Обрывается звонок через 30 секунд разговора (WARNING: Retransmission timeout......)
Asterisk (версия 1.8.6.0) разрывает соединение через 32 секунды после начала разговора, причем как внутри сети так и при звонках на мобилки.
В дебаге пишет следующее:
Retransmitting #10 (NAT) to XX.XXX.X.X:43412:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XXX.X.X:43412;branch=z9hG4bK-d8754z-34b2cf72fd37ba4a-1---d8754z-;received=XX.XXX.X.X;rport=43412
From: ;tag=b64cf074
To: ;tag=as1368689b
Call-ID: Y2ZkMGIwZDg4YTI5YzRiNDFhOTk4YzVlNGUyMTUzMjM.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 312
v=0
o=root 312848739 312848739 IN IP4 YY.YYY.YYY.YYY
s=Asterisk PBX 1.8.6.0
c=IN IP4 my_domain_ip
t=0 0
m=audio 15264 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Sep 21 10:54:12] WARNING[8260]: chan_sip.c:3620 retrans_pkt: Retransmission timeout reached on transmission Y2ZkMGIwZDg4YTI5YzRiNDFhOTk4YzVlNGUyMTUzMjM. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Sep 21 10:54:12] WARNING[8260]: chan_sip.c:3649 retrans_pkt: Hanging up call Y2ZkMGIwZDg4YTI5YzRiNDFhOTk4YzVlNGUyMTUzMjM. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog '5e19fa4031db1e4d07ef60e32c811121@my_domain_ip:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing for address/port to send to
set_destination: set destination to XX.XXX.X.X:35406
Reliably Transmitting (NAT) to XX.XXX.X.X:35406:
BYE sip:001*0016@XX.XXX.X.X:35406;rinstance=4f5fc3514ab4b4b2 SIP/2.0
Via: SIP/2.0/UDP my_domain_ip:5060;branch=z9hG4bK72bc8d49;rport
Max-Forwards: 70
From: "001*0014" ;tag=as74fd9a0a
To: ;tag=3dca843a
Call-ID: 5e19fa4031db1e4d07ef60e32c811121@YY.YYY.YYY.YYY:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (phones, 001*0016, 2) exited non-zero on 'SIP/001*0014-00000000'
Scheduling destruction of SIP dialog 'Y2ZkMGIwZDg4YTI5YzRiNDFhOTk4YzVlNGUyMTUzMjM.' in 32000 ms (Method: INVITE)
set_destination: Parsing for address/port to send to
set_destination: set destination to XX.XXX.X.X:43412
Reliably Transmitting (NAT) to XX.XXX.X.X:43412:
BYE sip:001*0014@XX.XXX.X.X:43412 SIP/2.0
Via: SIP/2.0/UDP my_domain_ip:5060;branch=z9hG4bK44281bd4;rport
Max-Forwards: 70
From: ;tag=as1368689b
To: ;tag=b64cf074
Call-ID: Y2ZkMGIwZDg4YTI5YzRiNDFhOTk4YzVlNGUyMTUzMjM.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.6.0
Proxy-Authorization: Digest username="001*0014", realm="asterisk", algorithm=MD5, uri="sip:mydomainname", nonce="", response="9fb88acc9877e7963cba612863104824"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
---
Asterisk стоит за NAT, localnet и externip прописаны. Проблема похоже с сетью,возможно нестабильное интернет соединение, но не могу понять как ее решить.
И еще непонятна строка:
Proxy-Authorization: Digest username="001*0014", realm="asterisk", algorithm=MD5, uri="sip:mydomainname", nonce="", response="9fb88acc9877e7963cba612863104824"
настроек прокси никаких не делал, но в консоли также часто вижу предупреждение:
WARNING[8260]: chan_sip.c:26618 build_peer: no value given for outbound proxy on line 0 of sip.conf.
Помогите разобраться.
Last edited by tor.zntu on Wed Sep 28, 2011 15:51
