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Blind transfer + NAT = Retransmission timeout

Проблемы Asterisk без вэб-оболочек и их решения

Модераторы: april22, Zavr2008

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sadm
Сообщения: 21
Зарегистрирован: 17 авг 2011, 21:41

Blind transfer + NAT = Retransmission timeout

Сообщение sadm »

Добрый день. Прошу помощи с *. Подключился к SIP-провайдеру Arctel. Пытаюсь настроить blind transfer входящих звонков от него между абонентами станции. При переводе, связь длится в среднем 4 секунды, после чего прерывается с такими ошибками в логах:

Код: Выделить всё

    -- Started music on hold, class 'default', on SIP/arctel1_out-00000000
    -- <SIP/12-00000004> Playing 'pbx-transfer.gsm' (language 'ru')
    -- Blind transferring SIP/arctel1_out-00000000 to '17' (context office) priority 1
    -- Stopped music on hold on SIP/arctel1_out-00000000
    -- fixed jitterbuffer destroyed on channel SIP/12-00000004
    -- Executing [17@office:1] Dial("SIP/arctel1_out-00000000", "SIP/17,,rTt") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/17
    -- SIP/17-00000005 is ringing
    -- SIP/17-00000005 is ringing
[Nov 22 10:18:53] WARNING[9664]: chan_sip.c:3696 retrans_pkt: Retransmission timeout reached on transmission SBCeccc1ac7cc2cc1bb34ebce3bc5a6ef2a@172.21.0.2 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[Nov 22 10:18:53] WARNING[9664]: chan_sip.c:3725 retrans_pkt: Hanging up call SBCeccc1ac7cc2cc1bb34ebce3bc5a6ef2a@172.21.0.2 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (office, 17, 1) exited non-zero on 'SIP/arctel1_out-00000000'
    -- fixed jitterbuffer destroyed on channel SIP/arctel1_out-00000000
Asterisk версии 1.8.18.0. Сервер и клиенты находятся на НАТ-ом. Перепробовал кучу настроек. без особого результата. Входящие/исходящие идут нормально, но на blind transfer всё ломается.
В чем может быть дело?
Вернуться к началу
awsswa
Сообщения: 2390
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: Blind transfer + NAT = Retransmission timeout

Сообщение awsswa »

почитать про nat=yes
платный суппорт по мере возможностей
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sadm
Сообщения: 21
Зарегистрирован: 17 авг 2011, 21:41

Re: Blind transfer + NAT = Retransmission timeout

Сообщение sadm »

awsswa писал(а):почитать про nat=yes
Читал и про nat=yes, и про qualify, и про canreinvite, и про "asterisk и клиенты за NAT", расставлял эти опции в разных порядках в разных местах - результат 4 секунды после blind transfer и ошибка "Retransmission timeout reached ".
Может я что-то не то читаю? Или не понимаю что.
Вернуться к началу
ded
Сообщения: 15626
Зарегистрирован: 26 авг 2010, 19:00

Re: Blind transfer + NAT = Retransmission timeout

Сообщение ded »

Эт точно!
И команда
CLI> sip show peers
покажет это, а если сделать ещё
CLI> sip set debug peer 17
то можно увидеть как Астериск много-много раз ретрансмитить инвайты ему.
Вернуться к началу
sadm
Сообщения: 21
Зарегистрирован: 17 авг 2011, 21:41

Re: Blind transfer + NAT = Retransmission timeout

Сообщение sadm »

Я выставил nat=yes глобально в sip.conf, и в блоках описания sip-провайдеров. В описании килентов стоит nat=no, - клиенты находятся в одной локальной подсети с сервером астериск (впрочем, выставление nat=yes на клиентах не сильно поменяло картину).

Код: Выделить всё

CLI> sip show peers

Name/username              Host                                    Dyn Forcerport ACL Port     Status
10/11                      (Unspecified)                            D                 0        UNKNOWN
11/11                      (Unspecified)                            D                 0        UNKNOWN
12/12                      192.168.0.100                            D                 51544    OK (107 ms)
14/14                      192.168.0.103                            D                 1045     OK (102 ms)
15/15                      192.168.0.102                            D                 49681    OK (7 ms)
17/17                      192.168.0.117                            D                 52789    OK (105 ms)
arctel1_out/POrgRus1    86.110.4.148                                 N             5060     OK (29 ms)
arctel2_out/POrgRus2    86.110.4.148                                 N             5060     OK (30 ms)
sipmarket_out/102621       85.17.222.134                                N             5060     OK (64 ms)
дебаг для 17 пира с nat=yes и no:

Ткните меня пожалуйста носом - в каком месте проблемы.

