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ответил 2011-11-16 17:41:57 +0400

DJs3000 Gravatar DJs3000 flag of Russian Federation

http://retrogamesbattle.c...

TCP дамп при звонке

16:34:15.571930 IP (tos 0x60, ttl 63, id 60064, offset 0, flags [none], proto UDP (17), length 861)
    213.247.249.2.5060 > 86.110.4.148.5060: SIP, length: 833
        INVITE sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK3c0008bf;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@213.247.249.2>;tag=as6a3e84d6
        To: <sip:89250347252@86.110.4.148>
        Contact: <sip:HellKlUsov6@213.247.249.2>
        Call-ID: 0511b43a7c4756430a386a62139d3dc9@213.247.249.2
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX 1.6.2.13
        Date: Wed, 16 Nov 2011 13:34:57 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 260

        v=0
        o=root 481902525 481902525 IN IP4 213.247.249.2
        s=Asterisk PBX 1.6.2.13
        c=IN IP4 213.247.249.2
        t=0 0
        m=audio 13908 RTP/AVP 18 101
        a=rtpmap:18 G729/8000
        a=fmtp:18 annexb=no
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv

16:34:15.580670 IP (tos 0xb8, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 60)
    213.247.249.2.65238 > 86.110.4.148.23260: UDP, length 32
16:34:15.581297 IP (tos 0x60, ttl 63, id 60065, offset 0, flags [none], proto UDP (17), length 436)
    213.247.249.2.5060 > 86.110.4.148.5060: SIP, length: 408
        ACK sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK3c0008bf;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@213.247.249.2>;tag=as6a3e84d6
        To: <sip:89250347252@86.110.4.148>;tag=11eeb31d
        Contact: <sip:HellKlUsov6@213.247.249.2>
        Call-ID: 0511b43a7c4756430a386a62139d3dc9@213.247.249.2
        CSeq: 102 ACK
        User-Agent: Asterisk PBX 1.6.2.13
        Content-Length: 0


16:34:15.600782 IP (tos 0xb8, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 60)
    213.247.249.2.65238 > 86.110.4.148.23260: UDP, length 32

TCP дамп при звонке

16:34:15.571930 IP (tos 0x60, ttl 63, id 60064, offset 0, flags [none], proto UDP (17), length 861)
    213.247.249.2.5060 213.247.249.x.5060 > 86.110.4.148.5060: SIP, length: 833
        INVITE sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK3c0008bf;rport
213.247.249.x:5060;branch=z9hG4bK3c0008bf;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@213.247.249.2>;tag=as6a3e84d6
<sip:HellKlUsov6@213.247.249.x>;tag=as6a3e84d6
        To: <sip:89250347252@86.110.4.148>
        Contact: <sip:HellKlUsov6@213.247.249.2>
<sip:HellKlUsov6@213.247.249.x>
        Call-ID: 0511b43a7c4756430a386a62139d3dc9@213.247.249.2
0511b43a7c4756430a386a62139d3dc9@213.247.249.x
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX 1.6.2.13
        Date: Wed, 16 Nov 2011 13:34:57 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 260

        v=0
        o=root 481902525 481902525 IN IP4 213.247.249.2
213.247.249.x
        s=Asterisk PBX 1.6.2.13
        c=IN IP4 213.247.249.2
213.247.249.x
        t=0 0
        m=audio 13908 RTP/AVP 18 101
        a=rtpmap:18 G729/8000
        a=fmtp:18 annexb=no
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv

16:34:15.580670 IP (tos 0xb8, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 60)
    213.247.249.2.65238 213.247.249.x.65238 > 86.110.4.148.23260: UDP, length 32
16:34:15.581297 IP (tos 0x60, ttl 63, id 60065, offset 0, flags [none], proto UDP (17), length 436)
    213.247.249.2.5060 213.247.249.x.5060 > 86.110.4.148.5060: SIP, length: 408
        ACK sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK3c0008bf;rport
213.247.249.x:5060;branch=z9hG4bK3c0008bf;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@213.247.249.2>;tag=as6a3e84d6
<sip:HellKlUsov6@213.247.249.x>;tag=as6a3e84d6
        To: <sip:89250347252@86.110.4.148>;tag=11eeb31d
        Contact: <sip:HellKlUsov6@213.247.249.2>
<sip:HellKlUsov6@213.247.249.x>
        Call-ID: 0511b43a7c4756430a386a62139d3dc9@213.247.249.2
0511b43a7c4756430a386a62139d3dc9@213.247.249.x
        CSeq: 102 ACK
        User-Agent: Asterisk PBX 1.6.2.13
        Content-Length: 0


