Hi everybody,

I'm trying to put together a basic click to call. A servlet reads http parameters for caller/called and invokes CallFactory.create().initiate().
I can see CE sending a REFER to AMS, and AMS accepting that and creating an INVITE as a result of that:

INVITE sip:ClickToCall@10.135.63.69:5061;transport=TLS;ams-cid=183c63a1-b86b-4b16-a5b8-dde74ab5919b;av-svc-fea=ClickToCall

however for some reason that INVITE does not seem to reach my application. I've put CallListeners and SipMessageListeners to log any callback they receive and they log nothing.
It looks like they are being detected by CE:

2014-10-09 09:41:40,101 [SoapConnectorThreadPool : 1] ClickToCall INFO - ClickToCall-1.0.0.0.1 - Listener class "com.avaya.ept.clicktocall.MyMessageListener" found with annotation "TheSipMessageListener" and implementation of "SipMessageListener"
2014-10-09 09:41:40,102 [SoapConnectorThreadPool : 1] ClickToCall INFO - ClickToCall-1.0.0.0.1 - Listener class "com.avaya.ept.clicktocall.MyCallListener" found with annotation "TheCallListener" and implementation of "CallListener"

on my CARRule.xml I have pretty much the default as created by the archetype, which includes:
<TerminatingServiceRule desc="Want to receive incoming INVITEs with ClickToCall feature URI">
<FeatureURI>ClickToCall</FeatureURI>
</TerminatingServiceRule>

and I believe that's what I'd need. (I also have rules for imsorig and imsterm but I don't think that should be an issue, should it?).

It looks like CE knows it should invoke my application eventually:

[10/9/14 9:53:59:444 MDT] 000001d5 ApplicationPa 3 ApplicationPathSelector findSippletMatch appinfo: javax.servlet.sip.ar.SipApplicationRouterInfo@42aebe81; Data: nextApplicationName=pfa, sipRouteModifier=NO_ROUTE, sipApplicationRoutingRegion = Label = sd Type = TERMINATING, stateInfo = RequestState [stack=[Service=pfa Label = sd Type = TERMINATING], serviceList=[pfa, ClickToCall-1.0.0.0.1], requestUriUserStr=ClickToCall, requestUriHostStr=10.135.63.69, paiHostStr=snapin.avaya.com, paiUserStr=ce-msml, applicationId=null, sentToExitPtfService=false], subscriberURI = sip:ClickToCall@10.135.63.69:5061;transport=TLS;ams-cid=183c63a1-b86b-4b16-a5b8-dde74ab5919b;av-svc-fea=ClickToCall

but for some reason my request is never received, and other than the 100 Trying there is no final response sent by anybody.
I can see the following exception, not sure if it's related:

[10/9/14 9:53:59:510 MDT] 000001d7 ThreadPool 1 An unhandled exception occured on a thread in Threadpool: SipContainerPool
java.lang.NullPointerException
at com.ibm.ws.sip.container.tu.TransactionUserBase.setHandler(TransactionUserBase.java:1044)
at com.ibm.ws.sip.container.tu.TransactionUserWrapper.setHandler(TransactionUserWrapper.java:1123)
at com.ibm.ws.sip.container.servlets.SipSessionImplementation.setHandler(SipSessionImplementation.java:945)
at com.ibm.ws.sip.container.servlets.SipSessionImplementation.init(SipSessionImplementation.java:244)
at com.ibm.ws.sip.container.servlets.SipSessionImplementation.<init>(SipSessionImplementation.java:184)
at com.ibm.ws.sip.container.servlets.DRSSipSessionImpl.<init>(DRSSipSessionImpl.java:92)
at com.ibm.ws.sip.container.servlets.SIPSessionFactory.createSIPSession(SIPSessionFactory.java:99)
at com.ibm.ws.sip.container.tu.TransactionUserBase.getSipSession(TransactionUserBase.java:1080)
at com.ibm.ws.sip.container.tu.TransactionUserBase.getAllSipSessions(TransactionUserBase.java:740)
at com.ibm.ws.sip.container.tu.TransactionUserWrapper.getAllSipSessions(TransactionUserWrapper.java:1348)
at com.ibm.ws.sip.container.servlets.SipApplicationSessionImpl.getAllSIPSessions(SipApplicationSessionImpl.java:817)
at com.ibm.ws.sip.container.servlets.SipApplicationSessionImpl.getAllSIPSessions(SipApplicationSessionImpl.java:788)
at com.ibm.ws.sip.container.servlets.SipApplicationSessionImpl.getSessions(SipApplicationSessionImpl.java:844)


can you please suggest how should I go about troubleshooting this?

thanks John. So just to fully confirm, regardless of how many VDNs/vectors will do adjunct routing to my application and how many concurrent requests my application will receive as a result of the adjunct routing, I only need 1 tsapi advanced license and no basic licenses?

thanks,

H

I'm writting an application that will monitor N vdns for adjunt route requests, and I'm not clear what kind of licenses will I need. I'm expecting the application to handle M concurrent adjunct route requests.
Can somebody please help?

a) 1 tsapi advanced license + N tsapi basic licenses?
b) 1 tsapi advanced license + M tsapi basic licenses?
c) 1 tsapi advance license only?
d) none of the above?

thanks!

H

Hi Craig,

could you please also email me more information on this? I'm trying to do the same thing.

thanks,

H

thanks Ross. that's what I'm trying but it's not working.
I can see several known and unknown headers, but Call-Info is not showing up anywhere.

Is there any way to read the Call-Info header from ccxml or a DD/OD application?

Thanks Ross and Shweta. You're right, I didn't realise I wasn't looking at the final specification.

thanks,

H

I'm trying to use 'fetch' for a synchronous fetch.
Looking at the specification, the synch attribute should allow me to do it but so far I'm unable to do so.
I've set the synch attribute to a variable name, both a complex and simple variable, but AVP seems to block the execution.

Does anybody have a sample fetch that uses the synch attribute?

thanks,

H

sorry I'm not sure I understand you.
Why would that be an issue? are you saying this approach could cause problems somewhere? where/why?

thank you,

Horaci

I have developed a static ccxml application and I'm trying to configure it in VoicePortal so that HTTP access is not required. For example, using URL file://home/www/static.ccxml
I've been doing some tests and so far it's not working for me, can somebody provide an example of how to do so or confirm whether file access (not through HTTP) is possible to retrieve ccxml/vxml resources?

thanks,

Horaci

Does anybody have a sample ccxml page that uses custom SIP headers when making an outbound call through "createcall" ?

Also, is there any document describing all the "hints" createcall (and other ccxml elements) can take when running inside VoicePortal?

thanks,

H

In case anybody is having the same question, use RouteAddress.ALL_ROUTE_ADDRESS as the vdn to register to.

I'm trying to write an application that will register for route requests with ALL the vdns in a CM. Is there any way to do this, without having to go through all the vdns calling registerRouteCallback on each of them?
Something like a wildcard vdn so I receive all route requests for that CTI link?

thanks,

Horaci

I'm trying to get mutual authentication working in order to test http io processor events.
I downloaded the calling card sample application but I'm having trouble setting up the mutual authentication.
First I tried to follow the pdf mentioned on the sample application "Creating and Using Certificates for HTTPS Communication between Avaya Voice Portal and Tomcat Application Server - Issue 1.0" but I keep getting handshake errors on the application trying to send the event:
javax.net.ssl.SSLHandshakeException: Received fatal alert: handshake_failure
   at com.sun.net.ssl.internal.ssl.Alerts.getSSLException(Alerts.java:174)
   at com.sun.net.ssl.internal.ssl.Alerts.getSSLException(Alerts.java:136)

Then I tried using DD 5.1's support for certificates. I exported the runtime config for tomcat after having added the VP certificate (importing the certificate from https://<ip address>:8443) but that doesn't seem to make any difference.
Can somebody please comment what steps are required in order to test http io processor events with DD 5.1?

thanks,

Horaci

great, I'll have a look at that.

thanks again John,

H