Код: Выделить всё

    -- SIP/12-0000000f connected line has changed. Saving it until answer for SIP/arctel1_out-0000000b
    -- SIP/12-0000000f answered SIP/arctel1_out-0000000b
Scheduling destruction of SIP dialog '61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.0.117:52789:
CANCEL sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK67758461
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as0804f71d
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
Call-ID: 61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.18.0
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060' in 6400 ms (Method: INVITE)
    -- fixed jitterbuffer created on channel SIP/arctel1_out-0000000b
Retransmitting #1 (no NAT) to 192.168.0.117:52789:
CANCEL sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK67758461
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as0804f71d
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
Call-ID: 61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.18.0
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK67758461
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=9e543824
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as0804f71d
Call-ID: 61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060
CSeq: 102 CANCEL
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK67758461
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=9e543824
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as0804f71d
Call-ID: 61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.117:52789:
ACK sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK67758461
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as0804f71d
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=9e543824
Contact: <sip:Anonymous@192.168.0.250:5060>
Call-ID: 61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.18.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK67758461
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=65234a34
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as0804f71d
Call-ID: 61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060
CSeq: 102 CANCEL
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
    -- fixed jitterbuffer created on channel SIP/12-0000000f

<--- SIP read from UDP:192.168.0.117:52789 --->


<------------->
Really destroying SIP dialog '61d0c93534bc458c696c8030216bd8d3@192.168.0.250:5060' Method: INVITE
Really destroying SIP dialog 'OGJiY2FkZTU2MzMxZmFmZGVkYTE4YjMyMTFiOGU3Mzg.' Method: REGISTER
    -- Started music on hold, class 'default', on SIP/arctel1_out-0000000b
    -- <SIP/12-0000000f> Playing 'pbx-transfer.gsm' (language 'ru')
    -- Blind transferring SIP/arctel1_out-0000000b to '17' (context office) priority 1
    -- Stopped music on hold on SIP/arctel1_out-0000000b
    -- fixed jitterbuffer destroyed on channel SIP/12-0000000f
    -- Executing [17@office:1] Dial("SIP/arctel1_out-0000000b", "SIP/17,,rTt") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 12432
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
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 192.168.0.117:52789:
INVITE sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK61172e0d
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as3f096286
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
Contact: <sip:Anonymous@192.168.0.250:5060>
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.18.0
Date: Fri, 23 Nov 2012 10:29:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 331

v=0
o=root 813974312 813974312 IN IP4 192.168.0.250
s=Asterisk PBX 1.8.18.0
c=IN IP4 192.168.0.250
t=0 0
m=audio 12432 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
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/17
Retransmitting #1 (no NAT) to 192.168.0.117:52789:
INVITE sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK61172e0d
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as3f096286
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
Contact: <sip:Anonymous@192.168.0.250:5060>
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.18.0
Date: Fri, 23 Nov 2012 10:29:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 331

v=0
o=root 813974312 813974312 IN IP4 192.168.0.250
s=Asterisk PBX 1.8.18.0
c=IN IP4 192.168.0.250
t=0 0
m=audio 12432 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
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:192.168.0.117:52789 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK61172e0d
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=4f7d8343
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as3f096286
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK61172e0d
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=4f7d8343
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as3f096286
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
    -- SIP/17-00000010 is ringing
    -- SIP/17-00000010 is ringing

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK61172e0d
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=4f7d8343
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as3f096286
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 257

v=0
o=3cxVCE 74699595 390052620 IN IP4 192.168.0.117
s=3cxVCE Audio Call
c=IN IP4 192.168.0.117
t=0 0
m=audio 40044 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.117:40044
list_route: hop: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
set_destination: Parsing <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8> for address/port to send to
set_destination: set destination to 192.168.0.117:52789
Transmitting (no NAT) to 192.168.0.117:52789:
ACK sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK39aa39ff
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as3f096286
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=4f7d8343
Contact: <sip:Anonymous@192.168.0.250:5060>
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.18.0
Content-Length: 0