16:34:15.600782 IP (tos 0xb8, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 60)
    213.247.249.2.65238 213.247.249.x.65238 > 86.110.4.148.23260: UDP, length 32

TCP дамп при звонке

16:34:15.571930 IP (tos 0x60, ttl 63, id 60064, offset 0, flags [none], proto UDP (17), length 861)
    213.247.249.x.5060 > 86.110.4.148.5060: SIP, length: 833
        INVITE sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.x:5060;branch=z9hG4bK3c0008bf;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@213.247.249.x>;tag=as6a3e84d6
        To: <sip:89250347252@86.110.4.148>
        Contact: <sip:HellKlUsov6@213.247.249.x>
        Call-ID: 0511b43a7c4756430a386a62139d3dc9@213.247.249.x
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX 1.6.2.13
        Date: Wed, 16 Nov 2011 13:34:57 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 260

        v=0
        o=root 481902525 481902525 IN IP4 213.247.249.x
        s=Asterisk PBX 1.6.2.13
        c=IN IP4 213.247.249.x
        t=0 0
        m=audio 13908 RTP/AVP 18 101
        a=rtpmap:18 G729/8000
        a=fmtp:18 annexb=no
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv

16:34:15.580670 IP (tos 0xb8, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 60)
    213.247.249.x.65238 > 86.110.4.148.23260: UDP, length 32
16:34:15.581297 IP (tos 0x60, ttl 63, id 60065, offset 0, flags [none], proto UDP (17), length 436)
    213.247.249.x.5060 > 86.110.4.148.5060: SIP, length: 408
        ACK sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.x:5060;branch=z9hG4bK3c0008bf;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@213.247.249.x>;tag=as6a3e84d6
        To: <sip:89250347252@86.110.4.148>;tag=11eeb31d
        Contact: <sip:HellKlUsov6@213.247.249.x>
        Call-ID: 0511b43a7c4756430a386a62139d3dc9@213.247.249.x
        CSeq: 102 ACK
        User-Agent: Asterisk PBX 1.6.2.13
        Content-Length: 0


16:34:15.600782 IP (tos 0xb8, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 60)
    213.247.249.x.65238 > 86.110.4.148.23260: UDP, length 32

так же выкладываю сип дебагу этого же звонка:

<--- SIP read from UDP:192.168.9.230:5060 --->
INVITE sip:89250347252@192.168.9.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKda29fad8ea3b3cdd
From: <sip:306@192.168.9.50>;tag=7dcdfc5bc6f28f89
To: <sip:89250347252@192.168.9.50>
Contact: <sip:306@192.168.9.230:5060;transport=udp>
Supported: replaces, timer, path
P-Early-Media: Supported
Call-ID: 2ffacbc5661f42fe@192.168.9.230
CSeq: 14822 INVITE
User-Agent: Grandstream GXP280 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 349

v=0
o=306 8000 8000 IN IP4 192.168.9.230
s=SIP Call
c=IN IP4 192.168.9.230
t=0 0
m=audio 5092 RTP/AVP 0 8 4 18 2 97 9 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20

<------------->
--- (14 headers 17 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Sending to 192.168.9.230 : 5060 (NAT)
Using INVITE request as basis request - 2ffacbc5661f42fe@192.168.9.230
Found peer '306' for '306' from 192.168.9.230:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 3
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 G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format GSM for ID 3
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x50e (gsm|ulaw|alaw|g729|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.9.230:5092
Looking for 89250347252 in from-internal (domain 192.168.9.50)
list_route: hop: <sip:306@192.168.9.230:5060;transport=udp>

