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Не слышно друг друга

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

Модераторы: april22, Zavr2008

Ответить
sip_novichek
Сообщения: 21
Зарегистрирован: 05 июн 2013, 12:48

Re: Не слышно друг друга

Сообщение sip_novichek »

Я же написал:
to_virus_net
что не работает.

Код: Выделить всё

 * Name       : 101
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : office
  Language     : ru
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  MOH Suggest  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 2147483647
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : 3329
  Insecure     : no
  Force rport  : No
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       :
  Addr->IP     : 192.168.9.101:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 101
  SIP Options  : (none)
  Codecs       : 0x10c (ulaw|alaw|g729)
  Codec Order  : (ulaw:20,alaw:20,g729:20)
  Auto-Framing :  No
  Status       : OK (29 ms)
  Useragent    : Linksys/SPA921-5.1.8
  Reg. Contact : sip:101@192.168.9.101:5060
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No
Вот и я Вам все об этом же, что не работает, что при параметрах которым советуете, всё равно

Код: Выделить всё

Got  RTP packet from    192.168.9.101:16420 (type 00, seq 006227, ts 284687092, len 000160)
Sent RTP packet to      192.168.1.1:17186 (type 00, seq 059618, ts 284687088, len 000160)
Got  RTP packet from    192.168.9.101:16420 (type 00, seq 006228, ts 284687252, len 000160)
Sent RTP packet to      192.168.1.1:17186 (type 00, seq 059619, ts 284687248, len 000160)
Got  RTP packet from    192.168.9.101:16420 (type 00, seq 006229, ts 284687412, len 000160)
Я их сам все пробовал, но не работает, поэтому и обратился на форум.
Вернуться к началу
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Не слышно друг друга

Сообщение Vlad1983 »

последний шанс:
выставить на Linksys
RTP Packet Size: 0.020 (обычно делают в первую очередь)
ЛС: @rostel
Вернуться к началу
sip_novichek
Сообщения: 21
Зарегистрирован: 05 июн 2013, 12:48

Re: Не слышно друг друга

Сообщение sip_novichek »

Выставил, но всё тоже самое.
Вернуться к началу
virus_net
Сообщения: 2337
Зарегистрирован: 05 июн 2013, 08:12
Откуда: Москва

Re: Не слышно друг друга

Сообщение virus_net »

sip_novichek писал(а):Вот и я Вам все об этом же, что не работает, что при параметрах которым советуете, всё равно
а мы тебе все о том же, что при таких настройках не должно быть прямого RTP, а раз он по прежнему есть, то значит что-то у тя все же настроено не так ;)
Покажи весь свой sip.conf и снова покажи дебаг сип вызова.
И как по мне было бы лучше смотреть его tcpdump`ом, а не в консоли астера:

Код: Выделить всё

# tcpdump -nvs0 -i IFACE_NAME port 5060
где IFACE_NAME это имя интерфейса сервера, через который проходит вызов.

А так же оставь тока один кодек ulaw в шаблоне office.
мой SIP URI sip:virus_net@asterisk.ru
bitname.ru - Домены .bit (namecoin) .emc .coin .lib .bazar (emercoin)
ENUMER - звони бесплатно и напрямую.
Вернуться к началу
sip_novichek
Сообщения: 21
Зарегистрирован: 05 июн 2013, 12:48

Re: Не слышно друг друга

Сообщение sip_novichek »

Сейчас так:
sip.conf

Код: Выделить всё

[general]
context=default
allowguest=no
allowoverlap=no
localnet=192.168.1.0/24
localnet=192.168.5.0/24
localnet=192.168.9.0/24
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
directmedia=no
nat=no
language=ru
callevents=yes

[addpac]
type=peer
defaultuser=addpac
password=123321
host=192.168.1.1
context=potok
insecure=invite,port
disallow=all
allow=ulaw
canreinvite=no
qualify=yes

[office]
type=peer
host=dynamic
canreinvite=no
context=telephone
callcounter=yes
secret=2332
disallow=all
allow=ulaw
canreinvite=no
qualify=yes

