为什么星号无法正常使用Android的SIP客户端的工作？ [英] Why asterisk not properly working with android sip client?
本文介绍了为什么星号无法正常使用Android的SIP客户端的工作？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
Asterisk的= 1.8.11.0
=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（Android原生的SIP客户端），现在我可以听到双方的声音，但是当我拨打电话，从机器人到PC（zoiper / X-Lite）我不能听到在Android任何从我的电脑打电话。
在另一方面，我已经测试过的Elastix这种情况下（也使用星号1.8.11.0）与音频没有问题。
PC（zoiper）IP 192.168.15.27
Android的IP 192.168.15.71
星号服务器的ip 192.168.15.118
在Android打电话来zoiper当 SIP调试。
＆LT; --- SIP从UDP阅读：192.168.15.71：45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK5996b0d9;rport=5060;received=192.168.15.118
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as05233e7d
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
呼叫ID：14151c2d29c039983aa449b56ce419e0@192.168.15.118：5060
这个Cseq：102 OPTIONS
内容长度：0＆LT;＆------------- GT;
---（7头0线）---
真正摧毁SIP对话'14151c2d29c039983aa449b56ce419e0@192.168.15.118：5060'方法：OPTIONS＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
OPTIONS SIP：192.168.15.118 SIP / 2.0
呼叫ID：5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71
这个Cseq：7757 OPTIONS
从211＆LT; SIP：211@192.168.15.118> ;;标签= 1758376458
要：211＆LT; SIP：211@192.168.15.118>
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bK6c57f04e92e9e34153367ca0e6e29675353134;rport
马克斯 - 前锋：70
用户代理：SIPAUA / 0.1.001
内容长度：0＆LT;＆------------- GT;
---（9头0线）---
在默认情况下寻找S（域192.168.15.118）＆LT; ---发射（NAT）来192.168.15.71:45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bK6c57f04e92e9e34153367ca0e6e29675353134;received=192.168.15.71;rport=45616
从211＆LT; SIP：211@192.168.15.118> ;;标签= 1758376458
要：211＆LT; SIP：211@192.168.15.118> ;;标签= as6a8e1b47
呼叫ID：5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71
这个Cseq：7757 OPTIONS
服务器：的Asterisk PBX 1.8.11.0
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
联系方式：其中，SIP：192.168.15.118：5060＆GT;
接受：应用/ SDP
内容长度：0
＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK1ecea84c;rport=5060;received=192.168.15.118
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as167765df
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
呼叫ID：5f4d210800aa07065fd35fba215e9783@192.168.15.118：5060
这个Cseq：102 OPTIONS
内容长度：0＆LT;＆------------- GT;
---（7头0线）---
真正摧毁SIP对话'5f4d210800aa07065fd35fba215e9783@192.168.15.118：5060'方法：OPTIONS
真正摧毁SIP对话5e5f98ad4818911a86d4b438d054e39f@192.168.15.71'方法：OPTIONS
可靠地发射（NAT）来192.168.15.71:45616：
OPTIONS SIP：211@192.168.15.71：45616;运输= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK15f7593b;rport
马克斯 - 前锋：70
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as53340ecf
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
联系方式：其中，SIP：asterisk@192.168.15.118：5060＆GT;
呼叫ID：3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118：5060
这个Cseq：102 OPTIONS
用户代理：的Asterisk PBX 1.8.11.0
日期：星期六，2013年1月12日19时44分34秒GMT
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
内容长度：0
---＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK15f7593b;rport=5060;received=192.168.15.118
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as53340ecf
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
呼叫ID：3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118：5060
这个Cseq：102 OPTIONS
内容长度：0＆LT;＆------------- GT;
---（7头0线）---
真正摧毁SIP对话'3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118：5060'方法：OPTIONS＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
BYE SIP：215@192.168.15.118：5060 SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134
这个Cseq：5511 BYE
从211＆LT; SIP：211@192.168.15.118> ;;标签= 2465683119
要：其中，SIP：215@192.168.15.118> ;;标签= as573c52b3
呼叫ID：d188757cd6c044783fd413d11f8a982f@192.168.15.71
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
马克斯 - 前锋：70
