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Errore nelle chiamate in uscita 403 Forbidden

https://www.qnapclub.it/viewtopic.php?t=10034

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Errore nelle chiamate in uscita 403 Forbidden

Inviato: 25 set 2014, 11:58
da andreaqnap
Ciao a tutti,

sono un perfetto ignorante quindi probabilmente la soluzione al mio problema è dietro l'angolo ma non riesco a vederla.

Ho installato l'Asterisk Digium sul mio QNAP TS459. Il trunk si registra correttamente, così come tutti i telefoni SIP fisici (yealink T20). Le chiamate in ingresso le ricevo senza problemi, così come effettuo chiamate fra interni o il trasferimento di una chiamata da un telefono ad un altro.

Non riesco invece ad effettuare chiamate in uscita: il messaggio è "403 forbidden". Ho provato ad usare anche un softphone (Zoiper) ma ottengo lo stesso errore. Il pattern per le chiamate in uscita prevede la composizione dello 0 per prendere la linea e poi il numero da chiamare: in Digium è stato impostato quindi _0.

Incollo di seguito il log del CLI e la configurazione di una estensione:

[DAL CLI]
<--- SIP read from UDP:192.168.2.5:5062 --->
INVITE sip:03496753439@192.168.2.63 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.5:5062;branch=z9hG4bK1853633956
From: "Andrea Uff" <sip:6005@192.168.2.63>;tag=607636310
To: <sip:03496753439@192.168.2.63>
Call-ID: 1958915248@192.168.2.5
CSeq: 1 INVITE
Contact: <sip:6005@192.168.2.5:5062>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T20P 9.60.0.100
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 292
v=0
o=- 20054 20054 IN IP4 192.168.2.5
s=SDP data
c=IN IP4 192.168.2.5
t=0 0
m=audio 11790 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (14 headers 14 lines) ---
Sending to 192.168.2.5:5062 (NAT)
Using INVITE request as basis request - 1958915248@192.168.2.5
Found peer '6005' for '6005' from 192.168.2.5:5062
<--- Reliably Transmitting (NAT) to 192.168.2.5:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.5:5062;branch=z9hG4bK1853633956;received=192.168.2.5;rport=5062
From: "Andrea Uff" <sip:6005@192.168.2.63>;tag=607636310
To: <sip:03496753439@192.168.2.63>;tag=as0709e1ae
Call-ID: 1958915248@192.168.2.5
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1586be21"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1958915248@192.168.2.5' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.2.5:5062 --->
ACK sip:03496753439@192.168.2.63 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.5:5062;branch=z9hG4bK1853633956
From: "Andrea Uff" <sip:6005@192.168.2.63>;tag=607636310
To: <sip:03496753439@192.168.2.63>;tag=as0709e1ae
Call-ID: 1958915248@192.168.2.5
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:192.168.2.5:5062 --->
INVITE sip:03496753439@192.168.2.63 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.5:5062;branch=z9hG4bK451938962
From: "Andrea Uff" <sip:6005@192.168.2.63>;tag=607636310
To: <sip:03496753439@192.168.2.63>
Call-ID: 1958915248@192.168.2.5
CSeq: 2 INVITE
Contact: <sip:6005@192.168.2.5:5062>
Authorization: Digest username="6005", realm="asterisk", nonce="1586be21", uri="sip:03496753439@192.168.2.63", response="a7961645122f6c2db69d4eedcb193bb0", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T20P 9.60.0.100
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 292
v=0
o=- 20054 20054 IN IP4 192.168.2.5
s=SDP data
c=IN IP4 192.168.2.5
t=0 0
m=audio 11790 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (15 headers 14 lines) ---
Sending to 192.168.2.5:5062 (NAT)
Using INVITE request as basis request - 1958915248@192.168.2.5
Found peer '6005' for '6005' from 192.168.2.5:5062
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.5:11790
Looking for 03496753439 in DLPN_TEST (domain 192.168.2.63)
list_route: hop: <sip:6005@192.168.2.5:5062>
<--- Transmitting (NAT) to 192.168.2.5:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.5:5062;branch=z9hG4bK451938962;received=192.168.2.5;rport=5062
From: "Andrea Uff" <sip:6005@192.168.2.63>;tag=607636310
To: <sip:03496753439@192.168.2.63>
Call-ID: 1958915248@192.168.2.5
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:03496753439@192.168.2.63:5060>
Content-Length: 0
<------------>
-- Executing [03496753439@DLPN_TEST:1] Macro("SIP/6005-0000002a", "trunkdial-failover-0.3,SIP/trunk_1/3496753439,,trunk_1,") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf("SIP/6005-0000002a", "0?1-fmsetcid,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6005-0000002a", "0?1-setgbobname,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/6005-0000002a", "CALLERID(num)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:4] Set("SIP/6005-0000002a", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:5] GotoIf("SIP/6005-0000002a", "0?1-dial,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:6] Set("SIP/6005-0000002a", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:7] Set("SIP/6005-0000002a", "CALLERID(all)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:8] Goto("SIP/6005-0000002a", "1-dial,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/6005-0000002a", "SIP/trunk_1/3496753439") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/trunk_1/3496753439
[Sep 25 11:22:54] WARNING[14802]: chan_sip.c:21076 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:289415655@cust.sip.twt.it>;tag=as1c0218c5'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6005-0000002a", "0 > 0 ?1-CHANUNAVAIL,1:1-out,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-out,1)
-- Executing [1-out@macro-trunkdial-failover-0.3:1] Hangup("SIP/6005-0000002a", "") in new stack
== Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on 'SIP/6005-0000002a' in macro 'trunkdial-failover-0.3'
== Spawn extension (DLPN_TEST, 03496753439, 1) exited non-zero on 'SIP/6005-0000002a'
Scheduling destruction of SIP dialog '1958915248@192.168.2.5' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 192.168.2.5:5062 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.2.5:5062;branch=z9hG4bK451938962;received=192.168.2.5;rport=5062
From: "Andrea Uff" <sip:6005@192.168.2.63>;tag=607636310
To: <sip:03496753439@192.168.2.63>;tag=as50d9b83d
Call-ID: 1958915248@192.168.2.5
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.2.5:5062 --->
ACK sip:03496753439@192.168.2.63 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.5:5062;branch=z9hG4bK451938962
From: "Andrea Uff" <sip:6005@192.168.2.63>;tag=607636310
To: <sip:03496753439@192.168.2.63>;tag=as50d9b83d
Call-ID: 1958915248@192.168.2.5
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:192.168.2.5:5062 --->
<------------->

[ESTENSIONE 6005]
fullname=Andrea
registersip=no
host=dynamic
callgroup=1
mailbox=6005
call-limit=100
type=peer
username=6005
transfer=yes
callcounter=yes
context=DLPN_TEST
cid_number=6005
hasvoicemail=no
vmsecret=
email=
threewaycalling=no
hasdirectory=no
callwaiting=no
hasmanager=no
hasagent=no
hassip=yes
hasiax=no
secret=andreacsraee
nat=yes
canreinvite=no
dtmfmode=rfc2833
insecure=no
pickupgroup=1
disallow=all
allow=ulaw,gsm
macaddress=5
autoprov=yes
label=6005
linenumber=1
LINEKEYS=1

Qualcuno mi può dare una mano? Grazie mille, Andrea

Re: Errore nelle chiamate in uscita 403 Forbidden

Inviato: 08 gen 2015, 20:28
da alby
buonasera, Andrea hai poi risolto il tuo problema???
Ma dimmi che versione di firmware hai sul ts 459 ???
: Blink :
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