---
    -- SIP/17-00000010 answered SIP/arctel1_out-0000000b
    -- fixed jitterbuffer created on channel SIP/17-00000010
[Nov 23 14:30:01] WARNING[14808]: chan_sip.c:3696 retrans_pkt: Retransmission timeout reached on transmission SBCc0661b7dd7a2074fc48ce8041ecdfffc@172.21.0.2 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Nov 23 14:30:01] WARNING[14808]: chan_sip.c:3725 retrans_pkt: Hanging up call SBCc0661b7dd7a2074fc48ce8041ecdfffc@172.21.0.2 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog '1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8> for address/port to send to
set_destination: set destination to 192.168.0.117:52789
Reliably Transmitting (no NAT) to 192.168.0.117:52789:
BYE sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK40051b34
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as3f096286
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=4f7d8343
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.18.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
    -- fixed jitterbuffer destroyed on channel SIP/17-00000010
  == Spawn extension (office, 17, 1) exited non-zero on 'SIP/arctel1_out-0000000b'
    -- fixed jitterbuffer destroyed on channel SIP/arctel1_out-0000000b
Retransmitting #1 (no NAT) to 192.168.0.117:52789:
BYE sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK40051b34
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as3f096286
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=4f7d8343
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.18.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK40051b34
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=4f7d8343
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as3f096286
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 103 BYE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060' Method: INVITE

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK40051b34
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=4f7d8343
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as3f096286
Call-ID: 1d6a96c15ebbcaad3ff1f4467c9add6a@192.168.0.250:5060
CSeq: 103 BYE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

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

Код: Выделить всё

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [POrgRus1@arctel1_out:1] Dial("SIP/arctel1_out-00000000", "SIP/15&SIP/17&SIP/14&SIP/12,,rTt") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/15
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 17120
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.117:52789:
INVITE sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK3b6a944b;rport
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as3742dc10
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
Contact: <sip:Anonymous@192.168.0.250:5060>
Call-ID: 75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.18.0
Date: Fri, 23 Nov 2012 10:09:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1502898565 1502898565 IN IP4 192.168.0.250
s=Asterisk PBX 1.8.18.0
c=IN IP4 192.168.0.250
t=0 0
m=audio 17120 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
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/17
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/14
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/12
    -- SIP/15-00000001 connected line has changed. Saving it until answer for SIP/arctel1_out-00000000
    -- SIP/17-00000002 connected line has changed. Saving it until answer for SIP/arctel1_out-00000000
    -- SIP/14-00000003 connected line has changed. Saving it until answer for SIP/arctel1_out-00000000
    -- SIP/12-00000004 connected line has changed. Saving it until answer for SIP/arctel1_out-00000000

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK3b6a944b;rport=5060
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=423bfb05
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as3742dc10
Call-ID: 75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
    -- SIP/17-00000002 is ringing
    -- SIP/14-00000003 is ringing
    -- SIP/14-00000003 is ringing
    -- SIP/15-00000001 is ringing
    -- SIP/15-00000001 is ringing
    -- SIP/12-00000004 is ringing
    -- SIP/12-00000004 is ringing
    -- SIP/12-00000004 connected line has changed. Saving it until answer for SIP/arctel1_out-00000000
    -- SIP/12-00000004 answered SIP/arctel1_out-00000000
Scheduling destruction of SIP dialog '75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060' in 6784 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.0.117:52789:
CANCEL sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK3b6a944b;rport
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as3742dc10
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
Call-ID: 75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.18.0
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060' in 6784 ms (Method: INVITE)
    -- fixed jitterbuffer created on channel SIP/arctel1_out-00000000

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK3b6a944b;rport=5060
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=423bfb05
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as3742dc10
Call-ID: 75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060
CSeq: 102 CANCEL
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK3b6a944b;rport=5060
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=423bfb05
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as3742dc10
Call-ID: 75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.0.117:52789:
ACK sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK3b6a944b;rport
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as3742dc10
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=423bfb05
Contact: <sip:Anonymous@192.168.0.250:5060>
Call-ID: 75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.18.0
Content-Length: 0


