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Звонок обрывается через 18-20 секунд [закрыт]

0

Прошу помощи в решении проблемы. Есть asterisk 1.8.22.0. Без ната. Астрериск---роутер--свитч-телефоны(или софтфоны). Все порты 5060, 10000-20000 открыты. траффик ходит через ВПН. При звонках на внутренний номер устанавливается соединение, проходит 20 сек. и разрыв

[2014-06-12 12:20:13] WARNING[1637]: chan_sip.c:3979 retrans_pkt: Retransmission timeout reached on transmission ZWMzNTczMWRlMTU0ZWExMjZlMjI5NGY0NzdjYjU2OWQ. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 25985ms with no response
[2014-06-12 12:20:13] WARNING[1637]: chan_sip.c:4008 retrans_pkt: Hanging up call ZWMzNTczMWRlMTU0ZWExMjZlMjI5NGY0NzdjYjU2OWQ. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Скрины из wireshark: http://saveimg.ru/show-image.php?id=359f48c25679a91f5d7a905dd85108e8 http://saveimg.ru/show-image.php?id=697c10b9c4ddc85474ea78502080b167 Вместе http://saveimg.ru/show-image.php?id=b5293a69791717d76c2c61990df5e24d

Настройки peer: asterisk*CLI> sip show peer 3100

  * Name       : 3100
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : from-internal
  Subscr.Cont. : <Not set>
  Language     : 
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  MOH Suggest  : 
  Mailbox      : 3100@device
  VM Extension : *97
  LastMsgsSent : 0/0
  Call limit   : 2147483647
  Max forwards : 0
  Dynamic      : Yes
  Callerid     : "3100" <3100>
  MaxCallBR    : 384 kbps
  Expire       : 3596
  Insecure     : no
  Force rport  : No
  ACL          : Yes
  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   : Yes
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 
  Addr->IP     : 192.168.194.250:5620
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 3100
  SIP Options  : (none)
  Codecs       : 0xc (ulaw|alaw)
  Codec Order  : (ulaw:20,alaw:20)
  Auto-Framing :  No 
  Status       : OK (10 ms)
  Useragent    : X-Lite release 4.6.1 stamp 73073 ff7d322c-W
  Reg. Contact : sip:3100@192.168.194.250:5620;rinstance=71bccf3d02b51ac0
  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

в той сети только 1 комп(192.168.194.250) вот sip debug звонка softphone Zoiper(3101)-->Xlite(3100)

 == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 11812
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.194.250:5620:
INVITE sip:3100@192.168.194.250:5620;rinstance=84211f663855424d SIP/2.0
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK411c54d8
Max-Forwards: 70
From: "3101" <sip:3101@192.168.199.13>;tag=as16bfa2a2
To: <sip:3100@192.168.194.250:5620;rinstance=84211f663855424d>
Contact: <sip:3101@192.168.199.13:5060>
Call-ID: 15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.22.0)
Date: Thu, 12 Jun 2014 14:15:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2101538408 2101538408 IN IP4 192.168.199.13
s=Asterisk PBX 1.8.22.0
c=IN IP4 192.168.199.13
t=0 0
m=audio 11812 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/3100

<--- Transmitting (no NAT) to 192.168.194.250:38752 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-d1005562babd8466-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
To: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3100@192.168.199.13:5060>
Content-Length: 0


<------------>
    -- Connected line update to SIP/3101-00000016 prevented.
Retransmitting #1 (no NAT) to 192.168.194.250:5620:
INVITE sip:3100@192.168.194.250:5620;rinstance=84211f663855424d SIP/2.0
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK411c54d8
Max-Forwards: 70
From: "3101" <sip:3101@192.168.199.13>;tag=as16bfa2a2
To: <sip:3100@192.168.194.250:5620;rinstance=84211f663855424d>
Contact: <sip:3101@192.168.199.13:5060>
Call-ID: 15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.22.0)
Date: Thu, 12 Jun 2014 14:15:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2101538408 2101538408 IN IP4 192.168.199.13
s=Asterisk PBX 1.8.22.0
c=IN IP4 192.168.199.13
t=0 0
m=audio 11812 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.194.250:5620 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK411c54d8
To: <sip:3100@192.168.194.250:5620;rinstance=84211f663855424d>
From: "3101" <sip:3101@192.168.199.13>;tag=as16bfa2a2
Call-ID: 15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:192.168.194.250:5620 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK411c54d8
To: <sip:3100@192.168.194.250:5620;rinstance=84211f663855424d>
From: "3101" <sip:3101@192.168.199.13>;tag=as16bfa2a2
Call-ID: 15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:192.168.194.250:5620 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK411c54d8
Contact: <sip:3100@192.168.194.250:5620>
To: <sip:3100@192.168.194.250:5620;rinstance=84211f663855424d>;tag=8153f605
From: "3101"<sip:3101@192.168.199.13>;tag=as16bfa2a2
Call-ID: 15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 4.6.1 stamp 73073 ff7d322c-W
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:3100@192.168.194.250:5620>
    -- SIP/3100-00000017 is ringing

<--- Transmitting (no NAT) to 192.168.194.250:38752 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-d1005562babd8466-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
To: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3100@192.168.199.13:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.194.250:5620 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK411c54d8
Contact: <sip:3100@192.168.194.250:5620>
To: <sip:3100@192.168.194.250:5620;rinstance=84211f663855424d>;tag=8153f605
From: "3101"<sip:3101@192.168.199.13>;tag=as16bfa2a2
Call-ID: 15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.6.1 stamp 73073 ff7d322c-W
Content-Length: 217

v=0
o=- 13047056130267137 3 IN IP4 192.168.194.250
s=X-Lite release 4.6.1 stamp 73073
c=IN IP4 192.168.194.250
t=0 0
m=audio 55964 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.194.250:55964
list_route: hop: <sip:3100@192.168.194.250:5620>
set_destination: Parsing <sip:3100@192.168.194.250:5620> for address/port to send to
set_destination: set destination to 192.168.194.250:5620
Transmitting (no NAT) to 192.168.194.250:5620:
ACK sip:3100@192.168.194.250:5620 SIP/2.0
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK5bdff269
Max-Forwards: 70
From: "3101" <sip:3101@192.168.199.13>;tag=as16bfa2a2
To: <sip:3100@192.168.194.250:5620;rinstance=84211f663855424d>;tag=8153f605
Contact: <sip:3101@192.168.199.13:5060>
Call-ID: 15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(1.8.22.0)
Content-Length: 0


---
    -- Connected line update to SIP/3101-00000016 prevented.
    -- SIP/3100-00000017 answered SIP/3101-00000016
Audio is at 10338
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.194.250:38752 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-d1005562babd8466-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
To: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3100@192.168.199.13:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1187272524 1187272524 IN IP4 192.168.199.13
s=Asterisk PBX 1.8.22.0
c=IN IP4 192.168.199.13
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (no NAT) to 192.168.194.250:38752:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-d1005562babd8466-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
To: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3100@192.168.199.13:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1187272524 1187272524 IN IP4 192.168.199.13
s=Asterisk PBX 1.8.22.0
c=IN IP4 192.168.199.13
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.194.250:38752 --->
PUBLISH sip:3101@192.168.199.13;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-e435177cbede6c16-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:3101@118.200.191.119:38752;transport=UDP>
To: "3101"<sip:3101@192.168.199.13;transport=UDP>
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=e9613901
Call-ID: OWFkMzE5ZmExMmRhYzZhODc3MWI0YjEyZTgwOTZmNmI.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Event: presence
Allow-Events: presence, kpml
Content-Length: 268

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:3101@192.168.199.13;transport=UDP"> <tuple id="3101" > <status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---

<--- Transmitting (no NAT) to 192.168.194.250:38752 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-e435177cbede6c16-1---d8754z-;rport;received=192.168.194.250
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=e9613901
To: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=as785db222
Call-ID: OWFkMzE5ZmExMmRhYzZhODc3MWI0YjEyZTgwOTZmNmI.
CSeq: 1 PUBLISH
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'OWFkMzE5ZmExMmRhYzZhODc3MWI0YjEyZTgwOTZmNmI.' Method: PUBLISH

<--- SIP read from UDP:192.168.194.250:38752 --->
SUBSCRIBE sip:3101@192.168.199.13;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-c62a51bdecac6dfd-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:3101@118.200.191.119:38752;transport=UDP>
To: "3101"<sip:3101@192.168.199.13;transport=UDP>
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=5b267f5d
Call-ID: ZDU1YzUzZmE2ZjMxYjVjZDllZjYzZTk3OTg4NzQwYmY.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Creating new subscription
Sending to 192.168.194.250:38752 (no NAT)
list_route: hop: <sip:3101@118.200.191.119:38752;transport=UDP>
Found peer '3101' for '3101' from 192.168.194.250:38752

<--- Transmitting (no NAT) to 192.168.194.250:38752 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-c62a51bdecac6dfd-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=5b267f5d
To: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=as5d77a6a8
Call-ID: ZDU1YzUzZmE2ZjMxYjVjZDllZjYzZTk3OTg4NzQwYmY.
CSeq: 1 SUBSCRIBE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0be7881e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZDU1YzUzZmE2ZjMxYjVjZDllZjYzZTk3OTg4NzQwYmY.' in 6400 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.194.250:38752 --->
SUBSCRIBE sip:3101@192.168.199.13;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-3be4bfdd29f7706a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:3101@118.200.191.119:38752;transport=UDP>
To: "3101"<sip:3101@192.168.199.13;transport=UDP>
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=5b267f5d
Call-ID: ZDU1YzUzZmE2ZjMxYjVjZDllZjYzZTk3OTg4NzQwYmY.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Authorization: Digest username="3101",realm="asterisk",nonce="0be7881e",uri="sip:3101@192.168.199.13;transport=UDP",response="0e1d938a04852d296b82a8a20878aac4",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 192.168.194.250:38752 (no NAT)
Found peer '3101' for '3101' from 192.168.194.250:38752
Looking for 3101 in from-internal (domain 192.168.199.13)

<--- Transmitting (no NAT) to 192.168.194.250:38752 --->
SIP/2.0 489 Bad Event (format application/watcherinfo+xml)
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-3be4bfdd29f7706a-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=5b267f5d
To: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=as5d77a6a8
Call-ID: ZDU1YzUzZmE2ZjMxYjVjZDllZjYzZTk3OTg4NzQwYmY.
CSeq: 2 SUBSCRIBE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2014-06-12 22:15:30] WARNING[3968]: chan_sip.c:25438 handle_request_subscribe: SUBSCRIBE failure: unrecognized format:'application/watcherinfo+xml' pvt: subscribed: 0, stateid: -1, laststate: 0,dialogver: 0, subscribecont: '', subscribeuri: ''
Really destroying SIP dialog 'ZDU1YzUzZmE2ZjMxYjVjZDllZjYzZTk3OTg4NzQwYmY.' Method: SUBSCRIBE
Retransmitting #2 (no NAT) to 192.168.194.250:38752:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-d1005562babd8466-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
To: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3100@192.168.199.13:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1187272524 1187272524 IN IP4 192.168.199.13
s=Asterisk PBX 1.8.22.0
c=IN IP4 192.168.199.13
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #3 (no NAT) to 192.168.194.250:38752:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-d1005562babd8466-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
To: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3100@192.168.199.13:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1187272524 1187272524 IN IP4 192.168.199.13
s=Asterisk PBX 1.8.22.0
c=IN IP4 192.168.199.13
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #4 (no NAT) to 192.168.194.250:38752:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-d1005562babd8466-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
To: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3100@192.168.199.13:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1187272524 1187272524 IN IP4 192.168.199.13
s=Asterisk PBX 1.8.22.0
c=IN IP4 192.168.199.13
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #5 (no NAT) to 192.168.194.250:38752:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-d1005562babd8466-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
To: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3100@192.168.199.13:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1187272524 1187272524 IN IP4 192.168.199.13
s=Asterisk PBX 1.8.22.0
c=IN IP4 192.168.199.13
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #6 (no NAT) to 192.168.194.250:38752:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-d1005562babd8466-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
To: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 2 INVITE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3100@192.168.199.13:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1187272524 1187272524 IN IP4 192.168.199.13
s=Asterisk PBX 1.8.22.0
c=IN IP4 192.168.199.13
t=0 0
m=audio 10338 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[2014-06-12 22:15:36] WARNING[3968]: chan_sip.c:3979 retrans_pkt: Retransmission timeout reached on transmission OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2014-06-12 22:15:36] WARNING[3968]: chan_sip.c:4008 retrans_pkt: Hanging up call OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Executing [h@macro-dial-one:1] Macro("SIP/3101-00000016", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/3101-00000016", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/3101-00000016", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] Hangup("SIP/3101-00000016", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/3101-00000016' in macro 'hangupcall'
  == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/3101-00000016'
Scheduling destruction of SIP dialog '15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:3100@192.168.194.250:5620> for address/port to send to
set_destination: set destination to 192.168.194.250:5620
Reliably Transmitting (no NAT) to 192.168.194.250:5620:
BYE sip:3100@192.168.194.250:5620 SIP/2.0
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK122fde40
Max-Forwards: 70
From: "3101" <sip:3101@192.168.199.13>;tag=as16bfa2a2
To: <sip:3100@192.168.194.250:5620;rinstance=84211f663855424d>;tag=8153f605
Call-ID: 15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(1.8.22.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'SIP/3101-00000016' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/3101-00000016' in macro 'exten-vm'
  == Spawn extension (from-internal, 3100, 2) exited non-zero on 'SIP/3101-00000016'
Scheduling destruction of SIP dialog 'OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:3101@118.200.191.119:38752;transport=UDP> for address/port to send to
set_destination: set destination to 118.200.191.119:38752
Reliably Transmitting (no NAT) to 118.200.191.119:38752:
BYE sip:3101@118.200.191.119:38752;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK57165704;rport
Max-Forwards: 70
From: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
To: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
CSeq: 102 BYE
User-Agent: FPBX-2.11.0(1.8.22.0)
Proxy-Authorization: Digest username="3101", realm="asterisk", algorithm=MD5, uri="sip:192.168.199.13", nonce="", response="133d1f882d75955922e05253eed358e5"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---
  == MixMonitor close filestream
  == End MixMonitor Recording SIP/3101-00000016