[100](office)
[101](office)
[102](office)
[103](office)
[104](office)
[105](office)
[106](office)
[107](office)
[108](office)
[109](office)
sip debug

Код: Выделить всё

<--- SIP read from UDP:192.168.9.101:5060 --->
SIP/2.0 200 OK
To: <sip:101@192.168.9.101:5060>;tag=a0acb4a612f7c0c8i0
From: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK0fbcea76
Contact: "User1" <sip:101@192.168.9.101:5060>
Server: Linksys/SPA921-5.1.8
Content-Length: 210
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 1256119 1256119 IN IP4 192.168.9.101
s=-
c=IN IP4 192.168.9.101
t=0 0
m=audio 16440 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.9.101:16440
list_route: hop: <sip:101@192.168.9.101:5060>
set_destination: Parsing <sip:101@192.168.9.101:5060> for address/port to send to
set_destination: set destination to 192.168.9.101:5060
Transmitting (no NAT) to 192.168.9.101:5060:
ACK sip:101@192.168.9.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK5fabe114
Max-Forwards: 70
From: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
To: <sip:101@192.168.9.101:5060>;tag=a0acb4a612f7c0c8i0
Contact: <sip:201@192.168.5.254:5060>
Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.22.0 
Content-Length: 0


---
    -- SIP/101-00000005 answered SIP/cisco-00000004
Audio is at 12384
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3FC129C;received=192.168.1.1
From: <sip:201@192.168.1.1>;tag=4269F36C-2153
To: <sip:101@192.168.5.254>;tag=as4c5e395b
Call-ID: FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.22.0 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:101@192.168.5.254:5060>
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1830088190 1830088190 IN IP4 192.168.5.254
s=Asterisk PBX 1.8.22.0 
c=IN IP4 192.168.5.254
t=0 0
m=audio 12384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<--- SIP read from UDP:192.168.1.1:64454 --->
ACK sip:101@192.168.5.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3FD16E6
From: <sip:201@192.168.1.1>;tag=4269F36C-2153
To: <sip:101@192.168.5.254>;tag=as4c5e395b
Date: Wed, 05 Jun 2013 12:05:05 GMT
Call-ID: FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.9.101:5060 --->
BYE sip:201@192.168.5.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.9.101:5060;branch=z9hG4bK-6d1f4502
From: <sip:101@192.168.9.101:5060>;tag=a0acb4a612f7c0c8i0
To: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/SPA921-5.1.8
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.9.101:5060 (no NAT)
Scheduling destruction of SIP dialog '6fc065473312f673090ef3882d0af102@192.168.5.254:5060' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.9.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.9.101:5060;branch=z9hG4bK-6d1f4502;received=192.168.9.101
From: <sip:101@192.168.9.101:5060>;tag=a0acb4a612f7c0c8i0
To: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
CSeq: 101 BYE
Server: Asterisk PBX 1.8.22.0 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (cisco-phones, 101, 1) exited non-zero on 'SIP/cisco-00000004'
Scheduling destruction of SIP dialog 'FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:201@192.168.1.1:5060> for address/port to send to
set_destination: set destination to 192.168.1.1:5060
Reliably Transmitting (no NAT) to 192.168.1.1:5060:
BYE sip:201@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK4d867a6d
Max-Forwards: 70
From: <sip:101@192.168.5.254>;tag=as4c5e395b
To: <sip:201@192.168.1.1>;tag=4269F36C-2153
Call-ID: FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.22.0 
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK4d867a6d
From: <sip:101@192.168.5.254>;tag=as4c5e395b
To: <sip:201@192.168.1.1>;tag=4269F36C-2153
Date: Wed, 05 Jun 2013 12:05:10 GMT
Call-ID: FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 102 BYE

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1' Method: ACK
Reliably Transmitting (no NAT) to 192.168.1.1:5060:
OPTIONS sip:192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK6446d8f1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.5.254>;tag=as3bc7bc40
To: <sip:192.168.1.1>
Contact: <sip:asterisk@192.168.5.254:5060>
Call-ID: 3c6def870e534fa7229d347e397e67c3@192.168.5.254:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.22.0 
Date: Wed, 05 Jun 2013 12:05:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (no NAT) to 192.168.9.101:5060:
OPTIONS sip:101@192.168.9.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK120bb9f8
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.5.254>;tag=as6944fc7a
To: <sip:101@192.168.9.101:5060>
Contact: <sip:asterisk@192.168.5.254:5060>
Call-ID: 7e87f06f1d2f086441dc79985012a588@192.168.5.254:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.22.0 
Date: Wed, 05 Jun 2013 12:05:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK6446d8f1
From: "asterisk" <sip:asterisk@192.168.5.254>;tag=as3bc7bc40
To: <sip:192.168.1.1>;tag=426A17FC-1C1D
Date: Wed, 05 Jun 2013 12:05:14 GMT
Call-ID: 3c6def870e534fa7229d347e397e67c3@192.168.5.254:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 167
Content-Type: application/sdp

v=0
o=CiscoSystemsSIP-GW-UserAgent 8889 2366 IN IP4 192.168.1.1
s=SIP Call
c=IN IP4 192.168.1.1
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 192.168.1.1
<------------->
--- (14 headers 7 lines) ---
Really destroying SIP dialog '3c6def870e534fa7229d347e397e67c3@192.168.5.254:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.9.101:5060 --->
SIP/2.0 200 OK
To: <sip:101@192.168.9.101:5060>;tag=9979c86ab422f7bci0
From: "asterisk" <sip:asterisk@192.168.5.254>;tag=as6944fc7a
Call-ID: 7e87f06f1d2f086441dc79985012a588@192.168.5.254:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK120bb9f8
Server: Linksys/SPA921-5.1.8
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7e87f06f1d2f086441dc79985012a588@192.168.5.254:5060' Method: OPTIONS
tcpdump

Код: Выделить всё

       Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK0fbcea76
        Max-Forwards: 70
        From: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
        To: <sip:101@192.168.9.101:5060>
        Contact: <sip:201@192.168.5.254:5060>
        Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX 1.8.22.0 
        Date: Wed, 05 Jun 2013 12:05:47 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 237

        v=0
        o=root 1197840599 1197840599 IN IP4 192.168.5.254
        s=Asterisk PBX 1.8.22.0 
        c=IN IP4 192.168.5.254
        t=0 0
        m=audio 10704 RTP/AVP 0 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv

16:05:47.797636 IP (tos 0x0, ttl 245, id 3651, offset 0, flags [none], proto UDP (17), length 322)
    192.168.9.101.sip > 192.168.5.254.sip: SIP, length: 294
        SIP/2.0 100 Trying
        To: <sip:101@192.168.9.101:5060>
        From: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
        Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
        CSeq: 102 INVITE
        Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK0fbcea76
        Server: Linksys/SPA921-5.1.8
        Content-Length: 0


16:05:47.820560 IP (tos 0x0, ttl 245, id 3652, offset 0, flags [none], proto UDP (17), length 346)
    192.168.9.101.sip > 192.168.5.254.sip: SIP, length: 318
        SIP/2.0 180 Ringing
        To: <sip:101@192.168.9.101:5060>;tag=a0acb4a612f7c0c8i0
        From: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
        Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
        CSeq: 102 INVITE
        Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK0fbcea76
        Server: Linksys/SPA921-5.1.8
        Content-Length: 0


16:05:47.822245 IP (tos 0x0, ttl 64, id 43410, offset 0, flags [none], proto UDP (17), length 498)
    192.168.5.254.sip > 192.168.1.1.sip: SIP, length: 470
        SIP/2.0 180 Ringing
        Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3FC129C;received=192.168.1.1
        From: <sip:201@192.168.1.1>;tag=4269F36C-2153
        To: <sip:101@192.168.5.254>;tag=as4c5e395b
        Call-ID: FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1
        CSeq: 101 INVITE
        Server: Asterisk PBX 1.8.22.0 
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:101@192.168.5.254:5060>
        Content-Length: 0