<--- Transmitting (NAT) to 192.168.9.230:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKda29fad8ea3b3cdd;received=192.168.9.230
From: <sip:306@192.168.9.50>;tag=7dcdfc5bc6f28f89
To: <sip:89250347252@192.168.9.50>
Call-ID: 2ffacbc5661f42fe@192.168.9.230
CSeq: 14822 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89250347252@192.168.9.50>
Content-Length: 0


<------------>
    -- Executing [89250347252@from-internal:1] Dial("SIP/306-000000b9", "Sip/arctel/89250347252") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Audio is at 213.247.249.2 port 17866
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 86.110.4.148:5060:
INVITE sip:89250347252@86.110.4.148 SIP/2.0
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK0cad963a;rport
Max-Forwards: 70
From: "device" <sip:HellKlUsov6@213.247.249.2>;tag=as0d4115c1
To: <sip:89250347252@86.110.4.148>
Contact: <sip:HellKlUsov6@213.247.249.2>
Call-ID: 5c3e81c42f16630d35e41d32300afce3@213.247.249.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Wed, 16 Nov 2011 13:47:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 521641283 521641283 IN IP4 213.247.249.2
s=Asterisk PBX 1.6.2.13
c=IN IP4 213.247.249.2
t=0 0
m=audio 17866 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called arctel/89250347252

<--- SIP read from UDP:86.110.4.148:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK0cad963a;rport=5060
Call-ID: 5c3e81c42f16630d35e41d32300afce3@213.247.249.2
From: "device"<sip:HellKlUsov6@213.247.249.2>;tag=as0d4115c1
To: <sip:89250347252@86.110.4.148>
CSeq: 102 INVITE
Content-Length: 0


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

<--- SIP read from UDP:86.110.4.148:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK0cad963a;rport=5060
Call-ID: 5c3e81c42f16630d35e41d32300afce3@213.247.249.2
From: "device"<sip:HellKlUsov6@213.247.249.2>;tag=as0d4115c1
To: <sip:89250347252@86.110.4.148>;tag=1bdefcac
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from 86.110.4.148
Transmitting (NAT) to 86.110.4.148:5060:
ACK sip:89250347252@86.110.4.148 SIP/2.0
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK0cad963a;rport
Max-Forwards: 70
From: "device" <sip:HellKlUsov6@213.247.249.2>;tag=as0d4115c1
To: <sip:89250347252@86.110.4.148>;tag=1bdefcac
Contact: <sip:HellKlUsov6@213.247.249.2>
Call-ID: 5c3e81c42f16630d35e41d32300afce3@213.247.249.2
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0


---
    -- SIP/arctel-000000ba is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [89250347252@from-internal:2] Hangup("SIP/306-000000b9", "") in new stack
  == Spawn extension (from-internal, 89250347252, 2) exited non-zero on 'SIP/306-000000b9'
    -- Executing [h@from-internal:1] Macro("SIP/306-000000b9", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/306-000000b9", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/306-000000b9", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/306-000000b9", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/306-000000b9", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/306-000000b9", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,12)
    -- Executing [s@macro-hangupcall:12] Hangup("SIP/306-000000b9", "") in new stack
  == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/306-000000b9' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/306-000000b9'
Scheduling destruction of SIP dialog '2ffacbc5661f42fe@192.168.9.230' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.9.230:5060 --->
SIP/2.0 500 Server internal failure
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKda29fad8ea3b3cdd;received=192.168.9.230
From: <sip:306@192.168.9.50>;tag=7dcdfc5bc6f28f89
To: <sip:89250347252@192.168.9.50>;tag=as6b4c86f8
Call-ID: 2ffacbc5661f42fe@192.168.9.230
CSeq: 14822 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.9.230:5060 --->
ACK sip:89250347252@192.168.9.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKda29fad8ea3b3cdd
From: <sip:306@192.168.9.50>;tag=7dcdfc5bc6f28f89
To: <sip:89250347252@192.168.9.50>;tag=as6b4c86f8
Contact: <sip:306@192.168.9.230:5060;transport=udp>
Supported: path
Call-ID: 2ffacbc5661f42fe@192.168.9.230
CSeq: 14822 ACK
User-Agent: Grandstream GXP280 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '5c3e81c42f16630d35e41d32300afce3@213.247.249.2' Method: INVITE

