Вход | Регистрация
Вы здесь: Главная / Форум / Главный форум по Asterisk / Конфигурация и настройка / Рвётся связь

Рвётся связь

Рвётся связь время от времени
<123 4
Intent
Сообщений: 8

Re: Рвётся связь

Да, правильно. Поступает внешний звонок, * кидает его в очередь, трубку берёт клиент, подключенный к *, сип диалог устанавливается, но в один момент(но далеко не всегда!) клиент посылает BYE сообщение по причине того, что нет подтверждающего ACK. В логах * в этот момент следующее:


<------------->
[Sep 8 13:05:40] --- (11 headers 9 lines) ---
[Sep 8 13:05:43]
<--- SIP read from UDP:192.168.0.3:43552 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK11add036;rport=5060
To: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>;tag=fd445c05
From: "number"<sip:number@192.168.0.2>;tag=as790e2dc7
Call-ID: 77b2c68b4de3a2b1119f7c1672d691b4@192.168.0.2
CSeq: 102 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
[Sep 8 13:05:43] --- (8 headers 0 lines) ---
[Sep 8 13:05:44]
<--- SIP read from UDP:192.168.0.3:43552 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK28026e76;rport=5060
Contact: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>
To: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>;tag=c85d982b
From: "number"<sip:number@192.168.0.2>;tag=as3182e926
Call-ID: 2671ad37741768d0314f47a3081b140c@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 2 2 IN IP4 192.168.0.3
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.3
t=0 0
m=audio 40306 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 8 13:05:44] --- (11 headers 9 lines) ---
[Sep 8 13:05:48]
<--- SIP read from UDP:192.168.0.3:43552 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK28026e76;rport=5060
Contact: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>
To: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>;tag=c85d982b
From: "number"<sip:number@192.168.0.2>;tag=as3182e926
Call-ID: 2671ad37741768d0314f47a3081b140c@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 2 2 IN IP4 192.168.0.3
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.3
t=0 0
m=audio 40306 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 8 13:05:48] --- (11 headers 9 lines) ---
[Sep 8 13:05:51] NOTICE[27711]: chan_sip.c:12779 check_auth: Correct auth, but based on stale nonce received from '"505"<sip:505@ast.pronto.krsk.info>;tag=081b6a65'
[Sep 8 13:05:52]
<--- SIP read from UDP:192.168.0.3:43552 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK28026e76;rport=5060
Contact: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>
To: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>;tag=c85d982b
From: "number"<sip:number@192.168.0.2>;tag=as3182e926
Call-ID: 2671ad37741768d0314f47a3081b140c@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 2 2 IN IP4 192.168.0.3
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.3
t=0 0
m=audio 40306 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 8 13:05:52] --- (11 headers 9 lines) ---
[Sep 8 13:05:56]
<--- SIP read from UDP:192.168.0.3:43552 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK28026e76;rport=5060
Contact: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>
To: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>;tag=c85d982b
From: "number"<sip:number@192.168.0.2>;tag=as3182e926
Call-ID: 2671ad37741768d0314f47a3081b140c@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 2 2 IN IP4 192.168.0.3
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.3
t=0 0
m=audio 40306 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 8 13:05:56] --- (11 headers 9 lines) ---
[Sep 8 13:05:56]
<--- SIP read from UDP:192.168.0.3:43552 --->
BYE sip:number@192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.3:43552;branch=z9hG4bK-d8754z-c0751d7e4608a320-1---d8754z-
Max-Forwards: 70
Contact: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>
To: "number"<sip:number@192.168.0.2>;tag=as3182e926
From: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>;tag=c85d982b
Call-ID: 2671ad37741768d0314f47a3081b140c@192.168.0.2
CSeq: 2 BYE
User-Agent: X-Lite release 1104o stamp 56125
Reason: SIP;description="ACK not received"
Content-Length: 0


<------------->
[Sep 8 13:05:56] --- (11 headers 0 lines) ---
[Sep 8 13:05:56] Sending to 192.168.0.3 : 43552 (no NAT)
[Sep 8 13:05:56]
<--- Transmitting (no NAT) to 192.168.0.3:43552 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.3:43552;branch=z9hG4bK-d8754z-c0751d7e4608a320-1---d8754z-;received=192.168.0.3
From: <sip:501@192.168.0.3:43552;rinstance=d0a25e2200e4722d>;tag=c85d982b
To: "number"<sip:number@192.168.0.2>;tag=as3182e926
Call-ID: 2671ad37741768d0314f47a3081b140c@192.168.0.2
CSeq: 2 BYE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
Ubuntu 10.04, Asterisk PBX 1.8.0
2010-09-10 12:17

switch
Avatara of switch
Откуда: Уфа
Сообщений: 5856

Re: Рвётся связь

SIP/2.0 487 Request Terminated
Это отвечает на инвайт ваш Хлайт.
может вы Хлайт коряво настроили?
попробуйте другой софтфон
http://www.lynks.ru - Решения телефонии, мини-АТС, VoIP на основе Trixbox и Asterisk
2010-09-10 12:33

Intent
Сообщений: 8

Re: Рвётся связь

switch: пробывал уже десятки софтфонов - одна и та же история. В SJPhone вообще при разрыве выскакивает ошибка "Ack time out"
Ubuntu 10.04, Asterisk PBX 1.8.0
2010-09-10 12:37

cron333
Сообщений: 1573

Re: Рвётся связь

Вы привели диалог с того момента, когда X-lite делает - SIP/2.0 487 Request Terminated (SIP/2.0 487 Request Terminated - запрос отменен, обычно приходит при отмене вызова).

Покажите начало этого диалога, с запроса INVITE от * .

В любом случае нужно разбираться, почему ваш X-Lite делает это ...
2010-09-10 12:45

kronos
Откуда: Minsk
Сообщений: 55

Re: Рвётся связь

Похоже я не одинок :)
У меня похожая проблема с софтофонами. Рвут соединение либо по истечению 30 секунд, либо дают чуть больше времени.

Софтофоны пробовал разные, при этом время разрыва чуть варьировалось. XLite рвал через 30 сек, ZoIPer через 40.

При этом, подключенные к Астериску сип адаптеры работают без проблем.
Глубоко проблему пока не копал, нужно было косяки с регистрацией прикрыть :)
2010-09-10 19:07

Alexcr
Откуда: AST
Сообщений: 280

Re: Рвётся связь

;--------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
rtpkeepalive=30 ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)
2010-09-10 19:55

cron333
Сообщений: 1573

Re: Рвётся связь

До RTP здесь процесс не доходит ...


P.S. Кстати проверьте, нет ли в этот момент проблем с DNS (как у kronos).

P.P.S. * очень чувствителен к проблемам с DNS.
2010-09-11 01:47

Intent
Сообщений: 8

Re: Рвётся связь

Kronos: Если рвется связь через какой-то определённый интервал, скорее всего срабатывают таймеры какие-то.

вот полный дебаг звонка с обрывом, 777 - очередь, 505 - клиет с X-lite:


<------------->
[Sep 13 13:01:12] --- (9 headers 0 lines) ---
[Sep 13 13:01:13]
<--- SIP read from UDP:provIP:5060 --->
INVITE sip:777@MyEXTIP SIP/2.0
Via: SIP/2.0/UDP provIP:5060;branch=z9hG4bK-44ff120019ffff10ff0000600866ffff
From: <sip:8XXXXXXXXX@provIP;user=phone>;tag=ffff120019ffff10ff0000600866ffff
To: <sip:myNUMBER@MyEXTIP:5060;user=phone>
Call-ID: e2b5120019b08d10800000600866b880@vs.krsk.info
CSeq: 1 INVITE
Contact: <sip:8XXXXXXXXX@provIP;user=phone>
Max-Forwards: 70
User-Agent: MERA MSIP v.1.0.2
Cisco-Guid: 3285810685-3190428127-3142295812-2898732256
Content-Type: application/sdp
Content-Length: 313

v=0
o=- 1284354073 1284354073 IN IP4 provIP
s=-
c=IN IP4 provIP
t=0 0
m=audio 28860 RTP/AVP 18 8 0 4 4 3 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
[Sep 13 13:01:13] --- (12 headers 14 lines) ---
[Sep 13 13:01:13] Sending to provIP : 5060 (NAT)
[Sep 13 13:01:13] Using INVITE request as basis request - e2b5120019b08d10800000600866b880@vs.krsk.info
[Sep 13 13:01:13] Found peer 'provIP' for '8XXXXXXXXX' from provIP:5060
[Sep 13 13:01:13] Found RTP audio format 18
[Sep 13 13:01:13] Found RTP audio format 8
[Sep 13 13:01:13] Found RTP audio format 0
[Sep 13 13:01:13] Found RTP audio format 4
[Sep 13 13:01:13] Found RTP audio format 4
[Sep 13 13:01:13] Found RTP audio format 3
[Sep 13 13:01:13] Found RTP audio format 101
[Sep 13 13:01:13] Found audio description format G729 for ID 18
[Sep 13 13:01:13] Found audio description format PCMA for ID 8
[Sep 13 13:01:13] Found audio description format PCMU for ID 0
[Sep 13 13:01:13] Found audio description format G723 for ID 4
[Sep 13 13:01:13] Found audio description format G723 for ID 4
[Sep 13 13:01:13] Found audio description format gsm for ID 3
[Sep 13 13:01:13] Found audio description format telephone-event for ID 101
[Sep 13 13:01:13] Capabilities: us - 0x8 (alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Sep 13 13:01:13] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 13 13:01:13] Peer audio RTP is at port provIP:28860
[Sep 13 13:01:13] Looking for 777 in sip-dialout (domain MyEXTIP)
[Sep 13 13:01:13] list_route: hop: <sip:8XXXXXXXXX@provIP;user=phone>
[Sep 13 13:01:13]
<--- Transmitting (NAT) to provIP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP provIP:5060;branch=z9hG4bK-44ff120019ffff10ff0000600866ffff;received=provIP
From: <sip:8XXXXXXXXX@provIP;user=phone>;tag=ffff120019ffff10ff0000600866ffff
To: <sip:myNUMBER@MyEXTIP:5060;user=phone>
Call-ID: e2b5120019b08d10800000600866b880@vs.krsk.info
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:777@MyEXTIP>
Content-Length: 0


<------------>
[Sep 13 13:01:13] Audio is at MyEXTIP port 41284
[Sep 13 13:01:13] Adding codec 0x8 (alaw) to SDP
[Sep 13 13:01:13] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 13 13:01:13]
<--- Reliably Transmitting (NAT) to provIP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP provIP:5060;branch=z9hG4bK-44ff120019ffff10ff0000600866ffff;received=provIP
From: <sip:8XXXXXXXXX@provIP;user=phone>;tag=ffff120019ffff10ff0000600866ffff
To: <sip:myNUMBER@MyEXTIP:5060;user=phone>;tag=as7f812fe8
Call-ID: e2b5120019b08d10800000600866b880@vs.krsk.info
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:777@MyEXTIP>
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1148844926 1148844926 IN IP4 MyEXTIP
s=Asterisk PBX 1.6.2.11
c=IN IP4 MyEXTIP
t=0 0
m=audio 41284 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Sep 13 13:01:13]
<--- SIP read from UDP:provIP:5060 --->
ACK sip:777@MyEXTIP SIP/2.0
Via: SIP/2.0/UDP provIP:5060;branch=z9hG4bK-44ff120019ffff10ff0000600866ffff;received=provIP
From: <sip:8XXXXXXXXX@provIP;user=phone>;tag=ffff120019ffff10ff0000600866ffff
To: <sip:myNUMBER@MyEXTIP:5060;user=phone>;tag=as7f812fe8
Call-ID: e2b5120019b08d10800000600866b880@vs.krsk.info
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: MERA MSIP v.1.0.2
Content-Length: 0


<------------->
[Sep 13 13:01:19] --- (12 headers 0 lines) ---
[Sep 13 13:01:19] Really destroying SIP dialog '740232223ee968282d31f0ff1d677fc3@192.168.0.2' Method: OPTIONS
[Sep 13 13:01:25] Audio is at 192.168.0.2 port 40040
[Sep 13 13:01:25] Adding codec 0x8 (alaw) to SDP
[Sep 13 13:01:25] Adding codec 0x4 (ulaw) to SDP
[Sep 13 13:01:25] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 13 13:01:25] Reliably Transmitting (no NAT) to 192.168.0.6:41318:
INVITE sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport
Max-Forwards: 70
From: "8XXXXXXXXX" <sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
Contact: <sip:8XXXXXXXXX@192.168.0.2>
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Mon, 13 Sep 2010 05:01:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1170556874 1170556874 IN IP4 192.168.0.2
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.0.2
t=0 0
m=audio 40040 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep 13 13:01:26] Retransmitting #1 (no NAT) to 192.168.0.6:41318:
INVITE sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport
Max-Forwards: 70
From: "8XXXXXXXXX" <sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
Contact: <sip:8XXXXXXXXX@192.168.0.2>
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Mon, 13 Sep 2010 05:01:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1170556874 1170556874 IN IP4 192.168.0.2
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.0.2
t=0 0
m=audio 40040 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Sep 13 13:01:26]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
[Sep 13 13:01:26] --- (9 headers 0 lines) ---
[Sep 13 13:01:29]
<--- SIP read from UDP:192.168.0.219:54808 --->



<------------->
[Sep 13 13:01:34] NOTICE[13100]: chan_sip.c:11539 sip_reregister: -- Re-registration for myNUMBER@provIP
[Sep 13 13:01:34] REGISTER 11 headers, 0 lines
[Sep 13 13:01:34] Reliably Transmitting (NAT) to provIP:5060:
REGISTER sip:provIP SIP/2.0
Via: SIP/2.0/UDP MyEXTIP:5060;branch=z9hG4bK045b51c3;rport
Max-Forwards: 70
From: <sip:myNUMBER@provIP>;tag=as0c4b68bb
To: <sip:myNUMBER@provIP>
Call-ID: 2a3390dc482adb3320812b4d2c66150b@provIP
CSeq: 194 REGISTER
User-Agent: Asterisk PBX 1.6.2.11
Authorization: Digest username="myNUMBER", realm="provIP", algorithm=MD5, uri="sip:provIP", nonce="50f78f00c88a8d10800000600866b880@vs.krsk.info", response="b2df02a42b539f4a9129151ab054c109"
Expires: 120
Contact: <sip:777@MyEXTIP>
Content-Length: 0


---
[Sep 13 13:01:35] Retransmitting #1 (NAT) to provIP:5060:
REGISTER sip:provIP SIP/2.0
Via: SIP/2.0/UDP MyEXTIP:5060;branch=z9hG4bK045b51c3;rport
Max-Forwards: 70
From: <sip:myNUMBER@provIP>;tag=as0c4b68bb
To: <sip:myNUMBER@provIP>
Call-ID: 2a3390dc482adb3320812b4d2c66150b@provIP
CSeq: 194 REGISTER
User-Agent: Asterisk PBX 1.6.2.11
Authorization: Digest username="myNUMBER", realm="provIP", algorithm=MD5, uri="sip:provIP", nonce="50f78f00c88a8d10800000600866b880@vs.krsk.info", response="b2df02a42b539f4a9129151ab054c109"
Expires: 120
Contact: <sip:777@MyEXTIP>
Content-Length: 0


---
[Sep 13 13:01:35]
<--- SIP read from UDP:provIP:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP MyEXTIP:5060;branch=z9hG4bK045b51c3;rport
From: <sip:myNUMBER@provIP>;tag=as0c4b68bb
To: <sip:myNUMBER@provIP>
Call-ID: 2a3390dc482adb3320812b4d2c66150b@provIP
CSeq: 194 REGISTER
Contact: <sip:777@MyEXTIP>
Expires: 120
Content-Length: 0


<------------->
[Sep 13 13:01:35] --- (9 headers 0 lines) ---
[Sep 13 13:01:35]
<--- SIP read from UDP:provIP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP MyEXTIP:5060;branch=z9hG4bK045b51c3;rport
From: <sip:myNUMBER@provIP>;tag=as0c4b68bb
To: <sip:myNUMBER@provIP>
Call-ID: 2a3390dc482adb3320812b4d2c66150b@provIP
CSeq: 194 REGISTER
Contact: <sip:777@MyEXTIP>;expires=120
Expires: 120
Date: Mon, 13 Sep 2010 05:01:34 GMT
Content-Length: 0



<------------>
[Sep 13 13:01:39] Scheduling destruction of SIP dialog 'YzFmY2IwNzMzOTEyZDZhNTU5ZTRmMGIwMDA5ODZmY2M.' in 32000 ms (Method: REGISTER)
[Sep 13 13:01:42]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:01:42] --- (11 headers 9 lines) ---
[Sep 13 13:01:42] Found RTP audio format 8
[Sep 13 13:01:42] Found RTP audio format 0
[Sep 13 13:01:42] Found RTP audio format 101
[Sep 13 13:01:42] Found audio description format telephone-event for ID 101
[Sep 13 13:01:42] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Sep 13 13:01:42] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 13 13:01:42] Peer audio RTP is at port 192.168.0.6:43376
[Sep 13 13:01:42] list_route: hop: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
[Sep 13 13:01:42] set_destination: Parsing <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74> for address/port to send to
[Sep 13 13:01:42] set_destination: set destination to 192.168.0.6, port 41318
[Sep 13 13:01:42] Transmitting (no NAT) to 192.168.0.6:41318:
ACK sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK484ea6c5;rport
Max-Forwards: 70
From: "8XXXXXXXXX" <sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
Contact: <sip:8XXXXXXXXX@192.168.0.2>
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0


---
[Sep 13 13:01:43]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:01:43] --- (11 headers 9 lines) ---
[Sep 13 13:01:44]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:01:44] --- (11 headers 9 lines) ---
[Sep 13 13:01:46]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:01:46] --- (11 headers 9 lines) ---
[Sep 13 13:01:47]
<--- SIP read from UDP:192.168.0.72:5060 --->



<------------->
[Sep 13 13:01:50]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:01:54] --- (9 headers 0 lines) ---
[Sep 13 13:01:54] Really destroying SIP dialog '5931a6990109511e4582f79d5be5d1de@192.168.0.2' Method: INVITE
[Sep 13 13:01:54]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:01:54] --- (11 headers 9 lines) ---
[Sep 13 13:01:58]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:01:58] --- (11 headers 9 lines) ---
[Sep 13 13:01:59]
<--- SIP read from UDP:192.168.0.219:54808 --->



<------------->
[Sep 13 13:02:02]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv


<------------->
[Sep 13 13:02:05] --- (8 headers 0 lines) ---
[Sep 13 13:02:06]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:02:08] WARNING[13100]: translate.c:160 framein: no samples for alawtolin
[Sep 13 13:02:10]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:02:10] --- (11 headers 9 lines) ---
[Sep 13 13:02:11] Really destroying SIP dialog 'YzFmY2IwNzMzOTEyZDZhNTU5ZTRmMGIwMDA5ODZmY2M.' Method: REGISTER
[Sep 13 13:02:14]
<--- SIP read from UDP:192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK631588e8;rport=5060
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
From: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
ontent-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 183

v=0
o=- 7 2 IN IP4 192.168.0.6
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.6
t=0 0
m=audio 43376 RTP/AVP 8 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
[Sep 13 13:02:14] --- (11 headers 9 lines) ---
[Sep 13 13:02:14]
<--- SIP read from UDP:192.168.0.6:41318 --->
BYE sip:8XXXXXXXXX@192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:41318;branch=z9hG4bK-d8754z-6825e87db93c9453-1---d8754z-
Max-Forwards: 70
Contact: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>
To: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
From: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 2 BYE
User-Agent: X-Lite release 1104o stamp 56125
Reason: SIP;description="ACK not received"
Content-Length: 0


