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    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/10004-00000183", "SIP/office/90031,300,Ttr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 18016
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to officeip:5060:
INVITE sip:90031@officeip SIP/2.0
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK5f165e0b;rport
Max-Forwards: 70
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>
Contact: <sip:10004@filialip:5060>
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.13.1)
Date: Tue, 03 Jun 2014 07:21:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 194525798 194525798 IN IP4 filialip
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 filialip
t=0 0
m=audio 18016 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

---
    -- Called SIP/office/90031

<--- SIP read from UDP:officeip:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK5f165e0b;received=filialip;rport=5060
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>;tag=as79445bbf
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.6.2.9)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c2ae67b"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:90031@officeip> for address/port to send to
set_destination: set destination to officeip:5060
Transmitting (NAT) to officeip:5060:
ACK sip:90031@officeip SIP/2.0
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK5f165e0b;rport
Max-Forwards: 70
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>;tag=as79445bbf
Contact: <sip:10004@filialip:5060>
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(1.8.13.1)
Content-Length: 0

---
Audio is at 18016
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to officeip:5060:
INVITE sip:90031@officeip SIP/2.0
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK7de64492;rport
Max-Forwards: 70
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>
Contact: <sip:10004@filialip:5060>
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0(1.8.13.1)
Authorization: Digest username="office", realm="asterisk", algorithm=MD5, uri="sip:90031@officeip", nonce="4c2ae67b", response="df103ca6ce7459aa4f1dc34f935cee99"
Date: Tue, 03 Jun 2014 07:21:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 194525798 194525799 IN IP4 filialip
s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
c=IN IP4 filialip
t=0 0
m=audio 18016 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

---

<--- SIP read from UDP:officeip:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK7de64492;received=filialip;rport=5060
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>;tag=as79445bbf
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 103 INVITE
Server: FPBX-2.9.0(1.6.2.9)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:90031@officeip> for address/port to send to
set_destination: set destination to officeip:5060
Transmitting (NAT) to officeip:5060:
ACK sip:90031@officeip SIP/2.0
Via: SIP/2.0/UDP filialip:5060;branch=z9hG4bK7de64492;rport
Max-Forwards: 70
From: "Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2
To: <sip:90031@officeip>;tag=as79445bbf
Contact: <sip:10004@filialip:5060>
Call-ID: 6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(1.8.13.1)
Content-Length: 0

---
[2014-06-03 14:21:29] WARNING[26892]: chan_sip.c:20366 handle_response_invite: Received response: "Forbidden" from '"Dispetcher 1" <sip:10004@filialip>;tag=as529e86d2'
Scheduling destruction of SIP dialog '6c2e8c540ca35c2a6fac3ea51b551a7b@filialip:5060' in 32000 ms (Method: INVITE)

I have 2 servers with Asterisks on them: 192.168.241.98 and 192.168.243.112.

There is a valid registration on the first:

register => wagateway:qwerty@192.168.243.112:5060

CLI output:

CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time                 
192.168.243.112:5060                    N      wagateway          105 Registered           Wed, 26 Jun 2013 16:42:42

And peers on 243.112 are just fine:

CLI> sip show peers
Name/username             Host                                    Dyn Forcerport ACL Port     Status      Description                      
wacaller/wacaller         192.168.242.235                          D   a             5062     OK (13 ms)                                          
wagateway/s               192.168.241.98                           D   a             5060     OK (1 ms)

extensions.conf on 243.112:

[watest]

exten => 123123123,1,NoOp()
exten => 123123123,n,Dial(SIP/wagateway)
exten => 123123123,n,Hangup()

sip.conf on 243.112:

[wacaller]
type=friend
secret=qwerty
host=dynamic
context=watest
qualify=yes
allow=ulaw
allow=alaw

[wagateway]
type=friend
secret=qwerty
fromuser=wagateway
host=dynamic
context=watest
qualify=yes
allow=ulaw
allow=alaw

Now I try to call 123123123 from wacaller’s Grandstream phone.

