SIP -> rejected because extension not found

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ccfritz

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Hallo alle Helfer! ;-)

unsere Asterisk hängt auf einem SIP trunk an unserem provider. daher ergibt sich folgendes in der sip.conf

Code: 
sip.conf =>

[Incoming-SIP]
type=friend   
host=XXX.XXX.XXX.XXX

extensions.conf

Code: 
[Incoming-SIP]
exten => _+X.125,1,Answer()
exten => _+X.125,n,Dial(SIP/userAA)


Fehlermeldung:

Code: 
phone*CLI> 
  == Using SIP RTP CoS mark 5
[Feb  9 13:15:32] NOTICE[4884]: chan_sip.c:20044 handle_request_invite: Call from 'Incoming-SIP' to extension '+49123456789' rejected because extension not found.

Asterisk V1.6

hat jemand eine Idee ? sicher nur eine Kleinigkeit...

danke!
 
was willst Du mit dem Pattern
Code: 
_+X.[COLOR="Red"]125[/COLOR]
erreichen? Demnach müßten alle eingehenden Anrufe auf 125 enden.

in der sip.conf fehlt Dir ein "context=Incoming-SIP"
Zum Vergrößern anklicken....

Das sieht nur so aus. Anhand der Fehlermeldung ist eindeutig erkennbar, daß diese aus dem richtgen Context kommt! Das mit dem "+" kann durchaus notwendig sein, es gibt wirklich Provider, die eine Rufnummer mit + am Anfang schicken.
 
@indianer: richitg, soltle auch der fall sein. _+x. beduet ja, alle herienkommenden anrufe die mit z.b. +49xxx kommen.. und mit 125 enden, da diese nummer ja angerufen wurde.
sprich die itneren druchwahl 125

habe jetzt mal auf ganz reset gemacht:

Code: 
sip.conf =>

[Incoming-SIP]
type=peer   
host=XXX.XXX.XXX.XXX
context=Incoming-SIP

extensions.conf

Code: 
[general]
static=yes
writeprotect=no
priorityjumping=no
autofallthrough=yes
clearglobalvars=no

[globals]


;Hereinkommende Anrufe die über SIP kommen
[Incoming-SIP]

exten => s,1,Answer()
exten => s,n,Playback(hello-world)
exten => s,n,Hangup()

selbe fehler....
 
Das mit der 125 nach dem X. wird vermutlich nicht funktionieren.

Falls die Länge der Nummer vor der 125 immer gleich und bekannt ist, solltest Du einfach mal mit _+XXXXXXXX125 testen, wobei jedes X für eine Stelle vor der 125 steht - also entsprechend viele X verwenden.

Die "neue" sip.conf kann so nicht funktionieren, wenn der wiedergegebene Teil wirklich alles ist.
 
habe die exte.conf nun geändert...

Code: 
phone*CLI> dialplan show
[ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ]
  's' =>            1. NoOp()                                     [app_dial]

[ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ]
  's' =>            1. NoOp()                                     [app_queue]

[ Context 'parkedcalls' created by 'features' ]
  '700' =>          1. Park()                                     [features]

[ Context 'Incoming-SIP' created by 'pbx_ael' ]
  '_X.' =>          1. Answer()                                   [pbx_ael]
                    2. Playback(hello-world)                      [pbx_ael]
                    3. Hangup()                                   [pbx_ael]

-= 4 extensions (6 priorities) in 4 contexts. =-


@VoIP_Indianer: wieso kann die sip.conf so nicht stimmen?

trunks sind online ... und die calls kommen auch über den richtigen context rein...
 
so habe mal einen debug aus einem call geholt:

Code: 
<--- SIP read from UDP:213.XXX.XXX.XX7:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 213.XXX.XXX.XX7:5060;branch=z9hG4bK26ff7a2a;rport
From: "+49167576" <sip:[email protected]>;tag=as39b83cf1
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: silver.sip
Max-Forwards: 70
Remote-Party-ID: "+49167576" <sip:[email protected]>;privacy=off;screen=no
Date: Tue, 09 Feb 2010 13:44:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 28264 28264 IN IP4 213.XXX.XXX.XX7
s=session
c=IN IP4 213.XXX.XXX.XX7
t=0 0
m=audio 14452 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  0 [ 47]: INVITE sip:[email protected] SIP/2.0
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 213.XXX.XXX.XX7:5060;branch=z9hG4bK26ff7a2a;rport
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  2 [ 72]: From: "+49167576" <sip:[email protected]>;tag=as39b83cf1
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  3 [ 38]: To: <sip:[email protected]>
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  4 [ 44]: Contact: <sip:[email protected]>
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  5 [ 57]: Call-ID: [email protected]
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  6 [ 16]: CSeq: 102 INVITE
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  7 [ 22]: User-Agent: silver.sip
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  8 [ 16]: Max-Forwards: 70
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  9 [ 90]: Remote-Party-ID: "+49167576" <sip:[email protected]>;privacy=off;screen=no
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header 10 [ 35]: Date: Tue, 09 Feb 2010 13:44:10 GMT
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header 11 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header 12 [ 29]: Content-Type: application/sdp
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header 13 [ 19]: Content-Length: 269
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header 14 [  0]: 
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  0 [  3]: v=0
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  1 [ 41]: o=root 28264 28264 IN IP4 213.XXX.XXX.XX7
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  2 [  9]: s=session
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  3 [ 24]: c=IN IP4 213.XXX.XXX.XX7
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  4 [  5]: t=0 0
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  5 [ 31]: m=audio 14452 RTP/AVP 8 0 3 101
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  6 [ 20]: a=rtpmap:8 PCMA/8000
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  7 [ 20]: a=rtpmap:0 PCMU/8000
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  8 [ 19]: a=rtpmap:3 GSM/8000
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body  9 [ 33]: a=rtpmap:101 telephone-event/8000
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body 10 [ 15]: a=fmtp:101 0-16
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:    Body 11 [ 25]: a=silenceSupp:off - - - -
--- (14 headers 12 lines) ---
[Feb  9 14:43:21] DEBUG[5412]: acl.c:499 ast_ouraddrfor: Found IP address for this socket
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:3677 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 78.142.183.42:5060
  == Using SIP RTP CoS mark 5
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:5000 do_setnat: Setting NAT on RTP to Off
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7289 sip_alloc: Allocating new SIP dialog for [email protected] - INVITE (With RTP)
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:21606 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
Sending to 213.XXX.XXX.XX7 : 5060 (no NAT)
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:19924 handle_request_invite: Initializing initreq for method INVITE - callid [email protected]
Using INVITE request as basis request - [email protected]
Found peer 'Incoming-SIP' for '+49167576' from 213.XXX.XXX.XX7:5060
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:5000 do_setnat: Setting NAT on RTP to Off
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8194 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8194 process_sdp: Processing session-level SDP o=root 28264 28264 IN IP4 213.XXX.XXX.XX7... UNSUPPORTED.
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8194 process_sdp: Processing session-level SDP s=session... UNSUPPORTED.
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8194 process_sdp: Processing session-level SDP c=IN IP4 213.XXX.XXX.XX7... OK.
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8194 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMA for ID 8
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8358 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
Found audio description format PCMU for ID 0
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8358 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
Found audio description format GSM for ID 3
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8358 process_sdp: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
Found audio description format telephone-event for ID 101
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8358 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8358 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED.
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8358 process_sdp: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED.
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 213.XXX.XXX.XX7:14452
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:8543 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw)
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:20016 handle_request_invite: Checking SIP call limits for device 
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:5640 update_call_counter: Updating call counter for incoming call
Looking for +491234567125 in Incoming-SIP (domain 78.142.183.42)

<--- Reliably Transmitting (no NAT) to 213.XXX.XXX.XX7:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 213.XXX.XXX.XX7:5060;branch=z9hG4bK26ff7a2a;received=213.XXX.XXX.XX7;rport=5060
From: "+49167576" <sip:[email protected]>;tag=as39b83cf1
To: <sip:[email protected]>;tag=as3d9db869
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX SVN-branch-1.6.2-r244555
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:3908 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id  #54
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:3556 __sip_xmit: Trying to put 'SIP/2.0 404' onto UDP socket destined for 213.XXX.XXX.XX7:5060
[Feb  9 14:43:21] NOTICE[5412]: chan_sip.c:20044 handle_request_invite: Call from 'Incoming-SIP' to extension '+491234567125' rejected because extension not found.
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:5640 update_call_counter: Updating call counter for incoming call
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
phone*CLI> 
<--- SIP read from UDP:213.XXX.XXX.XX7:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 213.XXX.XXX.XX7:5060;branch=z9hG4bK26ff7a2a;rport
From: "+49167576" <sip:[email protected]>;tag=as39b83cf1
To: <sip:[email protected]>;tag=as3d9db869
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: silver.sip
Max-Forwards: 70
Remote-Party-ID: "+49167576" <sip:[email protected]>;privacy=off;screen=no
Content-Length: 0


<------------->
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  0 [ 44]: ACK sip:[email protected] SIP/2.0
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  1 [ 66]: Via: SIP/2.0/UDP 213.XXX.XXX.XX7:5060;branch=z9hG4bK26ff7a2a;rport
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  2 [ 72]: From: "+49167576" <sip:[email protected]>;tag=as39b83cf1
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  3 [ 53]: To: <sip:[email protected]>;tag=as3d9db869
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  4 [ 44]: Contact: <sip:[email protected]>
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  5 [ 57]: Call-ID: [email protected]
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  6 [ 13]: CSeq: 102 ACK
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  7 [ 22]: User-Agent: silver.sip
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  8 [ 16]: Max-Forwards: 70
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header  9 [ 90]: Remote-Party-ID: "+49167576" <sip:[email protected]>;privacy=off;screen=no
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header 10 [ 17]: Content-Length: 0
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:7815 parse_request:  Header 11 [  0]: 
--- (11 headers 0 lines) ---
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:21606 handle_incoming: **** Received ACK (6) - Command in SIP ACK
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:4060 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #54
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:4092 __sip_ack: Stopping retransmission on '[email protected]' of Response 102: Match Found
[Feb  9 14:43:21] DEBUG[5412]: chan_sip.c:5787 sip_destroy: Destroying SIP dialog [email protected]
Really destroying SIP dialog '[email protected]' Method: ACK
phone*CLI>
 
gelöst:

das "+" Zeichen also plus zeichen wird bei "X" übergangen...

exten => _+X!,1,Goto(${EXTEN:1})


löst das Problem...

quelle: support meines providers...
 
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