小贝子编程

为什么星号不能正常工作与安卓sip客户端



Asterisk= 1.8.11.0

Android = 2.3/4.0.3

Android Sip client=Native Android Sip client/sipdemo

当我使用zoiper/xlite从我的pc呼叫到android(原生android sip客户端)现在我可以听到来自双方的音频，但当我从android呼叫到pc (zoiper/xlite)我不能听到任何android上的声音。另一方面，我在elastix(也使用星号1.8.11.0)上测试了这个场景，音频没有问题。Pc (zoiper) IP 192.168.15.27Android IP 192.168.15.71星号服务器IP 192.168.15.118

从android调用到zoiper时的Sip调试

<--- SIP read from UDP:192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5996b0d9;rport=5060;received=192.168.15.118
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as05233e7d
To: <sip:211@192.168.15.71:45616;transport=udp>
Call-ID: 14151c2d29c039983aa449b56ce419e0@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '14151c2d29c039983aa449b56ce419e0@192.168.15.118:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.15.71:45616 --->
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: 5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71
CSeq: 7757 OPTIONS
From: "211" <sip:211@192.168.15.118>;tag=1758376458
To: "211" <sip:211@192.168.15.118>
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK6c57f04e92e9e34153367ca0e6e29675353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Looking for s in default (domain 192.168.15.118)
<--- Transmitting (NAT) to 192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK6c57f04e92e9e34153367ca0e6e29675353134;received=192.168.15.71;rport=45616
From: "211" <sip:211@192.168.15.118>;tag=1758376458
To: "211" <sip:211@192.168.15.118>;tag=as6a8e1b47
Call-ID: 5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71
CSeq: 7757 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.15.118:5060>
Accept: application/sdp
Content-Length: 0

<--- SIP read from UDP:192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1ecea84c;rport=5060;received=192.168.15.118
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as167765df
To: <sip:211@192.168.15.71:45616;transport=udp>
Call-ID: 5f4d210800aa07065fd35fba215e9783@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '5f4d210800aa07065fd35fba215e9783@192.168.15.118:5060' Method: OPTIONS
Really destroying SIP dialog '5e5f98ad4818911a86d4b438d054e39f@192.168.15.71' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK15f7593b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as53340ecf
To: <sip:211@192.168.15.71:45616;transport=udp>
Contact: <sip:asterisk@192.168.15.118:5060>
Call-ID: 3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
<--- SIP read from UDP:192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK15f7593b;rport=5060;received=192.168.15.118
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as53340ecf
To: <sip:211@192.168.15.71:45616;transport=udp>
Call-ID: 3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.15.71:45616 --->
BYE sip:215@192.168.15.118:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134
CSeq: 5511 BYE
From: "211" <sip:211@192.168.15.118>;tag=2465683119
To: <sip:215@192.168.15.118>;tag=as573c52b3
Call-ID: d188757cd6c044783fd413d11f8a982f@192.168.15.71
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Max-Forwards: 70
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.15.71:45616 (NAT)
Scheduling destruction of SIP dialog 'd188757cd6c044783fd413d11f8a982f@192.168.15.71' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134;received=192.168.15.71;rport=45616
From: "211" <sip:211@192.168.15.118>;tag=2465683119
To: <sip:215@192.168.15.118>;tag=as573c52b3
Call-ID: d188757cd6c044783fd413d11f8a982f@192.168.15.71
CSeq: 5511 BYE
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '0d06690561757f98637241394cc8dbba@192.168.15.118:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP> for address/port to send to
set_destination: set destination to 115.167.21.82:5060
Reliably Transmitting (NAT) to 192.168.15.27:5060:
BYE sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport
Max-Forwards: 70
From: "device" <sip:211@192.168.15.118>;tag=as404f0eb0
To: <sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP>;tag=96055240
Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.11.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (incoming-calls-wildcard, 215, 1) exited non-zero on 'SIP/211-00000008'
Retransmitting #1 (NAT) to 192.168.15.27:5060:
BYE sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport
Max-Forwards: 70
From: "device" <sip:211@192.168.15.118>;tag=as404f0eb0
To: <sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP>;tag=96055240
Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.11.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:192.168.15.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060
Contact: <sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP>
To: <sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP>;tag=96055240
From: "device"<sip:211@192.168.15.118>;tag=as404f0eb0
Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
CSeq: 103 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0
<------------->
-- (9 headers 0 lines) ---
Really destroying SIP dialog '0d06690561757f98637241394cc8dbba@192.168.15.118:5060' Method: INVITE
<--- SIP read from UDP:192.168.15.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060
Contact: <sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP>
To: <sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP>;tag=96055240
From: "device"<sip:211@192.168.15.118>;tag=as404f0eb0
Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
CSeq: 103 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0
<------------->
-- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK30d9d70f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as4f0724aa
To: <sip:211@192.168.15.71:45616;transport=udp>
Contact: <sip:asterisk@192.168.15.118:5060>
Call-ID: 05f27bf10f0a0759637b90252dc4366a@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
<--- SIP read from UDP:192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK30d9d70f;rport=5060;received=192.168.15.118
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as4f0724aa
To: <sip:211@192.168.15.71:45616;transport=udp>
Call-ID: 05f27bf10f0a0759637b90252dc4366a@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '05f27bf10f0a0759637b90252dc4366a@192.168.15.118:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.15.71:45616 --->
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: a5a311df861221d42844a8c485d4fee8@192.168.15.71
CSeq: 5815 OPTIONS
From: "211" <sip:211@192.168.15.118>;tag=3109248316
To: "211" <sip:211@192.168.15.118>
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK1e09e6f6c30b4b74ddc93c590abd3383353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Looking for s in default (domain 192.168.15.118)
<--- Transmitting (NAT) to 192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK1e09e6f6c30b4b74ddc93c590abd3383353134;received=192.168.15.71;rport=45616
From: "211" <sip:211@192.168.15.118>;tag=3109248316
To: "211" <sip:211@192.168.15.118>;tag=as51223faf
Call-ID: a5a311df861221d42844a8c485d4fee8@192.168.15.71
CSeq: 5815 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.15.118:5060>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'a5a311df861221d42844a8c485d4fee8@192.168.15.71' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1f8e44dc;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as7a9a1ea3
To: <sip:211@192.168.15.71:45616;transport=udp>
Contact: <sip:asterisk@192.168.15.118:5060>
Call-ID: 7ebcafc7159379fd047075a85c424588@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
<--- SIP read from UDP:192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1f8e44dc;rport=5060;received=192.168.15.118
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as7a9a1ea3
To: <sip:211@192.168.15.71:45616;transport=udp>
Call-ID: 7ebcafc7159379fd047075a85c424588@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '7ebcafc7159379fd047075a85c424588@192.168.15.118:5060' Method: OPTIONS
Really destroying SIP dialog 'd188757cd6c044783fd413d11f8a982f@192.168.15.71' Method: BYE
Really destroying SIP dialog 'a81e6a5f591141abd73f9dad478a6b56@192.168.15.71' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK017f2b0a;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as5367b37c
To: <sip:211@192.168.15.71:45616;transport=udp>
Contact: <sip:asterisk@192.168.15.118:5060>
Call-ID: 1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK017f2b0a;rport=5060;received=192.168.15.118
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as5367b37c
To: <sip:211@192.168.15.71:45616;transport=udp>
Call-ID: 1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118:5060' Method: OPTIONS

从pc (zoiper)呼叫android

<--- SIP read from UDP:192.168.15.71:45616 --->
BYE sip:215@192.168.15.118:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134
CSeq: 1 BYE
From: <sip:211@192.168.15.71:45616;transport=udp>;tag=4162167884
To: "device" <sip:215@192.168.15.118>;tag=as5805dc66
Call-ID: 2732e4564ce8534c5765a456045a9960@192.168.15.118:5060
Max-Forwards: 70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Sending to 192.168.15.71:45616 (NAT)
Scheduling destruction of SIP dialog '2732e4564ce8534c5765a456045a9960@192.168.15.118:5060' in 8576 ms (Method: BYE)
<--- Transmitting (NAT) to 192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134;received=192.168.15.71;rport=45616
From: <sip:211@192.168.15.71:45616;transport=udp>;tag=4162167884
To: "device" <sip:215@192.168.15.118>;tag=as5805dc66
Call-ID: 2732e4564ce8534c5765a456045a9960@192.168.15.118:5060
CSeq: 1 BYE
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (incoming-calls-wildcard, 211, 1) exited non-zero on 'SIP/215-0000000a'
Scheduling destruction of SIP dialog 'MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:215@115.167.21.82:5060;transport=UDP> for address/port to send to
set_destination: set destination to 115.167.21.82:5060
Reliably Transmitting (NAT) to 192.168.15.27:5060:
BYE sip:215@115.167.21.82:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
Max-Forwards: 70
From: <sip:211@192.168.15.118;transport=UDP>;tag=as10377813
To: <sip:215@192.168.15.118;transport=UDP>;tag=50312112
Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.11.0
Proxy-Authorization: Digest username="215", realm="asterisk", algorithm=MD5, uri="sip:192.168.15.118", nonce="", response="c897390cc8e4f674d7e9cd1efa7319a6"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
Retransmitting #1 (NAT) to 192.168.15.27:5060:
BYE sip:215@115.167.21.82:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
Max-Forwards: 70
From: <sip:211@192.168.15.118;transport=UDP>;tag=as10377813
To: <sip:215@192.168.15.118;transport=UDP>;tag=50312112
Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.11.0
Proxy-Authorization: Digest username="215", realm="asterisk", algorithm=MD5, uri="sip:192.168.15.118", nonce="", response="c897390cc8e4f674d7e9cd1efa7319a6"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
<--- SIP read from UDP:192.168.15.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060
Contact: <sip:215@115.167.21.82:5060;transport=UDP>
To: <sip:215@192.168.15.118;transport=UDP>;tag=50312112
From: <sip:211@192.168.15.118;transport=UDP>;tag=as10377813
Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
CSeq: 102 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.' Method: ACK
<--- SIP read from UDP:192.168.15.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060
Contact: <sip:215@115.167.21.82:5060;transport=UDP>
To: <sip:215@192.168.15.118;transport=UDP>;tag=50312112
From: <sip:211@192.168.15.118;transport=UDP>;tag=as10377813
Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
CSeq: 102 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK01e7e47d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as73902c1e
To: <sip:211@192.168.15.71:45616;transport=udp>
Contact: <sip:asterisk@192.168.15.118:5060>
Call-ID: 7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
<--- SIP read from UDP:192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK01e7e47d;rport=5060;received=192.168.15.118
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as73902c1e
To: <sip:211@192.168.15.71:45616;transport=udp>
Call-ID: 7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.15.71:45616 --->
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71
CSeq: 9273 OPTIONS
From: "211" <sip:211@192.168.15.118>;tag=740019322
To: "211" <sip:211@192.168.15.118>
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK5ac51b5cca5ad84320fc342692fdbb2d353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Looking for s in default (domain 192.168.15.118)
<--- Transmitting (NAT) to 192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK5ac51b5cca5ad84320fc342692fdbb2d353134;received=192.168.15.71;rport=45616
From: "211" <sip:211@192.168.15.118>;tag=740019322
To: "211" <sip:211@192.168.15.118>;tag=as1bed6ef2
Call-ID: c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71
CSeq: 9273 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.15.118:5060>
Accept: application/sdp
Content-Length: 0
<--- SIP read from UDP:192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK3c24b1cb;rport=5060;received=192.168.15.118
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as54c6581a
To: <sip:211@192.168.15.71:45616;transport=udp>
Call-ID: 7742af6c24e3e9a33efa0ba73051a29e@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '7742af6c24e3e9a33efa0ba73051a29e@192.168.15.118:5060' Method: OPTIONS
<--- SIP read from UDP:192.168.15.71:45616 --->
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: 18347a0db6841423591b08250847a1e0@192.168.15.71
CSeq: 3824 OPTIONS
From: "211" <sip:211@192.168.15.118>;tag=841349553
To: "211" <sip:211@192.168.15.118>
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK79bb575a5d0f832291b861987849d9a2353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Looking for s in default (domain 192.168.15.118)