243.112 CLI says:

[Jun 27 09:27:54] WARNING[20447][C-0000000b]: chan_sip.c:23213 handle_response_invite: Received response: "Forbidden" from '"WACaller" <sip:wagateway@192.168.243.112>;tag=as30b27eae'

Sip debug on 243.112:

<--- SIP read from UDP:192.168.242.235:5062 --->
INVITE sip:123123123@192.168.243.112 SIP/2.0
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;rport
From: "WACaller" <sip:wacaller@192.168.243.112>;tag=1014197566
To: <sip:123123123@192.168.243.112>
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 240 INVITE
Contact: "WACaller" <sip:wacaller@192.168.242.235:5062>
Max-Forwards: 70
User-Agent: Grandstream GXP1400 1.0.4.13
Privacy: none
P-Preferred-Identity: "WACaller" <sip:wacaller@192.168.243.112>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 412

v=0
o=wacaller 8000 8000 IN IP4 192.168.242.235
s=SIP Call
c=IN IP4 192.168.242.235
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 19 lines) ---
Sending to 192.168.242.235:5062 (no NAT)
Sending to 192.168.242.235:5062 (no NAT)
Using INVITE request as basis request - 298833112-5062-25@BJC.BGI.CEC.CDF
Found peer 'wacaller' for 'wacaller' from 192.168.242.235:5062

<--- Reliably Transmitting (no NAT) to 192.168.242.235:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;received=192.168.242.235;rport=5062
From: "WACaller" <sip:wacaller@192.168.243.112>;tag=1014197566
To: <sip:123123123@192.168.243.112>;tag=as5a3de236
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 240 INVITE
Server: Asterisk PBX SVN-trunk-r385782
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f84bef0"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '298833112-5062-25@BJC.BGI.CEC.CDF' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.242.235:5062 --->
ACK sip:123123123@192.168.243.112 SIP/2.0
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;rport
From: "WACaller" <sip:wacaller@192.168.243.112>;tag=1014197566
To: <sip:123123123@192.168.243.112>;tag=as5a3de236
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 240 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.242.235:5062 --->
INVITE sip:123123123@192.168.243.112 SIP/2.0
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK1881861609;rport
From: "WACaller" <sip:wacaller@192.168.243.112>;tag=1014197566
To: <sip:123123123@192.168.243.112>
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 241 INVITE
Contact: "WACaller" <sip:wacaller@192.168.242.235:5062>
Authorization: Digest username="wacaller", realm="asterisk", nonce="4f84bef0", uri="sip:123123123@192.168.243.112", response="53cdb5b8c1822c80870faab15a6dc6d2", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXP1400 1.0.4.13
Privacy: none
P-Preferred-Identity: "WACaller" <sip:wacaller@192.168.243.112>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 412

v=0
o=wacaller 8000 8000 IN IP4 192.168.242.235
s=SIP Call
c=IN IP4 192.168.242.235
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 19 lines) ---
Sending to 192.168.242.235:5062 (no NAT)
Using INVITE request as basis request - 298833112-5062-25@BJC.BGI.CEC.CDF
Found peer 'wacaller' for 'wacaller' from 192.168.242.235:5062
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 2
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 G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 97
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.242.235:5004
Looking for 123123123 in watest (domain 192.168.243.112)
list_route: route/path hop: <sip:wacaller@192.168.242.235:5062>

<--- Transmitting (no NAT) to 192.168.242.235:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK1881861609;received=192.168.242.235;rport=5062
From: "WACaller" <sip:wacaller@192.168.243.112>;tag=1014197566
To: <sip:123123123@192.168.243.112>
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 241 INVITE
Server: Asterisk PBX SVN-trunk-r385782
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:123123123@192.168.243.112:5060>
Content-Length: 0


<------------>
Audio is at 17372
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.241.98:5060:
INVITE sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;rport
Max-Forwards: 70
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as3f5f372a
To: <sip:s@192.168.241.98:5060>
Contact: <sip:wagateway@192.168.243.112:5060>
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Date: Wed, 26 Jun 2013 08:31:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 2059284449 2059284449 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 17372 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.241.98:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;received=192.168.243.112;rport=5060
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as3f5f372a
To: <sip:s@192.168.241.98:5060>;tag=as22eeeac0
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="603b4bbf"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 192.168.241.98:5060:
ACK sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;rport
Max-Forwards: 70
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as3f5f372a
To: <sip:s@192.168.241.98:5060>;tag=as22eeeac0
Contact: <sip:wagateway@192.168.243.112:5060>
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0


