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Retransmission timeout reached on transmission в Asterisk

Newbies/FAQ Forum 4 сообщений -
#1
Здравствуйте,

Возникла проблема следующего характера:

При звонке например с экстеншена 100 (подключаюсь к нему из Интернета) на 101 (который внутри сети) разговор идет, а примерно через 6 секунд в логах Asterisk вываливается ошибка:
Код:
[Jan 30 08:04:58] WARNING[2119]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission 52IXBJSBuDga-b7X7wM2-OHXdPZOnxw1 for seqno 8311 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Jan 30 08:04:58] WARNING[2119]: chan_sip.c:4288 retrans_pkt: Hanging up call 52IXBJSBuDga-b7X7wM2-OHXdPZOnxw1 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).


В Интернет Asterisk смотрит через роутер. Наружу проброшены порты sip 5060 (tcp/udp) и rtp 10000-20000 (udp).

rtp.conf
Код:
[general]
rtpstart = 10000
rtpend = 20000


sip.conf
Код:
[general]
tcpenable = yes
disallow = all
allow = alaw,ulaw

[phones](!)
type = friend
context = phones
host = dynamic
nat = no
qualify = yes

[100](phones)
defaultuser = 100
secret = *****

[101](phones)
defaultuser = 101
secret = *****


Прикладываю лог sip set debug on в момент начала звонка:
Код:
[Jan 30 08:26:09]
[Jan 30 08:26:09] INVITE sip:1000@davydenko.no-ip.org SIP/2.0
[Jan 30 08:26:09] Via: SIP/2.0/UDP 192.168.9.113:62312;rport;branch=z9hG4bKPjd7BKtjkTFBw0c7xZZYV4fLKZNT56ShJz
[Jan 30 08:26:09] Max-Forwards: 70
[Jan 30 08:26:09] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:09] To:
[Jan 30 08:26:09] Contact:
[Jan 30 08:26:09] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] CSeq: 8491 INVITE
[Jan 30 08:26:09] Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[Jan 30 08:26:09] Supported: replaces, 100rel, timer, norefersub
[Jan 30 08:26:09] Session-Expires: 1800
[Jan 30 08:26:09] Min-SE: 90
[Jan 30 08:26:09] User-Agent: Telephone 1.1.4
[Jan 30 08:26:09] Content-Type: application/sdp
[Jan 30 08:26:09] Content-Length: 481
[Jan 30 08:26:09]
[Jan 30 08:26:09] v=0
[Jan 30 08:26:09] o=- 3600135449 3600135449 IN IP4 86.102.40.95
[Jan 30 08:26:09] s=pjmedia
[Jan 30 08:26:09] b=AS:84
[Jan 30 08:26:09] t=0 0
[Jan 30 08:26:09] a=X-nat:0
[Jan 30 08:26:09] m=audio 4022 RTP/AVP 103 102 104 109 3 0 8 9 101
[Jan 30 08:26:09] c=IN IP4 86.102.40.95
[Jan 30 08:26:09] b=TIAS:64000
[Jan 30 08:26:09] a=rtcp:4023 IN IP4 192.168.1.4
[Jan 30 08:26:09] a=sendrecv
[Jan 30 08:26:09] a=rtpmap:103 speex/16000
[Jan 30 08:26:09] a=rtpmap:102 speex/8000
[Jan 30 08:26:09] a=rtpmap:104 speex/32000
[Jan 30 08:26:09] a=rtpmap:109 iLBC/8000
[Jan 30 08:26:09] a=fmtp:109 mode=30
[Jan 30 08:26:09] a=rtpmap:3 GSM/8000
[Jan 30 08:26:09] a=rtpmap:0 PCMU/8000
[Jan 30 08:26:09] a=rtpmap:8 PCMA/8000
[Jan 30 08:26:09] a=rtpmap:9 G722/8000
[Jan 30 08:26:09] a=rtpmap:101 telephone-event/8000
[Jan 30 08:26:09] a=fmtp:101 0-15
[Jan 30 08:26:09]
[Jan 30 08:26:09] --- (15 headers 22 lines) ---
[Jan 30 08:26:09] Sending to 86.102.40.95:62312 (NAT)
[Jan 30 08:26:09] Sending to 86.102.40.95:62312 (NAT)
[Jan 30 08:26:09] Using INVITE request as basis request - OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] Found peer '101' for '101' from 86.102.40.95:62312
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09] SIP/2.0 401 Unauthorized
[Jan 30 08:26:09] Via: SIP/2.0/UDP 192.168.9.113:62312;branch=z9hG4bKPjd7BKtjkTFBw0c7xZZYV4fLKZNT56ShJz;received=86.102.40.95;rport=62312
[Jan 30 08:26:09] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:09] To: ;tag=as6edf2125
[Jan 30 08:26:09] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] CSeq: 8491 INVITE
[Jan 30 08:26:09] Server: Asterisk PBX 12.0.0
[Jan 30 08:26:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 30 08:26:09] Supported: replaces, timer
[Jan 30 08:26:09] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f06f8e9"
[Jan 30 08:26:09] Content-Length: 0
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09] Scheduling destruction of SIP dialog 'OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC' in 6400 ms (Method: INVITE)
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09] ACK sip:1000@davydenko.no-ip.org SIP/2.0
[Jan 30 08:26:09] Via: SIP/2.0/UDP 192.168.9.113:62312;rport;branch=z9hG4bKPjd7BKtjkTFBw0c7xZZYV4fLKZNT56ShJz
[Jan 30 08:26:09] Max-Forwards: 70
[Jan 30 08:26:09] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:09] To: ;tag=as6edf2125
[Jan 30 08:26:09] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] CSeq: 8491 ACK
[Jan 30 08:26:09] Content-Length: 0
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09] --- (8 headers 0 lines) ---
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09] INVITE sip:1000@davydenko.no-ip.org SIP/2.0
[Jan 30 08:26:09] Via: SIP/2.0/UDP 192.168.9.113:62312;rport;branch=z9hG4bKPj--2YwwGTp.6jhoB8jE4xUcTsAS5ytHEZ
[Jan 30 08:26:09] Max-Forwards: 70
[Jan 30 08:26:09] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:09] To:
[Jan 30 08:26:09] Contact:
[Jan 30 08:26:09] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] CSeq: 8492 INVITE
[Jan 30 08:26:09] Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
[Jan 30 08:26:09] Supported: replaces, 100rel, timer, norefersub
[Jan 30 08:26:09] Session-Expires: 1800
[Jan 30 08:26:09] Min-SE: 90
[Jan 30 08:26:09] User-Agent: Telephone 1.1.4
[Jan 30 08:26:09] Authorization: Digest username="101", realm="asterisk", nonce="3f06f8e9", uri="sip:1000@davydenko.no-ip.org", response="f7928fbcd348b7f075eb6074d114433c", algorithm=MD5
[Jan 30 08:26:09] Content-Type: application/sdp
[Jan 30 08:26:09] Content-Length: 481
[Jan 30 08:26:09]
[Jan 30 08:26:09] v=0
[Jan 30 08:26:09] o=- 3600135449 3600135449 IN IP4 86.102.40.95
[Jan 30 08:26:09] s=pjmedia
[Jan 30 08:26:09] b=AS:84
[Jan 30 08:26:09] t=0 0
[Jan 30 08:26:09] a=X-nat:0
[Jan 30 08:26:09] m=audio 4022 RTP/AVP 103 102 104 109 3 0 8 9 101
[Jan 30 08:26:09] c=IN IP4 86.102.40.95
[Jan 30 08:26:09] b=TIAS:64000
[Jan 30 08:26:09] a=rtcp:4023 IN IP4 192.168.1.4
[Jan 30 08:26:09] a=sendrecv
[Jan 30 08:26:09] a=rtpmap:103 speex/16000
[Jan 30 08:26:09] a=rtpmap:102 speex/8000
[Jan 30 08:26:09] a=rtpmap:104 speex/32000
[Jan 30 08:26:09] a=rtpmap:109 iLBC/8000
[Jan 30 08:26:09] a=fmtp:109 mode=30
[Jan 30 08:26:09] a=rtpmap:3 GSM/8000
[Jan 30 08:26:09] a=rtpmap:0 PCMU/8000
[Jan 30 08:26:09] a=rtpmap:8 PCMA/8000
[Jan 30 08:26:09] a=rtpmap:9 G722/8000
[Jan 30 08:26:09] a=rtpmap:101 telephone-event/8000
[Jan 30 08:26:09] a=fmtp:101 0-15
[Jan 30 08:26:09]
[Jan 30 08:26:09] --- (16 headers 22 lines) ---
[Jan 30 08:26:09] Sending to 86.102.40.95:62312 (NAT)
[Jan 30 08:26:09] Using INVITE request as basis request - OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] Found peer '101' for '101' from 86.102.40.95:62312
[Jan 30 08:26:09] == Using SIP RTP CoS mark 5
[Jan 30 08:26:09] Found RTP audio format 103
[Jan 30 08:26:09] Found RTP audio format 102
[Jan 30 08:26:09] Found RTP audio format 104
[Jan 30 08:26:09] Found RTP audio format 109
[Jan 30 08:26:09] Found RTP audio format 3
[Jan 30 08:26:09] Found RTP audio format 0
[Jan 30 08:26:09] Found RTP audio format 8
[Jan 30 08:26:09] Found RTP audio format 9
[Jan 30 08:26:09] Found RTP audio format 101
[Jan 30 08:26:09] Found audio description format speex for ID 103
[Jan 30 08:26:09] Found audio description format speex for ID 102
[Jan 30 08:26:09] Found audio description format speex for ID 104
[Jan 30 08:26:09] Found audio description format iLBC for ID 109
[Jan 30 08:26:09] Found audio description format GSM for ID 3
[Jan 30 08:26:09] Found audio description format PCMU for ID 0
[Jan 30 08:26:09] Found audio description format PCMA for ID 8
[Jan 30 08:26:09] Found audio description format G722 for ID 9
[Jan 30 08:26:09] Found audio description format telephone-event for ID 101
[Jan 30 08:26:09] Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Jan 30 08:26:09] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jan 30 08:26:09] Peer audio RTP is at port 86.102.40.95:4022
[Jan 30 08:26:09] Looking for 1000 in phones (domain davydenko.no-ip.org)
[Jan 30 08:26:09] list_route: route/path hop:
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09] SIP/2.0 100 Trying
[Jan 30 08:26:09] Via: SIP/2.0/UDP 192.168.9.113:62312;branch=z9hG4bKPj--2YwwGTp.6jhoB8jE4xUcTsAS5ytHEZ;received=86.102.40.95;rport=62312
[Jan 30 08:26:09] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:09] To:
[Jan 30 08:26:09] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] CSeq: 8492 INVITE
[Jan 30 08:26:09] Server: Asterisk PBX 12.0.0
[Jan 30 08:26:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 30 08:26:09] Supported: replaces, timer
[Jan 30 08:26:09] Session-Expires: 1800;refresher=uas
[Jan 30 08:26:09] Contact:
[Jan 30 08:26:09] Content-Length: 0
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09] -- Executing [1000@phones:1] Verbose("SIP/101-0000000b", "Call to operators queue") in new stack
[Jan 30 08:26:09] Call to operators queue
[Jan 30 08:26:09] -- Executing [1000@phones:2] Answer("SIP/101-0000000b", "") in new stack
[Jan 30 08:26:09] Audio is at 17378
[Jan 30 08:26:09] Adding codec 100004 (alaw) to SDP
[Jan 30 08:26:09] Adding codec 100003 (ulaw) to SDP
[Jan 30 08:26:09] Adding non-codec 0x1 (telephone-event) to SDP
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09] SIP/2.0 200 OK
[Jan 30 08:26:09] Via: SIP/2.0/UDP 192.168.9.113:62312;branch=z9hG4bKPj--2YwwGTp.6jhoB8jE4xUcTsAS5ytHEZ;received=86.102.40.95;rport=62312
[Jan 30 08:26:09] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:09] To: ;tag=as208e6a8b
[Jan 30 08:26:09] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] CSeq: 8492 INVITE
[Jan 30 08:26:09] Server: Asterisk PBX 12.0.0
[Jan 30 08:26:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 30 08:26:09] Supported: replaces, timer
[Jan 30 08:26:09] Session-Expires: 1800;refresher=uas
[Jan 30 08:26:09] Contact:
[Jan 30 08:26:09] Content-Type: application/sdp
[Jan 30 08:26:09] Require: timer
[Jan 30 08:26:09] Content-Length: 257
[Jan 30 08:26:09]
[Jan 30 08:26:09] v=0
[Jan 30 08:26:09] o=root 1248685486 1248685486 IN IP4 192.168.1.3
[Jan 30 08:26:09] s=Asterisk PBX 12.0.0
[Jan 30 08:26:09] c=IN IP4 192.168.1.3
[Jan 30 08:26:09] t=0 0
[Jan 30 08:26:09] m=audio 17378 RTP/AVP 8 0 101
[Jan 30 08:26:09] a=rtpmap:8 PCMA/8000
[Jan 30 08:26:09] a=rtpmap:0 PCMU/8000
[Jan 30 08:26:09] a=rtpmap:101 telephone-event/8000
[Jan 30 08:26:09] a=fmtp:101 0-16
[Jan 30 08:26:09] a=ptime:20
[Jan 30 08:26:09] a=sendrecv
[Jan 30 08:26:09]
[Jan 30 08:26:09]
[Jan 30 08:26:09] Retransmitting #1 (NAT) to 86.102.40.95:62312:
[Jan 30 08:26:09] SIP/2.0 200 OK
[Jan 30 08:26:09] Via: SIP/2.0/UDP 192.168.9.113:62312;branch=z9hG4bKPj--2YwwGTp.6jhoB8jE4xUcTsAS5ytHEZ;received=86.102.40.95;rport=62312
[Jan 30 08:26:09] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:09] To: ;tag=as208e6a8b
[Jan 30 08:26:09] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] CSeq: 8492 INVITE
[Jan 30 08:26:09] Server: Asterisk PBX 12.0.0
[Jan 30 08:26:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 30 08:26:09] Supported: replaces, timer
[Jan 30 08:26:09] Session-Expires: 1800;refresher=uas
[Jan 30 08:26:09] Contact:
[Jan 30 08:26:09] Content-Type: application/sdp
[Jan 30 08:26:09] Require: timer
[Jan 30 08:26:09] Content-Length: 257
[Jan 30 08:26:09]
[Jan 30 08:26:09] v=0
[Jan 30 08:26:09] o=root 1248685486 1248685486 IN IP4 192.168.1.3
[Jan 30 08:26:09] s=Asterisk PBX 12.0.0
[Jan 30 08:26:09] c=IN IP4 192.168.1.3
[Jan 30 08:26:09] t=0 0
[Jan 30 08:26:09] m=audio 17378 RTP/AVP 8 0 101
[Jan 30 08:26:09] a=rtpmap:8 PCMA/8000
[Jan 30 08:26:09] a=rtpmap:0 PCMU/8000
[Jan 30 08:26:09] a=rtpmap:101 telephone-event/8000
[Jan 30 08:26:09] a=fmtp:101 0-16
[Jan 30 08:26:09] a=ptime:20
[Jan 30 08:26:09] a=sendrecv
[Jan 30 08:26:09]
[Jan 30 08:26:09] ---
[Jan 30 08:26:09] Retransmitting #2 (NAT) to 86.102.40.95:62312:
[Jan 30 08:26:09] SIP/2.0 200 OK
[Jan 30 08:26:09] Via: SIP/2.0/UDP 192.168.9.113:62312;branch=z9hG4bKPj--2YwwGTp.6jhoB8jE4xUcTsAS5ytHEZ;received=86.102.40.95;rport=62312
[Jan 30 08:26:09] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:09] To: ;tag=as208e6a8b
[Jan 30 08:26:09] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] CSeq: 8492 INVITE
[Jan 30 08:26:09] Server: Asterisk PBX 12.0.0
[Jan 30 08:26:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 30 08:26:09] Supported: replaces, timer
[Jan 30 08:26:09] Session-Expires: 1800;refresher=uas
[Jan 30 08:26:09] Contact:
[Jan 30 08:26:09] Content-Type: application/sdp
[Jan 30 08:26:09] Require: timer
[Jan 30 08:26:09] Content-Length: 257
[Jan 30 08:26:09]
[Jan 30 08:26:09] v=0
[Jan 30 08:26:09] o=root 1248685486 1248685486 IN IP4 192.168.1.3
[Jan 30 08:26:09] s=Asterisk PBX 12.0.0
[Jan 30 08:26:09] c=IN IP4 192.168.1.3
[Jan 30 08:26:09] t=0 0
[Jan 30 08:26:09] m=audio 17378 RTP/AVP 8 0 101
[Jan 30 08:26:09] a=rtpmap:8 PCMA/8000
[Jan 30 08:26:09] a=rtpmap:0 PCMU/8000
[Jan 30 08:26:09] a=rtpmap:101 telephone-event/8000
[Jan 30 08:26:09] a=fmtp:101 0-16
[Jan 30 08:26:09] a=ptime:20
[Jan 30 08:26:09] a=sendrecv
[Jan 30 08:26:09]
[Jan 30 08:26:09] ---
[Jan 30 08:26:09] -- Executing [1000@phones:3] Queue("SIP/101-0000000b", "operators") in new stack
[Jan 30 08:26:09] -- Started music on hold, class 'default', on SIP/101-0000000b
[Jan 30 08:26:09] -- Stopped music on hold on SIP/101-0000000b
[Jan 30 08:26:09] -- Playing 'queue-youarenext.slin' (language 'ru')
[Jan 30 08:26:09] Retransmitting #3 (NAT) to 86.102.40.95:62312:
[Jan 30 08:26:09] SIP/2.0 200 OK
[Jan 30 08:26:09] Via: SIP/2.0/UDP 192.168.9.113:62312;branch=z9hG4bKPj--2YwwGTp.6jhoB8jE4xUcTsAS5ytHEZ;received=86.102.40.95;rport=62312
[Jan 30 08:26:09] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:09] To: ;tag=as208e6a8b
[Jan 30 08:26:09] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:09] CSeq: 8492 INVITE
[Jan 30 08:26:09] Server: Asterisk PBX 12.0.0
[Jan 30 08:26:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 30 08:26:09] Supported: replaces, timer
[Jan 30 08:26:09] Session-Expires: 1800;refresher=uas
[Jan 30 08:26:09] Contact:
[Jan 30 08:26:09] Content-Type: application/sdp
[Jan 30 08:26:09] Require: timer
[Jan 30 08:26:09] Content-Length: 257
[Jan 30 08:26:09]
[Jan 30 08:26:09] v=0
[Jan 30 08:26:09] o=root 1248685486 1248685486 IN IP4 192.168.1.3
[Jan 30 08:26:09] s=Asterisk PBX 12.0.0
[Jan 30 08:26:09] c=IN IP4 192.168.1.3
[Jan 30 08:26:09] t=0 0
[Jan 30 08:26:09] m=audio 17378 RTP/AVP 8 0 101
[Jan 30 08:26:09] a=rtpmap:8 PCMA/8000
[Jan 30 08:26:09] a=rtpmap:0 PCMU/8000
[Jan 30 08:26:09] a=rtpmap:101 telephone-event/8000
[Jan 30 08:26:09] a=fmtp:101 0-16
[Jan 30 08:26:09] a=ptime:20
[Jan 30 08:26:09] a=sendrecv
[Jan 30 08:26:09]
[Jan 30 08:26:09] ---
[Jan 30 08:26:10] Retransmitting #4 (NAT) to 86.102.40.95:62312:
[Jan 30 08:26:10] SIP/2.0 200 OK
[Jan 30 08:26:10] Via: SIP/2.0/UDP 192.168.9.113:62312;branch=z9hG4bKPj--2YwwGTp.6jhoB8jE4xUcTsAS5ytHEZ;received=86.102.40.95;rport=62312
