Question

Asterisk = 1.8.11.0

Android = 2.3 / 4.0.3

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

当我从我的电脑使用zoiper / xlite调用android（原生android sip客户端）时，现在我可以听到来自双方的音频但是当我从android到pc（zoiper / xlite）打电话时我听不到任何关于android的内容。另一方面，我已经在elastix（也使用星号1.8.11.0）上测试了这个场景，音频没问题。 pc（zoiper）ip 192.168.15.27 android ip 192.168.15.71 星号服务器ip 192.168.15.118

从android到zoiper调用时调试调试。

<--- 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）调用到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

Answer 1

感谢本论坛中的每一个人都期待...我已经解决了这个问题...两个设备“xlite / zoiper”和“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BroadVoice-32

zoiper的默认编解码器是GSM

android的默认编解码器是G.711 uLaw

由于这些设备应该在相互通信时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，这导致单向音频（当从android调用到xlite / zoiper时）。在sip.conf中创建SIP帐户时，我们可以强制两个通信客户端以下列方式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)

我们还可以通过在两端选择相同的音频编解码器来配置客户端的音频编解码器设置。

Answer 2

好吧，

配置方面，您的配置似乎正常。但是，对我来说似乎很奇怪的是跟踪内的以下IP地址 - 115.167.21.82

请详细说明这个IP地址是什么？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，请在/etc/asterisk/sip.conf中查找“localnet”定义，并定义本地网络地址范围。

使用Elastix时，它可以自动设置 - 但是在进行手动安装时，您必须设置一个。

Nir S

Answer 3

看起来你有nat问题：

http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

为什么星号不能正常使用android sip客户端？

3 个答案: