我已经安装了Asterisk 11，两个wifi手机可以通过星号服务器进行通话。然而，wifi电话和LTE（4G）电话无法发出声音。
Asterisk 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=ulaw ; Allow codecs in order of preference
allow=alaw
register => 12121111111:1234:11111111@sipauth.deltathree.com/1000
srvlookup=no
directrtpsetup=yes
trustpid=yes
sendrpid=no
qualify=yes
callevents=yes
insecure=invite
pedantic=no
videosupport=yes
canreinvite=yes
nat=yes
externip=XXX.XXX.91.12
localnet=10.7.21.4/255.255.255.0
qualify=yes
directmedia=yes
啜饮设置
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: Yes
User Agent: Asterisk PBX 11.8.1
SDP Session Name: Asterisk PBX 11.8.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: On
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Enabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Enabled using externhost
Externhost: XXX.52.91.12:0
Externaddr: XXX.52.91.12:0
Externrefresh: 600
Localnet: XX.7.21.0/255.255.255.0
XX.7.21.0/255.255.255.0
Global Signalling Settings:
---------------------------
Codecs: (ulaw|alaw)
Codec Order: ulaw:20,alaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 2000
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
Realtime SIP Settings:
----------------------
Realtime Peers: Yes
Realtime Regs: No
Cache Friends: No
Update: Yes
Ignore Reg. Expire: No
Save sys. name: No
Auto Clear: 120 (Disabled)
啜饮日志 当我看着sip日志时，它看起来很好。我只看到一个＆＃34;邀请＆＃34;从服务器到wifi电话。
interface: eth0 (10.7.21.0/255.255.255.0)
filter: ( port 5060 ) and (ip or ip6)
#
U 2014/04/16 22:46:28.514023 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
INVITE sip:2000@asterisk-sip-domain.com SIP/2.0.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;rport.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>.
CSeq: 20 INVITE.
Call-ID: Z6lXHBKOyd.
Max-Forwards: 70.
Supported: replaces, outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Content-Type: application/sdp.
Content-Length: 372.
Contact: <sip:1000@//WIFI-PUBLIC-IP//:1495>;+sip.instance="<urn:uuid:41bf1699-9e9a-4817-8b8c-e51f7b4ae2dc>".
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
.
v=0.
o=1000 2350 2859 IN IP4 //WIFI-PRIVATE-IP//.
s=Talk.
c=IN IP4 //WIFI-PRIVATE-IP//.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 124 120 111 110 0 8 101.
a=rtpmap:124 opus/48000.
a=fmtp:124 useinbandfec=1; usedtx=1.
a=rtpmap:120 SILK/16000.
a=rtpmap:111 speex/16000.
a=fmtp:111 vbr=on.
a=rtpmap:110 speex/8000.
a=fmtp:110 vbr=on.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 2014/04/16 22:46:28.517399 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;received=//WIFI-PUBLIC-IP//;rport=1495.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>.
Call-ID: Z6lXHBKOyd.
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:2000@//AMAZON-EC2-SERVER//:5060>.
Content-Length: 0.
.
#
U 2014/04/16 22:46:28.522887 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
INVITE sip:2000@//LTE-PHONE-PUBLIC-IP//:63968 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
Max-Forwards: 70.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>.
Contact: <sip:1000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Wed, 16 Apr 2014 13:46:28 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 1526682879 1526682879 IN IP4 //WIFI-PRIVATE-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //WIFI-PRIVATE-IP//.
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 2014/04/16 22:46:29.022450 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
INVITE sip:2000@//LTE-PHONE-PUBLIC-IP//:63968 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
Max-Forwards: 70.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>.
Contact: <sip:1000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Wed, 16 Apr 2014 13:46:28 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 1526682879 1526682879 IN IP4 //WIFI-PRIVATE-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //WIFI-PRIVATE-IP//.
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 2014/04/16 22:46:29.113047 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: sip:2000@//LTE-PHONE-PUBLIC-IP//:63968.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
.
#
U 2014/04/16 22:46:29.426139 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
.
#
U 2014/04/16 22:46:29.426158 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
.
#
U 2014/04/16 22:46:29.427976 f:5060 -> //WIFI-PUBLIC-IP//:1495
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;received=//WIFI-PUBLIC-IP//;rport=1495.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
Call-ID: Z6lXHBKOyd.
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:2000@//AMAZON-EC2-SERVER//:5060>.
Content-Length: 0.
.
** (WHY IT MAKES ONE MORE INVITE FROM SERVER TO WIFI-PHONE???)**
#
U 2014/04/16 22:46:30.307448 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:48504
INVITE sip:1000@//WIFI-PUBLIC-IP//:48504 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK451726cf;rport.
Max-Forwards: 70.
From: <sip:2000@//AMAZON-EC2-SERVER//>;tag=as30b8a8a5.
To: <sip:1000@//WIFI-PUBLIC-IP//:48504>.
Contact: <sip:2000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: 1c2fd2cd6a4ac372408845e8077ba2b5@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.8.1.
Date: Wed, 16 Apr 2014 13:46:14 GMT.
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 741350827 741350827 IN IP4 //LTE-PHONE-PUBLIC-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //LTE-PHONE-PUBLIC-IP//.
t=0 0.
m=audio 30390 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 2014/04/16 22:46:30.816230 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK12ab34f9;rport.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Contact: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;+sip.instance="<urn:uuid:8afceca3-368f-4f57-a586-6056d3492371>".
Content-Type: application/sdp.
Content-Length: 183.
.
v=0.
o=2000 2310 1562 IN IP4 //LTE-PHONE-PUBLIC-IP//.
s=Talk.
c=IN IP4 //LTE-PHONE-PUBLIC-IP//.
b=AS:380.
t=0 0.
m=audio 30390 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 2014/04/16 22:46:30.816888 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
ACK sip:2000@//LTE-PHONE-PUBLIC-IP//:63968 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK680dd0d2;rport.
Max-Forwards: 70.
From: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
To: <sip:2000@//LTE-PHONE-PUBLIC-IP//:63968>;tag=zZBSo25.
Contact: <sip:1000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.
#
U 2014/04/16 22:46:30.817278 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
SIP/2.0 200 OK.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;branch=z9hG4bK.FuKwv3ZMW;received=//WIFI-PUBLIC-IP//;rport=1495.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
Call-ID: Z6lXHBKOyd.
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:2000@//AMAZON-EC2-SERVER//:5060>.
Content-Type: application/sdp.
Content-Length: 261.
.
v=0.
o=root 1551912347 1551912347 IN IP4 //LTE-PHONE-PUBLIC-IP//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //LTE-PHONE-PUBLIC-IP//.
t=0 0.
m=audio 30390 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 2014/04/16 22:46:30.925455 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
ACK sip:2000@//AMAZON-EC2-SERVER//:5060 SIP/2.0.
Via: SIP/2.0/UDP //WIFI-PRIVATE-IP//:48504;rport;branch=z9hG4bK.qV8rz6rI4.
From: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
To: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
CSeq: 20 ACK.
Call-ID: Z6lXHBKOyd.
Max-Forwards: 70.
.
#
U 2014/04/16 22:46:35.277987 //LTE-PHONE-PUBLIC-IP//:63968 -> //AMAZON-EC2-PRIVATE-IP//:5060
BYE sip:1000@//AMAZON-EC2-SERVER//:5060 SIP/2.0.
Via: SIP/2.0/UDP //LTE-PHONE-PUBLIC-IP//:63968;branch=z9hG4bK.Jfn1vpiLT;rport.
From: <sip:2000@//LTE-PHONE-PUBLIC-IP//>;tag=zZBSo25.
To: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
CSeq: 111 BYE.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
Max-Forwards: 70.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
.
