403 Forbidden после первого звонка по SIP транку

[bahek2462774]

Добрый день.
помогите =) Проходит в SIP транк один звонок ( причем неважно входящий или исходящий ) и потом все , вываливается **403 forbidden**
sip провайдер - "Укртелеком"

вот **sip.conf**

Код: Выделить всё

   [general]
    bindport=5060 
    bindaddr=0.0.0.0  
    allowguest=no
    dtmf=rfc2833 
    context=default
    ;externip=195.88.113.188
    localnet=10.10.10.0/255.255.255.0
    register => 380892506511:XXXXXXX@sip.ukrtel.net/380892506511
    
    [ukrtelecom]
    type=friend
    secret=XXXXXXX
    username=380892506511
    fromuser=380892506511
    defaultuser=380892506511
    callerid="ukrtel" <380892506511>
    host=sip.ukrtel.net
    fromdomain=sip.ukrtel.net
    nat=yes
    canreinvite=no
    qualify=yes
    context=sip_all_in
    disallow=all
    allow=alaw
    allow=ulaw
    insecure=invite,port
    
    [office](!)
    type=friend
    context=phones
    secret=XXXXXXX
    disallow=all
    allow=ulaw
    allow=alaw
    ; ##### nat=yes
    ; ##### qualify=yes
    ; ##### canreinvite=no
    ; ##### sipreinvite=no
    host=dynamic
    
    [100](office)
    [101](office)
    host=192.168.0.233

а вот **extensions.conf**

Код: Выделить всё

    [globals]
    
    [general]
    autofallthrough=yes
    context=default
    allowguest=no
    bindport=5060
    bindaddr=0.0.0.0
    
    [default]
    exten => s,1,Answer()
    exten => s,n,Playback(hello-world)
    exten => s,n,Hangup()
    
    [sip_all_in]
    exten => _38.,1,Verbose(########## call from the SIP_ALL_IN ###########)
    exten => _38.,n,Dial(SIP/100&SIP/101)
    
    [sip_all_out]
    exten => _9.,1,Verbose(########## call from the SIP_ALL_OUT ###########)
    exten => _9.,n,Dial(SIP/ukrtelecom/${EXTEN:1},120)
    
    
    ; ################################### INSIDE ##########################
    [internal]
    exten => 100, 1, Verbose(1\Extension 1000)
    exten => 100, n, Dial(SIP/1000,30)
    exten => 100, n, Hangup()
    exten => 101, 1, Verbose(1\Extension 1001)
    exten => 101, n, Dial(SIP/1001,30)
    exten => 101, n, Hangup()
    
    
    ; ######### test ########3
    
    exten => 500,1,Verbose(########### HELLO - WORDL ######### )
    exten => 500,n,background(hello-world)
    
    [phones]
    include => sip_all_in
    include => internal
    include => sip_all_out

вот дамп когда **звонок не проходит**

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

-- Executing [90800506800@phones:1] Verbose("SIP/101-00000026", "########## call from the SIP_ALL_OUT ###########") in new stack
########## call from the SIP_ALL_OUT ###########
-- Executing [90800506800@phones:2] Dial("SIP/101-00000026", "SIP/ukrtelecom/0800506800,120") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.10.10.110 port 13754
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 195.5.0.83:5060:
INVITE sip:0800506800@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK4281921c;rport
Max-Forwards: 70
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as286e18f2
To: <sip:0800506800@sip.ukrtel.net>
Contact: <sip:380892506511@10.10.10.110>
Call-ID: 4536d4150392dd0648af6a86289234a9@sip.ukrtel.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Date: Mon, 30 Jul 2012 12:04:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 1986983298 1986983298 IN IP4 10.10.10.110
s=Asterisk PBX 1.6.2.9-2+squeeze6
c=IN IP4 10.10.10.110
t=0 0
m=audio 13754 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 ukrtelecom/0800506800

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK4281921c;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as286e18f2
To: <sip:0800506800@195.5.0.83>
Call-ID: 4536d4150392dd0648af6a86289234a9@sip.ukrtel.net
CSeq: 102 INVITE


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

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK4281921c;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as286e18f2
To: <sip:0800506800@195.5.0.83>;tag=aprqngfrt-0omsq010000c6
Call-ID: 4536d4150392dd0648af6a86289234a9@sip.ukrtel.net
CSeq: 102 INVITE


<------------->
--- (6 headers 0 lines) ---
Transmitting (NAT) to 195.5.0.83:5060:
ACK sip:0800506800@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK4281921c;rport
Max-Forwards: 70
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as286e18f2
To: <sip:0800506800@sip.ukrtel.net>;tag=aprqngfrt-0omsq010000c6
Contact: <sip:380892506511@10.10.10.110>
Call-ID: 4536d4150392dd0648af6a86289234a9@sip.ukrtel.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Content-Length: 0


---
[Jul 30 15:04:07] WARNING[1767]: chan_sip.c:17994 handle_response_invite: Received response: "Forbidden" from '"101" <sip:380892506511@sip.ukrtel.net>;tag=as286e18f2'
-- SIP/ukrtelecom-00000027 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/101-00000026' status is 'CONGESTION'
Really destroying SIP dialog '4536d4150392dd0648af6a86289234a9@sip.ukrtel.net' Method: INVITE
energotourserver*CLI>

вот дамп когда **звонок проходит**

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK43d6f2c9;rport=60638
From: <sip:380892506511@sip.ukrtel.net>;tag=as4b83d39a
To: <sip:380892506511@195.5.0.83>;tag=aprqcvrsqu2-p2r94110000u6
Call-ID: 7fae610b155ea1e37031b9577cd780e1@127.0.1.1
CSeq: 111 REGISTER
Contact: <sip:380892506511@195.5.0.83>;expires=30


<------------->
--- (7 headers 0 lines) ---
Scheduling destruction of SIP dialog '7fae610b155ea1e37031b9577cd780e1@127.0.1.1' in 32000 ms (Method: REGISTER)
[Jul 30 15:05:31] NOTICE[1767]: chan_sip.c:18399 handle_response_register: Outbound Registration: Expiry for sip.ukrtel.net is 120 sec (Scheduling reregistration in 105 s)
== Using SIP RTP CoS mark 5
-- Executing [90800506800@phones:1] Verbose("SIP/101-00000028", "########## call from the SIP_ALL_OUT ###########") in new stack
########## call from the SIP_ALL_OUT ###########
-- Executing [90800506800@phones:2] Dial("SIP/101-00000028", "SIP/ukrtelecom/0800506800,120") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.10.10.110 port 15890
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 195.5.0.83:5060:
INVITE sip:0800506800@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK671ecf1e;rport
Max-Forwards: 70
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>
Contact: <sip:380892506511@10.10.10.110>
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Date: Mon, 30 Jul 2012 12:05:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 1856866922 1856866922 IN IP4 10.10.10.110
s=Asterisk PBX 1.6.2.9-2+squeeze6
c=IN IP4 10.10.10.110
t=0 0
m=audio 15890 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 ukrtelecom/0800506800

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK671ecf1e;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 102 INVITE


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

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 407 Proxy authentication required
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK671ecf1e;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=896FD33D68B66E113617ECF08AC04D4013436499839011779
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 102 INVITE
Content-Length: 0
Proxy-Authenticate: Digest realm="sip.ukrtel.net",domain="sip.ukrtel.net",nonce="MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==",algorithm=MD5
Organization: Ukrtelecom


<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 195.5.0.83:5060:
ACK sip:0800506800@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK671ecf1e;rport
Max-Forwards: 70
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=896FD33D68B66E113617ECF08AC04D4013436499839011779
Contact: <sip:380892506511@10.10.10.110>
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Content-Length: 0


---
Audio is at 10.10.10.110 port 15890
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 195.5.0.83:5060:
INVITE sip:0800506800@sip.ukrtel.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK69060425;rport
Max-Forwards: 70
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>
Contact: <sip:380892506511@10.10.10.110>
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Proxy-Authorization: Digest username="380892506511", realm="sip.ukrtel.net", algorithm=MD5, uri="sip.ukrtel.net", nonce="MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==", response="ba9ce3cbc3d573ec0fb047860c6b1091"
Date: Mon, 30 Jul 2012 12:05:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 1856866922 1856866923 IN IP4 10.10.10.110
s=Asterisk PBX 1.6.2.9-2+squeeze6
c=IN IP4 10.10.10.110
t=0 0
m=audio 15890 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK69060425;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 103 INVITE


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

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK69060425;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 103 INVITE
Contact: <sip:0800506800@195.5.0.83:5060;transport=udp>
Allow: INVITE
Allow: ACK
Allow: PRACK
Allow: SUBSCRIBE
Allow: BYE
Allow: CANCEL
Allow: NOTIFY
Allow: INFO
Allow: REFER
Allow: UPDATE
Content-Type: application/sdp
Content-Length: 151

v=0
o=- 1 2 IN IP4 195.5.0.83
s=-
c=IN IP4 195.5.0.83
t=0 0
m=audio 20854 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (19 headers 8 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
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 195.5.0.83:20854
-- SIP/ukrtelecom-00000029 is ringing
-- SIP/ukrtelecom-00000029 is making progress passing it to SIP/101-00000028

