Asterisk Now和桑给巴尔OpenIVR之间的集成

Question

我对AsterikNow还很陌生，但我的目的是集成传入的SIP呼叫，通过Asteriknow和freePBX服务器路由到OpenIVR服务器，这进一步指向了我的应用程序服务器，该服务器使用vxml内容进行响应。

我遵循了OpenIVR网站上提供的集成说明(http://www.spokentech.org/aik.html)。让我详细介绍一下环境：

服务器1:Zanzibar OpenIVR+Apache Tomcat(IP:192.168.44.134)-两个服务器都已启动并运行

服务器2:CentOS6服务器，带有AsterikNow和FreePBX(IP:192.168.44.133)-服务器已启动，Asteriknow和FreePBX运行良好

AsterikNow侧用于集成的配置文件：

SIP.conf

#include sip_custom_post.conf
[Zanzibar]
type=peer
host=192.168.44.134
port=5090
dtmfmode=info
canreinvite=no
[shovan]
type=friend
host=dynamic
secret=password
context=users
deny=0.0.0.0/0.0.0.0
permit=192.168.44.0/255.255.255.0

Extension.conf

#include extensions_override_freepbx.conf
#include extensions_additional.conf
#include extensions_custom.conf
;--------------------------------------------------------------------------------;
[Zanzibar]
exten=>8000,1,SIPAddHeader(x-channel:${CHANNEL})
exten=>8000,2,SIPAddHeader("x-
application:vxml|
http://192.168.44.134:8084/VoiceDesk/VoiceDeskController")
exten=>8000,3,Dial(SIP/Zanzibar)
exten=>6001,1,Dial(SIP/shovan,20)

manager_Additional.conf

[manager]
secret = password
deny=192.168.128.0/255.255.255.0
permit=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

OpenIVR端的配置文件：

democonfig-withcairo.xml

<bean id="sipService" class="org.speechforge.zanzibar.sip.SipServer"
init-method="startup" destroy-method="shutdown">
<property name="dialogService"><ref bean="dialogService"/></property>
<property name="mySipAddress">
<value>sip:[e-mail]</value>
</property>
<property name="stackName">
<value>Agi Sip Stack</value>
</property>
<property name="port">
<value>5090</value>
</property>
<property name="transport">
<value>UDP</value>
</property>
<property name="cairoSipAddress">
<value>sip:[e-mail]</value>
</property>
<property name="cairoSipHostName">
<value>192.168.44.134</value>
</property>
<property name="cairoSipPort">
<value>5050</value>
</property>
</bean>

和

<bean id="_8000"
class="org.speechforge.zanzibar.jvoicexml.impl.VoiceXmlSessionProcessor"
singleton="false">
<property name="appUrl">
<value>http://192.168.44.134:8084/VoiceDesk/VoiceDeskController</value>
</property>
</bean>

pbxconfig.xml

<bean id="callControl" class="org.speechforge.zanzibar.asterisk.CallControl"
init-method="startup" destroy-method="shutdown">
<property name="address">
<value>192.168.44.133</value>
</property>
<property name="name">
<value>manager</value>
</property>
<property name="password">
<value>password</value>
</property>
<property name="disabled">
<value>false</value>
</property>
</bean>

抱歉把它写得太长了。但在配置结束时，每当我试图用Zoiper SIP Phone(安装在桌面上，ip为192.168.44.1)用一个名为"shovan"的注册帐户调用扩展8000时，我都会收到以下错误(shovan@192.168.44.133)

错误：

Connected to Asterisk 11.7.0 currently running on localhost (pid = 1549)
[2013-12-30 09:50:30] NOTICE[1712][C-0000000b]: 
chan_sip.c:25450 handle_request_invite: Call from 'shovan'
(192.168.44.1:56418) to    extension '8000' rejected because 
extension not found in context 'users'.
localhost*CLI>

请给我指引，让我更进一步。

谢谢肖万

Answer 1

检查您的sip.conf

context=users

并且在您的扩展.conf 中没有名为users的上下文

尝试更改

context=Zanzibar

在肖万领导下的sip.conf或者在扩展名为8000 的extension.conf中具有上下文名称[users]

有关更多信息，请阅读此链接。。。。。