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Не работают исходящие DIALSTATUS = CONGESTION and HANGUPCAUSE = 21

0

Настройки исходящих
username=712020707
type=friend
secret=PASS
nat=no
qualify=yes
insecure=port,invite
host=195.158.10.102
disallow=all
allow=ulaw&alaw&gsm
canreinvite=no

Помогите разобраться.

[May 1 20:09:20] VERBOSE[3652] chansip.c:
<--- SIP read from UDP:192.168.1.106:1372 --->
INVITE sip:2547687@192.168.1.245:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:1372;branch=z9hG4bK-d8754z-d67e9823981eca67-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:110@192.168.1.106:1372;rinstance=0e9e16c46309e74e>
To: <sip:2547687@192.168.1.245:5060>
From: "110"<sip:110@192.168.1.245:5060>;tag=320e9301
Call-ID: NzY1MDdiNWQ5ZjU0N2QxZGEzZDM2YmFhYTEwYTgzODU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 282

v=0
o=3cxVCE 183742980 324364875 IN IP4 192.168.1.106
s=3cxVCE Audio Call
c=IN IP4 192.168.1.106
t=0 0
m=audio 40008 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: --- (13 headers 13 lines) ---
[May 1 20:09:20] VERBOSE[3652] netsock.c: == Using UDPTL TOS bits 184
[May 1 20:09:20] VERBOSE[3652] netsock.c: == Using UDPTL CoS mark 5
[May 1 20:09:20] VERBOSE[3652] chansip.c: Sending to 192.168.1.106:1372 (no NAT)
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: Using INVITE request as basis request - NzY1MDdiNWQ5ZjU0N2QxZGEzZDM2YmFhYTEwYTgzODU.
[May 1 20:09:20] VERBOSE[3652] chansip.c: Found peer '110' for '110' from 192.168.1.106:1372
[May 1 20:09:20] VERBOSE[3652] chan
sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.1.106:1372 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.106:1372;branch=z9hG4bK-d8754z-d67e9823981eca67-1---d8754z-;received=192.168.1.106;rport=1372
From: "110"<sip:110@192.168.1.245:5060>;tag=320e9301
To: <sip:2547687@192.168.1.245:5060>;tag=as6d54a732
Call-ID: NzY1MDdiNWQ5ZjU0N2QxZGEzZDM2YmFhYTEwYTgzODU.
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66dc672e"
Content-Length: 0


<------------>
[May 1 20:09:20] VERBOSE[3652] chansip.c: Scheduling destruction of SIP dialog 'NzY1MDdiNWQ5ZjU0N2QxZGEzZDM2YmFhYTEwYTgzODU.' in 6528 ms (Method: INVITE)
[May 1 20:09:20] VERBOSE[3652] chan
sip.c:
<--- SIP read from UDP:192.168.1.106:1372 --->
ACK sip:2547687@192.168.1.245:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:1372;branch=z9hG4bK-d8754z-d67e9823981eca67-1---d8754z-;rport
Max-Forwards: 70
To: <sip:2547687@192.168.1.245:5060>;tag=as6d54a732
From: "110"<sip:110@192.168.1.245:5060>;tag=320e9301
Call-ID: NzY1MDdiNWQ5ZjU0N2QxZGEzZDM2YmFhYTEwYTgzODU.
CSeq: 1 ACK
Content-Length: 0

<------------->
[May 1 20:09:20] VERBOSE[3652] chansip.c: --- (8 headers 0 lines) ---
[May 1 20:09:20] VERBOSE[3652] chan
sip.c:
<--- SIP read from UDP:192.168.1.106:1372 --->
INVITE sip:2547687@192.168.1.245:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:1372;branch=z9hG4bK-d8754z-e178204028320269-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:110@192.168.1.106:1372;rinstance=0e9e16c46309e74e>
To: <sip:2547687@192.168.1.245:5060>
From: "110"<sip:110@192.168.1.245:5060>;tag=320e9301
Call-ID: NzY1MDdiNWQ5ZjU0N2QxZGEzZDM2YmFhYTEwYTgzODU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="110",realm="asterisk",nonce="66dc672e",uri="sip:2547687@192.168.1.245:5060",response="1d53ace890f293ce0d4f2a00c98a3c93",algorithm=MD5
Content-Length: 282