---
Scheduling destruction of SIP dialog '75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060' in 6784 ms (Method: INVITE)
    -- fixed jitterbuffer created on channel SIP/12-00000004
Really destroying SIP dialog 'OGJiY2FkZTU2MzMxZmFmZGVkYTE4YjMyMTFiOGU3Mzg.' Method: REGISTER
Really destroying SIP dialog '75edf1d679f117d80e151a6f7224a6a8@192.168.0.250:5060' Method: INVITE
    -- Started music on hold, class 'default', on SIP/arctel1_out-00000000
    -- <SIP/12-00000004> Playing 'pbx-transfer.gsm' (language 'ru')
    -- Blind transferring SIP/arctel1_out-00000000 to '17' (context office) priority 1
    -- Stopped music on hold on SIP/arctel1_out-00000000
    -- fixed jitterbuffer destroyed on channel SIP/12-00000004
    -- Executing [17@office:1] Dial("SIP/arctel1_out-00000000", "SIP/17,,rTt") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 19154
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.117:52789:
INVITE sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK51a441d6;rport
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as30c8c8d1
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
Contact: <sip:Anonymous@192.168.0.250:5060>
Call-ID: 1cd6e41900a708622fff64390e48a884@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.18.0
Date: Fri, 23 Nov 2012 10:10:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1134150322 1134150322 IN IP4 192.168.0.250
s=Asterisk PBX 1.8.18.0
c=IN IP4 192.168.0.250
t=0 0
m=audio 19154 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
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/17

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK51a441d6;rport=5060
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=bf017c41
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as30c8c8d1
Call-ID: 1cd6e41900a708622fff64390e48a884@192.168.0.250:5060
CSeq: 102 INVITE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
    -- SIP/17-00000005 is ringing

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK51a441d6;rport=5060
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=bf017c41
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as30c8c8d1
Call-ID: 1cd6e41900a708622fff64390e48a884@192.168.0.250:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 258

v=0
o=3cxVCE 253590990 192285720 IN IP4 192.168.0.117
s=3cxVCE Audio Call
c=IN IP4 192.168.0.117
t=0 0
m=audio 40024 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.117:40024
list_route: hop: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
set_destination: Parsing <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8> for address/port to send to
set_destination: set destination to 192.168.0.117:52789
Transmitting (NAT) to 192.168.0.117:52789:
ACK sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK0a55be7f;rport
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as30c8c8d1
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=bf017c41
Contact: <sip:Anonymous@192.168.0.250:5060>
Call-ID: 1cd6e41900a708622fff64390e48a884@192.168.0.250:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.18.0
Content-Length: 0


---
    -- SIP/17-00000005 answered SIP/arctel1_out-00000000
    -- fixed jitterbuffer created on channel SIP/17-00000005
[Nov 23 14:10:15] WARNING[14707]: chan_sip.c:3696 retrans_pkt: Retransmission timeout reached on transmission SBCe3bb6bfc6a11c1dcb89a697137e36803@172.21.0.2 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[Nov 23 14:10:15] WARNING[14707]: chan_sip.c:3725 retrans_pkt: Hanging up call SBCe3bb6bfc6a11c1dcb89a697137e36803@172.21.0.2 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog '1cd6e41900a708622fff64390e48a884@192.168.0.250:5060' in 6784 ms (Method: INVITE)
set_destination: Parsing <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8> for address/port to send to
set_destination: set destination to 192.168.0.117:52789
Reliably Transmitting (NAT) to 192.168.0.117:52789:
BYE sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK474b7b63;rport
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous@192.168.0.250>;tag=as30c8c8d1
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=bf017c41
Call-ID: 1cd6e41900a708622fff64390e48a884@192.168.0.250:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.18.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
    -- fixed jitterbuffer destroyed on channel SIP/17-00000005
  == Spawn extension (office, 17, 1) exited non-zero on 'SIP/arctel1_out-00000000'
    -- fixed jitterbuffer destroyed on channel SIP/arctel1_out-00000000

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK474b7b63;rport=5060
Contact: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=bf017c41
From: "Anonymous"<sip:Anonymous@192.168.0.250>;tag=as30c8c8d1
Call-ID: 1cd6e41900a708622fff64390e48a884@192.168.0.250:5060
CSeq: 103 BYE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1cd6e41900a708622fff64390e48a884@192.168.0.250:5060' Method: INVITE

<--- SIP read from UDP:192.168.0.117:52789 --->


<------------->
Reliably Transmitting (NAT) to 192.168.0.117:52789:
OPTIONS sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK2b8f9e1c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.250>;tag=as4f90e938
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>
Contact: <sip:asterisk@192.168.0.250:5060>
Call-ID: 5123352d3b8fdc3f71dd22ab7dcb018c@192.168.0.250:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.18.0
Date: Fri, 23 Nov 2012 10:10:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.117:52789 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bK2b8f9e1c;rport=5060
Contact: <sip:192.168.0.117:52789>
To: <sip:17@192.168.0.117:52789;rinstance=a45cbd4106e4a6c8>;tag=933ca71d
From: "asterisk"<sip:asterisk@192.168.0.250>;tag=as4f90e938
Call-ID: 5123352d3b8fdc3f71dd22ab7dcb018c@192.168.0.250:5060
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 '5123352d3b8fdc3f71dd22ab7dcb018c@192.168.0.250:5060' Method: OPTIONS
rusgate*CLI> quit
Executing last minute cleanups
Вернуться к началу
awsswa
Сообщения: 2390
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: Blind transfer + NAT = Retransmission timeout

Сообщение awsswa »

Что то мне подсказывает что не хватает localnet=192.168.0.0/16
Но лучше посмотреть ваш sip.conf
платный суппорт по мере возможностей
Вернуться к началу
ded
Сообщения: 15626
Зарегистрирован: 26 авг 2010, 19:00

Re: Blind transfer + NAT = Retransmission timeout

Сообщение ded »

Не согласный!
Несмотряна то, что внутренний номер 17 зарегистрирован с 192.168.0.117:52789 - при попытке
Transmitting (no NAT) to 192.168.0.117:52789
именно он ответил
SIP/2.0 481 Call/Transaction Does Not Exist
Что это за адость там стоит? Софтфон небось какой-нить?
Вернуться к началу
sadm
Сообщения: 21
Зарегистрирован: 17 авг 2011, 21:41

Re: Blind transfer + NAT = Retransmission timeout

Сообщение sadm »

awsswa писал(а):Что то мне подсказывает что не хватает localnet=192.168.0.0/16
Но лучше посмотреть ваш sip.conf
Вот он:

Код: Выделить всё

[general]
context = sipincoming
allowoverlap = no
allowtransfers = yes
allowguest=no
buggymwi = yes
checkmwi = 30
domain = 192.168.0.250
;defaultexpiry=3600
defaultexpiry=600
language = ru
localnet = 192.168.0.0/24
notifyringing = yes
notifyhold = yes
realm = rusiz.ru
rtcachefriends = yes
rtcache=yes
udpbindaddr = 0.0.0.0
srvlookup = no
tos_sip=cs3
tos_audio=ef
tos_video=af41
videosupport=no
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g729
t38pt_udptl=yes
t38pt_rtp=no
t38pt_tcp=no
tcpenable=yes
;externip=61.52.73.99
nat=yes


jbenable = yes
jbforce = yes
jbimpl = fixed
jblog = no
jbmaxsize = 150
jbresyncthreshold = 1000

register=100000:ххххххх:100000@85.17.222.134/sipmarket_in

register=User1:xxxxxxxx@86.110.4.148

[sipmarket_out]
type=peer
host=85.17.222.134
username=100000
secret=xxxxxxx
nat=yes
fromuser=100000
fromdomain=sipmarket.net
dtmfmode=rfc2833
insecure=invite
context=sipmarket_out
disallow=all
allow=g729
allow=alaw
allow=ulaw
qualify=yes
sendrpid=yes
canreinvite=no
call-limit=2
sipreinvite=no

[arctel1_out]
type=peer
host=86.110.4.148
username=User1
secret=xxxxxxxx
nat=yes
fromuser=User1
dtmfmode=rfc2833
insecure=invite,port
context=arctel1_out
disallow=all
allow=g729
qualify=yes
sendrpid=yes
canreinvite=no
sipreinvite=no
;call-limit=1
ded писал(а):Что это за адость там стоит? Софтфон небось какой-нить?
3CX Phone. Есть какие-то сложности с его эксплуатацией?
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Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: Blind transfer + NAT = Retransmission timeout

Сообщение Vlad1983 »

снимать полностью сигналку и RTP и смотреть что такого проскакивает, что не нравится арктелу
ЛС: @rostel
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awsswa
Сообщения: 2390
Зарегистрирован: 09 июн 2012, 10:52
Откуда: Россия, Пермь skype: yarick_perm

Re: Blind transfer + NAT = Retransmission timeout

Сообщение awsswa »

nat=yes раз у вас все за натом, ставиться глобально и на каждого
directmedia=no - тоже

rtp set debug on - звоните и смотрите адреса, внутренних не должно быть
и sip show peers сюда
платный суппорт по мере возможностей
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