Re-Invite-ы

alex1980 · Сообщение **alex1980** » 10 окт 2013, 20:31

Всем хай!

Народ, существует следующая проблема. Звоню с компа на телефон. Оба зареганы на одном и том же *. В итоге, когда с компа нажимаю во время разговора "послать факс", соответственно клиент посылает второй Invite на астериск. Астериск воспринимает этот Invite как следствие установить новый звонок на тот же номер. Т.е если я звоню с 1000 на 1001, и нажимаю отправить факс, то астер пытается установить еще один звонок с 1000 на 1001 параллельно. Первый же звонок при этом становится на Hold и я слышу MOH. Т.е. как я понимаю не отрабатывает функция Re-Invite-а.
Понимаю, что возможно подобная проблема уже где-то здесь поднималась, но поиск не дал результатов конкретно по моей проблеме. (( Подскажите пожалуйста, куда копать.. Бьюсь уже который день с этим ((

Vlad1983 · Сообщение **Vlad1983** » 10 окт 2013, 20:35

определенно расчет на телепатов

софтфон хоть какой?

alex1980 · Сообщение **alex1980** » 10 окт 2013, 20:36

Прошу прощения... Софтфон Zoiper. Аппарат Cisco IP Phone 7945. Астер 11.0.0-rc1

Vlad1983 · Сообщение **Vlad1983** » 10 окт 2013, 20:37

Zoiper не может отправлять факсы во время разговора.

alex1980 · Сообщение **alex1980** » 10 окт 2013, 20:41

Буквально пару дней назад у меня с Zoiper 3 эта фишка отработала... Звонили на аналоговый факс с Zoiper-а. Поговорили, нажали факс и все заработало. Очень странно.
Ок. Может посоветуете в этом случае какой софт позволяет слать факсы во время разговора, если все-таки проблема с Zoiper_ом?

Vlad1983 · Сообщение **Vlad1983** » 10 окт 2013, 20:51

как минимум софтина должна отправлять re-INVITE c тем же Call-ID, если другой - это уже другой вызов.

alex1980 · Сообщение **alex1980** » 10 окт 2013, 20:55

Так судя по логам она и отправляет с тем же Call-Id второй инвайт... поэтому-то и не понятно почему астериск стал считать второй Invite как сигнал для установления второго звонка.

Vlad1983 · Сообщение **Vlad1983** » 10 окт 2013, 20:59

предлагаете дальше гадать?
где дамп трафика?

alex1980 · Сообщение **alex1980** » 10 окт 2013, 21:09

Первый ивайт

Код: Выделить всё

<--- SIP read from UDP:192.168.0.2:5061 --->
INVITE sip:1001@192.168.0.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-2a6f6c9c2c2ba030-1---d8754z-
Max-Forwards: 70
Contact: <sip:1000@192.168.0.2:5061;transport=UDP>
To: <sip:1001@192.168.0.1;transport=UDP>
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Allow-Events: presence, kpml
Content-Length: 208

v=0
o=Z 0 0 IN IP4 192.168.0.2
s=Z
c=IN IP4 192.168.0.2
t=0 0
m=audio 15000 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.0.2:5061 (no NAT)
Using INVITE request as basis request - NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
Found peer '1000' for '1000' from 192.168.0.2:5061
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.2:15000
Looking for 1001 in from-internal (domain 192.168.0.1)
list_route: hop: <sip:1000@192.168.0.2:5061;transport=UDP>

<--- Transmitting (no NAT) to 192.168.0.2:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-2a6f6c9c2c2ba030-1---d8754z-;received=192.168.0.2
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
To: <sip:1001@192.168.0.1;transport=UDP>
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1001@192.168.0.1:5060>
Content-Length: 0

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

<--- Transmitting (no NAT) to 192.168.0.2:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-2a6f6c9c2c2ba030-1---d8754z-;received=192.168.0.2
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as302a27bb
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1001@192.168.0.1:5060>
Remote-Party-ID: "Shcherbakov Alexey" <sip:1001@192.168.0.1>;party=called;privacy=off;screen=no
Content-Length: 0

<------------>
Audio is at 18358
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.2:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-2a6f6c9c2c2ba030-1---d8754z-;received=192.168.0.2
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as302a27bb
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1001@192.168.0.1:5060>
Remote-Party-ID: "Shcherbakov Alexey" <sip:1001@192.168.0.1>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1043065622 1043065622 IN IP4 192.168.0.1
s=Asterisk PBX 11.0.0-rc1
c=IN IP4 192.168.0.1
t=0 0
m=audio 18358 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

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

<--- SIP read from UDP:192.168.0.2:5061 --->
ACK sip:1001@192.168.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-6d68fd25ab83e0e9-1---d8754z-
Max-Forwards: 70
Contact: <sip:1000@192.168.0.2:5061;transport=UDP>
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as302a27bb
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 1 ACK
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Content-Length: 0

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

Пришел второй инвайт

Код: Выделить всё

asterisk*CLI>
<--- SIP read from UDP:192.168.0.2:5061 --->
INVITE sip:1001@192.168.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-5dd87589f636b6ac-1---d8754z-
Max-Forwards: 70
Contact: <sip:1000@192.168.0.2:5061;transport=UDP>
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as302a27bb
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Allow-Events: presence, kpml
Content-Length: 208

v=0
o=Z 0 1 IN IP4 192.168.0.2
s=Z
c=IN IP4 192.168.0.2
t=0 0
m=audio 15000 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=inactive
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.0.2:5061 (no NAT)
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.2:15000

<--- Transmitting (no NAT) to 192.168.0.2:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-5dd87589f636b6ac-1---d8754z-;received=192.168.0.2
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as302a27bb
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1001@192.168.0.1:5060>
Content-Length: 0

<------------>
Audio is at 18358
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.2:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-5dd87589f636b6ac-1---d8754z-;received=192.168.0.2
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as302a27bb
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1001@192.168.0.1:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1043065622 1043065623 IN IP4 192.168.0.1
s=Asterisk PBX 11.0.0-rc1
c=IN IP4 192.168.0.1
t=0 0
m=audio 18358 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=inactive

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

<--- SIP read from UDP:192.168.0.2:5061 --->
INVITE sip:1001@192.168.0.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-35cc1445234ab9c3-1---d8754z-
Max-Forwards: 70
Contact: <sip:1000@192.168.0.2:5061;transport=UDP>
To: <sip:1001@192.168.0.1;transport=UDP>
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=f7352e7f
Call-ID: ODNmYTZlZTI0YmQyYjczOGIwMmZkNzM2NTFjODk2YmM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Allow-Events: presence, kpml
Content-Length: 208

v=0
o=Z 0 0 IN IP4 192.168.0.2
s=Z
c=IN IP4 192.168.0.2
t=0 0
m=audio 15002 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.0.2:5061 (no NAT)
Using INVITE request as basis request - ODNmYTZlZTI0YmQyYjczOGIwMmZkNzM2NTFjODk2YmM.
Found peer '1000' for '1000' from 192.168.0.2:5061
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.2:15002
Looking for 1001 in from-internal (domain 192.168.0.1)
list_route: hop: <sip:1000@192.168.0.2:5061;transport=UDP>

<--- Transmitting (no NAT) to 192.168.0.2:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-35cc1445234ab9c3-1---d8754z-;received=192.168.0.2
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=f7352e7f
To: <sip:1001@192.168.0.1;transport=UDP>
Call-ID: ODNmYTZlZTI0YmQyYjczOGIwMmZkNzM2NTFjODk2YmM.
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1001@192.168.0.1:5060>
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.2:5061 --->
ACK sip:1001@192.168.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-f288130d0dd3d603-1---d8754z-
Max-Forwards: 70
Contact: <sip:1000@192.168.0.2:5061;transport=UDP>
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as302a27bb
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 2 ACK
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.2:5061 --->
CANCEL sip:1001@192.168.0.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-35cc1445234ab9c3-1---d8754z-
Max-Forwards: 70
To: <sip:1001@192.168.0.1;transport=UDP>
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=f7352e7f
Call-ID: ODNmYTZlZTI0YmQyYjczOGIwMmZkNzM2NTFjODk2YmM.
CSeq: 1 CANCEL
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.0.2:5061 (no NAT)

<--- Reliably Transmitting (no NAT) to 192.168.0.2:5061 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-35cc1445234ab9c3-1---d8754z-;received=192.168.0.2
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=f7352e7f
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as3faf455c
Call-ID: ODNmYTZlZTI0YmQyYjczOGIwMmZkNzM2NTFjODk2YmM.
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<--- Transmitting (no NAT) to 192.168.0.2:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-35cc1445234ab9c3-1---d8754z-;received=192.168.0.2
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=f7352e7f
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as3faf455c
Call-ID: ODNmYTZlZTI0YmQyYjczOGIwMmZkNzM2NTFjODk2YmM.
CSeq: 1 CANCEL
Server: FPBX-2.11.0(11.0.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.2:5061 --->
ACK sip:1001@192.168.0.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5061;branch=z9hG4bK-d8754z-35cc1445234ab9c3-1---d8754z-
Max-Forwards: 70
To: <sip:1001@192.168.0.1;transport=UDP>;tag=as3faf455c
From: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=f7352e7f
Call-ID: ODNmYTZlZTI0YmQyYjczOGIwMmZkNzM2NTFjODk2YmM.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'ODNmYTZlZTI0YmQyYjczOGIwMmZkNzM2NTFjODk2YmM.' Method: ACK
Scheduling destruction of SIP dialog 'NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:1000@192.168.0.2:5061;transport=UDP> for address/port to send to
set_destination: set destination to 192.168.0.2:5061
Reliably Transmitting (no NAT) to 192.168.0.2:5061:
BYE sip:1000@192.168.0.2:5061;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK100e9d66
Max-Forwards: 70
From: <sip:1001@192.168.0.1;transport=UDP>;tag=as302a27bb
To: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 102 BYE
User-Agent: FPBX-2.11.0(11.0.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---

<--- SIP read from UDP:192.168.0.2:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK100e9d66
Contact: <sip:1000@192.168.0.2:5061;transport=UDP>
To: "From Fax"<sip:1000@192.168.0.1;transport=UDP>;tag=4a328950
From: <sip:1001@192.168.0.1;transport=UDP>;tag=as302a27bb
Call-ID: NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.
CSeq: 102 BYE
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NmVlMDkwMTk2MzAwOWYwZDA1YmIwZDU1MzgzNzNlZmI.' Method: ACK
asterisk*CLI> sip set debug off
SIP Debugging Disabled
asterisk*CLI>

alex1980 · Сообщение **alex1980** » 10 окт 2013, 21:26

Также заметил что при звонке с Zoiper-а нет данного заголовка в пакете:

Код: Выделить всё

X-asterisk-Info: SIP re-invite (External RTP bridge)

С VentaFax данный заголовок присутствует.

Re-Invite-ы

Re-Invite-ы

Re: Re-Invite-ы

Re: Re-Invite-ы

Re: Re-Invite-ы

Re: Re-Invite-ы

Re: Re-Invite-ы

Re: Re-Invite-ы

Re: Re-Invite-ы

Re: Re-Invite-ы

Re: Re-Invite-ы