Forbidden / circuit-busy / bei outbound calls

daguru2222 · 6 Okt 2010

Hallo Asterisk-Experten,

seit dem Neustarten des Asterisks sind auf einmal keine outbound calls mehr moeglich. Mir ist bewusst, dass das natuerlich nicht die Ursache sein kann, sondern etwas dafuer verantwortlich sein muss. Allerdings weiss ich dabei momentan nicht weiter und wuerde mich daher ueber jeden Rat freuen.

Inbound calls klappen problemlos, intern telefonieren geht auch, nur extern ueber den Trunk geht es nicht (wenn ich den Trunk hingegen direkt hier auf dem SNOM konfiguriere, kann ich auch outbound telefonieren, d.h. es muss mit dem Asterisk zu tun haben).

== Using SIP RTP CoS mark 5
-- Executing [0123456789@from-intern:1] Dial("SIP/9998-00000018", "SIP/0123456789@OUT_bellsip,45,r") in new stack
== Using SIP RTP CoS mark 5
-- Called 0123456789@OUT_bellsip
[Oct 6 17:14:44] WARNING[11736]: chan_sip.c:17929 handle_response_invite: Received response: "Forbidden" from '"Max" <sip:[email protected]>;tag=as6dd2be2f'
-- SIP/OUT_bellsip-00000019 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/9998-00000018' status is 'CONGESTION'

pbx10*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
9998/9998 111.111.111.111 D N A 62137 OK (48 ms)
OUT_bellsip/MaxMustermann 80.190.145.184 A 5060 OK (16 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
pbx10*CLI>

Die sip.conf:

Code:

[general]
context=default
bindport=5060
bindaddr=123.123.123.123
srvlookup=yes
alwaysauthreject=yes
allowguest=no
nat=yes
dtmfmode=auto
maxexpirey=3600
defaultexpirey=600
deny=0.0.0.0/0.0.0.0
registerattempts=10
; compactheaders=yes

register=MaxMustermann:[email protected]/MaxMustermann


[9998]
callerid=Max <9998>
host=dynamic
user=9998
secret=xxxxxxxxxxxx
type=friend
call-limit=3
mailbox=9998
vmexten=88
nat=yes
canreinvite=no
context=from-intern
qualify=yes

[OUT_bellsip]
disallow=all
host=bellsip.com
username=MaxMustermann
secret=xxxxxxxxxxxxx
; type=peer
type=friend
qualify=yes
; insecure=port,invite
permit=80.190.145.184/29
dtmfmode=rfc2833
; canreinvite=no
nat=no
; allow=gsm
allow=ulaw
allow=alaw
allow=g726
allow=g729

Die extensions.conf:

Code:

[lokal]
exten => _999X,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})

[macro-stdexten]
 exten => s,1,Dial(${ARG2},20)
 exten => s,2,Goto(s-${DIALSTATUS},1)
 exten => s-NOANSWER,1,Voicemail(${ARG1},us)
 exten => s-NOANSWER,2,Goto(default,s,1)
 exten => s-BUSY,1,Voicemail(${ARG1},us)
 exten => s-BUSY,2,Goto(default,s,1)
 exten => _s-.,1,Goto(s-NOANSWER,1)

[extlines_out]
exten => _0.,1,Dial(SIP/${EXTEN}@OUT_bellsip,45,r)

[incomming]
exten => OUT_bellsip,1,Macro(stdexten,9998,SIP/9998)

[echotest]
exten => 81,1,Answer
exten => 81,2,Wait(1)
exten => 81,3,Playback(demo-echotest)
exten => 81,4,Echo()
exten => 81,5,Playback(demo-echodone)
exten => 81,6,Hangup

[mailbox_own]
exten => 88,1,Answer
exten => 88,n,Wait(1)
exten => 88,n,VoicemailMain(s${CALLERID(num)})
exten => 88,n,hangup

[from-intern]
include => lokal
include => macro-ruf
include => echotest
include => mailbox_own
include => extlines_out

[default]
; Wer hier landet ist entweder schlecht konfiguriert oder hat keine "Rechte"
exten => _X.,1,Answer ()
exten => _X.,2,Verbose(D E F A U L T ==> ${CALLERID(num)} kam um ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} in DEFAULT an als er versuchte die Nummer ${EXTEN} anzurufen.)
; exten => _X.,3,Playback(/dein/weg/zum/benutzerdefinierten/sprachdir/keine_wahlregel)
exten => _X.,3,Playback(vm-goodbye)
exten => _X.,4,Hangup

Hier jetzt noch der Mitschnitt von tcpdump - wobei ich bei den Fehlermeldungen das Gefuehl entwickele, als ob es Probleme mit den Paketgroessen geben koennte ...

17:14:38.388930 IP (tos 0x0, ttl 64, id 62381, offset 0, flags [none], proto UDP (17), length 540) pbx10.sip > 80.190.145.184.sip: SIP, length: 512
OPTIONS sip:bellsip.com SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK3879d551;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as479b26d2
To: <sip:bellsip.com>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Wed, 06 Oct 2010 15:14:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

17:14:38.404930 IP (tos 0x0, ttl 56, id 0, offset 0, flags [DF], proto UDP (17), length 346) 80.190.145.184.sip > pbx10.sip: SIP, length: 318
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK3879d551;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as479b26d2
To: <sip:bellsip.com>;tag=dfaa3fabcb0d5d54782753a893ae1d7e.3b5f
Call-ID: [email protected]
CSeq: 102 OPTIONS
Content-Length: 0

17:14:38.788930 IP (tos 0x0, ttl 64, id 62382, offset 0, flags [none], proto UDP (17), length 628) pbx10.sip > 80.190.145.184.sip: SIP, length: 600
REGISTER sip:bellsip.com SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK7c255231;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as24e0a196
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 412 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Authorization: Digest username="MaxMustermann", realm="bellshare.com", algorithm=MD5, uri="sip:bellsip.com", nonce="4cac92129d16457ee34332d5e16830d55c00cf7f", response="a2a1876162dd0efc436f93b248a8f4fd"
Expires: 600
Contact: <sip:[email protected]>
Content-Length: 0

17:14:38.808930 IP (tos 0x0, ttl 56, id 0, offset 0, flags [DF], proto UDP (17), length 306) 80.190.145.184.sip > pbx10.sip: SIP, length: 278
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK7c255231;rport=5060
From: <sip:[email protected]>;tag=as24e0a196
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 412 REGISTER
Content-Length: 0

17:14:38.808930 IP (tos 0x0, ttl 56, id 0, offset 0, flags [DF], proto UDP (17), length 464) 80.190.145.184.sip > pbx10.sip: SIP, length: 436
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK7c255231;rport=5060
From: <sip:[email protected]>;tag=as24e0a196
To: <sip:[email protected]>;tag=dfaa3fabcb0d5d54782753a893ae1d7e.fe59
Call-ID: [email protected]
CSeq: 412 REGISTER
WWW-Authenticate: Digest realm="bellshare.com", nonce="4cac938a3adc43ab31b031824d9b6388683dc40d", stale=true
Content-Length: 0

17:14:38.808930 IP (tos 0x0, ttl 64, id 62383, offset 0, flags [none], proto UDP (17), length 628) pbx10.sip > 80.190.145.184.sip: SIP, length: 600
REGISTER sip:bellsip.com SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2025580f;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as4a5e8c46
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 413 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Authorization: Digest username="MaxMustermann", realm="bellshare.com", algorithm=MD5, uri="sip:bellsip.com", nonce="4cac938a3adc43ab31b031824d9b6388683dc40d", response="5475fd89f45d513a1746a145c38da6de"
Expires: 600
Contact: <sip:[email protected]>
Content-Length: 0

17:14:38.828930 IP (tos 0x0, ttl 56, id 0, offset 0, flags [DF], proto UDP (17), length 306) 80.190.145.184.sip > pbx10.sip: SIP, length: 278
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2025580f;rport=5060
From: <sip:[email protected]>;tag=as4a5e8c46
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 413 REGISTER
Content-Length: 0

17:14:38.828930 IP (tos 0x0, ttl 56, id 0, offset 0, flags [DF], proto UDP (17), length 433) 80.190.145.184.sip > pbx10.sip: SIP, length: 405
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2025580f;rport=5060
From: <sip:[email protected]>;tag=as4a5e8c46
To: <sip:[email protected]>;tag=dfaa3fabcb0d5d54782753a893ae1d7e.707a
Call-ID: [email protected]
CSeq: 413 REGISTER
Contact: <sip:[email protected]>;expires=600;received="sip:123.123.123.123:5060"
Content-Length: 0

17:14:44.420930 IP (tos 0x0, ttl 64, id 62384, offset 0, flags [none], proto UDP (17), length 873) pbx10.sip > 80.190.145.184.sip: SIP, length: 845
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK245c9e18;rport
Max-Forwards: 70
From: "Max" <sip:[email protected]>;tag=as6dd2be2f
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Wed, 06 Oct 2010 15:14:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 294

v=0
o=root 1944431118 1944431118 IN IP4 123.123.123.123
s=Asterisk PBX 1.6.2.13
c=IN IP4 123.123.123.123
t=0 0
m=audio 17776 RTP/AVP 0 8 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:14:44.464930 IP (tos 0x0, ttl 56, id 0, offset 0, flags [DF], proto UDP (17), length 467) 80.190.145.184.sip > pbx10.sip: SIP, length: 439
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK245c9e18;rport=5060
From: "Max" <sip:[email protected]>;tag=as6dd2be2f
To: <sip:[email protected]>;tag=dfaa3fabcb0d5d54782753a893ae1d7e.1d3d
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="bellshare.com", nonce="4cac939018af38e69e6568d9dafffb1a8e386f83"
Content-Length: 0

17:14:44.464930 IP (tos 0x0, ttl 64, id 62385, offset 0, flags [none], proto UDP (17), length 443) pbx10.sip > 80.190.145.184.sip: SIP, length: 415
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK245c9e18;rport
Max-Forwards: 70
From: "Max" <sip:[email protected]>;tag=as6dd2be2f
To: <sip:[email protected]>;tag=dfaa3fabcb0d5d54782753a893ae1d7e.1d3d
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0

17:14:44.464930 IP (tos 0x0, ttl 64, id 62386, offset 0, flags [none], proto UDP (17), length 1092) pbx10.sip > 80.190.145.184.sip: SIP, length: 1064
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK60d88e13;rport
Max-Forwards: 70
From: "Max" <sip:[email protected]>;tag=as6dd2be2f
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Proxy-Authorization: Digest username="MaxMustermann", realm="bellshare.com", algorithm=MD5, uri="sip:[email protected]", nonce="4cac939018af38e69e6568d9dafffb1a8e386f83", response="da8ab3751d7e575406f2deee982aafb6"
Date: Wed, 06 Oct 2010 15:14:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 294

v=0
o=root 1944431118 1944431119 IN IP4 123.123.123.123
s=Asterisk PBX 1.6.2.13
c=IN IP4 123.123.123.123
t=0 0
m=audio 17776 RTP/AVP 0 8 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

17:14:44.508930 IP (tos 0x0, ttl 56, id 0, offset 0, flags [DF], proto UDP (17), length 381) 80.190.145.184.sip > pbx10.sip: SIP, length: 353
SIP/2.0 403 From user does not match authenticated user
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK60d88e13;rport=5060
From: "Max" <sip:[email protected]>;tag=as6dd2be2f
To: <sip:[email protected]>;tag=dfaa3fabcb0d5d54782753a893ae1d7e.5de7
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0

17:14:44.508930 IP (tos 0x0, ttl 64, id 62387, offset 0, flags [none], proto UDP (17), length 443) pbx10.sip > 80.190.145.184.sip: SIP, length: 415
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK60d88e13;rport
Max-Forwards: 70
From: "Max" <sip:[email protected]>;tag=as6dd2be2f
To: <sip:[email protected]>;tag=dfaa3fabcb0d5d54782753a893ae1d7e.5de7
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0

17:14:47.004930 IP (tos 0x0, ttl 56, id 0, offset 0, flags [DF], proto UDP (17), length 32) 80.190.145.184.sip > pbx10.sip: SIP, length: 4
\000\000\000\000
^C
16 packets captured
16 packets received by filter
0 packets dropped by kernel
pbx10:~#

Hat jemand eine Idee, wie es zu der obigen Fehlermeldung kommt?

kombjuder · 6 Okt 2010

Steht doch im Debug:

From user does not match authenticated user

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Forbidden / circuit-busy / bei outbound calls

daguru2222

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