内容长度：0＆LT;＆------------- GT;
---（10头0线）---
发送到192.168.15.71:45616（NAT）
在6400毫秒调度SIP对话d188757cd6c044783fd413d11f8a982f@192.168.15.71破坏（方法：BYE）＆LT; ---发射（NAT）来192.168.15.71:45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134;received=192.168.15.71;rport=45616
从211＆LT; SIP：211@192.168.15.118> ;;标签= 2465683119
要：其中，SIP：215@192.168.15.118> ;;标签= as573c52b3
呼叫ID：d188757cd6c044783fd413d11f8a982f@192.168.15.71
这个Cseq：5511 BYE
服务器：的Asterisk PBX 1.8.11.0
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
内容长度：0
＆LT;＆------------ GT;
在6400毫秒：调度SIP对话5060 0d06690561757f98637241394cc8dbba@192.168.15.118破坏（方法：INVITE）
set_destination：解析＆LT; SIP：215@115.167.21.82：5060; rinstance = 49cb21467969bef8;运输= UDP＆GT;用于地址/端口发送到
set_destination：设定目的地115.167.21.82:5060
可靠地发射（NAT）来192.168.15.27:5060：
BYE SIP：215@115.167.21.82：5060; rinstance = 49cb21467969bef8;运输= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport
马克斯 - 前锋：70
从设备＆LT; SIP：211@192.168.15.118> ;;标签= as404f0eb0
要：其中，SIP：215@115.167.21.82：5060; rinstance = 49cb21467969bef8;运输= UDP取代;标签= 96055240
呼叫ID：0d06690561757f98637241394cc8dbba@192.168.15.118：5060
这个Cseq：103 BYE
用户代理：的Asterisk PBX 1.8.11.0
的X Asterisk的HangupCause：正常结算
的X Asterisk的HangupCause code：16
内容长度：0
---
==菌种扩展（来话通配符，215，1）SIP / 211-00000008'退出非零
重传＃1（NAT）来192.168.15.27:5060：
BYE SIP：215@115.167.21.82：5060; rinstance = 49cb21467969bef8;运输= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport
马克斯 - 前锋：70
从设备＆LT; SIP：211@192.168.15.118> ;;标签= as404f0eb0
要：其中，SIP：215@115.167.21.82：5060; rinstance = 49cb21467969bef8;运输= UDP取代;标签= 96055240
呼叫ID：0d06690561757f98637241394cc8dbba@192.168.15.118：5060
这个Cseq：103 BYE
用户代理：的Asterisk PBX 1.8.11.0
的X Asterisk的HangupCause：正常结算
的X Asterisk的HangupCause code：16
内容长度：0
---
＆LT; --- SIP从UDP阅读：192.168.15.27：5060 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060
联系方式：其中，SIP：215@115.167.21.82：5060; rinstance = 49cb21467969bef8;运输= UDP＆GT;
要：其中，SIP：215@115.167.21.82：5060; rinstance = 49cb21467969bef8;运输= UDP取代;标签= 96055240
从设备＆LT; SIP：211@192.168.15.118> ;;标签= as404f0eb0
呼叫ID：0d06690561757f98637241394cc8dbba@192.168.15.118：5060
这个Cseq：103 BYE
用户代理：Zoiper为Windows 2.39 r16838
内容长度：0
＆LT;＆------------- GT;
- （9头0线）---
真正摧毁SIP对话'0d06690561757f98637241394cc8dbba@192.168.15.118：5060'方法：INVITE＆LT; --- SIP从UDP阅读：192.168.15.27：5060 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060
联系方式：其中，SIP：215@115.167.21.82：5060; rinstance = 49cb21467969bef8;运输= UDP＆GT;
要：其中，SIP：215@115.167.21.82：5060; rinstance = 49cb21467969bef8;运输= UDP取代;标签= 96055240
从设备＆LT; SIP：211@192.168.15.118> ;;标签= as404f0eb0
呼叫ID：0d06690561757f98637241394cc8dbba@192.168.15.118：5060
这个Cseq：103 BYE
用户代理：Zoiper为Windows 2.39 r16838
内容长度：0＆LT;＆------------- GT;
- （9头0线）---
可靠地发射（NAT）来192.168.15.71:45616：
OPTIONS SIP：211@192.168.15.71：45616;运输= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK30d9d70f;rport
马克斯 - 前锋：70
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as4f0724aa
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
联系方式：其中，SIP：asterisk@192.168.15.118：5060＆GT;
呼叫ID：05f27bf10f0a0759637b90252dc4366a@192.168.15.118：5060
这个Cseq：102 OPTIONS
用户代理：的Asterisk PBX 1.8.11.0
日期：星期六，2013年1月12日19时44分37秒GMT
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
内容长度：0
---＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK30d9d70f;rport=5060;received=192.168.15.118
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as4f0724aa
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
呼叫ID：05f27bf10f0a0759637b90252dc4366a@192.168.15.118：5060
这个Cseq：102 OPTIONS
内容长度：0＆LT;＆------------- GT;
---（7头0线）---
真正摧毁SIP对话'05f27bf10f0a0759637b90252dc4366a@192.168.15.118：5060'方法：OPTIONS＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
OPTIONS SIP：192.168.15.118 SIP / 2.0
呼叫ID：a5a311df861221d42844a8c485d4fee8@192.168.15.71
这个Cseq：5815 OPTIONS
从211＆LT; SIP：211@192.168.15.118> ;;标签= 3109248316
要：211＆LT; SIP：211@192.168.15.118>
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bK1e09e6f6c30b4b74ddc93c590abd3383353134;rport
马克斯 - 前锋：70
用户代理：SIPAUA / 0.1.001
内容长度：0＆LT;＆------------- GT;
---（9头0线）---
在默认情况下寻找S（域192.168.15.118）＆LT; ---发射（NAT）来192.168.15.71:45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bK1e09e6f6c30b4b74ddc93c590abd3383353134;received=192.168.15.71;rport=45616
从211＆LT; SIP：211@192.168.15.118> ;;标签= 3109248316
要：211＆LT; SIP：211@192.168.15.118> ;;标签= as51223faf
呼叫ID：a5a311df861221d42844a8c485d4fee8@192.168.15.71
这个Cseq：5815 OPTIONS
服务器：的Asterisk PBX 1.8.11.0
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
联系方式：其中，SIP：192.168.15.118：5060＆GT;
接受：应用/ SDP
内容长度：0
＆LT;＆------------ GT;
在32000毫秒（：OPTIONS方法）调度SIP对话a5a311df861221d42844a8c485d4fee8@192.168.15.71破坏
可靠地发射（NAT）来192.168.15.71:45616：
OPTIONS SIP：211@192.168.15.71：45616;运输= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK1f8e44dc;rport
马克斯 - 前锋：70
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as7a9a1ea3
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
联系方式：其中，SIP：asterisk@192.168.15.118：5060＆GT;
呼叫ID：7ebcafc7159379fd047075a85c424588@192.168.15.118：5060
这个Cseq：102 OPTIONS
用户代理：的Asterisk PBX 1.8.11.0
日期：星期六，2013年1月12日19点44分41秒GMT
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
内容长度：0
---＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK1f8e44dc;rport=5060;received=192.168.15.118
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as7a9a1ea3
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
呼叫ID：7ebcafc7159379fd047075a85c424588@192.168.15.118：5060
这个Cseq：102 OPTIONS
内容长度：0＆LT;＆------------- GT;
---（7头0线）---
真正摧毁SIP对话'7ebcafc7159379fd047075a85c424588@192.168.15.118：5060'方法：OPTIONS
真正摧毁SIP对话d188757cd6c044783fd413d11f8a982f@192.168.15.71'方法：BYE
真正摧毁SIP对话a81e6a5f591141abd73f9dad478a6b56@192.168.15.71'方法：OPTIONS
可靠地发射（NAT）来192.168.15.71:45616：
OPTIONS SIP：211@192.168.15.71：45616;运输= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK017f2b0a;rport
马克斯 - 前锋：70
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as5367b37c
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
联系方式：其中，SIP：asterisk@192.168.15.118：5060＆GT;
呼叫ID：1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118：5060
这个Cseq：102 OPTIONS
用户代理：的Asterisk PBX 1.8.11.0
日期：星期六，2013年1月12日19时44分44秒GMT
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
内容长度：0---
＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK017f2b0a;rport=5060;received=192.168.15.118
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as5367b37c
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
呼叫ID：1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118：5060
这个Cseq：102 OPTIONS
内容长度：0＆LT;＆------------- GT;
---（7头0线）---
真正摧毁SIP对话'1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118：5060'方法：OPTIONS
从PC调用（zoiper）当到Android
＆LT; --- SIP从UDP阅读：192.168.15.71：45616 ---＆GT;
BYE SIP：215@192.168.15.118：5060 SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134
这个Cseq：1 BYE
来源：＆LT; SIP：211@192.168.15.71：45616;运输= UDP取代;标签= 4162167884
若要：设备＆LT; SIP：215@192.168.15.118> ;;标签= as5805dc66
呼叫ID：2732e4564ce8534c5765a456045a9960@192.168.15.118：5060
马克斯 - 前锋：70
内容长度：0＆LT;＆------------- GT;
---（8头0线）---
发送到192.168.15.71:45616（NAT）
在8576毫秒：调度SIP对话5060 2732e4564ce8534c5765a456045a9960@192.168.15.118破坏（方法：BYE）＆LT; ---发射（NAT）来192.168.15.71:45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134;received=192.168.15.71;rport=45616
来源：＆LT; SIP：211@192.168.15.71：45616;运输= UDP取代;标签= 4162167884
若要：设备＆LT; SIP：215@192.168.15.118> ;;标签= as5805dc66
呼叫ID：2732e4564ce8534c5765a456045a9960@192.168.15.118：5060
这个Cseq：1 BYE
服务器：的Asterisk PBX 1.8.11.0
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
内容长度：0
＆LT;＆------------ GT;
==菌种扩展（来话通配符，211，1）SIP / 215-0000000a退出非零
调度SIP对话的毁灭MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk。在6400毫秒（方法：ACK）
set_destination：解析＆LT; SIP：215@115.167.21.82：5060;交通= UDP＆GT;用于地址/端口发送到
set_destination：设定目的地115.167.21.82:5060
可靠地发射（NAT）来192.168.15.27:5060：
BYE SIP：215@115.167.21.82：5060;交通= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
马克斯 - 前锋：70
来源：＆LT; SIP：211@192.168.15.118;运输= UDP取代;标签= as10377813
要：其中，SIP：215@192.168.15.118;运输= UDP取代;标签= 50312112
呼叫ID：MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk。
这个Cseq：102 BYE
用户代理：的Asterisk PBX 1.8.11.0
代理授权：摘要用户名=215，境界=星号，算法MD5 =，= URISIP：192.168.15.118，现时=，回应=c897390cc8e4f674d7e9cd1efa7319a6
的X Asterisk的HangupCause：正常结算
的X Asterisk的HangupCause code：16
内容长度：0
---
重传＃1（NAT）来192.168.15.27:5060：
BYE SIP：215@115.167.21.82：5060;交通= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
马克斯 - 前锋：70
来源：＆LT; SIP：211@192.168.15.118;运输= UDP取代;标签= as10377813
要：其中，SIP：215@192.168.15.118;运输= UDP取代;标签= 50312112
呼叫ID：MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk。
这个Cseq：102 BYE
用户代理：的Asterisk PBX 1.8.11.0
代理授权：摘要用户名=215，境界=星号，算法MD5 =，= URISIP：192.168.15.118，现时=，回应=c897390cc8e4f674d7e9cd1efa7319a6
的X Asterisk的HangupCause：正常结算
的X Asterisk的HangupCause code：16
内容长度：0
---＆LT; --- SIP从UDP阅读：192.168.15.27：5060 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060
联系方式：其中，SIP：215@115.167.21.82：5060;交通= UDP＆GT;
要：其中，SIP：215@192.168.15.118;运输= UDP取代;标签= 50312112
来源：＆LT; SIP：211@192.168.15.118;运输= UDP取代;标签= as10377813
呼叫ID：MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk。
这个Cseq：102 BYE
用户代理：Zoiper为Windows 2.39 r16838
内容长度：0＆LT;＆------------- GT;
---（9头0线）---
对于来电对话BYE SIP响应消息到达
真正摧毁SIP对话MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk。方法：ACK＆LT; --- SIP从UDP阅读：192.168.15.27：5060 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060
联系方式：其中，SIP：215@115.167.21.82：5060;交通= UDP＆GT;
要：其中，SIP：215@192.168.15.118;运输= UDP取代;标签= 50312112
来源：＆LT; SIP：211@192.168.15.118;运输= UDP取代;标签= as10377813
呼叫ID：MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk。
这个Cseq：102 BYE
用户代理：Zoiper为Windows 2.39 r16838
内容长度：0＆LT;＆------------- GT;
---（9头0线）---
可靠地发射（NAT）来192.168.15.71:45616：
OPTIONS SIP：211@192.168.15.71：45616;运输= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK01e7e47d;rport
马克斯 - 前锋：70
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as73902c1e
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
联系方式：其中，SIP：asterisk@192.168.15.118：5060＆GT;
呼叫ID：7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118：5060
这个Cseq：102 OPTIONS
用户代理：的Asterisk PBX 1.8.11.0
日期：星期六，2013年1月12日19时54分09秒GMT
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
内容长度：0
---＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK01e7e47d;rport=5060;received=192.168.15.118
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as73902c1e
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
呼叫ID：7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118：5060
这个Cseq：102 OPTIONS
内容长度：0＆LT;＆------------- GT;
---（7头0线）---
真正摧毁SIP对话'7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118：5060'方法：OPTIONS＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
OPTIONS SIP：192.168.15.118 SIP / 2.0
呼叫ID：c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71
这个Cseq：9273 OPTIONS
从211＆LT; SIP：211@192.168.15.118> ;;标签= 740019322
要：211＆LT; SIP：211@192.168.15.118>
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bK5ac51b5cca5ad84320fc342692fdbb2d353134;rport
