sip android 客户端,为什么星号无法与android sip客户端正常工作？

最新推荐文章于 2024-06-23 16:38:35 发布
最新推荐文章于 2024-06-23 16:38:35 发布
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sip android 客户端

星号= 1.8.11.0

Android = 2.3 / 4.0.3

Android Sip客户端=本机Android sip客户端/ sipdemo

当我使用zoiper / xlite从我的PC呼叫到android(本地android sip客户端)时,现在我可以听到双方的音频,但是当我从android呼叫到pc(zoiper / xlite)时,我什么都听不到.

另一方面,我已经在elastix(也使用星号1.8.11.0)上测试了这种情况,音频没有问题.

pc(zoiper)IP 192.168.15.27

安卓ip 192.168.15.71

星号服务器ip 192.168.15.118

从android调用zoiper时进行Sip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: <211>211>

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

OPTIONS sip:192.168.15.118 SIP/2.0

Call-ID: 5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71

CSeq: 7757 OPTIONS

From: "211" <211>;tag=1758376458211>

To: "211" <211>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" <211>;tag=1758376458211>

To: "211" <211>;tag=as6a8e1b47211>

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: <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: <211>211>

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" ;tag=as53340ecf

To: <211>211>

Contact:

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/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: <211>211>

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

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" <211>;tag=2465683119211>

To: <215>;tag=as573c52b3215>

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)

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" <211>;tag=2465683119211>

To: <215>;tag=as573c52b3215>

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 <215> for address/port to send to215>

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" <211>;tag=as404f0eb0211>

To: <215>;tag=96055240215>

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" <211>;tag=as404f0eb0211>

To: <215>;tag=96055240215>

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/2.0 200 OK

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

Contact: <215>215>

To: <215>;tag=96055240215>

From: "device"<211>;tag=as404f0eb0211>

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/2.0 200 OK

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

Contact: <215>215>

To: <215>;tag=96055240215>

From: "device"<211>;tag=as404f0eb0211>

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" ;tag=as4f0724aa

To: <211>211>

Contact:

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/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: <211>211>

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

OPTIONS sip:192.168.15.118 SIP/2.0

Call-ID: a5a311df861221d42844a8c485d4fee8@192.168.15.71

CSeq: 5815 OPTIONS

From: "211" <211>;tag=3109248316211>

To: "211" <211>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" <211>;tag=3109248316211>

To: "211" <211>;tag=as51223faf211>

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: <192.168.15.118:5060>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" ;tag=as7a9a1ea3

To: <211>211>

Contact:

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/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: <211>211>

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" ;tag=as5367b37c

To: <211>211>

Contact:

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/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: <211>211>

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(Zooper)呼叫Android

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: <211>;tag=4162167884211>

To: "device" <215>;tag=as5805dc66215>

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)

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: <211>;tag=4162167884211>

To: "device" <215>;tag=as5805dc66215>

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 <215> for address/port to send to215>

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: <211>;tag=as10377813211>

To: <215>;tag=50312112215>

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: <211>;tag=as10377813211>

To: <215>;tag=50312112215>

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: <215>215>

To: <215>;tag=50312112215>

From: <211>;tag=as10377813211>

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: <215>215>

To: <215>;tag=50312112215>

From: <211>;tag=as10377813211>

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" ;tag=as73902c1e

To: <211>211>

Contact:

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/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: <211>211>

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

OPTIONS sip:192.168.15.118 SIP/2.0

Call-ID: c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71

CSeq: 9273 OPTIONS

From: "211" <211>;tag=740019322211>

To: "211" <211>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" <211>;tag=740019322211>

To: "211" <211>;tag=as1bed6ef2211>

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: <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: <211>211>

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

OPTIONS sip:192.168.15.118 SIP/2.0

Call-ID: 18347a0db6841423591b08250847a1e0@192.168.15.71

CSeq: 3824 OPTIONS

From: "211" <211>;tag=841349553211>

To: "211" <211>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" <211>;tag=4017391219211>

To: "211" <211>;tag=as52fe1845211>

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: <192.168.15.118:5060>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" ;tag=as6e6638f8

To: <211>211>

Contact:

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/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: <211>211>

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" ;tag=as76426de6

To: <211>211>

Contact:

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.

我在extensions.conf中的拨号计划是：

[incoming-calls-wildcard]

exten => _2XX,hint,(SIP/${EXTEN},,120)

exten => _2XX,1,Dial(SIP/${EXTEN},,120)

exten => _2XX,n,Hangup

我的SIP帐户是：

[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

陈炳好
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