<------------->
[Sep 13 13:02:14] --- (11 headers 0 lines) ---
[Sep 13 13:02:14] Sending to 192.168.0.6 : 41318 (no NAT)
[Sep 13 13:02:14]
<--- Transmitting (no NAT) to 192.168.0.6:41318 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.6:41318;branch=z9hG4bK-d8754z-6825e87db93c9453-1---d8754z-;received=192.168.0.6
From: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
To: "8XXXXXXXXX"<sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 2 BYE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Sep 13 13:02:14] Scheduling destruction of SIP dialog 'e2b5120019b08d10800000600866b880@vs.krsk.info' in 32000 ms (Method: ACK)
[Sep 13 13:02:14] set_destination: Parsing <sip:8XXXXXXXXX@provIP;user=phone> for address/port to send to
[Sep 13 13:02:14] set_destination: set destination to provIP, port 5060
[Sep 13 13:02:14] Reliably Transmitting (NAT) to provIP:5060:
BYE sip:8XXXXXXXXX@provIP;user=phone SIP/2.0
Via: SIP/2.0/UDP MyEXTIP:5060;branch=z9hG4bK6c8c10b2;rport
Max-Forwards: 70
From: <sip:myNUMBER@MyEXTIP:5060;user=phone>;tag=as7f812fe8
To: <sip:8XXXXXXXXX@provIP;user=phone>;tag=ffff120019ffff10ff0000600866ffff
Call-ID: e2b5120019b08d10800000600866b880@vs.krsk.info
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.11
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---
[Sep 13 13:02:14]
<--- SIP read from UDP:provIP:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP MyEXTIP:5060;branch=z9hG4bK6c8c10b2;rport
From: <sip:myNUMBER@MyEXTIP:5060;user=phone>;tag=as7f812fe8
To: <sip:8XXXXXXXXX@provIP;user=phone>;tag=ffff120019ffff10ff0000600866ffff
Call-ID: e2b5120019b08d10800000600866b880@vs.krsk.info
CSeq: 102 BYE
Server: MERA MSIP v.1.0.2
Content-Length: 0


<------------->
[Sep 13 13:02:14] --- (8 headers 0 lines) ---
[Sep 13 13:02:14] SIP Response message for INCOMING dialog BYE arrived
[Sep 13 13:02:14] Really destroying SIP dialog 'e2b5120019b08d10800000600866b880@vs.krsk.info' Method: ACK
[Sep 13 13:02:14] Really destroying SIP dialog '2cd746ff4608539964fe38d35fe5ff00@192.168.0.2' Method: BYE
Ubuntu 10.04, Asterisk PBX 1.8.0
2010-09-13 10:04

cron333
Сообщений: 1573

Re: Рвётся связь

ACK sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK484ea6c5;rport
Max-Forwards: 70
From: "8XXXXXXXXX" <sip:8XXXXXXXXX@192.168.0.2>;tag=as5607a8d1
To: <sip:505@192.168.0.6:41318;rinstance=eb2eb62885cadd74>;tag=e72f5e7d
Contact: <sip:8XXXXXXXXX@192.168.0.2>
Call-ID: 2cd746ff4608539964fe38d35fe5ff00@192.168.0.2
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0
* - отсылает ACK. Смотрите - почему не доходит ...

P.S. При дебаге определенного пира делайте так: CLI>sip set debug peer 505 ...

А так - половина ненужного мусора ...
2010-09-13 10:31

kronos
Откуда: Minsk
Сообщений: 55

Re: Рвётся связь

Как решал проблему я:

1. в настройках x-lite вырубил RTP тайм аут
2. в тех же настройках указал внутренний DNS сервер

забавно то, что простого отключения тайм аута было мало О_о

бред конечно, но пока работают люди. Посмотрим что будет дальше
2010-09-13 11:08

<123 4
Войдите чтобы оставить сообщение!
Добавить страницу в закладки:  Delicious Google Slashdot Yahoo Yandex.ru Reddit Digg Technorati Bobrdobr.ru Newsland.ru Smi2.ru Rumarkz.ru Vaau.ru Memori.ru Rucity.com Moemesto.ru News2.ru Mister-Wong.ru Myscoop.ru 100zakladok.ru