TCP дамп при звонке

16:34:15.571930   tcpdump: listening on ext0, link-type EN10MB (Ethernet), capture size 1500 bytes
16:54:36.879097 IP (tos 0x60, ttl 63, id 60064, 60091, offset 0, flags [none], proto UDP (17), length 861)
    213.247.249.x.5060 859)
    213.247.249.2.5060 > 86.110.4.148.5060: SIP, length: 833
831
        INVITE sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.x:5060;branch=z9hG4bK3c0008bf;rport
213.247.249.2:5060;branch=z9hG4bK3bf73b59;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@213.247.249.x>;tag=as6a3e84d6
<sip:HellKlUsov6@86.110.4.148>;tag=as6fefe118
        To: <sip:89250347252@86.110.4.148>
        Contact: <sip:HellKlUsov6@213.247.249.x>
<sip:HellKlUsov6@213.247.249.2>
        Call-ID: 0511b43a7c4756430a386a62139d3dc9@213.247.249.x
17f67b0527b2c8085cda1e7c3dc50c78@86.110.4.148
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX 1.6.2.13
        Date: Wed, 16 Nov 2011 13:34:57 13:55:18 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 260

        v=0
        o=root 481902525 481902525 898618919 898618919 IN IP4 213.247.249.x
213.247.249.2
        s=Asterisk PBX 1.6.2.13
        c=IN IP4 213.247.249.x
213.247.249.2
        t=0 0
        m=audio 13908 19816 RTP/AVP 18 101
        a=rtpmap:18 G729/8000
        a=fmtp:18 annexb=no
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv

16:34:15.580670 IP (tos 0xb8, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 60)
    213.247.249.x.65238 > 86.110.4.148.23260: UDP, length 32
16:34:15.581297 16:54:36.900329 IP (tos 0x60, ttl 63, id 60065, 60092, offset 0, flags [none], proto UDP (17), length 436)
    213.247.249.x.5060 434)
    213.247.249.2.5060 > 86.110.4.148.5060: SIP, length: 408
406
        ACK sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.x:5060;branch=z9hG4bK3c0008bf;rport
213.247.249.2:5060;branch=z9hG4bK3bf73b59;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@213.247.249.x>;tag=as6a3e84d6
<sip:HellKlUsov6@86.110.4.148>;tag=as6fefe118
        To: <sip:89250347252@86.110.4.148>;tag=11eeb31d
<sip:89250347252@86.110.4.148>;tag=79513e35
        Contact: <sip:HellKlUsov6@213.247.249.x>
<sip:HellKlUsov6@213.247.249.2>
        Call-ID: 0511b43a7c4756430a386a62139d3dc9@213.247.249.x
17f67b0527b2c8085cda1e7c3dc50c78@86.110.4.148
        CSeq: 102 ACK
        User-Agent: Asterisk PBX 1.6.2.13
        Content-Length: 0


16:34:15.600782 IP (tos 0xb8, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 60)
    213.247.249.x.65238 > 86.110.4.148.23260: UDP, length 32

так же выкладываю сип дебагу этого же звонка:

Сип дебага:
 
<--- SIP read from UDP:192.168.9.230:5060 --->
INVITE sip:89250347252@192.168.9.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKda29fad8ea3b3cdd
192.168.9.230:5060;branch=z9hG4bKc45d37a171d0c5f4
From: <sip:306@192.168.9.50>;tag=7dcdfc5bc6f28f89
<sip:306@192.168.9.50>;tag=b08b41cb32d5cd2c
To: <sip:89250347252@192.168.9.50>
Contact: <sip:306@192.168.9.230:5060;transport=udp>
Supported: replaces, timer, path
P-Early-Media: Supported
Call-ID: 2ffacbc5661f42fe@192.168.9.230
627d6895aa6f0f1a@192.168.9.230
CSeq: 14822 31552 INVITE
User-Agent: Grandstream GXP280 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 349

