Integration between AsteriskNow and Zanzibar OpenIVR

Question

i m fairly new to AsteriskNow but my purpose is to integrate incoming SIP Calls to route through AsteriskNow cum freePBX Server to OpenIVR Server which further points to my Application Server that responds with vxml content.

I have followed the Integration note available at OpenIVR Site (http://www.spokentech.org/aik.html). Let me give the Environment details:

Server 1: Zanzibar OpenIVR + Apache Tomcat (IP: 192.168.44.134) - Both the Server is up and running

Server 2: CentOS6 server with AsteriskNow and FreePBX (IP: 192.168.44.133) - Server is up and AsteriskNow and FreePBX are working fine

Configuration files at AsteriskNow Side for Integration:

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

Configuration Files at OpenIVR Side:

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:[email protected]</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:[email protected]</value>
</property>
<property name="cairoSipHostName">
<value>192.168.44.134</value>
</property>
<property name="cairoSipPort">
<value>5050</value>
</property>
</bean>

and

<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>

Sorry for making it lengthy. But end of the configuration, i get the following error whenever i try to call the extension 8000 with Zoiper SIP Phone (installed at desktop having ip: 192.168.44.1) with a registered account named "shovan" ([email protected])

Error:

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>

Please give me direction so i pursue further.

Thanks Shovan

Answer 1

Check your sip.conf

context=users

and there is not context named users in your extension.conf

try changing

context=Zanzibar

in sip.conf under Shovan or have a context name [users] in extension.conf with extension 8000

for more information read this link.....