v=0
o=3cxVCE 183742980 324364875 IN IP4 192.168.1.106
s=3cxVCE Audio Call
c=IN IP4 192.168.1.106
t=0 0
m=audio 40008 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[May 1 20:09:20] VERBOSE[3652] chansip.c: --- (14 headers 13 lines) ---
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: Sending to 192.168.1.106:1372 (no NAT)
[May 1 20:09:20] VERBOSE[3652] chansip.c: Using INVITE request as basis request - NzY1MDdiNWQ5ZjU0N2QxZGEzZDM2YmFhYTEwYTgzODU.
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: Found peer '110' for '110' from 192.168.1.106:1372
[May 1 20:09:20] VERBOSE[3652] netsock2.c: == Using SIP RTP TOS bits 184
[May 1 20:09:20] VERBOSE[3652] netsock2.c: == Using SIP RTP CoS mark 5
[May 1 20:09:20] VERBOSE[3652] chansip.c: Found RTP audio format 0
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: Found RTP audio format 8
[May 1 20:09:20] VERBOSE[3652] chansip.c: Found RTP audio format 3
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: Found RTP audio format 101
[May 1 20:09:20] VERBOSE[3652] chansip.c: Found audio description format PCMU for ID 0
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: Found audio description format PCMA for ID 8
[May 1 20:09:20] VERBOSE[3652] chansip.c: Found audio description format GSM for ID 3
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: Found audio description format telephone-event for ID 101
[May 1 20:09:20] VERBOSE[3652] chansip.c: Capabilities: us - 0x3c000e (gsm|ulaw|alaw|h261|h263|h263p|h264), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[May 1 20:09:20] VERBOSE[3652] chansip.c: Peer audio RTP is at port 192.168.1.106:40008
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: Peer doesn't provide video
[May 1 20:09:20] VERBOSE[3652] chansip.c: Looking for 2547687 in from-internal (domain 192.168.1.245:5060)
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: listroute: hop: <sip:110@192.168.1.106:1372;rinstance=0e9e16c46309e74e>
[May 1 20:09:20] VERBOSE[3652] chan
sip.c:
<--- Transmitting (no NAT) to 192.168.1.106:1372 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.106:1372;branch=z9hG4bK-d8754z-e178204028320269-1---d8754z-;received=192.168.1.106;rport=1372
From: "110"<sip:110@192.168.1.245:5060>;tag=320e9301
To: <sip:2547687@192.168.1.245:5060>
Call-ID: NzY1MDdiNWQ5ZjU0N2QxZGEzZDM2YmFhYTEwYTgzODU.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2547687@192.168.1.245:5060>
Content-Length: 0


<------------>
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [2547687@from-internal:1] Macro("SIP/110-00000759", "user-callerid,SKIPTTL,") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/110-00000759", "AMPUSER=110") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/110-00000759", "0?report") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/110-00000759", "1?Set(REALCALLERIDNUM=110)") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:4] Set("SIP/110-00000759", "AMPUSER=110") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:5] Set("SIP/110-00000759", "AMPUSERCIDNAME=operator1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/110-00000759", "0?report") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:7] Set("SIP/110-00000759", "AMPUSERCID=110") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:8] Set("SIP/110-00000759", "CALLERID(all)="operator1" <110>") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/110-00000759", "0?Set(CHANNEL(language)=)") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/110-00000759", "1?continue") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Goto (macro-user-callerid,s,19)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:19] Set("SIP/110-00000759", "CALLERID(number)=110") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:20] Set("SIP/110-00000759", "CALLERID(name)=operator1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-user-callerid:21] NoOp("SIP/110-00000759", "Using CallerID "operator1" <110>") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [2547687@from-internal:2] NoOp("SIP/110-00000759", "Calling Out Route: Gorod") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [2547687@from-internal:3] Set("SIP/110-00000759", "MOHCLASS=default") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [2547687@from-internal:4] Set("SIP/110-00000759", "NODEST=") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [2547687@from-internal:5] Macro("SIP/110-00000759", "record-enable,110,OUT,") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/110-00000759", "1?check") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Goto (macro-record-enable,s,4)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-record-enable:4] ExecIf("SIP/110-00000759", "0?MacroExit()") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-record-enable:5] GotoIf("SIP/110-00000759", "0?Group:OUT") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Goto (macro-record-enable,s,15)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-record-enable:15] GotoIf("SIP/110-00000759", "0?IN") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-record-enable:16] ExecIf("SIP/110-00000759", "1?MacroExit()") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [2547687@from-internal:6] Macro("SIP/110-00000759", "dialout-trunk,1,2547687,") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/110-00000759", "DIAL
TRUNK=1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/110-00000759", "0?sub-pincheck,s,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/110-00000759", "0?disabletrunk,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/110-00000759", "DIALNUMBER=2547687") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/110-00000759", "DIAL
TRUNKOPTIONS=trwTW") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/110-00000759", "OUTBOUND
GROUP=OUT1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/110-00000759", "0?nomax") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/110-00000759", "0?chanfull") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/110-00000759", "0?skipoutcid") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/110-00000759", "DIAL
TRUNKOPTIONS=WTwt") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/110-00000759", "outbound-callerid,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/110-00000759", "0?Set(CALLERPRES()=)") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/110-00000759", "0?Set(REALCALLERIDNUM=110)") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/110-00000759", "1?normcid") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Goto (macro-outbound-callerid,s,6)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/110-00000759", "USEROUTCID=") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/110-00000759", "EMERGENCYCID=") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/110-00000759", "TRUNKOUTCID=712020707") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/110-00000759", "1?trunkcid") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Goto (macro-outbound-callerid,s,12)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/110-00000759", "1?Set(CALLERID(all)=712020707)") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/110-00000759", "0?Set(CALLERID(all)=)") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/110-00000759", "1?Set(CALLERID(all)=712020707)") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/110-00000759", "0?Set(CALLERPRES()=prohib
passedscreen)") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/110-00000759", "1?sub-flp-1,s,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@sub-flp-1:1] ExecIf("SIP/110-00000759", "1?Return()") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/110-00000759", "OUTNUM=2547687") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/110-00000759", "custom=SIP/sip-2020707") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/110-00000759", "0?Set(DIAL
TRUNKOPTIONS=M(setmusic^default)WTwt)") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/110-00000759", "dialout-trunk-predial-hook,") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/110-00000759", "") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/110-00000759", "0?bypass,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/110-00000759", "0?customtrunk") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/110-00000759", "SIP/sip-2020707/2547687,300,WTwt") in new stack
[May 1 20:09:20] VERBOSE[684] netsock.c: == Using UDPTL TOS bits 184
[May 1 20:09:20] VERBOSE[684] netsock.c: == Using UDPTL CoS mark 5
[May 1 20:09:20] VERBOSE[684] netsock2.c: == Using SIP RTP TOS bits 184
[May 1 20:09:20] VERBOSE[684] netsock2.c: == Using SIP RTP CoS mark 5
[May 1 20:09:20] VERBOSE[684] chan
sip.c: Audio is at 5060
[May 1 20:09:20] VERBOSE[684] chansip.c: Adding codec 0x4 (ulaw) to SDP
[May 1 20:09:20] VERBOSE[684] chan
sip.c: Adding codec 0x8 (alaw) to SDP
[May 1 20:09:20] VERBOSE[684] chansip.c: Adding codec 0x2 (gsm) to SDP
[May 1 20:09:20] VERBOSE[684] chan
sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[May 1 20:09:20] VERBOSE[684] chansip.c: Reliably Transmitting (no NAT) to 195.158.10.102:5060:
INVITE sip:2547687@195.158.10.102 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.102:5060;branch=z9hG4bK64aaf956
Max-Forwards: 70
From: "712020707" <sip:712020707@10.0.1.102>;tag=as065535af
To: <sip:2547687@195.158.10.102>
Contact: <sip:712020707@10.0.1.102:5060>
Call-ID: 531016992fa41ba71e0c788e0fb2f903@10.0.1.102:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.7.0)
Date: Wed, 01 May 2013 15:09:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 881482853 881482853 IN IP4 10.0.1.102
s=Asterisk PBX 1.8.7.0
c=IN IP4 10.0.1.102
t=0 0
m=audio 14886 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[May 1 20:09:20] VERBOSE[684] app
dial.c: -- Called SIP/sip-2020707/2547687
[May 1 20:09:20] VERBOSE[3652] chansip.c:
<--- SIP read from UDP:195.158.10.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.102:5060;branch=z9hG4bK64aaf956
From: "712020707" <sip:712020707@10.0.1.102>;tag=as065535af
To: <sip:2547687@195.158.10.102>
Call-ID: 531016992fa41ba71e0c788e0fb2f903@10.0.1.102:5060
CSeq: 102 INVITE

<------------->
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: --- (6 headers 0 lines) ---
[May 1 20:09:20] VERBOSE[3652] chansip.c:
<--- SIP read from UDP:195.158.10.102:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.1.102:5060;branch=z9hG4bK64aaf956
From: "712020707" <sip:712020707@10.0.1.102>;tag=as065535af
To: <sip:2547687@195.158.10.102>;tag=aprqngfrt-tl5lt930000c6
Call-ID: 531016992fa41ba71e0c788e0fb2f903@10.0.1.102:5060
CSeq: 102 INVITE
Reason: Q.850;cause=55;text="Call Terminated"

<------------->
[May 1 20:09:20] VERBOSE[3652] chan
sip.c: --- (7 headers 0 lines) ---
[May 1 20:09:20] VERBOSE[3652] chansip.c: Transmitting (no NAT) to 195.158.10.102:5060:
ACK sip:2547687@195.158.10.102 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.102:5060;branch=z9hG4bK64aaf956
Max-Forwards: 70
From: "712020707" <sip:712020707@10.0.1.102>;tag=as065535af
To: <sip:2547687@195.158.10.102>;tag=aprqngfrt-tl5lt930000c6
Contact: <sip:712020707@10.0.1.102:5060>
Call-ID: 531016992fa41ba71e0c788e0fb2f903@10.0.1.102:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.7.0)
Content-Length: 0


---
[May 1 20:09:20] WARNING[3652] chan
sip.c: Received response: "Forbidden" from '"712020707" <sip:712020707@10.0.1.102>;tag=as065535af'
[May 1 20:09:20] VERBOSE[684] appdial.c: -- SIP/sip-2020707-0000075a is circuit-busy
[May 1 20:09:20] VERBOSE[684] app
dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/110-00000759", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-dialout-trunk:21] Goto("SIP/110-00000759", "s-CONGESTION,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Goto (macro-dialout-trunk,s-CONGESTION,1)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/110-00000759", "RC=21") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/110-00000759", "21,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Goto (macro-dialout-trunk,21,1)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [21@macro-dialout-trunk:1] Goto("SIP/110-00000759", "continue,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Goto (macro-dialout-trunk,continue,1)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/110-00000759", "1?noreport") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Goto (macro-dialout-trunk,continue,3)
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/110-00000759", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [continue@macro-dialout-trunk:4] Set("SIP/110-00000759", "CALLERID(number)=110") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [2547687@from-internal:7] Macro("SIP/110-00000759", "outisbusy,") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outisbusy:1] Progress("SIP/110-00000759", "") in new stack
[May 1 20:09:20] VERBOSE[684] chansip.c: Audio is at 5060
[May 1 20:09:20] VERBOSE[684] chan
sip.c: Adding codec 0x4 (ulaw) to SDP
[May 1 20:09:20] VERBOSE[684] chansip.c: Adding codec 0x2 (gsm) to SDP
[May 1 20:09:20] VERBOSE[684] chan
sip.c: Adding codec 0x8 (alaw) to SDP
[May 1 20:09:20] VERBOSE[684] chansip.c: Adding non-codec 0x1 (telephone-event) to SDP
[May 1 20:09:20] VERBOSE[684] chan
sip.c:
<--- Transmitting (no NAT) to 192.168.1.106:1372 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.106:1372;branch=z9hG4bK-d8754z-e178204028320269-1---d8754z-;received=192.168.1.106;rport=1372
From: "110"<sip:110@192.168.1.245:5060>;tag=320e9301
To: <sip:2547687@192.168.1.245:5060>;tag=as5d03095c
Call-ID: NzY1MDdiNWQ5ZjU0N2QxZGEzZDM2YmFhYTEwYTgzODU.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:2547687@192.168.1.245:5060>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1093149922 1093149922 IN IP4 192.168.1.245
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.245
t=0 0
m=audio 12654 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outisbusy:2] GotoIf("SIP/110-00000759", "0?emergency,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outisbusy:3] GotoIf("SIP/110-00000759", "0?intracompany,1") in new stack
[May 1 20:09:20] VERBOSE[684] pbx.c: -- Executing [s@macro-outisbusy:4] Playback("SIP/110-00000759", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[May 1 20:09:20] VERBOSE[684] file.c: -- <SIP/110-00000759> Playing 'all-circuits-busy-now.ulaw' (language 'en')

удалить закрыть спам изменить тег редактировать

спросил 2013-05-01 19:31:24 +0400

CorsarWish Gravatar CorsarWish
9 3 2

Comments

Спасибо!!! Помог fromdomain

CorsarWish ( 2013-05-02 00:02:09 +0400 )редактировать

1 Ответ

2

проверить формат набираемого номера, поэксперементировать с fromdomain, но самое простое - спросить у вышестоящего провайдера чего ему не нравится что он вам 403 дает.

ссылка удалить спам редактировать

ответил 2013-05-01 22:12:57 +0400

komrad123 Gravatar komrad123
3780 5 3 42

Ваш ответ

Please start posting your answer anonymously - your answer will be saved within the current session and published after you log in or create a new account. Please try to give a substantial answer, for discussions, please use comments and please do remember to vote (after you log in)!
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Задан: 2013-05-01 19:31:24 +0400

Просмотрен: 221 раз

Обновлен: May 01 '13

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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.