#
U 2014/04/16 22:46:35.278525 //AMAZON-EC2-PRIVATE-IP//:5060 -> //LTE-PHONE-PUBLIC-IP//:63968
SIP/2.0 200 OK.
Via: SIP/2.0/UDP //LTE-PHONE-PUBLIC-IP//:63968;branch=z9hG4bK.Jfn1vpiLT;received=//LTE-PHONE-PUBLIC-IP//;rport=63968.
From: <sip:2000@//LTE-PHONE-PUBLIC-IP//>;tag=zZBSo25.
To: <sip:1000@//AMAZON-EC2-SERVER//>;tag=as4a4e67da.
Call-ID: 60d3866362c2076357b37d2d4b930652@//AMAZON-EC2-SERVER//:5060.
CSeq: 111 BYE.
Server: Asterisk PBX 11.8.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
.
#
U 2014/04/16 22:46:35.278797 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
INVITE sip:1000@//WIFI-PUBLIC-IP//:1495 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK1930727f;rport.
Max-Forwards: 70.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Contact: <sip:2000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: Z6lXHBKOyd.
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 1551912347 1551912348 IN IP4 //AMAZON-EC2-SERVER//.
s=Asterisk PBX 11.8.1.
c=IN IP4 //AMAZON-EC2-SERVER//.
t=0 0.
m=audio 19500 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 2014/04/16 22:46:35.418765 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK1930727f;rport.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 102 INVITE.
.
#
U 2014/04/16 22:46:35.441248 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK1930727f;rport.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 102 INVITE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE.
Contact: <sip:1000@//WIFI-PUBLIC-IP//:1495>;+sip.instance="<urn:uuid:41bf1699-9e9a-4817-8b8c-e51f7b4ae2dc>".
Content-Type: application/sdp.
Content-Length: 180.
.
v=0.
o=1000 2350 2861 IN IP4 //WIFI-PRIVATE-IP//.
s=Talk.
c=IN IP4 //WIFI-PRIVATE-IP//.
b=AS:380.
t=0 0.
m=audio 7076 RTP/AVP 0 8 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 2014/04/16 22:46:35.441661 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
ACK sip:1000@//WIFI-PUBLIC-IP//:1495 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK7dd12d7d;rport.
Max-Forwards: 70.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Contact: <sip:2000@//AMAZON-EC2-SERVER//:5060>.
Call-ID: Z6lXHBKOyd.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 11.8.1.
Content-Length: 0.
.
#
U 2014/04/16 22:46:35.441754 //AMAZON-EC2-PRIVATE-IP//:5060 -> //WIFI-PUBLIC-IP//:1495
BYE sip:1000@//WIFI-PUBLIC-IP//:1495 SIP/2.0.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK48ef4999;rport.
Max-Forwards: 70.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 103 BYE.
User-Agent: Asterisk PBX 11.8.1.
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Content-Length: 0.
.
#
U 2014/04/16 22:46:35.474403 //WIFI-PUBLIC-IP//:1495 -> //AMAZON-EC2-PRIVATE-IP//:5060
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP //AMAZON-EC2-SERVER//:5060;branch=z9hG4bK48ef4999;rport.
From: "........." <sip:2000@asterisk-sip-domain.com>;tag=as380612c6.
To: <sip:1000@asterisk-sip-domain.com>;tag=8ClA8ivYF.
Call-ID: Z6lXHBKOyd.
CSeq: 103 BYE.
User-Agent: LinphoneAndroid/1.0.1 (belle-sip/1.3.1).
Supported: replaces, outbound.
.
exit
21 received, 0 dropped
你知道为什么当设备在LTE（4G）网络上时它无法发出声音吗？
答案 0 :(得分：0)
我在sip.conf和用户的conf中更改了以下代码。
canreinvite = yes
这一切都很好。但是，它通过Asterisk Server传送声音，这意味着服务器必须处理语音流量。