16:05:49.577952 IP (tos 0x0, ttl 245, id 3653, offset 0, flags [none], proto UDP (17), length 718)
    192.168.9.101.sip > 192.168.5.254.sip: SIP, length: 690
        SIP/2.0 200 OK
        To: <sip:101@192.168.9.101:5060>;tag=a0acb4a612f7c0c8i0
        From: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
        Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
        CSeq: 102 INVITE
        Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK0fbcea76
        Contact: "Test-SPB" <sip:101@192.168.9.101:5060>
        Server: Linksys/SPA921-5.1.8
        Content-Length: 210
        Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
        Supported: replaces
        Content-Type: application/sdp

        v=0
        o=- 1256119 1256119 IN IP4 192.168.9.101
        s=-
        c=IN IP4 192.168.9.101
        t=0 0
        m=audio 16440 RTP/AVP 0 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-15
        a=ptime:20
        a=sendrecv

16:05:49.578482 IP (tos 0x0, ttl 64, id 59688, offset 0, flags [none], proto UDP (17), length 423)
    192.168.5.254.sip > 192.168.9.101.sip: SIP, length: 395
        ACK sip:101@192.168.9.101:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK5fabe114
        Max-Forwards: 70
        From: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
        To: <sip:101@192.168.9.101:5060>;tag=a0acb4a612f7c0c8i0
        Contact: <sip:201@192.168.5.254:5060>
        Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
        CSeq: 102 ACK
        User-Agent: Asterisk PBX 1.8.22.0 
        Content-Length: 0


16:05:49.580626 IP (tos 0x0, ttl 64, id 43411, offset 0, flags [none], proto UDP (17), length 763)
    192.168.5.254.sip > 192.168.1.1.sip: SIP, length: 735
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK3FC129C;received=192.168.1.1
        From: <sip:201@192.168.1.1>;tag=4269F36C-2153
        To: <sip:101@192.168.5.254>;tag=as4c5e395b
        Call-ID: FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1
        CSeq: 101 INVITE
        Server: Asterisk PBX 1.8.22.0 
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:101@192.168.5.254:5060>
        Content-Type: application/sdp
        Content-Length: 237

        v=0
        o=root 1830088190 1830088190 IN IP4 192.168.5.254
        s=Asterisk PBX 1.8.22.0 
        c=IN IP4 192.168.5.254
        t=0 0
        m=audio 12384 RTP/AVP 0 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv

16:05:49.618241 IP (tos 0x0, ttl 250, id 5, offset 0, flags [none], proto UDP (17), length 367)
    192.168.1.1.64454 > 192.168.5.254.sip: SIP, length: 339
        ACK sip:101@192.168.5.254:5060 SIP/2.0
        Via: SIP/2.0/UDP  192.168.1.1:5060;branch=z9hG4bK3FD16E6
        From: <sip:201@192.168.1.1>;tag=4269F36C-2153
        To: <sip:101@192.168.5.254>;tag=as4c5e395b
        Date: Wed, 05 Jun 2013 12:05:05 GMT
        Call-ID: FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1
        Max-Forwards: 70
        CSeq: 101 ACK
        Content-Length: 0


16:05:53.037087 IP (tos 0x0, ttl 245, id 3825, offset 0, flags [none], proto UDP (17), length 385)
    192.168.9.101.sip > 192.168.5.254.sip: SIP, length: 357
        BYE sip:201@192.168.5.254:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.9.101:5060;branch=z9hG4bK-6d1f4502
        From: <sip:101@192.168.9.101:5060>;tag=a0acb4a612f7c0c8i0
        To: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
        Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
        CSeq: 101 BYE
        Max-Forwards: 70
        User-Agent: Linksys/SPA921-5.1.8
        Content-Length: 0


16:05:53.037668 IP (tos 0x0, ttl 64, id 59689, offset 0, flags [none], proto UDP (17), length 475)
    192.168.5.254.sip > 192.168.9.101.sip: SIP, length: 447
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.9.101:5060;branch=z9hG4bK-6d1f4502;received=192.168.9.101
        From: <sip:101@192.168.9.101:5060>;tag=a0acb4a612f7c0c8i0
        To: "201" <sip:201@192.168.5.254>;tag=as0cacad9f
        Call-ID: 6fc065473312f673090ef3882d0af102@192.168.5.254:5060
        CSeq: 101 BYE
        Server: Asterisk PBX 1.8.22.0 
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Length: 0


16:05:53.042161 IP (tos 0x0, ttl 64, id 43412, offset 0, flags [none], proto UDP (17), length 438)
    192.168.5.254.sip > 192.168.1.1.sip: SIP, length: 410
        BYE sip:201@192.168.1.1:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK4d867a6d
        Max-Forwards: 70
        From: <sip:101@192.168.5.254>;tag=as4c5e395b
        To: <sip:201@192.168.1.1>;tag=4269F36C-2153
        Call-ID: FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1
        CSeq: 102 BYE
        User-Agent: Asterisk PBX 1.8.22.0 
        X-Asterisk-HangupCause: Normal Clearing
        X-Asterisk-HangupCauseCode: 16
        Content-Length: 0


16:05:53.069856 IP (tos 0x0, ttl 250, id 6874, offset 0, flags [none], proto UDP (17), length 361)
    192.168.1.1.sip > 192.168.5.254.sip: SIP, length: 333
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK4d867a6d
        From: <sip:101@192.168.5.254>;tag=as4c5e395b
        To: <sip:201@192.168.1.1>;tag=4269F36C-2153
        Date: Wed, 05 Jun 2013 12:05:10 GMT
        Call-ID: FE1559D5-CD0E11E2-A3EDE10E-7B6264BA@192.168.1.1
        Server: Cisco-SIPGateway/IOS-12.x
        Content-Length: 0
        CSeq: 102 BYE
Вернуться к началу
sip_novichek
Сообщения: 21
Зарегистрирован: 05 июн 2013, 12:48

Re: Не слышно друг друга

Сообщение sip_novichek »

Out писал(а):абоненту 101 выставили nat=no?
Да, выставлял. Я это делал до обращения на форум.
Так как сам не смог разобраться поэтому и обратился за помощью.
Вернуться к началу
virus_net
Сообщения: 2337
Зарегистрирован: 05 июн 2013, 08:12
Откуда: Москва

Re: Не слышно друг друга

Сообщение virus_net »

sip_novichek писал(а):tcpdump
такс, это гуд.
Видно что астер и 101 устанавливают RTP между собой.
А как RTP потом по факту пошел ? Опять по прежнему напрямую между 1.1 и 9.101 ?

А ты можешь сдампать тоже самое, но прям перед 192.168.9.101 ? Т.е. все что приходит/уходит от 192.168.9.101 по 5060 при вызове ? Что бы понять если идет REINVITE то от кого и когда.
мой SIP URI sip:virus_net@asterisk.ru
bitname.ru - Домены .bit (namecoin) .emc .coin .lib .bazar (emercoin)
ENUMER - звони бесплатно и напрямую.
Вернуться к началу
sip_novichek
Сообщения: 21
Зарегистрирован: 05 июн 2013, 12:48

Re: Не слышно друг друга

Сообщение sip_novichek »

Сообщение Out » 4 минуты назад

[office]
type=peer
host=dynamic
canreinvite=no
context=telephone
callcounter=yes
secret=2332
disallow=all
allow=ulaw
canreinvite=no
qualify=yes

[100](office)
[101](office)


ГДЕ ТУТ NAT=NO?

Я же написал, что пробовал это вариант ДО того как обратился на форум.
Вернуться к началу
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Не слышно друг друга

Сообщение Vlad1983 »

неполные вырезки из логов

дамп в файл

Код: Выделить всё

tcpdump -i any -vvvnn -s0 -w /tmp/asdasd.cap udp
ссыль сюда
ЛС: @rostel
Вернуться к началу
sip_novichek
Сообщения: 21
Зарегистрирован: 05 июн 2013, 12:48

Re: Не слышно друг друга

Сообщение sip_novichek »

to virus_net
Вот полностью весь tcpdump вывел в файл

Код: Выделить всё

17:00:18.896546 IP (tos 0x0, ttl 250, id 0, offset 0, flags [none], proto UDP (17), length 1072)
    192.168.1.1.61149 > 192.168.5.254.sip: SIP, length: 1044
	INVITE sip:101@192.168.5.254:5060 SIP/2.0
	Via: SIP/2.0/UDP  192.168.1.1:5060;branch=z9hG4bK400131A
	From: <sip:201@192.168.1.1>;tag=429BDD3C-2331
	To: <sip:101@192.168.5.254>
	Date: Wed, 05 Jun 2013 12:59:36 GMT
	Call-ID: 9BD5EC66-CD1611E2-A42EE10E-7B6264BA@192.168.1.1
	Supported: 100rel,timer,replaces
	Min-SE:  1800
	Cisco-Guid: 2613808117-3440775650-2152923159-1521367144
	User-Agent: Cisco-SIPGateway/IOS-12.x
	Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
	CSeq: 101 INVITE
	Max-Forwards: 70
	Remote-Party-ID: <sip:201@192.168.1.1>;party=calling;screen=no;privacy=off
	Timestamp: 1370437176
	Contact: <sip:201@192.168.1.1:5060>
	Expires: 180
	Allow-Events: telephone-event
	Content-Type: application/sdp
	Content-Length: 247
	
	v=0
	o=CiscoSystemsSIP-GW-UserAgent 9584 5340 IN IP4 192.168.1.1
	s=SIP Call
	c=IN IP4 192.168.1.1
	t=0 0
	m=audio 19276 RTP/AVP 0 101
	c=IN IP4 192.168.1.1
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	
17:00:18.897298 IP (tos 0x0, ttl 64, id 43475, offset 0, flags [none], proto UDP (17), length 482)
    192.168.5.254.sip > 192.168.1.1.sip: SIP, length: 454
	SIP/2.0 100 Trying
	Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK400131A;received=192.168.1.1
	From: <sip:201@192.168.1.1>;tag=429BDD3C-2331
	To: <sip:101@192.168.5.254>
	Call-ID: 9BD5EC66-CD1611E2-A42EE10E-7B6264BA@192.168.1.1
	CSeq: 101 INVITE
	Server: Asterisk PBX 1.8.22.0 
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Contact: <sip:101@192.168.5.254:5060>
	Content-Length: 0
	
	
17:00:18.902954 IP (tos 0x0, ttl 64, id 60130, offset 0, flags [none], proto UDP (17), length 824)
    192.168.5.254.sip > 192.168.9.101.sip: SIP, length: 796
	INVITE sip:101@192.168.9.101:5060 SIP/2.0
	Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK0254dc4b
	Max-Forwards: 70
	From: "201" <sip:201@192.168.5.254>;tag=as679ec9b0
	To: <sip:101@192.168.9.101:5060>
	Contact: <sip:201@192.168.5.254:5060>
	Call-ID: 1f433337058b66812516efa825640f01@192.168.5.254:5060
	CSeq: 102 INVITE
	User-Agent: Asterisk PBX 1.8.22.0 
	Date: Wed, 05 Jun 2013 13:00:18 GMT
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Content-Type: application/sdp
	Content-Length: 237
	
	v=0
	o=root 1739612066 1739612066 IN IP4 192.168.5.254
	s=Asterisk PBX 1.8.22.0 
	c=IN IP4 192.168.5.254
	t=0 0
	m=audio 16588 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:00:18.932830 IP (tos 0x0, ttl 245, id 12, offset 0, flags [none], proto UDP (17), length 322)
    192.168.9.101.sip > 192.168.5.254.sip: SIP, length: 294
	SIP/2.0 100 Trying
	To: <sip:101@192.168.9.101:5060>
	From: "201" <sip:201@192.168.5.254>;tag=as679ec9b0
	Call-ID: 1f433337058b66812516efa825640f01@192.168.5.254:5060
	CSeq: 102 INVITE
	Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK0254dc4b
	Server: Linksys/SPA921-5.1.8
	Content-Length: 0
	
	
17:00:18.963394 IP (tos 0x0, ttl 245, id 13, offset 0, flags [none], proto UDP (17), length 346)
    192.168.9.101.sip > 192.168.5.254.sip: SIP, length: 318
	SIP/2.0 180 Ringing
	To: <sip:101@192.168.9.101:5060>;tag=d0dd9cc1854231e3i0
	From: "201" <sip:201@192.168.5.254>;tag=as679ec9b0
	Call-ID: 1f433337058b66812516efa825640f01@192.168.5.254:5060
	CSeq: 102 INVITE
	Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK0254dc4b
	Server: Linksys/SPA921-5.1.8
	Content-Length: 0
	
	
17:00:18.964660 IP (tos 0x0, ttl 64, id 43476, offset 0, flags [none], proto UDP (17), length 498)
    192.168.5.254.sip > 192.168.1.1.sip: SIP, length: 470
	SIP/2.0 180 Ringing
	Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK400131A;received=192.168.1.1
	From: <sip:201@192.168.1.1>;tag=429BDD3C-2331
	To: <sip:101@192.168.5.254>;tag=as7f3a9e69
	Call-ID: 9BD5EC66-CD1611E2-A42EE10E-7B6264BA@192.168.1.1
	CSeq: 101 INVITE
	Server: Asterisk PBX 1.8.22.0 
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Contact: <sip:101@192.168.5.254:5060>
	Content-Length: 0
	
	
17:00:20.056896 IP (tos 0x0, ttl 245, id 14, offset 0, flags [none], proto UDP (17), length 714)
    192.168.9.101.sip > 192.168.5.254.sip: SIP, length: 686
	SIP/2.0 200 OK
	To: <sip:101@192.168.9.101:5060>;tag=d0dd9cc1854231e3i0
	From: "201" <sip:201@192.168.5.254>;tag=as679ec9b0
	Call-ID: 1f433337058b66812516efa825640f01@192.168.5.254:5060
	CSeq: 102 INVITE
	Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK0254dc4b
	Contact: "Test-SPB" <sip:101@192.168.9.101:5060>
	Server: Linksys/SPA921-5.1.8
	Content-Length: 206
	Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
	Supported: replaces
	Content-Type: application/sdp
	
	v=0
	o=- 23857 23857 IN IP4 192.168.9.101
	s=-
	c=IN IP4 192.168.9.101
	t=0 0
	m=audio 16434 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-15
	a=ptime:20
	a=sendrecv
	
17:00:20.057285 IP (tos 0x0, ttl 64, id 60131, offset 0, flags [none], proto UDP (17), length 423)
    192.168.5.254.sip > 192.168.9.101.sip: SIP, length: 395
	ACK sip:101@192.168.9.101:5060 SIP/2.0
	Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK39ef4201
	Max-Forwards: 70
	From: "201" <sip:201@192.168.5.254>;tag=as679ec9b0
	To: <sip:101@192.168.9.101:5060>;tag=d0dd9cc1854231e3i0
	Contact: <sip:201@192.168.5.254:5060>
	Call-ID: 1f433337058b66812516efa825640f01@192.168.5.254:5060
	CSeq: 102 ACK
	User-Agent: Asterisk PBX 1.8.22.0 
	Content-Length: 0
	
	
17:00:20.058268 IP (tos 0x0, ttl 64, id 43477, offset 0, flags [none], proto UDP (17), length 763)
    192.168.5.254.sip > 192.168.1.1.sip: SIP, length: 735
	SIP/2.0 200 OK
	Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK400131A;received=192.168.1.1
	From: <sip:201@192.168.1.1>;tag=429BDD3C-2331
	To: <sip:101@192.168.5.254>;tag=as7f3a9e69
	Call-ID: 9BD5EC66-CD1611E2-A42EE10E-7B6264BA@192.168.1.1
	CSeq: 101 INVITE
	Server: Asterisk PBX 1.8.22.0 
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Contact: <sip:101@192.168.5.254:5060>
	Content-Type: application/sdp
	Content-Length: 237
	
	v=0
	o=root 1635001114 1635001114 IN IP4 192.168.5.254
	s=Asterisk PBX 1.8.22.0 
	c=IN IP4 192.168.5.254
	t=0 0
	m=audio 10960 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=ptime:20
	a=sendrecv
	
17:00:20.138882 IP (tos 0x0, ttl 250, id 1, offset 0, flags [none], proto UDP (17), length 366)
    192.168.1.1.61149 > 192.168.5.254.sip: SIP, length: 338
	ACK sip:101@192.168.5.254:5060 SIP/2.0
	Via: SIP/2.0/UDP  192.168.1.1:5060;branch=z9hG4bK401B5C
	From: <sip:201@192.168.1.1>;tag=429BDD3C-2331
	To: <sip:101@192.168.5.254>;tag=as7f3a9e69
	Date: Wed, 05 Jun 2013 12:59:36 GMT
	Call-ID: 9BD5EC66-CD1611E2-A42EE10E-7B6264BA@192.168.1.1
	Max-Forwards: 70
	CSeq: 101 ACK
	Content-Length: 0
	
	
17:00:23.397897 IP (tos 0x0, ttl 245, id 180, offset 0, flags [none], proto UDP (17), length 385)
    192.168.9.101.sip > 192.168.5.254.sip: SIP, length: 357
	BYE sip:201@192.168.5.254:5060 SIP/2.0
	Via: SIP/2.0/UDP 192.168.9.101:5060;branch=z9hG4bK-acc73c75
	From: <sip:101@192.168.9.101:5060>;tag=d0dd9cc1854231e3i0
	To: "201" <sip:201@192.168.5.254>;tag=as679ec9b0
	Call-ID: 1f433337058b66812516efa825640f01@192.168.5.254:5060
	CSeq: 101 BYE
	Max-Forwards: 70
	User-Agent: Linksys/SPA921-5.1.8
	Content-Length: 0
	
	
17:00:23.398428 IP (tos 0x0, ttl 64, id 60132, offset 0, flags [none], proto UDP (17), length 475)
    192.168.5.254.sip > 192.168.9.101.sip: SIP, length: 447
	SIP/2.0 200 OK
	Via: SIP/2.0/UDP 192.168.9.101:5060;branch=z9hG4bK-acc73c75;received=192.168.9.101
	From: <sip:101@192.168.9.101:5060>;tag=d0dd9cc1854231e3i0
	To: "201" <sip:201@192.168.5.254>;tag=as679ec9b0
	Call-ID: 1f433337058b66812516efa825640f01@192.168.5.254:5060
	CSeq: 101 BYE
	Server: Asterisk PBX 1.8.22.0 
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
	Supported: replaces, timer
	Content-Length: 0
	
	
17:00:23.401955 IP (tos 0x0, ttl 64, id 43478, offset 0, flags [none], proto UDP (17), length 438)
    192.168.5.254.sip > 192.168.1.1.sip: SIP, length: 410
	BYE sip:201@192.168.1.1:5060 SIP/2.0
	Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK261dd481
	Max-Forwards: 70
	From: <sip:101@192.168.5.254>;tag=as7f3a9e69
	To: <sip:201@192.168.1.1>;tag=429BDD3C-2331
	Call-ID: 9BD5EC66-CD1611E2-A42EE10E-7B6264BA@192.168.1.1
	CSeq: 102 BYE
	User-Agent: Asterisk PBX 1.8.22.0 
	X-Asterisk-HangupCause: Normal Clearing
	X-Asterisk-HangupCauseCode: 16
	Content-Length: 0
	
	
17:00:23.436811 IP (tos 0x0, ttl 250, id 6935, offset 0, flags [none], proto UDP (17), length 361)
    192.168.1.1.sip > 192.168.5.254.sip: SIP, length: 333
	SIP/2.0 200 OK
	Via: SIP/2.0/UDP 192.168.5.254:5060;branch=z9hG4bK261dd481
	From: <sip:101@192.168.5.254>;tag=as7f3a9e69
	To: <sip:201@192.168.1.1>;tag=429BDD3C-2331
	Date: Wed, 05 Jun 2013 12:59:40 GMT
	Call-ID: 9BD5EC66-CD1611E2-A42EE10E-7B6264BA@192.168.1.1
	Server: Cisco-SIPGateway/IOS-12.x
	Content-Length: 0
	CSeq: 102 BYE
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