VIDEOCHAT  ::   FAQ  ::   Поиск  ::   Регистрация  ::   Вход

BYE после SDP

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

Ответить
r1sh
Сообщения: 8
Зарегистрирован: 05 сен 2013, 07:21

BYE после SDP

Сообщение r1sh »

Добрый день!

Делаю базовую настройку нового сервера. Конфиги следующего содержания:

sip.conf:

Код: Выделить всё

;; SIP Provider
[Provider]
type=peer
username=*
defaultuser=*
fromuser=*
secret=*
host=80.87.*.*
nat=yes
disallow=all
allow=alaw
allow=ulaw
allow=g729
insecure=port,invite
qualify=yes

;;Template for local peers
[office]
type=friend
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=alaw
allow=g729
allow=ulaw
context=outgoing

;;Local peers
[100](office)
username=100
secret=qwe123qwe123
callerid="100" <100>
call-limit=2

[102](office)
username=102
secret=qwe123qwe123
callerid="102" <102>
call-limit=2
extensions.conf:

Код: Выделить всё

[general]
static=yes
writeprotect=no
clearglobalvars=no

[local]
exten => _XXX,1,Dial(SIP/${EXTEN},Tt)
exten => _XXX,n,Hangup

[outgoing]
exten => _XXXXXXXXXXX,1,Dial(SIP/lek/${EXTEN},180,T)
exten => _XXXXXXXX,1,Dial(SIP/lek/${EXTEN},180,T)
exten => _XXXXXXX,1,Dial(SIP/lek/${EXTEN},180,T)
Клиент регистрируется, при звонке на любой внешний номер идет обмен 10ю пакетами.
Он почему-то посылает BYE в момент инициализации звонка.

Код: Выделить всё

<--- SIP read from UDP:80.87.204.248:52046 --->
REGISTER sip:80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:52046;branch=z9hG4bK-d8754z-f67fb4096e188b24-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.48.172:52046;rinstance=e42d525fb0efdce0>
To: "100"<sip:100@80.87.204.246:5060>
From: "100"<sip:100@80.87.204.246:5060>;tag=3e116758
Call-ID: ZDcxMWE0OTI0MDA3M2Q3NGViZDU1NDQzM2MwYmY0NGU.
CSeq: 25 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="100",realm="asterisk",nonce="4688ae92",uri="sip:80.87.204.246:5060",response="f1cdef502c1f3b8337f698b1e7eff5d4",algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 80.87.204.248:52046 (NAT)

<--- Transmitting (NAT) to 80.87.204.248:52046 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.48.172:52046;branch=z9hG4bK-d8754z-f67fb4096e188b24-1---d8754z-;received=80.87.204.248;rport=52046
From: "100"<sip:100@80.87.204.246:5060>;tag=3e116758
To: "100"<sip:100@80.87.204.246:5060>;tag=as7d191680
Call-ID: ZDcxMWE0OTI0MDA3M2Q3NGViZDU1NDQzM2MwYmY0NGU.
CSeq: 25 REGISTER
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="71890a99"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZDcxMWE0OTI0MDA3M2Q3NGViZDU1NDQzM2MwYmY0NGU.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:80.87.204.248:52046 --->
REGISTER sip:80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:52046;branch=z9hG4bK-d8754z-ba00cf7084144126-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.48.172:52046;rinstance=e42d525fb0efdce0>
To: "100"<sip:100@80.87.204.246:5060>
From: "100"<sip:100@80.87.204.246:5060>;tag=3e116758
Call-ID: ZDcxMWE0OTI0MDA3M2Q3NGViZDU1NDQzM2MwYmY0NGU.
CSeq: 26 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="100",realm="asterisk",nonce="71890a99",uri="sip:80.87.204.246:5060",response="002c08c442f44e6c167816edb60c9bab",algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 80.87.204.248:52046 (NAT)
Reliably Transmitting (NAT) to 80.87.204.248:52046:
OPTIONS sip:100@192.168.48.172:52046;rinstance=e42d525fb0efdce0 SIP/2.0
Via: SIP/2.0/UDP 80.87.204.246:5060;branch=z9hG4bK2ead5b76;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@80.87.204.246>;tag=as19345089
To: <sip:100@192.168.48.172:52046;rinstance=e42d525fb0efdce0>
Contact: <sip:asterisk@80.87.204.246:5060>
Call-ID: 3ae9d00335dffdd65d4798da21145ce0@80.87.204.246:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.0.0-rc1
Date: Thu, 10 Oct 2013 16:21:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 80.87.204.248:52046 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.48.172:52046;branch=z9hG4bK-d8754z-ba00cf7084144126-1---d8754z-;received=80.87.204.248;rport=52046
From: "100"<sip:100@80.87.204.246:5060>;tag=3e116758
To: "100"<sip:100@80.87.204.246:5060>;tag=as7d191680
Call-ID: ZDcxMWE0OTI0MDA3M2Q3NGViZDU1NDQzM2MwYmY0NGU.
CSeq: 26 REGISTER
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: <sip:100@192.168.48.172:52046;rinstance=e42d525fb0efdce0>;expires=120
Date: Thu, 10 Oct 2013 16:21:45 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZDcxMWE0OTI0MDA3M2Q3NGViZDU1NDQzM2MwYmY0NGU.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:80.87.204.248:52046 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.87.204.246:5060;branch=z9hG4bK2ead5b76;rport=5060
Contact: <sip:192.168.48.172:52046>
To: <sip:100@192.168.48.172:52046;rinstance=e42d525fb0efdce0>;tag=5c16551c
From: "asterisk"<sip:asterisk@80.87.204.246>;tag=as19345089
Call-ID: 3ae9d00335dffdd65d4798da21145ce0@80.87.204.246:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '3ae9d00335dffdd65d4798da21145ce0@80.87.204.246:5060' Method: OPTIONS
    -- SIP/lek-0000001b is making progress passing it to SIP/100-0000001a
Audio is at 16852
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 80.87.204.248:52046 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.48.172:52046;branch=z9hG4bK-d8754z-a72c912501345e2b-1---d8754z-;received=80.87.204.248;rport=52046
From: "100"<sip:100@80.87.204.246:5060>;tag=a646042a
To: <sip:89500201650@80.87.204.246:5060>;tag=as1d3373f7
Call-ID: NzhhY2Q5MTJmODJhNTQ4MTI4MmJiMzRjMWUwZTZiNWM.
CSeq: 2 INVITE
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:89500201650@80.87.204.246:5060>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1952354595 1952354595 IN IP4 80.87.204.246
s=Asterisk PBX 11.0.0-rc1
c=IN IP4 80.87.204.246
t=0 0
m=audio 16852 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=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34
<------------>

<--- SIP read from UDP:80.87.204.248:52046 --->
BYE sip:89500201650@80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:52046;branch=z9hG4bK-d8754z-230b0828a32baf17-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.48.172:52046;rinstance=e42d525fb0efdce0>
To: <sip:89500201650@80.87.204.246:5060>;tag=as1d3373f7
From: "100"<sip:100@80.87.204.246:5060>;tag=a646042a
Call-ID: NzhhY2Q5MTJmODJhNTQ4MTI4MmJiMzRjMWUwZTZiNWM.
CSeq: 3 BYE
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="100",realm="asterisk",nonce="1db8288b",uri="sip:89500201650@80.87.204.246:5060",response="b94cf666687b72671f38a23372a5ff4e",algorithm=MD5
Content-Length: 0

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

<--- Reliably Transmitting (NAT) to 80.87.204.248:52046 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.48.172:52046;branch=z9hG4bK-d8754z-a72c912501345e2b-1---d8754z-;received=80.87.204.248;rport=52046
From: "100"<sip:100@80.87.204.246:5060>;tag=a646042a
To: <sip:89500201650@80.87.204.246:5060>;tag=as1d3373f7
Call-ID: NzhhY2Q5MTJmODJhNTQ4MTI4MmJiMzRjMWUwZTZiNWM.
CSeq: 2 INVITE
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Sending to 80.87.204.248:52046 (NAT)
Scheduling destruction of SIP dialog 'NzhhY2Q5MTJmODJhNTQ4MTI4MmJiMzRjMWUwZTZiNWM.' in 6784 ms (Method: BYE)

<--- Transmitting (NAT) to 80.87.204.248:52046 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.48.172:52046;branch=z9hG4bK-d8754z-230b0828a32baf17-1---d8754z-;received=80.87.204.248;rport=52046
From: "100"<sip:100@80.87.204.246:5060>;tag=a646042a
To: <sip:89500201650@80.87.204.246:5060>;tag=as1d3373f7
Call-ID: NzhhY2Q5MTJmODJhNTQ4MTI4MmJiMzRjMWUwZTZiNWM.
CSeq: 3 BYE
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (outgoing, 89500201650, 1) exited non-zero on 'SIP/100-0000001a'

<--- SIP read from UDP:80.87.204.248:52046 --->
ACK sip:89500201650@80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:52046;branch=z9hG4bK-d8754z-a72c912501345e2b-1---d8754z-;rport
Max-Forwards: 70
To: <sip:89500201650@80.87.204.246:5060>;tag=as1d3373f7
From: "100"<sip:100@80.87.204.246:5060>;tag=a646042a
Call-ID: NzhhY2Q5MTJmODJhNTQ4MTI4MmJiMzRjMWUwZTZiNWM.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'NzhhY2Q5MTJmODJhNTQ4MTI4MmJiMzRjMWUwZTZiNWM.' Method: BYE

<--- SIP read from UDP:80.87.204.248:52046 --->


<------------->
Т.е. гудок буквально проходит и все.

Подскажите, пожалуйста, в чем может быть проблема?
Вложения
sip.jpg
Вернуться к началу
ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: BYE после SDP

Сообщение ded »

;; SIP Provider
[Provider]
type=peer
username=*
defaultuser=*
fromuser=*
secret=*
host=80.87.*.*
context=????
nat=yes ; это навряд ли
disallow=all
allow=alaw
allow=ulaw
allow=g729
insecure=port,invite
qualify=yes

по поводу дебага: приходит
BYE от софтфона
3CXPhone 6.0.26523.0
думаю потому, что там m=video 0 RTP/AVP 34
попробуйте отключить видео.
Вернуться к началу
r1sh
Сообщения: 8
Зарегистрирован: 05 сен 2013, 07:21

Re: BYE после SDP

Сообщение r1sh »

Спасибо за ответ!

сделал:

context=outgoing
nat=no

Да вот этот BYE меня смущает, у меня этот софтфон зареген на другом астериске с продакшана, там все ок, а тут не хочет....ровно 10 пакетов каждый раз идет обмен
Вернуться к началу
ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: BYE после SDP

Сообщение ded »

Начинайте с эхо-теста.

контекст outgoing? При таком раскладе минмальный трабл - поймаете петлю (всё что приходит от провайдера туда же сразу и уйдёт автоматом)
и максимальный трабл - вас будут использовать как бесплатный телефон дозвона межгород и за рубеж.
Вернуться к началу
r1sh
Сообщения: 8
Зарегистрирован: 05 сен 2013, 07:21

Re: BYE после SDP

Сообщение r1sh »

хм...сделал номер 101.

Добавил в extensions.conf:

[outgoing]
exten => 101,1,Playback(demo-echotest)

Код: Выделить всё

<--- SIP read from UDP:80.87.204.248:64980 --->
INVITE sip:101@80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-0b04df039521aa61-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.48.172:64980;rinstance=12bd9902cd4b8210>
To: <sip:101@80.87.204.246:5060>
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 409

v=0
o=3cxVCE 372794310 52404525 IN IP4 192.168.48.172
s=3cxVCE Audio Call
c=IN IP4 192.168.48.172
t=0 0
m=audio 40048 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40010 RTP/AVP 34
c=IN IP4 192.168.48.172
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
Sending to 80.87.204.248:64980 (NAT)
Using INVITE request as basis request - OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
Found peer '100' for '100' from 80.87.204.248:64980

<--- Reliably Transmitting (NAT) to 80.87.204.248:64980 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-0b04df039521aa61-1---d8754z-;received=80.87.204.248;rport=64980
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
To: <sip:101@80.87.204.246:5060>;tag=as0ea04292
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 1 INVITE
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7ef2ac9f"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.' in 7296 ms (Method: INVITE)

<--- SIP read from UDP:80.87.204.248:64980 --->
ACK sip:101@80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-0b04df039521aa61-1---d8754z-;rport
Max-Forwards: 70
To: <sip:101@80.87.204.246:5060>;tag=as0ea04292
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:80.87.204.248:64980 --->
INVITE sip:101@80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-4261ee3d37331d28-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.48.172:64980;rinstance=12bd9902cd4b8210>
To: <sip:101@80.87.204.246:5060>
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="100",realm="asterisk",nonce="7ef2ac9f",uri="sip:101@80.87.204.246:5060",response="e769fef9a012f4378b47dacba9b59459",algorithm=MD5
Content-Length: 409

v=0
o=3cxVCE 372794310 52404525 IN IP4 192.168.48.172
s=3cxVCE Audio Call
c=IN IP4 192.168.48.172
t=0 0
m=audio 40048 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40010 RTP/AVP 34
c=IN IP4 192.168.48.172
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 80.87.204.248:64980 (NAT)
Using INVITE request as basis request - OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
Found peer '100' for '100' from 80.87.204.248:64980
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
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 GSM for ID 3
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - (ulaw|alaw|g729), peer - audio=(gsm|ulaw|alaw)/video=(h263)/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 192.168.48.172:40048
Looking for 101 in outgoing (domain 80.87.204.246)
list_route: hop: <sip:100@192.168.48.172:64980;rinstance=12bd9902cd4b8210>

