30秒后Asterisk呼叫掉线

作者：LEEstarmmmmm | 2023-06-13 13:31

如何解决《30秒后Asterisk呼叫掉线》经验，为你挑选了1个好方法。

我安装了Asterisk,并使用Android Zoiper应用程序拨打电话.它成功连接两个用户并听到声音,但30秒后呼叫掉线.

星号日志

[Apr 14 18:40:34] WARNING[27959]: chan_sip.c:4176 retrans_pkt: 
Retransmission timeout reached on transmission lPsW4atWG- for seqno 20 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Apr 14 18:40:34] WARNING[27959]: chan_sip.c:4205 retrans_pkt: 
Hanging up call lPsW4atWG- - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (from-sip, 1000, 1) exited non-zero on 'SIP/2000-0000000a'

Sip.conf

[general]
 context=default                       ; Default context for incoming calls
 ;
 bindport=5060                   ; bindport is the local UDP port that Asterisk will listen on
 bindaddr=0.0.0.0           ; IP address to bind to (0.0.0.0 binds to all)
 ;
 disallow=all                    ; First disallow all codecs
 allow=gsm
 allow=ulaw                      ; Allow codecs in order of preference
 ;
 register => 12121111111:1234:11111111@sipauth.deltathree.com/1000


allow=g729
allow=alaw
srvlookup=no
canreinvite=no
directrtpsetup=no
trustpid=yes
sendrpid=yes
qualify=yes
callevents=yes
insecure=invite
pedantic=no
useragent=Glastender PBX
videosupport=no
t38pt_udptl=no
t38pt_rtp=no
t38pt_tcp=no

nat=yes
media_address = XXX.52.91.XXX ; server ip address

看起来我需要在sip.conf上改变一些东西,并尝试了不同的配置.它还没有工作..你看到有什么问题吗？

SIP日志

interface: eth0 (10.7.21.0/255.255.255.0)
filter: ( port 5060 ) and (ip or ip6)
#
U 2014/04/15 00:22:15.941072 XX.53.122.134:5060 -> 10.8.21.XX:5060
INVITE sip:1000@sipdomain.com SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;rport.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com.
CSeq: 20 INVITE.
Call-ID: wh8Ai1e~0c.
Max-Forwards: 70.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Content-Type: application/sdp.
Content-Length: 280.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
Contact: ;+sip.instance="".
.
v=0.
o=2000 274 59 IN IP4 192.168.0.38.
s=Talk.
c=IN IP4 192.168.0.38.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
m=video 9078 RTP/AVP 103 99.
a=rtpmap:103 VP8/90000.
a=rtpmap:99 MP4V-ES/90000.
a=fmtp:99 profile-level-id=3.

#
U 2014/04/15 00:22:15.945220 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Length: 0.
.

#
U 2014/04/15 00:22:15.951499 10.8.21.XX:5060 -> 223.XX.130.50:40764
INVITE sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK70816646;rport.
Max-Forwards: 70.
From: ;tag=as679b5fe7.
To: .
Contact: .
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Mon, 14 Apr 2014 15:22:15 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 258.
.
v=0.
o=root 1811076761 1811076761 IN IP4 192.168.0.38.
s=Asterisk PBX 11.8.1.
c=IN IP4 192.168.0.38.
t=0 0.
m=audio 7076 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.

#
U 2014/04/15 00:22:16.045285 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK70816646;rport.
From: ;tag=as679b5fe7.
To: sip:1000@223.XX.130.50:40764.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 INVITE.
.

#
U 2014/04/15 00:22:16.445425 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK70816646;rport.
From: ;tag=as679b5fe7.
To: ;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
.

#
U 2014/04/15 00:22:16.447116 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Length: 0.
.

#
U 2014/04/15 00:22:16.838201 XX.53.122.134:5060 -> 10.8.21.XX:5060
.
.

#
U 2014/04/15 00:22:19.275720 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK70816646;rport.
From: ;tag=as679b5fe7.
To: ;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Contact: ;+sip.instance="".
Content-Type: application/sdp.
Content-Length: 176.
.
v=0.
o=1000 3792 2294 IN IP4 223.XX.130.50.
s=Talk.
c=IN IP4 223.XX.130.50.
b=AS:380.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

#
U 2014/04/15 00:22:19.276630 10.8.21.XX:5060 -> 223.XX.130.50:40764
ACK sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK730c16dd;rport.
Max-Forwards: 70.
From: ;tag=as679b5fe7.
To: ;tag=coOV3rP.
Contact: .
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.

#
U 2014/04/15 00:22:19.276978 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:19.776861 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:20.778018 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:22.777522 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:25.139894 XX.53.122.134:32840 -> 10.8.21.XX:5060
.
.

#
U 2014/04/15 00:22:26.777002 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:30.179568 XX.53.122.134:55180 -> 10.8.21.XX:5060
.
.

#
U 2014/04/15 00:22:30.777462 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:34.777660 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:38.777721 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:42.777667 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:46.776449 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:46.927655 XX.53.122.134:5060 -> 10.8.21.XX:5060
.
.

#
U 2014/04/15 00:22:50.776948 10.8.21.XX:5060 -> XX.53.122.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
From: ;tag=dGlp5o0FS.
To: sip:1000@sipdomain.com;tag=as1ba98ffc.
Call-ID: wh8Ai1e~0c.
CSeq: 20 INVITE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: .
Content-Type: application/sdp.
Content-Length: 287.
.
v=0.
o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
s=Asterisk PBX 11.8.1.
c=IN IP4 223.XX.130.50.
t=0 0.
m=audio 45068 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.
m=video 0 RTP/AVP 103 99.

#
U 2014/04/15 00:22:51.278124 10.8.21.XX:5060 -> XX.53.122.134:5060
INVITE sip:2000@XX.53.122.134 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK348a4dc2;rport.
Max-Forwards: 70.
From: sip:1000@sipdomain.com;tag=as1ba98ffc.
To: ;tag=dGlp5o0FS.
Contact: .
Call-ID: wh8Ai1e~0c.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 259.
.
v=0.
o=root 1836373944 1836373945 IN IP4 117.52.91.12.
s=Asterisk PBX 11.8.1.
c=IN IP4 117.52.91.12.
t=0 0.
m=audio 19152 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.

#
U 2014/04/15 00:22:51.278285 10.8.21.XX:5060 -> 223.XX.130.50:40764
INVITE sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK59c0124b;rport.
Max-Forwards: 70.
From: ;tag=as679b5fe7.
To: ;tag=coOV3rP.
Contact: .
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 259.
.
v=0.
o=root 1811076761 1811076762 IN IP4 117.52.91.12.
s=Asterisk PBX 11.8.1.
c=IN IP4 117.52.91.12.
t=0 0.
m=audio 15858 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.

#
U 2014/04/15 00:22:51.344965 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK59c0124b;rport.
From: ;tag=as679b5fe7.
To: ;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 103 INVITE.
.

#
U 2014/04/15 00:22:51.355122 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK59c0124b;rport.
From: ;tag=as679b5fe7.
To: ;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 103 INVITE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Contact: ;+sip.instance="".
Content-Type: application/sdp.
Content-Length: 176.
.
v=0.
o=1000 3792 2296 IN IP4 223.XX.130.50.
s=Talk.
c=IN IP4 223.XX.130.50.
b=AS:380.
t=0 0.
m=audio 45068 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

#
U 2014/04/15 00:22:51.355539 10.8.21.XX:5060 -> 223.XX.130.50:40764
ACK sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK144199ce;rport.
Max-Forwards: 70.
From: ;tag=as679b5fe7.
To: ;tag=coOV3rP.
Contact: .
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 103 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.

#
U 2014/04/15 00:22:51.355619 10.8.21.XX:5060 -> 223.XX.130.50:40764
BYE sip:1000@223.XX.130.50:40764 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK0ac3adc4;rport.
Max-Forwards: 70.
From: ;tag=as679b5fe7.
To: ;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 104 BYE.
User-Agent: Asterisk PBX 11.8.1.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.

#
U 2014/04/15 00:22:51.408414 XX.53.122.134:5060 -> 10.8.21.XX:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK348a4dc2;rport.
From: ;tag=as1ba98ffc.
To: ;tag=dGlp5o0FS.
Call-ID: wh8Ai1e~0c.
CSeq: 102 INVITE.
.

#
U 2014/04/15 00:22:51.408837 XX.53.122.134:5060 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK348a4dc2;rport.
From: ;tag=as1ba98ffc.
To: ;tag=dGlp5o0FS.
Call-ID: wh8Ai1e~0c.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Contact: ;+sip.instance="".
Content-Type: application/sdp.
Content-Length: 170.
.
v=0.
o=2000 274 61 IN IP4 192.168.0.38.
s=Talk.
c=IN IP4 192.168.0.38.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

#
U 2014/04/15 00:22:51.409343 10.8.21.XX:5060 -> XX.53.122.134:5060
ACK sip:2000@XX.53.122.134 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK04d7bdd5;rport.
Max-Forwards: 70.
From: sip:1000@sipdomain.com;tag=as1ba98ffc.
To: ;tag=dGlp5o0FS.
Contact: .
Call-ID: wh8Ai1e~0c.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.

#
U 2014/04/15 00:22:51.409471 10.8.21.XX:5060 -> XX.53.122.134:5060
BYE sip:2000@XX.53.122.134 SIP/2.0.
Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK1b9de0d9;rport.
Max-Forwards: 70.
From: sip:1000@sipdomain.com;tag=as1ba98ffc.
To: ;tag=dGlp5o0FS.
Call-ID: wh8Ai1e~0c.
CSeq: 103 BYE.
User-Agent: Asterisk PBX 11.8.1.
X-Asterisk-HangupCause: No user responding.
X-Asterisk-HangupCauseCode: 18.
Content-Length: 0.
.

#
U 2014/04/15 00:22:51.453121 223.XX.130.50:40764 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK0ac3adc4;rport.
From: ;tag=as679b5fe7.
To: ;tag=coOV3rP.
Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
CSeq: 104 BYE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
.

#
U 2014/04/15 00:22:51.495263 XX.53.122.134:5060 -> 10.8.21.XX:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK1b9de0d9;rport.
From: ;tag=as1ba98ffc.
To: ;tag=dGlp5o0FS.
Call-ID: wh8Ai1e~0c.
CSeq: 103 BYE.
User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
.

exit
37 received, 0 dropped

谢谢.

1> Vivek Raj..：

这个问题是由于防火墙和服务器上的问题引起的.你只需要按照以下步骤操作:1)首先进行防火墙设置并检查服务器的ip是否在那里列入白名单.2)如果您已经检查了以上几点,那么您肯定面临NAT问题,要解决此问题,您必须在sip.conf中添加以下参数

[general]
externip=XXX.XX.91.XX
localnet=10.2.32.12/255.255.255.0
nat=yes

30秒后Asterisk呼叫掉线

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