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ответил 2011-03-05 21:38:10 +0400

pilchard Gravatar pilchard

 
<------------->
--- (13 headers 0 lines) ---
    -- SIP/xxx.com-00000180 is ringing

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK5a60541f;received=195.90.168.
130;rport=5060
From: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
To: <sip:74951234567@sip.xxx.com>;tag=as215800d1
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
CSeq: 103 INVITE
Server: Xxx Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:74951234567@78.46.xxx.xxx>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 1172285262 1172285262 IN IP4 78.46.xxx.xxx
s=Asterisk PBX 1.6.2.13
c=IN IP4 78.46.xxx.xxx
t=0 0
m=audio 10806 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone
-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 78.46.xxx.xxx:10806
    -- SIP/xxx.com-00000180 is making progress passing it to SIP/240-0000017
f

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK5a60541f;received=195.90.168.
130;rport=5060
From: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
To: <sip:74951234567@sip.xxx.com>;tag=as215800d1
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
CSeq: 103 INVITE
Server: Xxx Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:74951234567@78.46.xxx.xxx>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 1172285262 1172285263 IN IP4 78.46.xxx.xxx
s=Asterisk PBX 1.6.2.13
c=IN IP4 78.46.xxx.xxx
t=0 0
m=audio 10806 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone
-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 78.46.xxx.xxx:10806
list_route: hop: <sip:74951234567@78.46.xxx.xxx>
set_destination: Parsing <sip:74951234567@78.46.xxx.xxx> for address/port to send
 to
set_destination: set destination to 78.46.xxx.xxx, port 5060
Transmitting (NAT) to 78.46.xxx.xxx:5060:
ACK sip:74951234567@78.46.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK4db756aa;rport
Max-Forwards: 70
From: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
To: <sip:74951234567@sip.xxx.com>;tag=as215800d1
Contact: <sip:26193@78.46.xxx.xxx>
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
CSeq: 103 ACK
User-Agent: FPBX-2.8.1(1.6.2.13)
Content-Length: 0


---
    -- SIP/xxx.com-00000180 answered SIP/240-0000017f
    -- Sending DTMF '2240' to the called party.
Reliably Transmitting (NAT) to 78.46.xxx.xxx:5060:
OPTIONS sip:sip.xxx.com SIP/2.0
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK59197096;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@78.46.xxx.xxx>;tag=as64386f91
To: <sip:sip.xxx.com>
Contact: <sip:Unknown@78.46.xxx.xxx>
Call-ID: 7edbee51265a285d58bfc71956081c04@78.46.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.6.2.13)
Date: Sat, 05 Mar 2011 18:37:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK59197096;received=195.90.168.
130;rport=5060
From: "Unknown" <sip:Unknown@78.46.xxx.xxx>;tag=as64386f91
To: <sip:sip.xxx.com>;tag=as76ce4def
Call-ID: 7edbee51265a285d58bfc71956081c04@78.46.xxx.xxx
CSeq: 102 OPTIONS
Server: Xxx Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '7edbee51265a285d58bfc71956081c04@78.46.xxx.xxx' M
ethod: OPTIONS
Really destroying SIP dialog '6219e2d966610dbc19eea1d26bb46baf@127.0.0.1' Method
: REGISTER

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
BYE sip:26193@78.46.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK79b46316;rport
Max-Forwards: 70
From: <sip:74951234567@sip.xxx.com>;tag=as215800d1
To: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
CSeq: 102 BYE
User-Agent: Xxx Voip
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 78.46.xxx.xxx : 5060 (NAT)

<--- Transmitting (NAT) to 78.46.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK79b46316;received=78.46.xxx.xxx;
rport=5060
From: <sip:74951234567@sip.xxx.com>;tag=as215800d1
To: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
CSeq: 102 BYE
Server: FPBX-2.8.1(1.6.2.13)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Executing [h@from-internal:1] Macro("SIP/240-0000017f", "hangupcall") in
new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/240-0000017f", "1?noautomon"
) in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/240-0000017f", "TOUCH_MONITOR_
OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/240-0000017f", "1?skiprg") i
n new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/240-0000017f", "1?skipblkvm"
) in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/240-0000017f", "1?theend")
in new stack
  == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/240-00000
17f' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/240-0000017f'
  == Spawn extension (from-internal, 7240, 1) exited non-zero on 'SIP/240-000001
7f'
    -- Executing [h@from-internal:1] Macro("SIP/240-0000017f", "hangupcall") in
new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/240-0000017f", "1?noautomon"
) in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/240-0000017f", "TOUCH_MONITOR_
OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/240-0000017f", "1?skiprg") i
n new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/240-0000017f", "1?skipblkvm"
) in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/240-0000017f", "1?theend")
in new stack
  == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/240-00000
17f' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/240-0000017f'
  == Extension Changed 240[ext-local] new state Idle for Notify User 240
Really destroying SIP dialog '5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com'
Method: BYE
ippbx*CLI> sip set debug off

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 78.46.xxx.xxx:5060:
ACK sip:74951234567@sip.xxx.com SIP/2.0
Via: SIP/2.0/UDP 195.90.xxx.xxx:5060;branch=z9hG4bK3d66a260;rport
Max-Forwards: 70
From: "device" <sip:26193@sip.xxx.com>;tag=as0a5ac802
To: <sip:74951234567@sip.xxx.com>;tag=as7332e815
Contact: <sip:26193@195.90.xxx.xxx>
Call-ID: 1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.6.2.13)
Content-Length: 0


---
Audio is at 195.90.xxx.xxx port 14400
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 78.46.xxx.xxx:5060:
INVITE sip:74951234567@sip.xxx.com SIP/2.0
Via: SIP/2.0/UDP 195.90.xxx.xxx:5060;branch=z9hG4bK4331ae0c;rport
Max-Forwards: 70
From: "device" <sip:26193@sip.xxx.com>;tag=as0a5ac802
To: <sip:74951234567@sip.xxx.com>
Contact: <sip:26193@195.90.xxx.xxx>
Call-ID: 1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com
CSeq: 103 INVITE
User-Agent: FPBX-2.8.1(1.6.2.13)
Authorization: Digest username="26193", realm="sip.xxx.com", algorithm=MD5, uri="sip:74951234567@sip.xxx.com", nonce="7df1454c", response="8e4345dffdc1942b525e467b04d
1741b"
Date: Sat, 05 Mar 2011 18:50:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1264701459 1264701460 IN IP4 195.90.xxx.xxx
s=Asterisk PBX 1.6.2.13
c=IN IP4 195.90.xxx.xxx
t=0 0
m=audio 14400 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.90.xxx.xxx:5060;branch=z9hG4bK4331ae0c;received=195.90.xxx.xxx;rport=5060
From: "device" <sip:26193@sip.xxx.com>;tag=as0a5ac802
To: <sip:74951234567@sip.xxx.com>
Call-ID: 1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com
CSeq: 103 INVITE
Server: Xxx Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:74951234567@78.46.xxx.xxx>
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
    -- SIP/xxx.com-00000180 is ringing

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 195.90.xxx.xxx:5060;branch=z9hG4bK4331ae0c;received=195.90.xxx.xxx;rport=5060
From: "device" <sip:26193@sip.xxx.com>;tag=as0a5ac802
To: <sip:74951234567@sip.xxx.com>;tag=as0b489ab6
Call-ID: 1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com
CSeq: 103 INVITE
Server: Xxx Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:74951234567@78.46.xxx.xxx>
Content-Length: 0


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

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK5a60541f;received=195.90.168.
130;rport=5060
195.90.xxx.xxx:5060;branch=z9hG4bK4331ae0c;received=195.90.xxx.xxx;rport=5060
From: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
<sip:26193@sip.xxx.com>;tag=as0a5ac802
To: <sip:74951234567@sip.xxx.com>;tag=as215800d1
<sip:74951234567@sip.xxx.com>;tag=as0b489ab6
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com
CSeq: 103 INVITE
Server: Xxx Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:74951234567@78.46.xxx.xxx>
Content-Type: application/sdp
Content-Length: 288
286

v=0
o=root 1172285262 1172285262 401752981 401752981 IN IP4 78.46.xxx.xxx
s=Asterisk PBX 1.6.2.13
c=IN IP4 78.46.xxx.xxx
t=0 0
m=audio 10806 10400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone
-event), (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 78.46.xxx.xxx:10806
    -- SIP/xxx.com-00000180 78.46.xxx.xxx:10400
    -- SIP/zadorma.com-00000184 is ringing
    -- SIP/zadorma.com-00000184 is making progress passing it to SIP/240-0000017
f
SIP/240-00000183

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK5a60541f;received=195.90.168.
130;rport=5060
195.90.xxx.xxx:5060;branch=z9hG4bK4331ae0c;received=195.90.xxx.xxx;rport=5060
From: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
<sip:26193@sip.xxx.com>;tag=as0a5ac802
To: <sip:74951234567@sip.xxx.com>;tag=as215800d1
<sip:74951234567@sip.xxx.com>;tag=as0b489ab6
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com
CSeq: 103 INVITE
Server: Xxx Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:74951234567@78.46.xxx.xxx>
Content-Type: application/sdp
Content-Length: 288
286

v=0
o=root 1172285262 1172285263 401752981 401752982 IN IP4 78.46.xxx.xxx
s=Asterisk PBX 1.6.2.13
c=IN IP4 78.46.xxx.xxx
t=0 0
m=audio 10806 10400 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone
-event), (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 78.46.xxx.xxx:10806
78.46.xxx.xxx:10400
list_route: hop: <sip:74951234567@78.46.xxx.xxx>
set_destination: Parsing <sip:74951234567@78.46.xxx.xxx> for address/port to send
 send to
set_destination: set destination to 78.46.xxx.xxx, port 5060
Transmitting (NAT) to 78.46.xxx.xxx:5060:
ACK sip:74951234567@78.46.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK4db756aa;rport
195.90.xxx.xxx:5060;branch=z9hG4bK6b34a4a9;rport
Max-Forwards: 70
From: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
<sip:26193@sip.xxx.com>;tag=as0a5ac802
To: <sip:74951234567@sip.xxx.com>;tag=as215800d1
<sip:74951234567@sip.xxx.com>;tag=as0b489ab6
Contact: <sip:26193@78.46.xxx.xxx>
<sip:26193@195.90.xxx.xxx>
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com
CSeq: 103 ACK
User-Agent: FPBX-2.8.1(1.6.2.13)
Content-Length: 0


---
    -- SIP/xxx.com-00000180 SIP/zadorma.com-00000184 answered SIP/240-0000017f
SIP/240-00000183
    -- Sending DTMF '2240' to the called party.
Reliably Transmitting (NAT) to 78.46.xxx.xxx:5060:
OPTIONS sip:sip.xxx.com SIP/2.0
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK59197096;rport
195.90.xxx.xxx:5060;branch=z9hG4bK4ec38dd7;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@78.46.xxx.xxx>;tag=as64386f91
<sip:Unknown@195.90.xxx.xxx>;tag=as51881757
To: <sip:sip.xxx.com>
Contact: <sip:Unknown@78.46.xxx.xxx>
<sip:Unknown@195.90.xxx.xxx>
Call-ID: 7edbee51265a285d58bfc71956081c04@78.46.xxx.xxx
0ee141141b0362fd6956dbab37c6dba8@195.90.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.6.2.13)
Date: Sat, 05 Mar 2011 18:37:24 18:50:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK59197096;received=195.90.168.
130;rport=5060
195.90.xxx.xxx:5060;branch=z9hG4bK4ec38dd7;received=195.90.xxx.xxx;rport=5060
From: "Unknown" <sip:Unknown@78.46.xxx.xxx>;tag=as64386f91
<sip:Unknown@195.90.xxx.xxx>;tag=as51881757
To: <sip:sip.xxx.com>;tag=as76ce4def
<sip:sip.xxx.com>;tag=as507b44f1
Call-ID: 7edbee51265a285d58bfc71956081c04@78.46.xxx.xxx
0ee141141b0362fd6956dbab37c6dba8@195.90.xxx.xxx
CSeq: 102 OPTIONS
Server: Xxx Voip
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '7edbee51265a285d58bfc71956081c04@78.46.xxx.xxx' M
ethod: '0ee141141b0362fd6956dbab37c6dba8@195.90.xxx.xxx' Method: OPTIONS
Really destroying SIP dialog '6219e2d966610dbc19eea1d26bb46baf@127.0.0.1' Method
: REGISTER