---
Audio is at 17372
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.241.98:5060:
INVITE sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;rport
Max-Forwards: 70
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as3f5f372a
To: <sip:s@192.168.241.98:5060>
Contact: <sip:wagateway@192.168.243.112:5060>
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Authorization: Digest username="s", realm="asterisk", algorithm=MD5, uri="sip:s@192.168.241.98:5060", nonce="603b4bbf", response="059cae207fb81fb76ea9061f71258895"
Date: Wed, 26 Jun 2013 08:31:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 2059284449 2059284450 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 17372 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.241.98:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;received=192.168.243.112;rport=5060
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as3f5f372a
To: <sip:s@192.168.241.98:5060>;tag=as22eeeac0
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.241.98:5060:
ACK sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;rport
Max-Forwards: 70
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as3f5f372a
To: <sip:s@192.168.241.98:5060>;tag=as22eeeac0
Contact: <sip:wagateway@192.168.243.112:5060>
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0


---
[Jun 26 16:31:48] WARNING[20447][C-0000000a]: chan_sip.c:23213 handle_response_invite: Received response: "Forbidden" from '"WACaller" <sip:wagateway@192.168.243.112>;tag=as3f5f372a'
Scheduling destruction of SIP dialog '758899861bee35980dadd87912ef805a@192.168.243.112:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '298833112-5062-25@BJC.BGI.CEC.CDF' in 6400 ms (Method: INVITE)

Sip debug on destination server:

<--- SIP read from UDP:192.168.243.112:5060 --->
INVITE sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;rport
Max-Forwards: 70
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as30b27eae
To: <sip:s@192.168.241.98:5060>
Contact: <sip:wagateway@192.168.243.112:5060>
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Date: Thu, 27 Jun 2013 01:27:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 1301894386 1301894386 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 15838 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 14 lines) ---
Sending to 192.168.243.112:5060 (NAT)
Using INVITE request as basis request - 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
Found peer 'wagateway' for 'wagateway' from 192.168.243.112:5060

<--- Reliably Transmitting (no NAT) to 192.168.243.112:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;received=192.168.243.112;rport=5060
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as30b27eae
To: <sip:s@192.168.241.98:5060>;tag=as671c0824
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0b63a660"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5dc37059030845ca3d974c513993876d@192.168.243.112:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.243.112:5060 --->
ACK sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;rport
Max-Forwards: 70
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as30b27eae
To: <sip:s@192.168.241.98:5060>;tag=as671c0824
Contact: <sip:wagateway@192.168.243.112:5060>
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.243.112:5060 --->
INVITE sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;rport
Max-Forwards: 70
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as30b27eae
To: <sip:s@192.168.241.98:5060>
Contact: <sip:wagateway@192.168.243.112:5060>
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Authorization: Digest username="s", realm="asterisk", algorithm=MD5, uri="sip:s@192.168.241.98:5060", nonce="0b63a660", response="537f37fe2fb8d0fd40733cb190ea70c8"
Date: Thu, 27 Jun 2013 01:27:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 1301894386 1301894387 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 15838 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 14 lines) ---
Sending to 192.168.243.112:5060 (no NAT)
Using INVITE request as basis request - 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
Found peer 'wagateway' for 'wagateway' from 192.168.243.112:5060

<--- Reliably Transmitting (no NAT) to 192.168.243.112:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;received=192.168.243.112;rport=5060
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as30b27eae
To: <sip:s@192.168.241.98:5060>;tag=as671c0824
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5dc37059030845ca3d974c513993876d@192.168.243.112:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.243.112:5060 --->
ACK sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;rport
Max-Forwards: 70
From: "WACaller" <sip:wagateway@192.168.243.112>;tag=as30b27eae
To: <sip:s@192.168.241.98:5060>;tag=as671c0824
Contact: <sip:wagateway@192.168.243.112:5060>
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
dev-ast*CLI> sip set debug off
SIP Debugging Disabled

Any help?

Откуда: 123123. Москва, Какоенибудьтам шоссе, д. 111.

