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спросил 2011-11-28 00:48:09 +0400

ilyindn Gravatar ilyindn flag of Russian Federation

Panasnic tde600 и Asterisk

Panasonic KX-TDE600 соеденина с двумя Asterisk-aми(1.6 и 1.8)через V-SIPGW16. Звонки с Астеров на ТДЕ и обратно, по внутренним номерам, ходят без проблем; с астера через ТДЕ в город, тоже без проблем. Звонки из города приходят на ТДЕ потоками Е1, далее обрабатываются голосовой почтой TVM200, далее набираешь внутренний номер и звонит телефон, но если этот номер на АСТЕРИСКЕ, то при поднятии трубки телефон соединяется и тут же говорит "hangupcall" и далее BYE: кусок лога прилогается. Куда копать??? Клиетами у Астера шлюзы LinkSys b D-Link, и Ip телефоны Linksys. Пробывал оставлять толко один кодек,не помогло.

SIP/2223-0000004e answered SIP/utde-0000004d
    -- Executing [h@macro-dial:1] Macro("SIP/utde-0000004d", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/utde-0000004d", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/utde-0000004d", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/utde-0000004d", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/utde-0000004d", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/utde-0000004d' in macro 'hangupcall'
Scheduling destruction of SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Reliably Transmitting (NAT) to 10.0.7.187:5060:
BYE sip:2223@10.0.7.187:5060 SIP/2.0

Panasnic Panasonic tde600 и Asterisk

Panasonic KX-TDE600 соеденина с двумя Asterisk-aми(1.6 и 1.8)через V-SIPGW16. Звонки с Астеров на ТДЕ и обратно, по внутренним номерам, ходят без проблем; с астера через ТДЕ в город, тоже без проблем. Звонки из города приходят на ТДЕ потоками Е1, далее обрабатываются голосовой почтой TVM200, далее набираешь внутренний номер и звонит телефон, но если этот номер на АСТЕРИСКЕ, то при поднятии трубки телефон соединяется и тут же говорит "hangupcall" и далее BYE: кусок лога прилогается. Куда копать??? Клиетами у Астера шлюзы LinkSys b D-Link, и Ip телефоны Linksys. Пробывал оставлять толко один кодек,не помогло.

SIP/2223-0000004e answered SIP/utde-0000004d
    -- Executing [h@macro-dial:1] Macro("SIP/utde-0000004d", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/utde-0000004d", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/utde-0000004d", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/utde-0000004d", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/utde-0000004d", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/utde-0000004d' in macro 'hangupcall'
Scheduling destruction of SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Reliably Transmitting (NAT) to 10.0.7.187:5060:
BYE sip:2223@10.0.7.187:5060 SIP/2.0

Panasonic tde600 и Asterisk

Panasonic KX-TDE600 соеденина с двумя Asterisk-aми(1.6 и 1.8)через V-SIPGW16. Звонки с Астеров на ТДЕ и обратно, по внутренним номерам, ходят без проблем; с астера через ТДЕ в город, тоже без проблем. Звонки из города приходят на ТДЕ потоками Е1, далее обрабатываются голосовой почтой TVM200, далее набираешь внутренний номер и звонит телефон, но если этот номер на АСТЕРИСКЕ, то при поднятии трубки телефон соединяется и тут же говорит "hangupcall" и далее BYE: кусок лога прилогается. Куда копать??? Клиетами у Астера шлюзы LinkSys b D-Link, и Ip телефоны Linksys. Пробывал оставлять толко один кодек,не помогло.помогло. Вот SIP Debug:

localhost*CLI> sip set debug ip 10.0.7.187
SIP Debugging Enabled for IP: 10.0.7.187
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [2223@from-internal:1] Macro("SIP/utde-0000004d", "exten-vm,novm,2223") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/utde-0000004d", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/utde-0000004d", "AMPUSER=917YYYYYYY") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/utde-0000004d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/utde-0000004d", "1?Set(REALCALLERIDNUM=917YYYYYYY)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/utde-0000004d", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/utde-0000004d", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/utde-0000004d", "1?report") in new stack
    -- Goto (macro-user-callerid,s,9)
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/utde-0000004d", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/utde-0000004d", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/utde-0000004d", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] NoOp("SIP/utde-0000004d", "Using CallerID "" <917YYYYYYY>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/utde-0000004d", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/utde-0000004d", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/utde-0000004d", "EXTTOCALL=2223") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/utde-0000004d", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/utde-0000004d", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/utde-0000004d", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/utde-0000004d", "record-enable,2223,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/utde-0000004d", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/utde-0000004d", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/utde-0000004d", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,16)
    -- Executing [s@macro-record-enable:16] GotoIf("SIP/utde-0000004d", "1?IN") in new stack
    -- Goto (macro-record-enable,s,21)
    -- Executing [s@macro-record-enable:21] ExecIf("SIP/utde-0000004d", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/utde-0000004d", "dial,,tT,2223") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/utde-0000004d", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/utde-0000004d", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is 'unknown' number is '917YYYYYYY'
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 2223 to extension map
    -- dialparties.agi: Extension 2223 cf is disabled
    -- dialparties.agi: Extension 2223 do not disturb is disabled
 dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
    -- dialparties.agi: dbset CALLTRACE/2223 to 917YYYYYYY
    -- dialparties.agi: Filtered ARG3: 2223
    -- <SIP/utde-0000004d>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/utde-0000004d", "SIP/2223,,tT") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 10.0.0.151 port 19030
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.0.7.187:5060:
INVITE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:11:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 536398297 536398297 IN IP4 10.0.0.151
s=Asterisk PBX 1.6.2.20
c=IN IP4 10.0.0.151
t=0 0
m=audio 19030 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

---
    -- Called 2223

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 100 Trying
To: <sip:2223@10.0.7.187:5060>
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


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

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 180 Ringing
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/2223-0000004e is ringing

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Contact: 2223 <sip:2223@10.0.7.187:5060>
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 249
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 129266 129266 IN IP4 10.0.7.187
s=-
c=IN IP4 10.0.7.187
t=0 0
m=audio 16432 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.7.187:16432
list_route: hop: <sip:2223@10.0.7.187:5060>
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Transmitting (NAT) to 10.0.7.187:5060:
ACK sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2ab87a29;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0


---
    -- SIP/2223-0000004e answered SIP/utde-0000004d
    -- Executing [h@macro-dial:1] Macro("SIP/utde-0000004d", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/utde-0000004d", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/utde-0000004d", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/utde-0000004d", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/utde-0000004d", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/utde-0000004d' in macro 'hangupcall'
Scheduling destruction of SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Reliably Transmitting (NAT) to 10.0.7.187:5060:
BYE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.20
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/utde-0000004d' i
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/utde-0000004
  == Spawn extension (from-internal, 2223, 1) exited non-zero on 'SIP/utde-00000

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' Metho
Reliably Transmitting (NAT) to 10.0.7.187:5060:
OPTIONS sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:Unknown@10.0.0.151>
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:12:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=6c8d39bdb2e07d9fi0
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5734428373f8a3c920aefe186a032a9e@10.0.0.151' Method: OPTIONS

Panasonic tde600 и Asterisk

Panasonic KX-TDE600 соеденина с двумя Asterisk-aми(1.6 и 1.8)через V-SIPGW16. Звонки с Астеров на ТДЕ и обратно, по внутренним номерам, ходят без проблем; с астера через ТДЕ в город, тоже без проблем. Звонки из города приходят на ТДЕ потоками Е1, далее обрабатываются голосовой почтой TVM200, далее набираешь внутренний номер и звонит телефон, но если этот номер на АСТЕРИСКЕ, то при поднятии трубки телефон соединяется и тут же говорит "hangupcall" и далее BYE: кусок лога прилогается. Куда копать??? Клиетами у Астера шлюзы LinkSys b D-Link, и Ip телефоны Linksys. Пробывал оставлять толко один кодек,не помогло. помогло.NAT-а НЕТ всё в одной сети. Да и при чём тут NAT если звонки не проходят только из города, а с панаса на астер проходят. Вот SIP Debug:

localhost*CLI> sip set debug ip 10.0.7.187
SIP Debugging Enabled for IP: 10.0.7.187
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [2223@from-internal:1] Macro("SIP/utde-0000004d", "exten-vm,novm,2223") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/utde-0000004d", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/utde-0000004d", "AMPUSER=917YYYYYYY") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/utde-0000004d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/utde-0000004d", "1?Set(REALCALLERIDNUM=917YYYYYYY)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/utde-0000004d", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/utde-0000004d", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/utde-0000004d", "1?report") in new stack
    -- Goto (macro-user-callerid,s,9)
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/utde-0000004d", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/utde-0000004d", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/utde-0000004d", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] NoOp("SIP/utde-0000004d", "Using CallerID "" <917YYYYYYY>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/utde-0000004d", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/utde-0000004d", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/utde-0000004d", "EXTTOCALL=2223") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/utde-0000004d", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/utde-0000004d", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/utde-0000004d", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/utde-0000004d", "record-enable,2223,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/utde-0000004d", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/utde-0000004d", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/utde-0000004d", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,16)
    -- Executing [s@macro-record-enable:16] GotoIf("SIP/utde-0000004d", "1?IN") in new stack
    -- Goto (macro-record-enable,s,21)
    -- Executing [s@macro-record-enable:21] ExecIf("SIP/utde-0000004d", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/utde-0000004d", "dial,,tT,2223") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/utde-0000004d", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/utde-0000004d", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is 'unknown' number is '917YYYYYYY'
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 2223 to extension map
    -- dialparties.agi: Extension 2223 cf is disabled
    -- dialparties.agi: Extension 2223 do not disturb is disabled
 dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
    -- dialparties.agi: dbset CALLTRACE/2223 to 917YYYYYYY
    -- dialparties.agi: Filtered ARG3: 2223
    -- <SIP/utde-0000004d>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/utde-0000004d", "SIP/2223,,tT") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 10.0.0.151 port 19030
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.0.7.187:5060:
INVITE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:11:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 536398297 536398297 IN IP4 10.0.0.151
s=Asterisk PBX 1.6.2.20
c=IN IP4 10.0.0.151
t=0 0
m=audio 19030 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

---
    -- Called 2223

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 100 Trying
To: <sip:2223@10.0.7.187:5060>
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


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

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 180 Ringing
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/2223-0000004e is ringing

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Contact: 2223 <sip:2223@10.0.7.187:5060>
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 249
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 129266 129266 IN IP4 10.0.7.187
s=-
c=IN IP4 10.0.7.187
t=0 0
m=audio 16432 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.7.187:16432
list_route: hop: <sip:2223@10.0.7.187:5060>
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Transmitting (NAT) to 10.0.7.187:5060:
ACK sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2ab87a29;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0


---
    -- SIP/2223-0000004e answered SIP/utde-0000004d
    -- Executing [h@macro-dial:1] Macro("SIP/utde-0000004d", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/utde-0000004d", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/utde-0000004d", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/utde-0000004d", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/utde-0000004d", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/utde-0000004d' in macro 'hangupcall'
Scheduling destruction of SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Reliably Transmitting (NAT) to 10.0.7.187:5060:
BYE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.20
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/utde-0000004d' i
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/utde-0000004
  == Spawn extension (from-internal, 2223, 1) exited non-zero on 'SIP/utde-00000

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' Metho
Reliably Transmitting (NAT) to 10.0.7.187:5060:
OPTIONS sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:Unknown@10.0.0.151>
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:12:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=6c8d39bdb2e07d9fi0
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5734428373f8a3c920aefe186a032a9e@10.0.0.151' Method: OPTIONS

Panasonic tde600 и Asterisk

Panasonic KX-TDE600 соеденина с двумя Asterisk-aми(1.6 и 1.8)через V-SIPGW16. Звонки с Астеров на ТДЕ и обратно, по внутренним номерам, ходят без проблем; с астера через ТДЕ в город, тоже без проблем. Звонки из города приходят на ТДЕ потоками Е1, далее обрабатываются голосовой почтой TVM200, далее набираешь внутренний номер и звонит телефон, но если этот номер на АСТЕРИСКЕ, то при поднятии трубки телефон соединяется и тут же говорит "hangupcall" и далее BYE: кусок лога прилогается. Куда копать??? Клиетами у Астера шлюзы LinkSys b и D-Link, и Ip телефоны Linksys. Пробывал оставлять толко один кодек,не помогло.NAT-а НЕТ всё в одной сети. Да и при чём тут NAT если звонки не проходят только из города, а с панаса на астер проходят. Вот SIP Debug:

localhost*CLI> sip set debug ip 10.0.7.187
SIP Debugging Enabled for IP: 10.0.7.187
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [2223@from-internal:1] Macro("SIP/utde-0000004d", "exten-vm,novm,2223") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/utde-0000004d", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/utde-0000004d", "AMPUSER=917YYYYYYY") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/utde-0000004d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/utde-0000004d", "1?Set(REALCALLERIDNUM=917YYYYYYY)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/utde-0000004d", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/utde-0000004d", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/utde-0000004d", "1?report") in new stack
    -- Goto (macro-user-callerid,s,9)
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/utde-0000004d", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/utde-0000004d", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/utde-0000004d", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] NoOp("SIP/utde-0000004d", "Using CallerID "" <917YYYYYYY>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/utde-0000004d", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/utde-0000004d", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/utde-0000004d", "EXTTOCALL=2223") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/utde-0000004d", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/utde-0000004d", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/utde-0000004d", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/utde-0000004d", "record-enable,2223,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/utde-0000004d", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/utde-0000004d", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/utde-0000004d", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,16)
    -- Executing [s@macro-record-enable:16] GotoIf("SIP/utde-0000004d", "1?IN") in new stack
    -- Goto (macro-record-enable,s,21)
    -- Executing [s@macro-record-enable:21] ExecIf("SIP/utde-0000004d", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/utde-0000004d", "dial,,tT,2223") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/utde-0000004d", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/utde-0000004d", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is 'unknown' number is '917YYYYYYY'
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 2223 to extension map
    -- dialparties.agi: Extension 2223 cf is disabled
    -- dialparties.agi: Extension 2223 do not disturb is disabled
 dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
    -- dialparties.agi: dbset CALLTRACE/2223 to 917YYYYYYY
    -- dialparties.agi: Filtered ARG3: 2223
    -- <SIP/utde-0000004d>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/utde-0000004d", "SIP/2223,,tT") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 10.0.0.151 port 19030
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.0.7.187:5060:
INVITE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:11:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 536398297 536398297 IN IP4 10.0.0.151
s=Asterisk PBX 1.6.2.20
c=IN IP4 10.0.0.151
t=0 0
m=audio 19030 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

---
    -- Called 2223

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 100 Trying
To: <sip:2223@10.0.7.187:5060>
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


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

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 180 Ringing
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/2223-0000004e is ringing

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Contact: 2223 <sip:2223@10.0.7.187:5060>
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 249
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 129266 129266 IN IP4 10.0.7.187
s=-
c=IN IP4 10.0.7.187
t=0 0
m=audio 16432 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.7.187:16432
list_route: hop: <sip:2223@10.0.7.187:5060>
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Transmitting (NAT) to 10.0.7.187:5060:
ACK sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2ab87a29;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0


---
    -- SIP/2223-0000004e answered SIP/utde-0000004d
    -- Executing [h@macro-dial:1] Macro("SIP/utde-0000004d", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/utde-0000004d", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/utde-0000004d", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/utde-0000004d", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/utde-0000004d", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/utde-0000004d' in macro 'hangupcall'
Scheduling destruction of SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Reliably Transmitting (NAT) to 10.0.7.187:5060:
BYE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.20
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/utde-0000004d' i
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/utde-0000004
  == Spawn extension (from-internal, 2223, 1) exited non-zero on 'SIP/utde-00000

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' Metho
Reliably Transmitting (NAT) to 10.0.7.187:5060:
OPTIONS sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:Unknown@10.0.0.151>
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:12:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=6c8d39bdb2e07d9fi0
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5734428373f8a3c920aefe186a032a9e@10.0.0.151' Method: OPTIONS

Помогите плиз Panasonic tde600 и Asterisk

Panasonic KX-TDE600 соеденина с двумя Asterisk-aми(1.6 и 1.8)через V-SIPGW16. Звонки с Астеров на ТДЕ и обратно, по внутренним номерам, ходят без проблем; с астера через ТДЕ в город, тоже без проблем. Звонки из города приходят на ТДЕ потоками Е1, далее обрабатываются голосовой почтой TVM200, далее набираешь внутренний номер и звонит телефон, но если этот номер на АСТЕРИСКЕ, то при поднятии трубки телефон соединяется и тут же говорит "hangupcall" и далее BYE: кусок лога прилогается. Куда копать??? Клиетами у Астера шлюзы LinkSys и D-Link, и Ip телефоны Linksys. Пробывал оставлять толко один кодек,не помогло.NAT-а НЕТ всё в одной сети. Да и при чём тут NAT если звонки не проходят только из города, а с панаса на астер проходят. Вот SIP Debug:

localhost*CLI> sip set debug ip 10.0.7.187
SIP Debugging Enabled for IP: 10.0.7.187
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [2223@from-internal:1] Macro("SIP/utde-0000004d", "exten-vm,novm,2223") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/utde-0000004d", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/utde-0000004d", "AMPUSER=917YYYYYYY") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/utde-0000004d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/utde-0000004d", "1?Set(REALCALLERIDNUM=917YYYYYYY)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/utde-0000004d", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/utde-0000004d", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/utde-0000004d", "1?report") in new stack
    -- Goto (macro-user-callerid,s,9)
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/utde-0000004d", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/utde-0000004d", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/utde-0000004d", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] NoOp("SIP/utde-0000004d", "Using CallerID "" <917YYYYYYY>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/utde-0000004d", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/utde-0000004d", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/utde-0000004d", "EXTTOCALL=2223") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/utde-0000004d", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/utde-0000004d", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/utde-0000004d", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/utde-0000004d", "record-enable,2223,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/utde-0000004d", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/utde-0000004d", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/utde-0000004d", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,16)
    -- Executing [s@macro-record-enable:16] GotoIf("SIP/utde-0000004d", "1?IN") in new stack
    -- Goto (macro-record-enable,s,21)
    -- Executing [s@macro-record-enable:21] ExecIf("SIP/utde-0000004d", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/utde-0000004d", "dial,,tT,2223") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/utde-0000004d", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/utde-0000004d", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is 'unknown' number is '917YYYYYYY'
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 2223 to extension map
    -- dialparties.agi: Extension 2223 cf is disabled
    -- dialparties.agi: Extension 2223 do not disturb is disabled
 dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
    -- dialparties.agi: dbset CALLTRACE/2223 to 917YYYYYYY
    -- dialparties.agi: Filtered ARG3: 2223
    -- <SIP/utde-0000004d>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/utde-0000004d", "SIP/2223,,tT") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 10.0.0.151 port 19030
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.0.7.187:5060:
INVITE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:11:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 536398297 536398297 IN IP4 10.0.0.151
s=Asterisk PBX 1.6.2.20
c=IN IP4 10.0.0.151
t=0 0
m=audio 19030 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

---
    -- Called 2223

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 100 Trying
To: <sip:2223@10.0.7.187:5060>
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


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

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 180 Ringing
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/2223-0000004e is ringing

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Contact: 2223 <sip:2223@10.0.7.187:5060>
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 249
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 129266 129266 IN IP4 10.0.7.187
s=-
c=IN IP4 10.0.7.187
t=0 0
m=audio 16432 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.7.187:16432
list_route: hop: <sip:2223@10.0.7.187:5060>
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Transmitting (NAT) to 10.0.7.187:5060:
ACK sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2ab87a29;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0


---
    -- SIP/2223-0000004e answered SIP/utde-0000004d
    -- Executing [h@macro-dial:1] Macro("SIP/utde-0000004d", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/utde-0000004d", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/utde-0000004d", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/utde-0000004d", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/utde-0000004d", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/utde-0000004d' in macro 'hangupcall'
Scheduling destruction of SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Reliably Transmitting (NAT) to 10.0.7.187:5060:
BYE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.20
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/utde-0000004d' i
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/utde-0000004
  == Spawn extension (from-internal, 2223, 1) exited non-zero on 'SIP/utde-00000

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' Metho
Reliably Transmitting (NAT) to 10.0.7.187:5060:
OPTIONS sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:Unknown@10.0.0.151>
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:12:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=6c8d39bdb2e07d9fi0
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5734428373f8a3c920aefe186a032a9e@10.0.0.151' Method: OPTIONS

Помогите плиз Решено Panasonic tde600 и AsteriskAsterisk по V-SIPGW16

Panasonic KX-TDE600 соеденина с двумя Asterisk-aми(1.6 и 1.8)через V-SIPGW16. Звонки с Астеров на ТДЕ и обратно, по внутренним номерам, ходят без проблем; с астера через ТДЕ в город, тоже без проблем. Звонки из города приходят на ТДЕ потоками Е1, далее обрабатываются голосовой почтой TVM200, далее набираешь внутренний номер и звонит телефон, но если этот номер на АСТЕРИСКЕ, то при поднятии трубки телефон соединяется и тут же говорит "hangupcall" и далее BYE: кусок лога прилогается. Куда копать??? Клиетами у Астера шлюзы LinkSys и D-Link, и Ip телефоны Linksys. Пробывал оставлять толко один кодек,не помогло.NAT-а НЕТ всё в одной сети. Да и при чём тут NAT если звонки не проходят только из города, а с панаса на астер проходят. Вот SIP Debug:проходят.

localhost*CLI> sip set debug ip 10.0.7.187
SIP Debugging Enabled for IP: 10.0.7.187
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [2223@from-internal:1] Macro("SIP/utde-0000004d", "exten-vm,novm,2223") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/utde-0000004d", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/utde-0000004d", "AMPUSER=917YYYYYYY") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/utde-0000004d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/utde-0000004d", "1?Set(REALCALLERIDNUM=917YYYYYYY)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/utde-0000004d", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/utde-0000004d", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/utde-0000004d", "1?report") in new stack
    -- Goto (macro-user-callerid,s,9)
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/utde-0000004d", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/utde-0000004d", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/utde-0000004d", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] NoOp("SIP/utde-0000004d", "Using CallerID "" <917YYYYYYY>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/utde-0000004d", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/utde-0000004d", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/utde-0000004d", "EXTTOCALL=2223") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/utde-0000004d", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/utde-0000004d", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/utde-0000004d", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/utde-0000004d", "record-enable,2223,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/utde-0000004d", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/utde-0000004d", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/utde-0000004d", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,16)
    -- Executing [s@macro-record-enable:16] GotoIf("SIP/utde-0000004d", "1?IN") in new stack
    -- Goto (macro-record-enable,s,21)
    -- Executing [s@macro-record-enable:21] ExecIf("SIP/utde-0000004d", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/utde-0000004d", "dial,,tT,2223") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/utde-0000004d", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/utde-0000004d", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is 'unknown' number is '917YYYYYYY'
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 2223 to extension map
    -- dialparties.agi: Extension 2223 cf is disabled
    -- dialparties.agi: Extension 2223 do not disturb is disabled
 dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
    -- dialparties.agi: dbset CALLTRACE/2223 to 917YYYYYYY
    -- dialparties.agi: Filtered ARG3: 2223
    -- <SIP/utde-0000004d>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/utde-0000004d", "SIP/2223,,tT") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 10.0.0.151 port 19030
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.0.7.187:5060:
INVITE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:11:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 536398297 536398297 IN IP4 10.0.0.151
s=Asterisk PBX 1.6.2.20
c=IN IP4 10.0.0.151
t=0 0
m=audio 19030 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