349
 
v=0
o=306 8000 8000 IN IP4 192.168.9.230
s=SIP Call
c=IN IP4 192.168.9.230
t=0 0
m=audio 5092 5010 RTP/AVP 0 8 4 18 2 97 9 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20

a=ptime:20
 
<------------->
--- (14 headers 17 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Sending to 192.168.9.230 : 5060 (NAT)
Using INVITE request as basis request - 2ffacbc5661f42fe@192.168.9.230
627d6895aa6f0f1a@192.168.9.230
Found peer '306' for '306' from 192.168.9.230:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 3
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 G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format GSM for ID 3
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x50e (gsm|ulaw|alaw|g729|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.9.230:5092
192.168.9.230:5010
Looking for 89250347252 in from-internal (domain 192.168.9.50)
list_route: hop: <sip:306@192.168.9.230:5060;transport=udp>

<sip:306@192.168.9.230:5060;transport=udp>
 
<--- Transmitting (NAT) to 192.168.9.230:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKda29fad8ea3b3cdd;received=192.168.9.230
192.168.9.230:5060;branch=z9hG4bKc45d37a171d0c5f4;received=192.168.9.230
From: <sip:306@192.168.9.50>;tag=7dcdfc5bc6f28f89
<sip:306@192.168.9.50>;tag=b08b41cb32d5cd2c
To: <sip:89250347252@192.168.9.50>
Call-ID: 2ffacbc5661f42fe@192.168.9.230
627d6895aa6f0f1a@192.168.9.230
CSeq: 14822 31552 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89250347252@192.168.9.50>
Content-Length: 0


0
 
<------------>
    -- Executing [89250347252@from-internal:1] Dial("SIP/306-000000b9", Dial("SIP/306-000000c6", "Sip/arctel/89250347252") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Audio is at 213.247.249.2 port 17866
17548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 86.110.4.148:5060:
INVITE sip:89250347252@86.110.4.148 SIP/2.0
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK0cad963a;rport
213.247.249.2:5060;branch=z9hG4bK20083305;rport
Max-Forwards: 70
From: "device" <sip:HellKlUsov6@213.247.249.2>;tag=as0d4115c1
<sip:hellklusov6@86.110.4.148>;tag=as719c8192
To: <sip:89250347252@86.110.4.148>
Contact: <sip:HellKlUsov6@213.247.249.2>
<sip:hellklusov6@213.247.249.2>
Call-ID: 5c3e81c42f16630d35e41d32300afce3@213.247.249.2
3d0e464d6474410d78ce607a34f8dac9@86.110.4.148
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Wed, 16 Nov 2011 13:47:54 13:58:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

262
 
v=0
o=root 521641283 521641283 1911200188 1911200188 IN IP4 213.247.249.2
s=Asterisk PBX 1.6.2.13
c=IN IP4 213.247.249.2
t=0 0
m=audio 17866 17548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    a=sendrecv
 
 
-- Called arctel/89250347252
 
 
<--- SIP read from UDP:86.110.4.148:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK0cad963a;rport=5060
213.247.249.2:5060;branch=z9hG4bK20083305;rport=5060
Call-ID: 5c3e81c42f16630d35e41d32300afce3@213.247.249.2
3d0e464d6474410d78ce607a34f8dac9@86.110.4.148
From: "device"<sip:HellKlUsov6@213.247.249.2>;tag=as0d4115c1
"device"<sip:hellklusov6@86.110.4.148>;tag=as719c8192
To: <sip:89250347252@86.110.4.148>
CSeq: 102 INVITE
Content-Length: 0


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

---
 
<--- SIP read from UDP:86.110.4.148:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK0cad963a;rport=5060
213.247.249.2:5060;branch=z9hG4bK20083305;rport=5060
Call-ID: 5c3e81c42f16630d35e41d32300afce3@213.247.249.2
3d0e464d6474410d78ce607a34f8dac9@86.110.4.148
From: "device"<sip:HellKlUsov6@213.247.249.2>;tag=as0d4115c1
"device"<sip:hellklusov6@86.110.4.148>;tag=as719c8192
To: <sip:89250347252@86.110.4.148>;tag=1bdefcac
<sip:89250347252@86.110.4.148>;tag=4eb000d2
CSeq: 102 INVITE
Content-Length: 0


0
 
<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from 86.110.4.148
Transmitting (NAT) to 86.110.4.148:5060:
ACK sip:89250347252@86.110.4.148 SIP/2.0
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK0cad963a;rport
213.247.249.2:5060;branch=z9hG4bK20083305;rport
Max-Forwards: 70
From: "device" <sip:HellKlUsov6@213.247.249.2>;tag=as0d4115c1
<sip:hellklusov6@86.110.4.148>;tag=as719c8192
To: <sip:89250347252@86.110.4.148>;tag=1bdefcac
<sip:89250347252@86.110.4.148>;tag=4eb000d2
Contact: <sip:HellKlUsov6@213.247.249.2>
<sip:hellklusov6@213.247.249.2>
Call-ID: 5c3e81c42f16630d35e41d32300afce3@213.247.249.2
3d0e464d6474410d78ce607a34f8dac9@86.110.4.148
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0


---
    -- SIP/arctel-000000ba 0
 
 
-- SIP/arctel-000000c7 is circuit-busy
 
 
== Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [89250347252@from-internal:2] Hangup("SIP/306-000000b9", Hangup("SIP/306-000000c6", "") in new stack
  == Spawn extension (from-internal, 89250347252, 2) exited non-zero on 'SIP/306-000000b9'
'SIP/306-000000c6'
    -- Executing [h@from-internal:1] Macro("SIP/306-000000b9", Macro("SIP/306-000000c6", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/306-000000b9", GotoIf("SIP/306-000000c6", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/306-000000b9", "TOUCH_MONITOR_OUTPUT=") NoOp("SIP/306-000000c6", "TOUCHMONITOROUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/306-000000b9", GotoIf("SIP/306-000000c6", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/306-000000b9", GotoIf("SIP/306-000000c6", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/306-000000b9", GotoIf("SIP/306-000000c6", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,12)
    -- Executing [s@macro-hangupcall:12] Hangup("SIP/306-000000b9", Hangup("SIP/306-000000c6", "") in new stack
  == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/306-000000b9' 'SIP/306-000000c6' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/306-000000b9'
'SIP/306-000000c6'
Scheduling destruction of SIP dialog '2ffacbc5661f42fe@192.168.9.230' '627d6895aa6f0f1a@192.168.9.230' in 6400 ms (Method: INVITE)

INVITE)
 
<--- Reliably Transmitting (NAT) to 192.168.9.230:5060 --->
SIP/2.0 500 Server internal failure
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKda29fad8ea3b3cdd;received=192.168.9.230
192.168.9.230:5060;branch=z9hG4bKc45d37a171d0c5f4;received=192.168.9.230
From: <sip:306@192.168.9.50>;tag=7dcdfc5bc6f28f89
<sip:306@192.168.9.50>;tag=b08b41cb32d5cd2c
To: <sip:89250347252@192.168.9.50>;tag=as6b4c86f8
<sip:89250347252@192.168.9.50>;tag=as01ccbeef
Call-ID: 2ffacbc5661f42fe@192.168.9.230
627d6895aa6f0f1a@192.168.9.230
CSeq: 14822 31552 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>

0
 
<------------>
 
<--- SIP read from UDP:192.168.9.230:5060 --->
ACK sip:89250347252@192.168.9.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKda29fad8ea3b3cdd
192.168.9.230:5060;branch=z9hG4bKc45d37a171d0c5f4
From: <sip:306@192.168.9.50>;tag=7dcdfc5bc6f28f89
<sip:306@192.168.9.50>;tag=b08b41cb32d5cd2c
To: <sip:89250347252@192.168.9.50>;tag=as6b4c86f8
<sip:89250347252@192.168.9.50>;tag=as01ccbeef
Contact: <sip:306@192.168.9.230:5060;transport=udp>
Supported: path
Call-ID: 2ffacbc5661f42fe@192.168.9.230
627d6895aa6f0f1a@192.168.9.230
CSeq: 14822 31552 ACK
User-Agent: Grandstream GXP280 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


0
 
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '5c3e81c42f16630d35e41d32300afce3@213.247.249.2' '3d0e464d6474410d78ce607a34f8dac9@86.110.4.148' Method: INVITE
INVITE

