Пожалуйста, войдите здесь. Часто задаваемые вопросы О нас
Задайте Ваш вопрос

403 Forbidden после первого звонка

0

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

[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

[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

    -- 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 

<--- 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.

удалить закрыть спам изменить тег редактировать

спросил 2012-07-30 16:11:56 +0400

анонимный пользователь

Аноним

Comments

Вышлите этот же лог в сапорт провайдеру. Никто лучше него не знает причины.

asteriskguru ( 2012-07-30 17:12:16 +0400 )редактировать

5 Ответов

Сорт. по » старые новые больше голосов
0

... на самом деле вот єто defaultexpiry=10 СОВСЕМ не помогает. В результате было выявленно решение:

  1. Проблема при преобразовании sip.ukrtel.net <=> 195.5.0.51 если прописывать именем - то транк падает каждые 30 секунд. НУЖНО ТОЛЬКО IP Адрес !!!
  2. Установить таймауты, для подстраховки. Выкладываю полностью рабочий конфиг для желающих и намучившихся ....

register=>380891111111:--tuta-parolik--@195.5.0.51

registerexpired=3600

registertimeout=20

registerattempts=0

registerretry403=yes

[ukrtel]

canreinvite=yes

host=195.5.0.51

port=5060

defaultuser=38089xxxxxxx

fromuser=38089xxxxxxx

username=38089xxxxxxx

secret=xxxxxxxxxxx

type=friend

insecure=port,invite

qualify=yes

nat=no

dtmfmode=rfc2833

context=ukrtel-trunk

fromdomain=195.5.0.51

diallow=all

allow=alaw

allow=ulaw

rtpkeepalive=20

registerattempts=0

диалплан

exten => _044XXXXXXX,1,Answer

exten => _044XXXXXXX,2,Dial(SIP/${EXTEN}@ukrtel)

exten => _044XXXXXXX,n,Hangup

для звонка по семизначному набору - подставляем код города (кому какой)

exten => _XXXXXXX,1,Answer

exten => _XXXXXXX,2,Dial(SIP/044${EXTEN}@ukrtel)

exten => _XXXXXXX,n,Hangup

Всем хорошего коннекта !!! Николай.

ссылка удалить спам редактировать

ответил 2016-07-30 19:49:49 +0400

1Nikolay Gravatar 1Nikolay
1

Comments

Через 4 года ждем продолжения.

zzuz ( 2016-07-31 16:37:54 +0400 )редактировать
0

Для Укртелекома время регистрации на их SIP-прокси не должно превышать 30 сек.

Чуть измените строку регистрации:

register => 38:XXXXXXX@sip.ukrtel.net/38~30

Defaultexpiry - отвечает за время, через которое клиент шлет повторный REGISTER для обновления своей регистрации на SIP-Registrar оператора. В данном случае - для Укртелекома достаточно значения до 30.

ссылка удалить спам редактировать

ответил 2012-07-30 17:09:16 +0400

mistral Gravatar mistral flag of Ukraine
370 ● 2 ● 5 ● 19

обновил 2012-08-02 18:08:55 +0400

0

Не могу почему то ответить с аккаунта своего ( спам распознает).

так вот - добавление ~30 - ситуацию не изменило. а в саппорт дозвонится трудно - у них там жесткий "испорченный телефон" - конкретно с инженерами поговорить не получается ((((

ссылка удалить спам редактировать

ответил 2012-07-30 17:33:56 +0400

bahek2462774 Gravatar bahek2462774
1 ● 2 ● 1 ● 2

Comments

значит так общаетесь с суппортом:) Суппорту достаточно оставить данные: номер А, номер Б, время неудачного звонка, (желательно несколько звонков), тогда суппорт сформирует заявку инженерам для рассмотрения. В течение суток проблему обычно решают. И еще - REGISTER тоже покажите.

mistral ( 2012-07-30 18:19:19 +0400 )редактировать
0

супоорт меня морозит , потому что с обычным софтфоном все работает ( пробывал с 3cxPhone). пробывал обновить астериск до версии 1.8.15 - результата ноль..эх...идеи кончаются (((

ссылка удалить спам редактировать

ответил 2012-07-31 14:38:17 +0400

bahek2462774 Gravatar bahek2462774
1 ● 2 ● 1 ● 2
0

проблему решило добавление в sip.conf в секцию general

defaultexpiry=10

но думаю это не правильный выход. Подскажите за что отвечает этот параметр ??

ссылка удалить спам редактировать

ответил 2012-08-02 15:49:10 +0400

bahek2462774 Gravatar bahek2462774
1 ● 2 ● 1 ● 2

Comments

Все правильно. Ответ выше.

mistral ( 2012-08-02 18:09:18 +0400 )редактировать

Ваш ответ

Please start posting your answer anonymously - your answer will be saved within the current session and published after you log in or create a new account. Please try to give a substantial answer, for discussions, please use comments and please do remember to vote (after you log in)!
[скрыть предварительный просмотр]

Закладки и информация

Добавить закладку
1 закладка

подписаться на rss ленту новостей

Статистика

Задан: 2012-07-30 16:11:56 +0400

Просмотрен: 3,730 раз

Обновлен: Jul 30 '16

Похожие вопросы:

Panasonic 403 forbidden при отбое

asterisk как sip клиент

GXV3501 и SIP-домофон

SIP/TCP софтфоны для x86

Tandberg&Asterisk

elastix + 2 sip trunka

Sip на динамически ip

asterisk & iskratel

Странный звонок и отзвон.

центральный сервер

О проекте | Часто задаваемые вопросы | политика конфиденциальности | оставить отзыв

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.