为什么星号不能正常使用 android sip 客户端? [英] Why asterisk not properly working with android sip client?

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本文介绍了为什么星号不能正常使用 android sip 客户端?的处理方法，对大家解决问题具有一定的参考价值，需要的朋友们下面随着小编来一起学习吧！

问题描述

星号= 1.8.11.0

Asterisk= 1.8.11.0

安卓= 2.3/4.0.3

Android= 2.3/4.0.3

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

Android Sip client=Native Android sip client/sipdemo

当我使用 zoiper/xlite 从我的电脑呼叫 android(原生 android sip 客户端)时，我现在可以听到来自双方的音频，但是当我从 android 呼叫 pc(zoiper/xlite)时，我在 android 上听不到任何声音.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 192.168.15.27安卓 ip 192.168.15.71星号服务器ip 192.168.15.118

When i call from my pc using zoiper/xlite to android (native android sip client) now i can hear audio from both sides but when i make call from android to pc (zoiper/xlite) i cannot hear anything on android. On the other hand i have tested this scenario on elastix (which also uses asterisk 1.8.11.0) with no problem in audio. pc(zoiper) ip 192.168.15.27 android ip 192.168.15.71 asterisk server ip 192.168.15.118

从 android 调用 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) 调用到 android 时

<--- 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....

I am using asterisk on local network (LAN)....

我在 extensions.conf 中的拨号方案是:

[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

推荐答案

感谢本论坛的每一位期待...我已经设法解决了这个问题...xlite/zoiper"和android"这两个设备native sip"客户端使用不同的默认音频编解码器.

Thanks for every one in this forum for anticipating...I have manged to solve the problem...The two devices "xlite/zoiper" and "android native sip" client uses different default audio codecs.

xlite 的默认编解码器是 BroadVoice-32

default codec for xlite is BroadVoice-32

zoiper 的默认编解码器是 GSM

default codec for zoiper is GSM

Android 的默认编解码器是 G.711 uLaw

default codec for android is 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 sip.conf 中创建 SIP 帐户时，我们可以通过以下方式强制两个通信客户端使用相同的音频编解码器.

As these devices should use same codec while communicating with each other..In my scenario these devices were using different codecs which results in one way audio (when calling from android to xlite/zoiper). While creating the SIP accounts in sip.conf while we can enforce two communicating clients to use same audio codec in following manner.

[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)

我们还可以通过在客户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..

We can also configure audio codec settings on client side by selecting the same audio codec on both sides..

这篇关于为什么星号不能正常使用 android sip 客户端?的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持IT屋！

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