TCP дамп при звонке

  tcpdump: listening on ext0, link-type EN10MB (Ethernet), capture size 1500 bytes
16:54:36.879097 IP (tos 0x60, ttl 63, id 60091, offset 0, flags [none], proto UDP (17), length 859)
    213.247.249.2.5060 > 86.110.4.148.5060: SIP, length: 831
        INVITE sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK3bf73b59;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@86.110.4.148>;tag=as6fefe118
        To: <sip:89250347252@86.110.4.148>
        Contact: <sip:HellKlUsov6@213.247.249.2>
        Call-ID: 17f67b0527b2c8085cda1e7c3dc50c78@86.110.4.148
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX 1.6.2.13
        Date: Wed, 16 Nov 2011 13:55:18 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 260

        v=0
        o=root 898618919 898618919 IN IP4 213.247.249.2
        s=Asterisk PBX 1.6.2.13
        c=IN IP4 213.247.249.2
        t=0 0
        m=audio 19816 RTP/AVP 18 101
        a=rtpmap:18 G729/8000
        a=fmtp:18 annexb=no
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv

16:54:36.900329 IP (tos 0x60, ttl 63, id 60092, offset 0, flags [none], proto UDP (17), length 434)
    213.247.249.2.5060 > 86.110.4.148.5060: SIP, length: 406
        ACK sip:89250347252@86.110.4.148 SIP/2.0
        Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK3bf73b59;rport
        Max-Forwards: 70
        From: "device" <sip:HellKlUsov6@86.110.4.148>;tag=as6fefe118
        To: <sip:89250347252@86.110.4.148>;tag=79513e35
        Contact: <sip:HellKlUsov6@213.247.249.2>
        Call-ID: 17f67b0527b2c8085cda1e7c3dc50c78@86.110.4.148
        CSeq: 102 ACK
        User-Agent: Asterisk PBX 1.6.2.13
        Content-Length: 0

Сип дебага:

<--- SIP read from UDP:192.168.9.230:5060 --->
INVITE sip:89250347252@192.168.9.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKc45d37a171d0c5f4
From: <sip:306@192.168.9.50>;tag=b08b41cb32d5cd2c
To: <sip:89250347252@192.168.9.50>
Contact: <sip:306@192.168.9.230:5060;transport=udp>
Supported: replaces, timer, path
P-Early-Media: Supported
Call-ID: 627d6895aa6f0f1a@192.168.9.230
CSeq: 31552 INVITE
User-Agent: Grandstream GXP280 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 349
 
349

v=0
o=306 8000 8000 IN IP4 192.168.9.230
s=SIP Call
c=IN IP4 192.168.9.230
t=0 0
m=audio 5010 RTP/AVP 0 8 4 18 2 97 9 3
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=ptime:20
 
a=ptime:20

<------------->
--- (14 headers 17 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Sending to 192.168.9.230 : 5060 (NAT)
Using INVITE request as basis request - 627d6895aa6f0f1a@192.168.9.230
Found peer '306' for '306' from 192.168.9.230:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 3
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 G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format GSM for ID 3
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x50e (gsm|ulaw|alaw|g729|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.9.230:5010
Looking for 89250347252 in from-internal (domain 192.168.9.50)
list_route: hop: <sip:306@192.168.9.230:5060;transport=udp>
 
<sip:306@192.168.9.230:5060;transport=udp>

<--- Transmitting (NAT) to 192.168.9.230:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKc45d37a171d0c5f4;received=192.168.9.230
From: <sip:306@192.168.9.50>;tag=b08b41cb32d5cd2c
To: <sip:89250347252@192.168.9.50>
Call-ID: 627d6895aa6f0f1a@192.168.9.230
CSeq: 31552 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89250347252@192.168.9.50>
Content-Length: 0
 
0


<------------>
    -- Executing [89250347252@from-internal:1] Dial("SIP/306-000000c6", "Sip/arctel/89250347252") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Audio is at 213.247.249.2 port 17548
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 86.110.4.148:5060:
INVITE sip:89250347252@86.110.4.148 SIP/2.0
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK20083305;rport
Max-Forwards: 70
From: "device" <sip:hellklusov6@86.110.4.148>;tag=as719c8192
<sip:HellKlUsov6@86.110.4.148>;tag=as719c8192
To: <sip:89250347252@86.110.4.148>
Contact: <sip:hellklusov6@213.247.249.2>
<sip:HellKlUsov6@213.247.249.2>
Call-ID: 3d0e464d6474410d78ce607a34f8dac9@86.110.4.148
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Wed, 16 Nov 2011 13:58:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262
 