<------------->
--- (9 headers 0 lines) ---
Looking for s in default (domain 192.168.15.118)
<--- Transmitting (NAT) to 192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK40d81d1686d23c9133e52f4180c2accb353134;received=192.168.15.71;rport=45616
From: "211" <sip:211@192.168.15.118>;tag=4017391219
To: "211" <sip:211@192.168.15.118>;tag=as52fe1845
Call-ID: 09809b7ed9a68b7f270f33f1d8de13dc@192.168.15.71
CSeq: 4619 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.15.118:5060>
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '09809b7ed9a68b7f270f33f1d8de13dc@192.168.15.71' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK13b8cca8;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as6e6638f8
To: <sip:211@192.168.15.71:45616;transport=udp>
Contact: <sip:asterisk@192.168.15.118:5060>
Call-ID: 76f6f373769041c3462ebb676641f1b7@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
<--- SIP read from UDP:192.168.15.71:45616 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK13b8cca8;rport=5060;received=192.168.15.118
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as6e6638f8
To: <sip:211@192.168.15.71:45616;transport=udp>
Call-ID: 76f6f373769041c3462ebb676641f1b7@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '76f6f373769041c3462ebb676641f1b7@192.168.15.118:5060' Method: OPTIONS
Really destroying SIP dialog '9eeee094f46eec920ac462e291314bde@192.168.15.71' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK47a8a134;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.15.118>;tag=as76426de6
To: <sip:211@192.168.15.71:45616;transport=udp>
Contact: <sip:asterisk@192.168.15.118:5060>
Call-ID: 3a98a25b41dc3b3e699ee4383669e984@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

我在本地网络(LAN)上使用星号....

我的拨号计划在extensions.conf是:

[incoming-calls-wildcard]
exten => _2XX,hint,(SIP/${EXTEN},,120)
exten => _2XX,1,Dial(SIP/${EXTEN},,120)
exten => _2XX,n,Hangup

我的sip帐号是:

[215]
deny=0.0.0.0/0.0.0.0
secret=very123
dtmfmode=rfc2833
canreinvite=no
context=incoming-calls-wildcard
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/215
mailbox=215@device
permit=0.0.0.0/0.0.0.0
callerid=device <215>
callcounter=yes
faxdetect=no

感谢论坛里的每一个人对…我已经设法解决了这个问题。两个设备"xlite/zoiper"one_answers"android native sip"客户端使用不同的默认音频编解码器。

xlite的默认编解码器是broadvice -32

zip文件默认的编解码器是GSM

android的默认编解码器是G.711 uLaw

因为这些设备在相互通信时应该使用相同的编解码器，在我的场景中，这些设备使用不同的编解码器，导致单向音频(当从android调用xlite/zoiper时)。当在SIP .conf中创建SIP帐号时，我们可以强制两个通信客户端使用相同的音频编解码器。

[211]
deny=0.0.0.0/0.0.0.0
secret=123456
dtmfmode=rfc2833
canreinvite=no
context=incoming-calls-wildcard
host=dynamic
nat=yes
type=friend
port=5060
qualify=yes
callgroup=
pickupgroup=
permit=0.0.0.0/0.0.0.0
callcounter=yes
faxdetect=no
disallow=all    (disable default audio codec)
allow=ulaw   (allow uLaw audio codec)

[215]
deny=0.0.0.0/0.0.0.0
secret=123456
dtmfmode=rfc2833
canreinvite=no
context=incoming-calls-wildcard
host=dynamic
nat=no
type=friend
port=5060
qualify=yes
callgroup=
pickupgroup=
permit=0.0.0.0/0.0.0.0
callcounter=yes
faxdetect=no
disallow=all    (disable default audio codec)
allow=ulaw   (allow uLaw audio codec)

我们也可以在客户端通过选择相同的音频编解码器来配置音频编解码器设置。

嗯，

配置方面，看起来您的配置是可以的。但是，在我看来奇怪的是，在您的跟踪中有以下IP地址- 115.167.21.82

你能详细说明这个IP地址是什么吗?我的假设是你位于NAT防火墙后面，这是你的外部IP地址。在这种情况下，查看/etc/asterisk/sip.conf中的"localnet"定义，并定义本地网络地址范围。

当使用Elastix时，它能够自动设置那个-但是当进行手动安装时，您必须设置那个。

Nir S

看起来你有NAT问题:

http://www.voip-info.org/wiki/view/Asterisk + SIP + NAT +解决方案

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