Сообщений: 8

SIP/2.0 403 Forbidden

Не работает исходящий SIP звонок. Не знаю куда копать.

trixbox1*CLI> sip show peers

Name/username Host Dyn Nat ACL Port Status

internetcalls/skynet66 77.72.169.129 5060 Unmonitored

gts-sip/5620640 89.232.125.48 N 5060 Unmonitored

105 (Unspecified) D N A 5060 UNKNOWN

104 (Unspecified) D N A 5060 UNKNOWN

103/103 192.168.101.154 D N A 5061 OK (16 ms)

102/102 192.168.101.154 D N A 5060 OK (16 ms)

101/101 192.168.101.153 D N A 5072 OK (15 ms)

100/100 192.168.101.153 D N A 5071 OK (15 ms)

8 sip peers [Monitored: 4 online, 2 offline Unmonitored: 2 online, 0 offline]

trixbox1*CLI> sip show registry

Host Username Refresh State Reg.Time

89.xxx.125.48:5060 5620640 101 Registered Fri, 19 Mar 2010 12:10:51

1 SIP registrations.

Really destroying SIP dialog ’11ce763328da47402307a074021634c5@127.0.0.1′ Method: REGISTER

PEER Details:

host=89.232.125.48

insecure=port,invite

nat=yes

fromuser=5620640

username=5620640

secret=xxxxxxxx

type=peer

disallow=all

allow=ulaw&g711a&g711

а это кусочек лога с SIP/2.0 403 от SIP прокси:

— Executing [s@macro-dialout-trunk:18] GotoIf(«SIP/103-08d2bf20», «0?customtrunk») in new stack

— Executing [s@macro-dialout-trunk:19] Dial(«SIP/103-08d2bf20», «SIP/gts-sip/2909294,300,») in new stack

== Using SIP RTP TOS bits 184

== Using SIP RTP CoS mark 5

== Using SIP VRTP TOS bits 136

== Using SIP VRTP CoS mark 6

Audio is at 192.168.101.180 port 16400

Adding codec 0x4 (ulaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (NAT) to 89.xxx.125.48:5060:

INVITE sip:2909294@89.xxx.125.48 SIP/2.0

Via: SIP/2.0/UDP 192.168.101.180:5060;branch=z9hG4bK0c37c7a5;rport

Max-Forwards: 70

From: «103» <sip:5620640@192.168.101.180>;tag=as4d695e92

To: <sip:2909294@89.xxx.125.48>

Contact: <sip:5620640@192.168.101.180>

Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180

CSeq: 102 INVITE

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

ate: Fri, 19 Mar 2010 09:02:49 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 278

v=0

o=root 1153224551 1153224551 IN IP4 192.168.101.180

s=Asterisk PBX 1.6.0.9-samy-r27

c=IN IP4 192.168.101.180

t=0 0

m=audio 16400 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off — — — —

a=ptime:20

a=sendrecv



— Called gts-sip/2909294

trixbox1*CLI>

<— SIP read from UDP://89.xxx.125.48:5060 —>

SIP/2.0 100 Trying

From: «103»<sip:5620640@192.168.101.180>;tag=as4d695e92

To: <sip:2909294@89.xxx.125.48>

Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180

CSeq: 102 INVITE

Via: SIP/2.0/UDP 192.168.101.180:5060;received=78.138.144.127;rport=5060;branch=z9hG4bK0c37c7a5

Content-Length: 0

<————->

— (7 headers 0 lines) —

trixbox1*CLI>

<— SIP read from UDP://89.xxx.125.48:5060 —>

SIP/2.0 403 Forbidden

From: «103»<sip:5620640@192.168.101.180>;tag=as4d695e92

To: <sip:2909294@89.xxx.125.48>;tag=2083351771

Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180

CSeq: 102 INVITE

Via: SIP/2.0/UDP 192.168.101.180:5060;received=78.138.144.127;rport=5060;branch=z9hG4bK0c37c7a5

contact: <sip:2909294@tattele.com:5060;maddr=89.xxx.125.48>

supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,com.nortelnetworks.im.encryption

Content-Length: 0

<————->

— (9 headers 0 lines) —

Transmitting (NAT) to 89.xx.125.48:5060:

ACK sip:2909294@89.xxx.125.48 SIP/2.0

Via: SIP/2.0/UDP 192.168.101.180:5060;branch=z9hG4bK0c37c7a5;rport

Max-Forwards: 70

From: «103» <sip:5620640@192.168.101.180>;tag=as4d695e92

To: <sip:2909294@89.xxx.125.48>;tag=2083351771

Contact: <sip:5620640@192.168.101.180>

Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180

CSeq: 102 ACK

User-Agent: Asterisk PBX 1.6.0.9-samy-r27

Content-Length: 0



— SIP/gts-sip-b79a2530 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)

— Executing [s@macro-dialout-trunk:20] Goto(«SIP/103-08d2bf20», «s-CONGESTION,1») in new stack

— Goto (macro-dialout-trunk,s-CONGESTION,1)

— Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf(«SIP/103-08d2bf20», «1?noreport») in new stack

— Goto (macro-dialout-trunk,s-CONGESTION,3)

— Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp(«SIP/103-08d2bf20», «TRUNK Dial failed due to CONGESTION — failing through to other trunks») in new stack

— Executing [78432909294@from-internal:5] Macro(«SIP/103-08d2bf20», «outisbusy,») in new stack

— Executing [s@macro-outisbusy:1] Playback(«SIP/103-08d2bf20», «all-circuits-busy-now,noanswer») in new stack

— <SIP/103-08d2bf20> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)

Really destroying SIP dialog ‘75408280436ca6a31df57c0f30f3e8d6@192.168.101.180’ Method: INVITE

— Executing [s@macro-outisbusy:2] Playback(«SIP/103-08d2bf20», «pls-try-call-later,noanswer») in new stack

— <SIP/103-08d2bf20> Playing ‘pls-try-call-later.ulaw’ (language ‘en’)

— Executing [s@macro-outisbusy:3] Macro(«SIP/103-08d2bf20», «hangupcall») in new stack

— Executing [s@macro-hangupcall:1] ResetCDR(«SIP/103-08d2bf20», «vw») in new stack

— Executing [s@macro-hangupcall:2] NoCDR(«SIP/103-08d2bf20», «») in new stack

— Executing [s@macro-hangupcall:3] GotoIf(«SIP/103-08d2bf20», «1?skiprg») in new stack

— Goto (macro-hangupcall,s,6)

— Executing [s@macro-hangupcall:6] GotoIf(«SIP/103-08d2bf20», «1?skipblkvm») in new stack

— Goto (macro-hangupcall,s,9)

— Executing [s@macro-hangupcall:9] GotoIf(«SIP/103-08d2bf20», «1?theend») in new stack

— Goto (macro-hangupcall,s,11)

— Executing [s@macro-hangupcall:11] Hangup(«SIP/103-08d2bf20», «») in new stack

== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/103-08d2bf20’ in macro ‘hangupcall’

== Spawn extension (macro-outisbusy, s, 3) exited non-zero on ‘SIP/103-08d2bf20’ in macro ‘outisbusy’

== Spawn extension (from-internal, 78432909294, 5) exited non-zero on ‘SIP/103-08d2bf20’

— Executing [h@from-internal:1] Macro(«SIP/103-08d2bf20», «hangupcall») in new stack

— Executing [s@macro-hangupcall:1] ResetCDR(«SIP/103-08d2bf20», «vw») in new stack

— Executing [s@macro-hangupcall:2] NoCDR(«SIP/103-08d2bf20», «») in new stack

— Executing [s@macro-hangupcall:3] GotoIf(«SIP/103-08d2bf20», «1?skiprg») in new stack

— Goto (macro-hangupcall,s,6)

— Executing [s@macro-hangupcall:6] GotoIf(«SIP/103-08d2bf20», «1?skipblkvm») in new stack

— Goto (macro-hangupcall,s,9)

— Executing [s@macro-hangupcall:9] GotoIf(«SIP/103-08d2bf20», «1?theend») in new stack

— Goto (macro-hangupcall,s,11)

— Executing [s@macro-hangupcall:11] Hangup(«SIP/103-08d2bf20», «») in new stack

== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/103-08d2bf20’ in macro ‘hangupcall’

== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/103-08d2bf20’

== End MixMonitor Recording SIP/103-08d2bf20

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