Android权限只读rwrr,为什么星号不能正确使用android sip客户端？

的Android 2.3 =/4.0.3

Android的SIP客户端=原生的Android SIP客户端/ sipdemo

当我使用zoiper/X-Lite到Android，从我的PC呼叫(本地android的sip客户端)现在我可以听到双方的音频，但是当我从android到pc(zoiper/xlite)拨打电话时，我无法听到任何关于android的声音。 另一方面，我测试了elastix(也使用星号1.8.11.0)这个场景，没有音频问题。 PC(zoiper)IP 192.168.15.27 安卓IP 192.168.15.71 星号服务器的ip从机器人打电话来zoiper时192.168.15.118

啜调试。

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" ;tag=as05233e7d

To:

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

Content-Length: 0

--- (7 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

OPTIONS sip:192.168.15.118 SIP/2.0

Call-ID: [email protected]

CSeq: 7757 OPTIONS

From: "211" ;tag=1758376458

To: "211"

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)

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" ;tag=1758376458

To: "211" ;tag=as6a8e1b47

Call-ID: [email protected]

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: <192.168.15.118:5060>192.168.15.118:5060>

Accept: application/sdp

Content-Length: 0

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" ;tag=as167765df

To:

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

Content-Length: 0

--- (7 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

Really destroying SIP dialog '[email protected]' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.15.71:45616:

OPTIONS sip:[email protected]:45616;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK15f7593b;rport

Max-Forwards: 70

From: "asterisk" ;tag=as53340ecf

To:

Contact:

Call-ID: [email protected]: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/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" ;tag=as53340ecf

To:

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

Content-Length: 0

--- (7 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

BYE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134

CSeq: 5511 BYE

From: "211" ;tag=2465683119

To: ;tag=as573c52b3

Call-ID: [email protected]

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 '[email protected]' in 6400 ms (Method: BYE)

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" ;tag=2465683119

To: ;tag=as573c52b3

Call-ID: [email protected]

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 '[email protected]:5060' in 6400 ms (Method: INVITE)

set_destination: Parsing 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:[email protected]: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" ;tag=as404f0eb0

To: ;tag=96055240

Call-ID: [email protected]: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:[email protected]: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" ;tag=as404f0eb0

To: ;tag=96055240

Call-ID: [email protected]: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/2.0 200 OK

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060

Contact:

To: ;tag=96055240

From: "device";tag=as404f0eb0

Call-ID: [email protected]:5060

CSeq: 103 BYE

User-Agent: Zoiper for Windows 2.39 r16838

Content-Length: 0

-- (9 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: INVITE

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060

Contact:

To: ;tag=96055240

From: "device";tag=as404f0eb0

Call-ID: [email protected]: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:[email protected]:45616;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK30d9d70f;rport

Max-Forwards: 70

From: "asterisk" ;tag=as4f0724aa

To:

Contact:

Call-ID: [email protected]: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/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" ;tag=as4f0724aa

To:

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

Content-Length: 0

--- (7 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

OPTIONS sip:192.168.15.118 SIP/2.0

Call-ID: [email protected]

CSeq: 5815 OPTIONS

From: "211" ;tag=3109248316

To: "211"

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)

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" ;tag=3109248316

To: "211" ;tag=as51223faf

Call-ID: [email protected]

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: <192.168.15.118:5060>192.168.15.118:5060>

Accept: application/sdp

Content-Length: 0

Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)

Reliably Transmitting (NAT) to 192.168.15.71:45616:

OPTIONS sip:[email protected]:45616;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1f8e44dc;rport

Max-Forwards: 70

From: "asterisk" ;tag=as7a9a1ea3

To:

Contact:

Call-ID: [email protected]: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/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" ;tag=as7a9a1ea3

To:

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

Content-Length: 0

--- (7 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

Really destroying SIP dialog '[email protected]' Method: BYE

Really destroying SIP dialog '[email protected]' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.15.71:45616:

OPTIONS sip:[email protected]:45616;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK017f2b0a;rport

Max-Forwards: 70

From: "asterisk" ;tag=as5367b37c

To:

Contact:

Call-ID: [email protected]: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/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" ;tag=as5367b37c

To:

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

Content-Length: 0

--- (7 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

当从PC(zoiper)打电话到Android

BYE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134

CSeq: 1 BYE

From: ;tag=4162167884

To: "device" ;tag=as5805dc66

Call-ID: [email protected]:5060

Max-Forwards: 70

Content-Length: 0

--- (8 headers 0 lines) ---

Sending to 192.168.15.71:45616 (NAT)

Scheduling destruction of SIP dialog '[email protected]:5060' in 8576 ms (Method: BYE)

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: ;tag=4162167884

To: "device" ;tag=as5805dc66

Call-ID: [email protected]: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 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:[email protected]:5060;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport

Max-Forwards: 70

From: ;tag=as10377813

To: ;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:[email protected]:5060;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport

Max-Forwards: 70

From: ;tag=as10377813

To: ;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/2.0 200 OK

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060

Contact:

To: ;tag=50312112

From: ;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/2.0 200 OK

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060

Contact:

To: ;tag=50312112

From: ;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:[email protected]:45616;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK01e7e47d;rport

Max-Forwards: 70

From: "asterisk" ;tag=as73902c1e

To:

Contact:

Call-ID: [email protected]: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/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" ;tag=as73902c1e

To:

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

Content-Length: 0

--- (7 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

OPTIONS sip:192.168.15.118 SIP/2.0

Call-ID: [email protected]

CSeq: 9273 OPTIONS

From: "211" ;tag=740019322

To: "211"

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)

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" ;tag=740019322

To: "211" ;tag=as1bed6ef2

Call-ID: [email protected]

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: <192.168.15.118:5060>192.168.15.118:5060>

Accept: application/sdp

Content-Length: 0

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" ;tag=as54c6581a

To:

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

Content-Length: 0

--- (7 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

OPTIONS sip:192.168.15.118 SIP/2.0

Call-ID: [email protected]

CSeq: 3824 OPTIONS

From: "211" ;tag=841349553

To: "211"

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)

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" ;tag=4017391219

To: "211" ;tag=as52fe1845

Call-ID: [email protected]

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: <192.168.15.118:5060>192.168.15.118:5060>

Accept: application/sdp

Content-Length: 0

Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)

Reliably Transmitting (NAT) to 192.168.15.71:45616:

OPTIONS sip:[email protected]:45616;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK13b8cca8;rport

Max-Forwards: 70

From: "asterisk" ;tag=as6e6638f8

To:

Contact:

Call-ID: [email protected]: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/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" ;tag=as6e6638f8

To:

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

Content-Length: 0

--- (7 headers 0 lines) ---

Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

Really destroying SIP dialog '[email protected]' Method: OPTIONS

Reliably Transmitting (NAT) to 192.168.15.71:45616:

OPTIONS sip:[email protected]:45616;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK47a8a134;rport

Max-Forwards: 70

From: "asterisk" ;tag=as76426de6

To:

Contact:

Call-ID: [email protected]: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

我使用局域网(LAN)上星号....在扩展

我的拨号计划.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账号是：

[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

[email protected]

permit=0.0.0.0/0.0.0.0

callerid=device <215>

callcounter=yes

faxdetect=no

2013-01-13

Zain