Oops, I misread your response. If you are specifying the sip URI directly in the request, which you are, then you do not need to have them provisioned against the user in ACE user profile. That is only needed if you are using the "ace:" URI.

The reason your called failed was due to the error "CallingPartyNoAnswer". This means that the calling party did not answer the phone. So this can be caused by a few different things:

1. You didn't answer the phone. Can I assume your soft client didn't ring? You need to answer the call before the second leg will proceed.

2. Your phone is registered with SM but for some reason it cannot accept incoming calls (config/network issue).

John.

That would explain it. ACE will give you a call ID if the call server accepts the call. If the call server later determines that the called party is unavailable the call will fail. In the case of TPCv3 using SIP, we give you a call ID once we know the addresses are routable. The call may still fail if the endpoints are not registered.

John.

The reverse transformation rules would explain the issue in the notification, but not the reason for call failure. Can you dial between the phones from the soft client?

Can you log into the ACE Web GUI and navigate Configuration->Service Providers. Then select the "Rule Validation" tab, and Third Party Call (v3) in the pull down.

Enter sip:20071@avaya.com and sip:20073@avaya.com into the URI fields. Press validate. What do your matching provider rules look like?

For some reason it looks like the domain is being stripped off the "sip:20071" number.

John.

Can you log into the ACE Web GUI and navigate Configuration->Service Providers. Then select the "Rule Validation" tab, and Third Party Call (v3) in the pull down.

Enter sip:20071@avaya.com and sip:20073@avaya.com into the URI fields. Press validate. What do your matching provider rules look like?

For some reason it looks like the domain is being stripped off the "sip:20071" number.

John.

TPCv2 is designed around a two party call model. If you need a multi-party call I would recommend switching to the TPCv3 API which is designed for multl-party conferencing.

John

TPCv2 is designed around a two party call model. If you need a multi-party call I would recommend switching to the TPCv3 API which is designed for multl-party conferencing.

John

TPCv2 is designed around a two party call model. If you need a multi-party call I would recommend switching to the TPCv3 API which is designed for multl-party conferencing.

John

So this does not involve the Avaya ACE product?

What product and which API are you using when you encounter this problem?

Sorry, we do not currently support this capability through ACE. I have captured your request and will pass it along to ACE product management team.

John.

Which provider are you using? Call notification is nat supported at the moment through the SIP provider. You must use a TR/87 or JTAPI provider. Do you have an AES?

From the OAM web GUI select menu Configuration -> Service Providers. Select the tab named "Rule Validation" then in the "Service Type" pulldown select Call Notification (v3.8). Enter the URI you are trying to monitor (sip:+1000@avaya.com) and hit Validate. This will tell you if your address is resolving to a service provider.

Perhaps you didn't configure the provider for Call Notification?

John

Are you just looking for simple makeCallSession, AudioCall, etc?

You are correct in that there is some overlap. The user profile web service provides full access and management of the user profile database, while the subscriber management interface provides a simple query API for user contact information.

We added the subscriber API for applications that just need to do fast read operations of subscriber data. We also give the location in the federation of the user across this interface, and also allow for the scoping of the queries.

For example, IBM Sametime integration does a local scope query based on Sametime contact ID to determine if the user is in the local ACE server, and to get the user's ACE ID for aggregated presence registration. Hotdesking also uses the subscriber interface to do distributed queries across a federation to find the user's home ACE so that the application can modify their contact data on the home ACE server, and set call forwarding states on their associated phone.

Are you using a federated ACE deployment or a stand alone installation?

Thank you,
John.