<--- SIP read from UDP:78.46.xxx.xxx:5060 --->
BYE sip:26193@78.46.xxx.xxx sip:26193@195.90.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK79b46316;rport
78.46.xxx.xxx:5060;branch=z9hG4bK4770ec70;rport
Max-Forwards: 70
From: <sip:74951234567@sip.xxx.com>;tag=as215800d1
<sip:74951234567@sip.xxx.com>;tag=as0b489ab6
To: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
<sip:26193@sip.xxx.com>;tag=as0a5ac802
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com
CSeq: 102 BYE
User-Agent: Xxx Voip
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 78.46.xxx.xxx : 5060 (NAT)

<--- Transmitting (NAT) to 78.46.xxx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 78.46.xxx.xxx:5060;branch=z9hG4bK79b46316;received=78.46.xxx.xxx;
rport=5060
78.46.xxx.xxx:5060;branch=z9hG4bK4770ec70;received=78.46.xxx.xxx;rport=5060
From: <sip:74951234567@sip.xxx.com>;tag=as215800d1
<sip:74951234567@sip.xxx.com>;tag=as0b489ab6
To: "device" <sip:26193@sip.xxx.com>;tag=as0f196f1f
<sip:26193@sip.xxx.com>;tag=as0a5ac802
Call-ID: 5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com
1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com
CSeq: 102 BYE
Server: FPBX-2.8.1(1.6.2.13)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Executing [h@from-internal:1] Macro("SIP/240-0000017f", Macro("SIP/240-00000183", "hangupcall") in
in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/240-0000017f", "1?noautomon"
) GotoIf("SIP/240-00000183", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/240-0000017f", "TOUCH_MONITOR_
OUTPUT=") NoOp("SIP/240-00000183", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/240-0000017f", GotoIf("SIP/240-00000183", "1?skiprg") i
n in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/240-0000017f", "1?skipblkvm"
) GotoIf("SIP/240-00000183", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/240-0000017f", "1?theend")
GotoIf("SIP/240-00000183", "1?theend") in new stack
  == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/240-00000
17f' 'SIP/240-00000183' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/240-0000017f'
'SIP/240-00000183'
  == Spawn extension (from-internal, 7240, 1) exited non-zero on 'SIP/240-000001
7f'
'SIP/240-00000183'
    -- Executing [h@from-internal:1] Macro("SIP/240-0000017f", Macro("SIP/240-00000183", "hangupcall") in
in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/240-0000017f", "1?noautomon"
) GotoIf("SIP/240-00000183", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/240-0000017f", "TOUCH_MONITOR_
OUTPUT=") NoOp("SIP/240-00000183", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/240-0000017f", GotoIf("SIP/240-00000183", "1?skiprg") i
n in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/240-0000017f", "1?skipblkvm"
) GotoIf("SIP/240-00000183", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/240-0000017f", "1?theend")
GotoIf("SIP/240-00000183", "1?theend") in new stack
  == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/240-00000
17f' 'SIP/240-00000183' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/240-0000017f'
'SIP/240-00000183'
  == Extension Changed 240[ext-local] new state Idle for Notify User 240
Really destroying SIP dialog '5137b7af42a39e345cf57f50247e2d0d@sip.xxx.com'
'1a3a5ca03a7a46f474f4553b117ee584@sip.xxx.com' Method: BYE
ippbx*CLI> sip set debug off
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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.