<--- SIP read from UDP:192.168.194.250:38752 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK57165704;rport=5060
Contact: <sip:3101@118.200.191.119:38752;transport=UDP>
To: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=9a014113
From: <sip:3100@192.168.199.13;transport=UDP>;tag=as300967ad
Call-ID: OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.
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 'OGE5OGU1ZjdkM2UzMjgyMGM2YTA2NTZhYmY5MDY3YzA.' Method: INVITE

<--- SIP read from UDP:192.168.194.250:5620 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK122fde40
Contact: <sip:3100@192.168.194.250:5620>
To: <sip:3100@192.168.194.250:5620;rinstance=84211f663855424d>;tag=8153f605
From: "3101"<sip:3101@192.168.199.13>;tag=as16bfa2a2
Call-ID: 15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060
CSeq: 103 BYE
User-Agent: X-Lite release 4.6.1 stamp 73073 ff7d322c-W
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '15cd9aa2457f55963b97abf8569b272f@192.168.199.13:5060' Method: INVITE

<--- SIP read from UDP:192.168.194.250:38752 --->
PUBLISH sip:3101@192.168.199.13;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-59047a9da50b2bea-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:3101@118.200.191.119:38752;transport=UDP>
To: "3101"<sip:3101@192.168.199.13;transport=UDP>
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=231dda39
Call-ID: OTFhNzY5N2U1MjRkZTA4MmM4Y2VjNjA2MDNlMTJhMjY.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Event: presence
Allow-Events: presence, kpml
Content-Length: 262

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:3101@192.168.199.13;transport=UDP"> <tuple id="3101" > <status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---

<--- Transmitting (no NAT) to 192.168.194.250:38752 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-59047a9da50b2bea-1---d8754z-;rport;received=192.168.194.250
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=231dda39
To: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=as4a8f68e2
Call-ID: OTFhNzY5N2U1MjRkZTA4MmM4Y2VjNjA2MDNlMTJhMjY.
CSeq: 1 PUBLISH
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'OTFhNzY5N2U1MjRkZTA4MmM4Y2VjNjA2MDNlMTJhMjY.' Method: PUBLISH

<--- SIP read from UDP:192.168.194.250:38752 --->
SUBSCRIBE sip:3101@192.168.199.13;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-1c0ecd8334cc1af5-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:3101@118.200.191.119:38752;transport=UDP>
To: "3101"<sip:3101@192.168.199.13;transport=UDP>
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=a37d8864
Call-ID: NDY3NzQ1MGE2Zjc2ZWM0NGQ3NWQzOWIxYjcwN2I3ODE.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Creating new subscription
Sending to 192.168.194.250:38752 (no NAT)
list_route: hop: <sip:3101@118.200.191.119:38752;transport=UDP>
Found peer '3101' for '3101' from 192.168.194.250:38752

<--- Transmitting (no NAT) to 192.168.194.250:38752 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-1c0ecd8334cc1af5-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=a37d8864
To: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=as6f8812ac
Call-ID: NDY3NzQ1MGE2Zjc2ZWM0NGQ3NWQzOWIxYjcwN2I3ODE.
CSeq: 1 SUBSCRIBE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6906180c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NDY3NzQ1MGE2Zjc2ZWM0NGQ3NWQzOWIxYjcwN2I3ODE.' in 6400 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.194.250:38752 --->
SUBSCRIBE sip:3101@192.168.199.13;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-9ce42072730bc373-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:3101@118.200.191.119:38752;transport=UDP>
To: "3101"<sip:3101@192.168.199.13;transport=UDP>
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=a37d8864
Call-ID: NDY3NzQ1MGE2Zjc2ZWM0NGQ3NWQzOWIxYjcwN2I3ODE.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Authorization: Digest username="3101",realm="asterisk",nonce="6906180c",uri="sip:3101@192.168.199.13;transport=UDP",response="28061b1f5662cd5f3eaebbc1c0c44d2f",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 192.168.194.250:38752 (no NAT)
Found peer '3101' for '3101' from 192.168.194.250:38752
Looking for 3101 in from-internal (domain 192.168.199.13)

<--- Transmitting (no NAT) to 192.168.194.250:38752 --->
SIP/2.0 489 Bad Event (format application/watcherinfo+xml)
Via: SIP/2.0/UDP 118.200.191.119:38752;branch=z9hG4bK-d8754z-9ce42072730bc373-1---d8754z-;received=192.168.194.250;rport=38752
From: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=a37d8864
To: "3101"<sip:3101@192.168.199.13;transport=UDP>;tag=as6f8812ac
Call-ID: NDY3NzQ1MGE2Zjc2ZWM0NGQ3NWQzOWIxYjcwN2I3ODE.
CSeq: 2 SUBSCRIBE
Server: FPBX-2.11.0(1.8.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2014-06-12 22:15:36] WARNING[3968]: chan_sip.c:25438 handle_request_subscribe: SUBSCRIBE failure: unrecognized format:'application/watcherinfo+xml' pvt: subscribed: 0, stateid: -1, laststate: 0,dialogver: 0, subscribecont: '', subscribeuri: ''
Really destroying SIP dialog 'NDY3NzQ1MGE2Zjc2ZWM0NGQ3NWQzOWIxYjcwN2I3ODE.' Method: SUBSCRIBE
Reliably Transmitting (no NAT) to 192.168.194.250:38752:
OPTIONS sip:3101@192.168.194.250:38752;rinstance=d4b1e527c4c9af15;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK2af1e964
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.199.13>;tag=as239964c9
To: <sip:3101@192.168.194.250:38752;rinstance=d4b1e527c4c9af15;transport=UDP>
Contact: <sip:Unknown@192.168.199.13:5060>
Call-ID: 055ba1b31bf715942882eee37721d8f3@192.168.199.13:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(1.8.22.0)
Date: Thu, 12 Jun 2014 14:15:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.194.250:38752 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.199.13:5060;branch=z9hG4bK2af1e964
Contact: <sip:192.168.194.250:38752>
To: <sip:3101@192.168.194.250:38752;rinstance=d4b1e527c4c9af15;transport=UDP>;tag=392f0c2c
From: "Unknown"<sip:Unknown@192.168.199.13>;tag=as239964c9
Call-ID: 055ba1b31bf715942882eee37721d8f3@192.168.199.13:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper Communicator 2.05.11136 rev.11135
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '055ba1b31bf715942882eee37721d8f3@192.168.199.13:5060' Method: OPTIONS
удалить переоткрыть спам изменить тег редактировать

спросил 2014-06-12 13:51:37 +0400

ipvinner Gravatar ipvinner
44 34 3 21

обновил 2014-06-12 19:08:56 +0400

Comments

Снимите дамп для начала. http://asterisk-support.ru/question/38386/kak-poniat-chto-proiskhodit-na-asteriske/ А также настройки пиров.

kostoprav ( 2014-06-12 13:56:05 +0400 )редактировать

1 Ответ

0

наиболее вероятно - не выставлен внешний адрес для ната

ссылка удалить спам редактировать

ответил 2014-06-12 17:03:17 +0400

meral Gravatar meral flag of Ukraine
21228 23 18 169
http://pro-sip.net/

Comments

схема сети следующая Asterisk(192.168.199.13) ------->роутер1-----роутер2------свитч-----телефоны(192.168.194.X) между роутерами поднят впн. не могу понять, зачем здесь нат? по Вашей рекомендации, конечно попытался менять настройки External IP Local Networks и для пиров ставил nat=yes, но это никак не отразилось на этой проблеме. Не вполне понимаю при чем здесь NAT. Судя по всем телефоны(или софтфоны пробовал и то и то) из сети 192.168.194.X не шлют ACK пакеты. не могу понять почему

ipvinner ( 2014-06-12 17:35:28 +0400 )редактировать

Настройки пиров покажите

kostoprav ( 2014-06-12 17:41:27 +0400 )редактировать

основные настройки пиров: http://saveimg.ru/show-image.php?id=f1304d5d045841afbaae0f378f3d0fca * Name : 3100 Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : from-internal Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : Mailbox : 3100@device VM Extension : *97 LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "3100" <3100> MaxCallBR : 384 kbps Expire : 3179 Insecure : no Force rport : No ACL : Yes 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 : Yes Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.194.250:10888 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 3100 SIP Options : replaces replace Codecs : 0xc (ulaw|alaw) Codec Order : (ulaw:20,alaw:20) Auto-Framing : No Status : OK (10 ms) Useragent : X-Lite 4.6.1 73073-16ec3173-W Reg. Contact : sip:3100@192.168.194.250:10888;rinstance=4d1a176537c8486f 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

ipvinner ( 2014-06-12 17:44:31 +0400 )редактировать

ну вот теперь ловите где у вас бросаются пакеты. два роутера, впн. на всех стыках. заодно может нат найдете. или вы надеетесь что за вас ктото это сделает? скрины вообще мало что дают. тем более с выпилиными адресами.

meral ( 2014-06-12 17:49:58 +0400 )редактировать

я проверял на роутерах в консолях все пакеты проходят без дропов, натов там нет никаких.

ipvinner ( 2014-06-12 18:42:23 +0400 )редактировать

ну ждите фею.

meral ( 2014-06-12 18:45:59 +0400 )редактировать

веет конструктивом, что мне выложить еще простыню sip debug? я Вас не понимаю.

ipvinner ( 2014-06-12 19:03:05 +0400 )редактировать

ну я ж сказал что сделать. посмотерть сип дебаг. там есть порты. посмотреть по всему пути где фаервол эти порты блочит. чем отличается собранный трейс во всех местах(может гдето sip ALG работает)

meral ( 2014-06-12 19:07:43 +0400 )редактировать

или ждать фею. феей вон иногда switch подрабатывает.

meral ( 2014-06-12 19:08:25 +0400 )редактировать

Все извиняюсь, на Fortigate роутере работал SIP-helper, я его сразу не заметил.

ipvinner ( 2014-06-12 19:24:09 +0400 )редактировать

ухты. вас посетила фея

meral ( 2014-06-13 01:58:39 +0400 )редактировать

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Задан: 2014-06-12 13:51:37 +0400

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Обновлен: Jun 12 '14

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