262

v=0
o=root 1911200188 1911200188 IN IP4 213.247.249.2
s=Asterisk PBX 1.6.2.13
c=IN IP4 213.247.249.2
t=0 0
m=audio 17548 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
 
 
a=sendrecv

---
    -- Called arctel/89250347252

 
 <--- SIP read from UDP:86.110.4.148:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK20083305;rport=5060
Call-ID: 3d0e464d6474410d78ce607a34f8dac9@86.110.4.148
From: "device"<sip:hellklusov6@86.110.4.148>;tag=as719c8192
"device"<sip:HellKlUsov6@86.110.4.148>;tag=as719c8192
To: <sip:89250347252@86.110.4.148>
CSeq: 102 INVITE
Content-Length: 0
 
0


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

<--- SIP read from UDP:86.110.4.148:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK20083305;rport=5060
Call-ID: 3d0e464d6474410d78ce607a34f8dac9@86.110.4.148
From: "device"<sip:hellklusov6@86.110.4.148>;tag=as719c8192
"device"<sip:HellKlUsov6@86.110.4.148>;tag=as719c8192
To: <sip:89250347252@86.110.4.148>;tag=4eb000d2
CSeq: 102 INVITE
Content-Length: 0
 
0


<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 500 "Server Internal Error" back from 86.110.4.148
Transmitting (NAT) to 86.110.4.148:5060:
ACK sip:89250347252@86.110.4.148 SIP/2.0
Via: SIP/2.0/UDP 213.247.249.2:5060;branch=z9hG4bK20083305;rport
Max-Forwards: 70
From: "device" <sip:hellklusov6@86.110.4.148>;tag=as719c8192
<sip:HellKlUsov6@86.110.4.148>;tag=as719c8192
To: <sip:89250347252@86.110.4.148>;tag=4eb000d2
Contact: <sip:hellklusov6@213.247.249.2>
<sip:HellKlUsov6@213.247.249.2>
Call-ID: 3d0e464d6474410d78ce607a34f8dac9@86.110.4.148
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0
 
 
0


---
    -- SIP/arctel-000000c7 is circuit-busy

 
 == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [89250347252@from-internal:2] Hangup("SIP/306-000000c6", "") in new stack
  == Spawn extension (from-internal, 89250347252, 2) exited non-zero on 'SIP/306-000000c6'
    -- Executing [h@from-internal:1] Macro("SIP/306-000000c6", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/306-000000c6", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/306-000000c6", "TOUCHMONITOROUTPUT=") "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/306-000000c6", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/306-000000c6", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/306-000000c6", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,12)
    -- Executing [s@macro-hangupcall:12] Hangup("SIP/306-000000c6", "") in new stack
  == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/306-000000c6' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/306-000000c6'
Scheduling destruction of SIP dialog '627d6895aa6f0f1a@192.168.9.230' in 6400 ms (Method: INVITE)
 
INVITE)

<--- Reliably Transmitting (NAT) to 192.168.9.230:5060 --->
SIP/2.0 500 Server internal failure
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKc45d37a171d0c5f4;received=192.168.9.230
From: <sip:306@192.168.9.50>;tag=b08b41cb32d5cd2c
To: <sip:89250347252@192.168.9.50>;tag=as01ccbeef
Call-ID: 627d6895aa6f0f1a@192.168.9.230
CSeq: 31552 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
 
<------------>
 
0


<------------>

<--- SIP read from UDP:192.168.9.230:5060 --->
ACK sip:89250347252@192.168.9.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230:5060;branch=z9hG4bKc45d37a171d0c5f4
From: <sip:306@192.168.9.50>;tag=b08b41cb32d5cd2c
To: <sip:89250347252@192.168.9.50>;tag=as01ccbeef
Contact: <sip:306@192.168.9.230:5060;transport=udp>
Supported: path
Call-ID: 627d6895aa6f0f1a@192.168.9.230
CSeq: 31552 ACK
User-Agent: Grandstream GXP280 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
 
0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '3d0e464d6474410d78ce607a34f8dac9@86.110.4.148' Method: INVITE
 INVITE
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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.