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK69060425;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 103 INVITE
Contact: "Main800506800" <sip:0800506800@195.5.0.83:5060;transport=udp>
Allow: INVITE
Allow: ACK
Allow: PRACK
Allow: SUBSCRIBE
Allow: BYE
Allow: CANCEL
Allow: NOTIFY
Allow: INFO
Allow: REFER
Allow: UPDATE
P-Asserted-Identity: "Main800506800" <sip:380892222600@corp.ukrtelecom.loc:5061>
Content-Type: application/sdp
Content-Length: 151

v=0
o=- 1 2 IN IP4 195.5.0.83
s=-
c=IN IP4 195.5.0.83
t=0 0
m=audio 20854 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (20 headers 8 lines) ---
-- SIP/ukrtelecom-00000029 is ringing
-- SIP/ukrtelecom-00000029 is making progress passing it to SIP/101-00000028

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK69060425;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 103 INVITE
Contact: "AIC Route to VP" <sip:0800506800@195.5.0.83:5060;transport=udp>
P-Asserted-Identity: "AIC Route to VP" <sip:380892222201@corp.ukrtelecom.loc:5061>
Allow: INVITE
Allow: ACK
Allow: PRACK
Allow: SUBSCRIBE
Allow: BYE
Allow: CANCEL
Allow: NOTIFY
Allow: INFO
Allow: REFER
Allow: UPDATE
Content-Type: application/sdp
Content-Length: 151

v=0
o=- 1 2 IN IP4 195.5.0.83
s=-
c=IN IP4 195.5.0.83
t=0 0
m=audio 20854 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (20 headers 8 lines) ---
list_route: hop: <sip:0800506800@195.5.0.83:5060;transport=udp>
set_destination: Parsing <sip:0800506800@195.5.0.83:5060;transport=udp> for address/port to send to
set_destination: set destination to 195.5.0.83, port 5060
Transmitting (NAT) to 195.5.0.83:5060:
ACK sip:0800506800@195.5.0.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK551d3960;rport
Max-Forwards: 70
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=113265522
Contact: <sip:380892506511@10.10.10.110>
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Content-Length: 0


---
-- SIP/ukrtelecom-00000029 answered SIP/101-00000028

<--- SIP read from UDP:195.5.0.83:5060 --->
UPDATE sip:380892506511@10.10.10.110 SIP/2.0
Via: SIP/2.0/UDP 195.5.0.83:5060;branch=z9hG4bKoqqhfo2088nh8fk4j5k0sm0000g00.1
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 1 UPDATE
From: <sip:0800506800@sip.ukrtel.net>;tag=113265522
To: "101" <sip:380892506511@195.5.0.83>;tag=as6f72688f
Content-Length: 0
Contact: "AIC Route to VP" <sip:0800506800@195.5.0.83:5060;transport=udp>
Max-Forwards: 8
Session-Expires: 1800
Supported: timer
P-Asserted-Identity: "Main800506800" <sip:892506511@10.254.10.17;user=phone>


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

<--- Transmitting (NAT) to 195.5.0.83:5060 --->
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 195.5.0.83:5060;branch=z9hG4bKoqqhfo2088nh8fk4j5k0sm0000g00.1;received=195.5.0.83
From: <sip:0800506800@sip.ukrtel.net>;tag=113265522
To: "101" <sip:380892506511@195.5.0.83>;tag=as6f72688f
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 1 UPDATE
Server: Asterisk PBX 1.6.2.9-2+squeeze6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Jul 30 15:05:48] NOTICE[1767]: chan_sip.c:22000 handle_incoming: Unknown SIP command 'UPDATE' from '195.5.0.83'

<--- SIP read from UDP:195.5.0.83:5060 --->
UPDATE sip:380892506511@10.10.10.110 SIP/2.0
Via: SIP/2.0/UDP 195.5.0.83:5060;branch=z9hG4bKoqqhfo2088nh8fk4j5k0sm0000010.1
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 2 UPDATE
From: <sip:0800506800@sip.ukrtel.net>;tag=113265522
To: "101" <sip:380892506511@195.5.0.83>;tag=as6f72688f
Content-Length: 0
Contact: "AIC Route to VP" <sip:0800506800@195.5.0.83:5060;transport=udp>
Max-Forwards: 8
Session-Expires: 1800
Supported: timer
P-Asserted-Identity: "VP0159" <sip:892506511@10.254.10.17;user=phone>


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

<--- Transmitting (NAT) to 195.5.0.83:5060 --->
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 195.5.0.83:5060;branch=z9hG4bKoqqhfo2088nh8fk4j5k0sm0000010.1;received=195.5.0.83
From: <sip:0800506800@sip.ukrtel.net>;tag=113265522
To: "101" <sip:380892506511@195.5.0.83>;tag=as6f72688f
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 2 UPDATE
Server: Asterisk PBX 1.6.2.9-2+squeeze6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Jul 30 15:05:48] NOTICE[1767]: chan_sip.c:22000 handle_incoming: Unknown SIP command 'UPDATE' from '195.5.0.83'
Scheduling destruction of SIP dialog '12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net' in 6400 ms (Method: UPDATE)
set_destination: Parsing <sip:0800506800@195.5.0.83:5060;transport=udp> for address/port to send to
set_destination: set destination to 195.5.0.83, port 5060
Reliably Transmitting (NAT) to 195.5.0.83:5060:
BYE sip:0800506800@195.5.0.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK46c7a376;rport
Max-Forwards: 70
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=113265522
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Proxy-Authorization: Digest username="380892506511", realm="sip.ukrtel.net", algorithm=MD5, uri="sip.ukrtel.net", nonce="MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==", response="49cd1e38a8fe27f4a910af7c1b757cc0"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== Spawn extension (phones, 90800506800, 2) exited non-zero on 'SIP/101-00000028'
Retransmitting #1 (NAT) to 195.5.0.83:5060:
BYE sip:0800506800@195.5.0.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK46c7a376;rport
Max-Forwards: 70
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=113265522
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Proxy-Authorization: Digest username="380892506511", realm="sip.ukrtel.net", algorithm=MD5, uri="sip.ukrtel.net", nonce="MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==", response="49cd1e38a8fe27f4a910af7c1b757cc0"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Retransmitting #2 (NAT) to 195.5.0.83:5060:
BYE sip:0800506800@195.5.0.83:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.110:5060;branch=z9hG4bK46c7a376;rport
Max-Forwards: 70
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@sip.ukrtel.net>;tag=113265522
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze6
Proxy-Authorization: Digest username="380892506511", realm="sip.ukrtel.net", algorithm=MD5, uri="sip.ukrtel.net", nonce="MTM0MzY0OTk4MzpTREZTZXJ2ZXJTZWNyZXRLZXk6MTM2NDM0OTE0Mg==", response="49cd1e38a8fe27f4a910af7c1b757cc0"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK46c7a376;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 104 BYE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net' Method: UPDATE

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK46c7a376;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 104 BYE
Content-Length: 0


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

<--- SIP read from UDP:195.5.0.83:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.110:5060;received=195.88.113.188;branch=z9hG4bK46c7a376;rport=60638
From: "101" <sip:380892506511@sip.ukrtel.net>;tag=as6f72688f
To: <sip:0800506800@195.5.0.83>;tag=113265522
Call-ID: 12d9dcd953de7a8f514a05ee6a480ebb@sip.ukrtel.net
CSeq: 104 BYE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
energotourserver*CLI>

Всем ответившим - Огромное спасибо.

[Vlad1983]

не проходит когда один канал уже занять? или после того как первый уже завершился?

[bahek2462774]

когда уже завершился.

[Vlad1983]

из
extensions.conf
удалите
[general]
autofallthrough=yes
context=default
allowguest=no
bindport=5060
bindaddr=0.0.0.0

оно там бестолку

в остальном трудно сказать, т.к. инвайты от вас одинаковые в обоих случаях
снимите полный дамп (киньте в личку ссылку если не хотите светить белыми IP) возможно что-то ещё мешает
отпишу здесь

возможно у вас увели логин и пароль и названивают со стороны соответственно блокируются лишние каналы
сверяться по трафику не пробовали?

[bahek2462774]

логин и пароль нет , не увели.....менял во первых только с утра , и щас проверил по трафику . все норм.


как снять полный дам ?

[bahek2462774]

Больше нет никаких идей ?

[Vlad1983]

идеи по снятию дампа?
обычно так
tcpdump -i any -vvvnn -s0 udp -w /tmp/dump.cap
но вариантов тьма

в идеале снять с зазеркаленого WAN маршрутизатора одновременно с дампом на PBX

если внешний IP белый и статичный, указать его в [general] sip.conf
nat=yes
externip=бла
localnet=все сети путь к которым без NAT
отключить SIP-ALG на маршрутизаторе

других соображений без дампа нет

[bahek2462774]

с роутером все нормально - один же звонок проходит..и разговаривается нормально.
со всеми настройками externip и localnet - тоже пробовал - результат не дало (((

дам снял. скачать дамп

[Vlad1983]

"слишком умный" маршрутизатор может ломать заголовки SIP, при этом оставляя в кеше теги, и подставлять их при следующем звонке, "думая" что это продолжение предыдущего.

[bahek2462774]

стоит DIR-825...

но внутри я подключаюсь без NAT-a,
т.е. сервер астериск - 192.168.0.233
ноут ( софтфон) - 192.168.233

а вообще схема такая

сервер астериск ===> dlink DIR-825 ===> модем DSL-2540U ===> укртелекомовская ТФОП