Question

我是Asterisk的新手;我使用的是Asterisk 11和X-Lite客户端软电话。我获得了成功的连接，但是在32秒之后，呼叫被丢弃并且连接被切断。

WARNING[3830]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission YjExYmF for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
WARNING[3896]: chan_sip.c:4204 retrans_pkt: Hanging up call YjExYmF - no reply to out critical packet (see See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). 
  == Spawn extension (TextMenu, start, 2) exited non-zero on 'SIP/Sip01-00000000'

我有一个非常基本的sip.conf文件

[general]
bindaddr=0.0.0.0
port=5060

[Line](!)
type=friend
host=dynamic
context=LocalSets

[Sip01](Line)
secret=password

[Sip02](Line)
secret=password

[Sip03](Line)
secret=password

从＆＃34; sip set debug on＆＃34; sip log （注意，我用XX.XXX隐藏了我的ip地址的尾端）我很确定这是从54.187.XX.XXX到172.31.XX.XXX的转移问题，但我不知道如何配置这个

sip set debug on
SIP Debugging enabled
*CLI>
<--- SIP read from UDP:208.181.XX.XXX:51732 --->
INVITE sip:201@54.187.XX.XXX SIP/2.0
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ffdd1930cfcc0045-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:Sip01@208.181.XX.XXX:51732>
To: <sip:201@54.187.XX.XXX>
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4.7.0 73589-02bfb00b-W
Content-Length: 305

v=0
o=- 13050786034548600 1 IN IP4 208.181.XX.XXX
s=X-Lite release 4.7.0 stamp 73589
c=IN IP4 208.181.XX.XXX
t=0 0
m=audio 55020 RTP/AVP 125 0 100 9 8 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
Sending to 208.181.XX.XXX:51732 (no NAT)
Sending to 208.181.XX.XXX:51732 (no NAT)
Using INVITE request as basis request - YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
Found peer 'Sip01' for 'Sip01' from 208.181.XX.XXX:51732

<--- Reliably Transmitting (no NAT) to 208.181.XX.XXX:51732 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ffdd1930cfcc0045-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as6e6be255
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 1 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6969372e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:208.181.XX.XXX:51732 --->
ACK sip:201@54.187.XX.XXX SIP/2.0
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ffdd1930cfcc0045-1---d8754z-;rport
Max-Forwards: 70
To: <sip:201@54.187.XX.XXX>;tag=as6e6be255
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:208.181.XX.XXX:51732 --->
INVITE sip:201@54.187.XX.XXX SIP/2.0
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:Sip01@208.181.XX.XXX:51732>
To: <sip:201@54.187.XX.XXX>
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4.7.0 73589-02bfb00b-W
Authorization: Digest username="Sip01",realm="asterisk",nonce="6969372e",uri="sip:201@54.187.XX.XXX",response="ea76602a3f2458bf83c85c2e6115a1bf",algorithm=MD5
Content-Length: 305

v=0
o=- 13050786034548600 1 IN IP4 208.181.XX.XXX
s=X-Lite release 4.7.0 stamp 73589
c=IN IP4 208.181.XX.XXX
t=0 0
m=audio 55020 RTP/AVP 125 0 100 9 8 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 208.181.XX.XXX:51732 (no NAT)
Using INVITE request as basis request - YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
Found peer 'Sip01' for 'Sip01' from 208.181.XX.XXX:51732
  == Using SIP RTP CoS mark 5
Found RTP audio format 125
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format opus for ID 125
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw|speex16|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 208.181.XX.XXX:55020
Looking for 201 in LocalSets (domain 54.187.XX.XXX)
list_route: hop: <sip:Sip01@208.181.XX.XXX:51732>

<--- Transmitting (no NAT) to 208.181.XX.XXX:51732 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Length: 0


<------------>
    -- Executing [201@LocalSets:1] Goto("SIP/Sip01-00000000", "TestMenu,start,1") in new stack
    -- Goto (TestMenu,start,1)
    -- Executing [start@TestMenu:1] Answer("SIP/Sip01-00000000", "") in new stack
Audio is at 11910
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 208.181.XX.XXX:51732 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