[Jan 30 08:26:10] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:10] To: ;tag=as208e6a8b
[Jan 30 08:26:10] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:10] CSeq: 8492 INVITE
[Jan 30 08:26:10] Server: Asterisk PBX 12.0.0
[Jan 30 08:26:10] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 30 08:26:10] Supported: replaces, timer
[Jan 30 08:26:10] Session-Expires: 1800;refresher=uas
[Jan 30 08:26:10] Contact:
[Jan 30 08:26:10] Content-Type: application/sdp
[Jan 30 08:26:10] Require: timer
[Jan 30 08:26:10] Content-Length: 257
[Jan 30 08:26:10]
[Jan 30 08:26:10] v=0
[Jan 30 08:26:10] o=root 1248685486 1248685486 IN IP4 192.168.1.3
[Jan 30 08:26:10] s=Asterisk PBX 12.0.0
[Jan 30 08:26:10] c=IN IP4 192.168.1.3
[Jan 30 08:26:10] t=0 0
[Jan 30 08:26:10] m=audio 17378 RTP/AVP 8 0 101
[Jan 30 08:26:10] a=rtpmap:8 PCMA/8000
[Jan 30 08:26:10] a=rtpmap:0 PCMU/8000
[Jan 30 08:26:10] a=rtpmap:101 telephone-event/8000
[Jan 30 08:26:10] a=fmtp:101 0-16
[Jan 30 08:26:10] a=ptime:20
[Jan 30 08:26:10] a=sendrecv
[Jan 30 08:26:10]
[Jan 30 08:26:10] ---
[Jan 30 08:26:12] Retransmitting #5 (NAT) to 86.102.40.95:62312:
[Jan 30 08:26:12] SIP/2.0 200 OK
[Jan 30 08:26:12] Via: SIP/2.0/UDP 192.168.9.113:62312;branch=z9hG4bKPj--2YwwGTp.6jhoB8jE4xUcTsAS5ytHEZ;received=86.102.40.95;rport=62312
[Jan 30 08:26:12] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:12] To: ;tag=as208e6a8b
[Jan 30 08:26:12] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:12] CSeq: 8492 INVITE
[Jan 30 08:26:12] Server: Asterisk PBX 12.0.0
[Jan 30 08:26:12] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 30 08:26:12] Supported: replaces, timer
[Jan 30 08:26:12] Session-Expires: 1800;refresher=uas
[Jan 30 08:26:12] Contact:
[Jan 30 08:26:12] Content-Type: application/sdp
[Jan 30 08:26:12] Require: timer
[Jan 30 08:26:12] Content-Length: 257
[Jan 30 08:26:12]
[Jan 30 08:26:12] v=0
[Jan 30 08:26:12] o=root 1248685486 1248685486 IN IP4 192.168.1.3
[Jan 30 08:26:12] s=Asterisk PBX 12.0.0
[Jan 30 08:26:12] c=IN IP4 192.168.1.3
[Jan 30 08:26:12] t=0 0
[Jan 30 08:26:12] m=audio 17378 RTP/AVP 8 0 101
[Jan 30 08:26:12] a=rtpmap:8 PCMA/8000
[Jan 30 08:26:12] a=rtpmap:0 PCMU/8000
[Jan 30 08:26:12] a=rtpmap:101 telephone-event/8000
[Jan 30 08:26:12] a=fmtp:101 0-16
[Jan 30 08:26:12] a=ptime:20
[Jan 30 08:26:12] a=sendrecv
[Jan 30 08:26:12]
[Jan 30 08:26:12] ---
[Jan 30 08:26:14] -- Told SIP/101-0000000b in operators their queue position (which was 1)
[Jan 30 08:26:14] -- Playing 'queue-thankyou.slin' (language 'ru')
[Jan 30 08:26:15] Retransmitting #6 (NAT) to 86.102.40.95:62312:
[Jan 30 08:26:15] SIP/2.0 200 OK
[Jan 30 08:26:15] Via: SIP/2.0/UDP 192.168.9.113:62312;branch=z9hG4bKPj--2YwwGTp.6jhoB8jE4xUcTsAS5ytHEZ;received=86.102.40.95;rport=62312
[Jan 30 08:26:15] From: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:15] To: ;tag=as208e6a8b
[Jan 30 08:26:15] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:15] CSeq: 8492 INVITE
[Jan 30 08:26:15] Server: Asterisk PBX 12.0.0
[Jan 30 08:26:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Jan 30 08:26:15] Supported: replaces, timer
[Jan 30 08:26:15] Session-Expires: 1800;refresher=uas
[Jan 30 08:26:15] Contact:
[Jan 30 08:26:15] Content-Type: application/sdp
[Jan 30 08:26:15] Require: timer
[Jan 30 08:26:15] Content-Length: 257
[Jan 30 08:26:15]
[Jan 30 08:26:15] v=0
[Jan 30 08:26:15] o=root 1248685486 1248685486 IN IP4 192.168.1.3
[Jan 30 08:26:15] s=Asterisk PBX 12.0.0
[Jan 30 08:26:15] c=IN IP4 192.168.1.3
[Jan 30 08:26:15] t=0 0
[Jan 30 08:26:15] m=audio 17378 RTP/AVP 8 0 101
[Jan 30 08:26:15] a=rtpmap:8 PCMA/8000
[Jan 30 08:26:15] a=rtpmap:0 PCMU/8000
[Jan 30 08:26:15] a=rtpmap:101 telephone-event/8000
[Jan 30 08:26:15] a=fmtp:101 0-16
[Jan 30 08:26:15] a=ptime:20
[Jan 30 08:26:15] a=sendrecv
[Jan 30 08:26:15]
[Jan 30 08:26:15] ---
[Jan 30 08:26:15] WARNING[2119]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC for seqno 8492 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Jan 30 08:26:15] WARNING[2119]: chan_sip.c:4288 retrans_pkt: Hanging up call OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Jan 30 08:26:15] == Spawn extension (phones, 1000, 3) exited non-zero on 'SIP/101-0000000b'
[Jan 30 08:26:15] Scheduling destruction of SIP dialog 'OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC' in 6400 ms (Method: INVITE)
[Jan 30 08:26:15] Reliably Transmitting (NAT) to 86.102.40.95:62312:
[Jan 30 08:26:15] BYE sip:101@86.102.40.95:62312;ob SIP/2.0
[Jan 30 08:26:15] Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK3b1b0fc4;rport
[Jan 30 08:26:15] Max-Forwards: 70
[Jan 30 08:26:15] From: ;tag=as208e6a8b
[Jan 30 08:26:15] To: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:15] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:15] CSeq: 102 BYE
[Jan 30 08:26:15] User-Agent: Asterisk PBX 12.0.0
[Jan 30 08:26:15] Proxy-Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:davydenko.no-ip.org", nonce="3f06f8e9", response="0a921b620feb123caf6b972bcea015b0"
[Jan 30 08:26:15] X-Asterisk-HangupCause: No user responding
[Jan 30 08:26:15] X-Asterisk-HangupCauseCode: 18
[Jan 30 08:26:15] Content-Length: 0
[Jan 30 08:26:15]
[Jan 30 08:26:15]
[Jan 30 08:26:15] ---
[Jan 30 08:26:15]
[Jan 30 08:26:15]
[Jan 30 08:26:15] SIP/2.0 200 OK
[Jan 30 08:26:15] Via: SIP/2.0/UDP 192.168.1.3:5060;rport=5060;received=77.34.246.47;branch=z9hG4bK3b1b0fc4
[Jan 30 08:26:15] Call-ID: OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC
[Jan 30 08:26:15] From: ;tag=as208e6a8b
[Jan 30 08:26:15] To: ;tag=6I6BcxzPq8NR3VYGE26iDISTo6c0n6yX
[Jan 30 08:26:15] CSeq: 102 BYE
[Jan 30 08:26:15] Content-Length: 0
[Jan 30 08:26:15]
[Jan 30 08:26:15]
[Jan 30 08:26:15] --- (7 headers 0 lines) ---
[Jan 30 08:26:15] SIP Response message for INCOMING dialog BYE arrived
[Jan 30 08:26:15] Really destroying SIP dialog 'OT3RZHvaI3VVJIM9tjgnfCmrC-oqj1TC' Method: INVITE
Asterisk*CLI> sip set debug off


davydenko.no-ip.org - DDNS адрес роутера за которым Asterisk
192.168.1.3 - внутренний IP адрес сервера Asterisk
86.102.40.95 - IP адрес клиента (которым регистрируюсь под экстеншеном 100)

SIP/1000 - это внутренний экстеншн который делает это:
Код:
exten => 1000, 1, Verbose(Telephone number)
same => n, Answer
same => n, Playback(zdravstujte)
same => n, Playback(your)
same => n, Playback(telephone-number)
same => n, SayDigits(${CALLERID(num)})
same => n, Hangup


Сейчас заметил везде rport=62312, в эту сторону копать? То есть как я понял Asterisk пытается переслать пакет не на 10000-20000 порты как указано в rtp.conf, а на 62312?

Подскажите куда копать и как быть, честно читал https://wiki.asterisk.org/wiki/display/AST/SIP+Ret... но ничего внятного не нашел про мой случай.

P.S.: Указывал в sip.conf externhost, externrefresh и localnet - на проблему не повлияло.

CentOS 6, Asterisk 12
#2
Цитата:
Сейчас заметил везде rport=62312, в эту сторону копать? То есть как я понял Asterisk пытается переслать пакет не на 10000-20000 порты как указано в rtp.conf, а на 62312?

Нет, это вообще не к RTP относится. RTP порты указываются в SDP body:
[Jan 30 08:26:09] m=audio 4022 RTP/AVP 103 102 104 109 3 0 8 9 101
[Jan 30 08:26:09] m=audio 17378 RTP/AVP 8 0 101
4022 и 17378 - вот RTP.

Смотрите в сторону Asterisk и NAT.
#3
Цитата:
: Указывал в sip.conf externhost, externrefresh и localnet - на проблему не повлияло.

Нужно ОБЯЗАТЕЛЬНО выставить в general sip.conf!
еще и nat=yes, directmedia=no у пира.
_________________
http://mh.otx.ru Гибкие SIP/E1 шлюзы Alvis-GW-2E1. Модернизация LDK300/TDA100:VoIP