<--- Transmitting (NAT) to 80.87.204.248:64980 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-4261ee3d37331d28-1---d8754z-;received=80.87.204.248;rport=64980
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
To: <sip:101@80.87.204.246:5060>
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 2 INVITE
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:101@80.87.204.246:5060>
Content-Length: 0


<------------>
Audio is at 18060
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 80.87.204.248:64980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-4261ee3d37331d28-1---d8754z-;received=80.87.204.248;rport=64980
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
To: <sip:101@80.87.204.246:5060>;tag=as33e63463
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 2 INVITE
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:101@80.87.204.246:5060>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1308243694 1308243694 IN IP4 80.87.204.246
s=Asterisk PBX 11.0.0-rc1
c=IN IP4 80.87.204.246
t=0 0
m=audio 18060 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=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34
<------------>
Retransmitting #1 (NAT) to 80.87.204.248:64980:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-4261ee3d37331d28-1---d8754z-;received=80.87.204.248;rport=64980
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
To: <sip:101@80.87.204.246:5060>;tag=as33e63463
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 2 INVITE
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:101@80.87.204.246:5060>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1308243694 1308243694 IN IP4 80.87.204.246
s=Asterisk PBX 11.0.0-rc1
c=IN IP4 80.87.204.246
t=0 0
m=audio 18060 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=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34
---

<--- SIP read from UDP:80.87.204.248:64980 --->
ACK sip:101@80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-5e05575c24360152-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.48.172:64980;rinstance=12bd9902cd4b8210>
To: <sip:101@80.87.204.246:5060>;tag=as33e63463
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 2 ACK
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="100",realm="asterisk",nonce="7ef2ac9f",uri="sip:101@80.87.204.246:5060",response="e769fef9a012f4378b47dacba9b59459",algorithm=MD5
Content-Length: 0

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

<--- SIP read from UDP:80.87.204.248:64980 --->
BYE sip:101@80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-8a613866c47b8e34-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.48.172:64980;rinstance=12bd9902cd4b8210>
To: <sip:101@80.87.204.246:5060>;tag=as33e63463
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 3 BYE
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="100",realm="asterisk",nonce="7ef2ac9f",uri="sip:101@80.87.204.246:5060",response="2b1c7a880d9d840cf1c94def3dd30c58",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 80.87.204.248:64980 (NAT)
Scheduling destruction of SIP dialog 'OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.' in 7296 ms (Method: BYE)

<--- Transmitting (NAT) to 80.87.204.248:64980 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-8a613866c47b8e34-1---d8754z-;received=80.87.204.248;rport=64980
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
To: <sip:101@80.87.204.246:5060>;tag=as33e63463
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 3 BYE
Server: Asterisk PBX 11.0.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:80.87.204.248:64980 --->
ACK sip:101@80.87.204.246:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.48.172:64980;branch=z9hG4bK-d8754z-5e05575c24360152-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:100@192.168.48.172:64980;rinstance=12bd9902cd4b8210>
To: <sip:101@80.87.204.246:5060>;tag=as33e63463
From: "100"<sip:100@80.87.204.246:5060>;tag=347bc631
Call-ID: OTEzYzY3N2JkNzA4M2RhODZmNTJlMTE4MDI5NGM1NDE.
CSeq: 2 ACK
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="100",realm="asterisk",nonce="7ef2ac9f",uri="sip:101@80.87.204.246:5060",response="e769fef9a012f4378b47dacba9b59459",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Вернуться к началу
ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: BYE после SDP

Сообщение ded »

Видно: телефон посылает Астериску
m=video 40010 RTP/AVP 34
в инвайте, а Астериск его не понимает, и отправляет ему (или кому нибудь дальше)
m=video 0 RTP/AVP 34
Вернуться к началу
Ответить

Вернуться в «Вопросы новичков»

© 2008 — 2025 Asterisk.ru
Digium, Asterisk and AsteriskNOW are registered trademarks of Digium, Inc.
Design and development by PostMet-Netzwerk GmbH