<------------>
Retransmitting #1 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
    -- Executing [start@TestMenu:2] BackGround("SIP/Sip01-00000000", "enter-ext-of-person") in new stack
    -- <SIP/Sip01-00000000> Playing 'enter-ext-of-person.gsm' (language 'en')
Retransmitting #2 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
[Jul 25 18:20:36] NOTICE[3952][C-00000000]: res_rtp_asterisk.c:4100 ast_rtp_read: Unknown RTP codec 126 received from '208.181.XX.XXX:55020'
    -- Executing [start@TestMenu:3] WaitExten("SIP/Sip01-00000000", "60") in new stack
Retransmitting #3 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
Retransmitting #4 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
Retransmitting #5 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
Retransmitting #6 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
Retransmitting #7 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
Retransmitting #8 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
Retransmitting #9 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
Retransmitting #10 (no NAT) to 208.181.XX.XXX:51732:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.181.XX.XXX:51732;branch=z9hG4bK-d8754z-ae549b31d20fda42-1---d8754z-;received=208.181.XX.XXX;rport=51732
From: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
To: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 2 INVITE
Server: Asterisk PBX 11.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:201@172.31.XX.XXX:5060>
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 696150323 696150323 IN IP4 172.31.XX.XXX
s=Asterisk PBX 11.11.0
c=IN IP4 172.31.XX.XXX
t=0 0
m=audio 11910 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

---
[Jul 25 18:21:05] WARNING[3943]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Jul 25 18:21:05] WARNING[3943]: chan_sip.c:4204 retrans_pkt: Hanging up call YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (TestMenu, start, 3) exited non-zero on 'SIP/Sip01-00000000'
Scheduling destruction of SIP dialog 'YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:Sip01@208.181.XX.XXX:51732> for address/port to send to
set_destination: set destination to 208.181.XX.XXX:51732
Reliably Transmitting (no NAT) to 208.181.XX.XXX:51732:
BYE sip:Sip01@208.181.XX.XXX:51732 SIP/2.0
Via: SIP/2.0/UDP 172.31.XX.XXX:5060;branch=z9hG4bK48b15afb;rport
Max-Forwards: 70
From: <sip:201@54.187.XX.XXX>;tag=as768cf196
To: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.11.0
Proxy-Authorization: Digest username="Sip01", realm="asterisk", algorithm=MD5, uri="sip:54.187.XX.XXX", nonce="6969372e", response="256e79bf45a166dfb059c99bccb14689"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---

<--- SIP read from UDP:208.181.XX.XXX:51732 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.XX.XXX:5060;branch=z9hG4bK48b15afb;rport=5060;received=54.187.XX.XXX
Contact: <sip:Sip01@208.181.XX.XXX:51732>
To: "Chris"<sip:Sip01@54.187.XX.XXX>;tag=f77d6c22
From: <sip:201@54.187.XX.XXX>;tag=as768cf196
Call-ID: YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE
CSeq: 102 BYE
User-Agent: X-Lite release 4.7.0 stamp 73589 02bfb00b-W
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'YzA0NmM5NjY5MzU0ZmU0YWM4NGZiMDJhY2ZjZGU0MzE' Method: INVITE
sip set debug off
SIP Debugging Disabled

Answer 1

不确定您的问题是否已修复。但是正在重传200 OK，因为它没有收到ACK消息。

通常根据协议，ACK将被发送到200 OK中的联系地址。但在你的情况下，它似乎是在NAT后面，因此应该将ACK发送到接收200 OK的NAT地址。

我希望这能解决问题。

Answer 2

重新发送意味着另一方无法获取数据包或者不想对其做出响应。

考虑到xlite被证明可以使用软电话，最有可能你的目的地的路由器或路由器启用了防火墙或sip-ALG并阻止数据包

如果星号未从另一方获得响应，则认为呼叫失败并断开连接。

星号电话在32秒后掉线

2 个答案: