new stack
    -- Executing GotoIf("SIP/100-081d0570", "0?start") in new stack
    -- Executing Set("SIP/100-081d0570", "REALCALLERIDNUM=100") in new stack
    -- Executing NoOp("SIP/100-081d0570", "REALCALLERIDNUM is 100") in new stack
    -- Executing Set("SIP/100-081d0570", "AMPUSER=100") in new stack
    -- Executing Set("SIP/100-081d0570", "AMPUSERCIDNAME=Cyraxx") in new stack
    -- Executing GotoIf("SIP/100-081d0570", "0?report") in new stack
    -- Executing Set("SIP/100-081d0570", "CALLERID(all)=Cyraxx <100>") in new stack
    -- Executing NoOp("SIP/100-081d0570", "Using CallerID "Cyraxx" <100>") in new stack
    -- Executing Macro("SIP/100-081d0570", "record-enable|100|OUT") in new stack
    -- Executing GotoIf("SIP/100-081d0570", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/100-081d0570", "recordingcheck|20061026-085337|1161845617.395") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20061026-085337|1161845617.395: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/100-081d0570", "No recording needed") in new stack
    -- Executing Macro("SIP/100-081d0570", "outbound-callerid|2") in new stack
    -- Executing GotoIf("SIP/100-081d0570", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing NoOp("SIP/100-081d0570", "REALCALLERIDNUM is 100") in new stack
    -- Executing Set("SIP/100-081d0570", "USEROUTCID=") in new stack
    -- Executing Set("SIP/100-081d0570", "EMERGENCYCID=") in new stack
    -- Executing Set("SIP/100-081d0570", "TRUNKOUTCID=") in new stack
    -- Executing GotoIf("SIP/100-081d0570", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing GotoIf("SIP/100-081d0570", "1?usercid") in new stack
    -- Goto (macro-outbound-callerid,s,13)
    -- Executing GotoIf("SIP/100-081d0570", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,17)
    -- Executing NoOp("SIP/100-081d0570", "CallerID set to "Cyraxx" <100>") in new stack
    -- Executing GotoIf("SIP/100-081d0570", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,12)
    -- Executing AGI("SIP/100-081d0570", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
  fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing Set("SIP/100-081d0570", "OUTNUM=02119843883") in new stack
    -- Executing Set("SIP/100-081d0570", "custom=SIP/arcor") in new stack
    -- Executing GotoIf("SIP/100-081d0570", "0?customtrunk") in new stack
    -- Executing Dial("SIP/100-081d0570", "SIP/arcor/02119*******|120|r") in new stack
Destroying call '[email protected]'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Goto("SIP/100-081d0570", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing NoOp("SIP/100-081d0570", "Dial failed due to CHANUNAVAIL") in new stack
    -- Executing Macro("SIP/100-081d0570", "outisbusy|") in new stack
    -- Executing Playback("SIP/100-081d0570", "all-circuits-busy-now|noanswer") in new stack
We're at 192.168.1.5 port 13182
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 192.168.1.2:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a80102000008d3454077a10000712400002c33;received=192.168.1.2
From: "unknown"<sip:[email protected]>;tag=9498910925858
To: <sip:02119*******@192.168.1.5>;tag=as2183240d
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02119*******@192.168.1.5>
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 5941 5941 IN IP4 192.168.1.5
s=session
c=IN IP4 192.168.1.5
t=0 0
m=audio 13182 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Playing 'all-circuits-busy-now' (language 'en')
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 212.144.24.22:5060:
REGISTER sip:0211.sip.arcor.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK02f8b5fd;rport
From: <sip:02115******@0211.sip.arcor.de>;tag=as6dbbcb90
To: <sip:02115******@0211.sip.arcor.de>
Call-ID: [email protected]
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="021156*****", realm="arcor.de", algorithm=MD5, uri="sip:0211.sip.arcor.de", nonce="45407704f08854d6ad33f538b55183b9a0e8b54c", response="bdf4b1d34b3d00ea496166f2abbe186a", opaque=""
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from 212.144.24.22:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;received=88.76.211.55;branch=z9hG4bK02f8b5fd;rport=5060
From: <sip:02115*******@0211.sip.arcor.de>;tag=as6dbbcb90
To: <sip:02115*******@0211.sip.arcor.de>
Call-ID: [email protected]
CSeq: 107 REGISTER
Contact: <sip:[email protected]>;expires=60


--- (7 headers 0 lines) ---
Scheduling destruction of call '[email protected]' in 32000 ms
    -- Executing Playback("SIP/100-081d0570", "pls-try-call-later|noanswer") in new stack
    -- Playing 'pls-try-call-later' (language 'en')
asterisk*CLI>
<-- SIP read from 192.168.1.2:5060:
OPTIONS sip:192.168.1.5:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a8010200000010454077a5000026e200002c36
Content-Length: 0
Call-ID: [email protected]
CSeq: 3303 OPTIONS
From: <sip:[email protected]>;tag=9499321827002
Max-Forwards: 70
To: <sip:192.168.1.5:5060>


--- (8 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.1.5)
Transmitting (no NAT) to 192.168.1.2:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a8010200000010454077a5000026e200002c36;received=192.168.1.2
From: <sip:[email protected]>;tag=9499321827002
To: <sip:192.168.1.5:5060>;tag=as22058826
Call-ID: [email protected]
CSeq: 3303 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.1.5>
Accept: application/sdp
Content-Length: 0


---
Destroying call '[email protected]'
    -- Executing Macro("SIP/100-081d0570", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/100-081d0570", "w") in new stack
    -- Executing NoCDR("SIP/100-081d0570", "") in new stack
    -- Executing Wait("SIP/100-081d0570", "5") in new stack
asterisk*CLI>
<-- SIP read from 192.168.1.3:5060:


--- (0 headers 0 lines) Nat keepalive ---
    -- Executing Hangup("SIP/100-081d0570", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/100-081d0570' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/100-081d0570' in macro 'outisbusy'
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/100-081d0570'
Scheduling destruction of call '[email protected]' in 32000 ms
Reliably Transmitting (no NAT) to 192.168.1.2:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a80102000008d3454077a10000712400002c33;received=192.168.1.2
From: "unknown"<sip:[email protected]>;tag=9498910925858
To: <sip:02119*******@192.168.1.5>;tag=as2183240d
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from 192.168.1.2:5060:
ACK sip:02119*******@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2;rport;branch=z9hG4bKc0a80102000008d3454077a10000712400002c33
Content-Length: 0
Call-ID: [email protected]
CSeq: 2 ACK
From: "unknown"<sip:[email protected]>;tag=9498910925858
Max-Forwards: 70
To: <sip:0211984[email protected]>;tag=as2183240d
User-Agent: SJphone/1.60.289a (SJ Labs)
Wenn ihr mehr Infos bräuchtet, sagt mir bitte was, ich verzweifle hier noch :-(
 
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Monotron

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12 Okt 2006
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Schade das nichts als Antwort kam.
Aber ich habe die Lösung mitlerweile auch gefunden.

In der Trunk bei Outbound folgendes einstellen, dann gehts:

Code:
allow=gsm&ilbc&alaw&ulaw
context=from-pstn
disallow=all
fromdomain=0211.sip.arcor.de
fromuser=02115*******
host=0211.sip.arcor.de
insecure=very
secret=********
type=friend
username=02115*******
 

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