马克斯 - 前锋：70
用户代理：SIPAUA / 0.1.001
内容长度：0＆LT;＆------------- GT;
---（9头0线）---
在默认情况下寻找S（域192.168.15.118）＆LT; ---发射（NAT）来192.168.15.71:45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bK5ac51b5cca5ad84320fc342692fdbb2d353134;received=192.168.15.71;rport=45616
从211＆LT; SIP：211@192.168.15.118> ;;标签= 740019322
要：211＆LT; SIP：211@192.168.15.118> ;;标签= as1bed6ef2
呼叫ID：c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71
这个Cseq：9273 OPTIONS
服务器：的Asterisk PBX 1.8.11.0
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
联系方式：其中，SIP：192.168.15.118：5060＆GT;
接受：应用/ SDP
内容长度：0＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK3c24b1cb;rport=5060;received=192.168.15.118
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as54c6581a
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
呼叫ID：7742af6c24e3e9a33efa0ba73051a29e@192.168.15.118：5060
这个Cseq：102 OPTIONS
内容长度：0＆LT;＆------------- GT;
---（7头0线）---
真正摧毁SIP对话'7742af6c24e3e9a33efa0ba73051a29e@192.168.15.118：5060'方法：OPTIONS＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
OPTIONS SIP：192.168.15.118 SIP / 2.0
呼叫ID：18347a0db6841423591b08250847a1e0@192.168.15.71
这个Cseq：3824 OPTIONS
从211＆LT; SIP：211@192.168.15.118> ;;标签= 841349553
要：211＆LT; SIP：211@192.168.15.118>
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bK79bb575a5d0f832291b861987849d9a2353134;rport
马克斯 - 前锋：70
用户代理：SIPAUA / 0.1.001
内容长度：0＆LT;＆------------- GT;
---（9头0线）---
在默认情况下寻找S（域192.168.15.118）
＆LT;＆------------- GT;
---（9头0线）---
在默认情况下寻找S（域192.168.15.118）＆LT; ---发射（NAT）来192.168.15.71:45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.71:45616;branch=z9hG4bK40d81d1686d23c9133e52f4180c2accb353134;received=192.168.15.71;rport=45616
从211＆LT; SIP：211@192.168.15.118> ;;标签= 4017391219
要：211＆LT; SIP：211@192.168.15.118> ;;标签= as52fe1845
呼叫ID：09809b7ed9a68b7f270f33f1d8de13dc@192.168.15.71
这个Cseq：4619 OPTIONS
服务器：的Asterisk PBX 1.8.11.0
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
联系方式：其中，SIP：192.168.15.118：5060＆GT;
接受：应用/ SDP
内容长度：0
＆LT;＆------------ GT;
在32000毫秒（：OPTIONS方法）SIP对话09809b7ed9a68b7f270f33f1d8de13dc@192.168.15.71调度破坏
可靠地发射（NAT）来192.168.15.71:45616：
OPTIONS SIP：211@192.168.15.71：45616;运输= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK13b8cca8;rport
马克斯 - 前锋：70
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as6e6638f8
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
联系方式：其中，SIP：asterisk@192.168.15.118：5060＆GT;
呼叫ID：76f6f373769041c3462ebb676641f1b7@192.168.15.118：5060
这个Cseq：102 OPTIONS
用户代理：的Asterisk PBX 1.8.11.0
日期：星期六，2013年1月12日19点54分31秒GMT
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
内容长度：0
---＆LT; ---从UDP读取SIP：192.168.15.71：45616 ---＆GT;
SIP / 2.0 200 OK
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK13b8cca8;rport=5060;received=192.168.15.118
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as6e6638f8
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
呼叫ID：76f6f373769041c3462ebb676641f1b7@192.168.15.118：5060
这个Cseq：102 OPTIONS
内容长度：0＆LT;＆------------- GT;
---（7头0线）---
真正摧毁SIP对话'76f6f373769041c3462ebb676641f1b7@192.168.15.118：5060'方法：OPTIONS
真正摧毁SIP对话9eeee094f46eec920ac462e291314bde@192.168.15.71'方法：OPTIONS
可靠地发射（NAT）来192.168.15.71:45616：
OPTIONS SIP：211@192.168.15.71：45616;运输= UDP SIP / 2.0
途经：SIP / 2.0 / UDP 192.168.15.118:5060;branch=z9hG4bK47a8a134;rport
马克斯 - 前锋：70
从星号＆LT; SIP：asterisk@192.168.15.118> ;;标签= as76426de6
要：其中，SIP：211@192.168.15.71：45616;运输= UDP＆GT;
联系方式：其中，SIP：asterisk@192.168.15.118：5060＆GT;
呼叫ID：3a98a25b41dc3b3e699ee4383669e984@192.168.15.118：5060
这个Cseq：102 OPTIONS
用户代理：的Asterisk PBX 1.8.11.0
日期：星期六，2013年1月12日19时54分34秒GMT
允许：INVITE，ACK，CANCEL，OPTIONS，BYE，REFER，订阅通知，信息，出版
支持单位：替换，定时
内容长度：0
我使用本地网络上的星号（LAN）......
中的extensions.conf我的拨号方案是：
[来话通配符]分配延长=＆GT; _2XX，提示，（SIP / $ {EXTEN} ,, 120）分配延长=＆GT; _2XX，1，拨打（SIP / $ {EXTEN} ,, 120）分配延长=＆GT; _2XX，N，挂断
我的SIP账号为：
[215]否认= 0.0.0.0 / 0.0.0.0秘密= very123dtmfmode = RFC2833canreinvite =无上下文=来话通配符主机=动态类型=朋友NAT = YES端口= 5060出线= YEScallgroup =pickupgroup =拨打= SIP / 215邮箱= 215 @设备许可证= 0.0.0.0 / 0.0.0.0来电显示=设备＆lt; 215 GT;callcounter = YESfaxdetect =无
解决方案
感谢每一个在这个论坛的期待...我已经manged解决问题...这两个设备X-Lite / zoiper和机器人本地SIP客户端使用不同的默认音频codeCS。
有关X-Lite默认codeC是BroadVoice-32
有关zoiper默认codeC是GSM
针对Android默认codeC是G.711 ULAW
由于同时与每个other..In我的情况下，这些设备都使用不同的codeCS导致单向语音（于Android打电话给X-Lite / zoiper时），通信这些设备应使用同一codeC。
在创建中sip.conf的SIP帐号，而我们可以执行两个通信客户下列方式使用相同的音频codeC。
[211]
否认= 0.0.0.0 / 0.0.0.0
秘密= 123456
dtmfmode = RFC2833
canreinvite =无
上下文=来话通配符
主机=动态
NAT = YES
类型=朋友
端口= 5060
出线= YES
callgroup =
pickupgroup =
许可证= 0.0.0.0 / 0.0.0.0
callcounter = YES
faxdetect =无
禁止= ALL（禁用默认音频codeC）
允许= ULAW（允许ULAW音频codeC）[215]
否认= 0.0.0.0 / 0.0.0.0
秘密= 123456
dtmfmode = RFC2833
canreinvite =无
上下文=来话通配符
主机=动态
NAT =无
类型=朋友
端口= 5060
出线= YES
callgroup =
pickupgroup =
许可证= 0.0.0.0 / 0.0.0.0
callcounter = YES
faxdetect =无
禁止= ALL（禁用默认音频codeC）
允许= ULAW（允许ULAW音频codeC）
我们还可以通过双方的..
选择相同的音频codeC配置客户端上的音频codeC设置Asterisk= 1.8.11.0
Android= 2.3/4.0.3
Android Sip client=Native Android sip client/sipdemo
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
Sip debug when calling from android to zoiper .
<--- 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
When calling from pc (zoiper) to 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
I am using asterisk on local network (LAN)....
My dial plan in extensions.conf is:
[incoming-calls-wildcard]
exten => _2XX,hint,(SIP/${EXTEN},,120)
exten => _2XX,1,Dial(SIP/${EXTEN},,120)
exten => _2XX,n,Hangup
My sip account is:
[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
解决方案
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.
default codec for xlite is BroadVoice-32
default codec for zoiper is GSM
default codec for android is G.711 uLaw
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)
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无法正常使用Android的SIP客户端的工作？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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