---
    -- Called 2223

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 100 Trying
To: <sip:2223@10.0.7.187:5060>
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


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

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 180 Ringing
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/2223-0000004e is ringing

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK3c44de44
Contact: 2223 <sip:2223@10.0.7.187:5060>
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 2223 <sip:2223@10.0.0.151>;screen=yes;party=called
Content-Length: 249
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 129266 129266 IN IP4 10.0.7.187
s=-
c=IN IP4 10.0.7.187
t=0 0
m=audio 16432 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.7.187:16432
list_route: hop: <sip:2223@10.0.7.187:5060>
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Transmitting (NAT) to 10.0.7.187:5060:
ACK sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2ab87a29;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Contact: <sip:917YYYYYYY@10.0.0.151>
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.20
Content-Length: 0


---
    -- SIP/2223-0000004e answered SIP/utde-0000004d
    -- Executing [h@macro-dial:1] Macro("SIP/utde-0000004d", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/utde-0000004d", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/utde-0000004d", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/utde-0000004d", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/utde-0000004d", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/utde-0000004d' in macro 'hangupcall'
Scheduling destruction of SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:2223@10.0.7.187:5060> for address/port to send to
set_destination: set destination to 10.0.7.187, port 5060
Reliably Transmitting (NAT) to 10.0.7.187:5060:
BYE sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82;rport
Max-Forwards: 70
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.20
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/utde-0000004d' i
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/utde-0000004
  == Spawn extension (from-internal, 2223, 1) exited non-zero on 'SIP/utde-00000

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=b63e3380a7728df1i0
From: "917YYYYYYY" <sip:917YYYYYYY@10.0.0.151>;tag=as50827177
Call-ID: 1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK533ecb82
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1c8b4ef47feec1b466eb31cf083ed797@10.0.0.151' Metho
Reliably Transmitting (NAT) to 10.0.7.187:5060:
OPTIONS sip:2223@10.0.7.187:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
To: <sip:2223@10.0.7.187:5060>
Contact: <sip:Unknown@10.0.0.151>
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.20
Date: Fri, 18 Nov 2011 22:12:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.7.187:5060 --->
SIP/2.0 200 OK
To: <sip:2223@10.0.7.187:5060>;tag=6c8d39bdb2e07d9fi0
From: "Unknown" <sip:Unknown@10.0.0.151>;tag=as6bbdd321
Call-ID: 5734428373f8a3c920aefe186a032a9e@10.0.0.151
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.0.151:5060;branch=z9hG4bK2dd8abd6
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5734428373f8a3c920aefe186a032a9e@10.0.0.151' Method: OPTIONS

Решено Panasonic tde600 + TVM200 и Asterisk по V-SIPGW16

Panasonic KX-TDE600 соеденина с двумя Asterisk-aми(1.6 и 1.8)через V-SIPGW16. Звонки с Астеров на ТДЕ и обратно, по внутренним номерам, ходят без проблем; с астера через ТДЕ в город, тоже без проблем. Звонки из города приходят на ТДЕ потоками Е1, далее обрабатываются голосовой почтой TVM200, далее набираешь внутренний номер и звонит телефон, но если этот номер на АСТЕРИСКЕ, то при поднятии трубки телефон соединяется и тут же говорит "hangupcall" и далее BYE: кусок лога прилогается. Куда копать??? Клиетами у Астера шлюзы LinkSys и D-Link, и Ip телефоны Linksys. Пробывал оставлять толко один кодек,не помогло.NAT-а НЕТ всё в одной сети. Да и при чём тут NAT если звонки не проходят только из города, а с панаса на астер проходят.

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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.