Question

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上更改一些内容，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记录

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: <sip:2000@sipdomain.com>;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:2000@XX.53.122.134>;+sip.instance="<urn:uuid:0b49a090-f01c-41a2-b771-bdc956e9b516>".
.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>.
Contact: <sip:2000@10.8.21.XX:5060>.
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: <sip:2000@10.8.21.XX>;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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;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:1000@223.XX.130.50:40764>;+sip.instance="<urn:uuid:307db642-fa79-44d8-835f-15152558c31a>".
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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Contact: <sip:2000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
Contact: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Contact: <sip:2000@10.8.21.XX:5060>.
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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;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:1000@223.XX.130.50:40764>;+sip.instance="<urn:uuid:307db642-fa79-44d8-835f-15152558c31a>".
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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;tag=coOV3rP.
Contact: <sip:2000@10.8.21.XX:5060>.
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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;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: <sip:1000@sipdomain.com>;tag=as1ba98ffc.
To: <sip:2000@sipdomain.com>;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: <sip:1000@sipdomain.com>;tag=as1ba98ffc.
To: <sip:2000@sipdomain.com>;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:2000@XX.53.122.134>;+sip.instance="<urn:uuid:0b49a090-f01c-41a2-b771-bdc956e9b516>".
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: <sip:2000@sipdomain.com>;tag=dGlp5o0FS.
Contact: <sip:1000@10.8.21.XX:5060>.
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: <sip:2000@sipdomain.com>;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: <sip:2000@10.8.21.XX>;tag=as679b5fe7.
To: <sip:1000@223.XX.130.50:40764>;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: <sip:1000@sipdomain.com>;tag=as1ba98ffc.
To: <sip:2000@sipdomain.com>;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。

Answer 1

这个问题是由防火墙和服务器上的问题引起的。您只需按照以下步骤操作： 1）首先进行防火墙设置，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

Answer 2

我发现了!!真棒！这是localnet地址问题。我想放置公共（外部）IP地址和私有IP地址。

sip.conf

allow=g729
srvlookup=no
directrtpsetup=yes
trustpid=yes
sendrpid=no
qualify=yes
callevents=yes
insecure=invite
pedantic=no
videosupport=yes
t38pt_udptl=no
t38pt_rtp=no
t38pt_tcp=no
canreinvite=yes
nat=yes
externip=XXX.XX.91.XX
localnet=10.2.32.12/255.255.255.0

Answer 3

       UA1              Your Asterisk Server     UA2
      (IPv4)            (IPv4/IPv6)             (IPv6)
        |                    |                    |
        |   F1 INVITE        |                    |
        |------------------->|      F2 INVITE     |
        |                    |------------------->|
        |    100 Trying      |                    |
        |<-------------------|                    |
        |                    |    F3 200 OK       |
        |    F4 200 OK       |<-------------------|
        |<-------------------|                    |
        |                    |                    |
        |       F5 ACK       |                    |
        |------------------->|       F6 ACK       |
        |                    |------------------->|
        |                    |                    |
        |                    |        F7 BYE      |
        |       F8 BYE       |<-------------------|
        |<-------------------|                    |

问题在于你的UA1没有从第二个UA2获得ACK。我有同样的问题，我知道每个sip拨号器默认30秒的sip呼叫超时，所以它在30秒后因为UA2没有收到ACK信号而挂起。通过cli上面的命令发布你的完整堆栈轨道，所以我可以帮你解决这个问题。

CLI> sip set debug on

我发现这背后的一个原因是NAT问题。您的设备位于NAT后面并且星号无法向您注册的设备发送ACK信号，因此它会为ACK信号提供重新传输超时。

Answer 4

看起来你启用了防火墙？

您需要调试SIP数据包。

在asterisk CLI上执行：

set sip debug on

Answer 5

我有同样的问题...

with debug

#set debug on

我发现修复问题的解决方案落后于 NAT 。在/ etc / host中添加你的iplocal，例如

192.168.0.X domain.com

/etc/host --> (add line 192.168.0.X domain/ddns, such as, 192.168.0.2 abdef.com) 
/etc/host --> remove line --> 127.0.0.1 localhost & other ip
/etc/hostname, change your default hostname with your domain or ddns or ip-public

Answer 6

非常简单在里面 sip.conf文件，您必须提及本地IP地址和子网掩码地址。举个例子。 localnet = 10.2.32.12 / 255.255.255.0

要获取系统的IP地址，请在以下命令中输入

：

对于Windows：ipconfig

对于ubuntu：ifconfig

30秒后Asterisk呼叫掉线

6 个答案: