I were working migrating a component to connect to AES 7.
For that we ve changed some configuration and migrating to the Avaya sdk7, nothing more.
We were able to make calls between h323 phones, also call SIP phones successfully.
Always using h323 phones as calling devices.
But In production we ll need to use SIPs as sources numbers.
And we have errors when we try to register a SIP phone into an AES session.
So we are handling different possibilities:

1. Its not possible to do using DMCC so we need to use another protocol (JTAPI)
2. We need to try something different in the code, like use first a h323 phone as a source and after transfer the call to a SIP number.
3. Its possible to do some trick in the AES for routing a simulated h323 phone to a SIP one.

I ll be waiting for any advice.
Thanks a lot

Between the lines I read that you want to register a SIP extension via an API and place calls form that extension. Why cant you utilize an actual desk phone or end user soft phone as the calling party?


DMCC can cause a SIP desk phone or soft phone to place a call to another party. Am I right that you don't have an existing hardphone or softphone to use as the originating extension? If you do have an existing Avaya built SIP device to use as the originator, you can use a DMCC call control services against that extension in to place a call from the SIP phone to another phone number. There are a number of pre-conditions that must be met for this to work. There is no need to register a DMCC device against the SIP device if all you want to use are call control services (this may be the problem you speak of -- you can only register a DMCC device in INDEPENDENT mode against a SIP device AND the set of DMCC services is significantly reduced with such a configuration).

Please review these two FAQs.
https://www.devconnectprogram.com/site/global/products_resources/avaya_aura_application_enablement_services/support/faq/index.gsp#860
https://www.devconnectprogram.com/site/global/products_resources/avaya_aura_application_enablement_services/support/faq/index.gsp#880

Avaya CTI works with Avaya devices, thus you have to use an Avaya SIP desk phone or soft phone as your originating entity.

JTAPI will not help you register a SIP device. It is just another way of invoking the MakeCall() functionality already available to you in DMCC and TSAPI, but if MakeCall() is what you need, it can facilitate placing a call from a SIP station to a destination (but as I said above DMCC can do that as well).

You could use a DMCC H.323 device to originate the call and transfer it to the actual party (or conference it in and drop out the DMCC device at the right moment). Keep in mind that one of the two calls must be 'stable' before you will be able to complete the conference or transfer. By stable I mean past the point of having been answered.

John.
I have a real SIP and I want to connect it to a final number using a third party call.
Should I do something before calling like register o monitoring the SIP extension?
Now Im using this way to get the SIP id :

                GetDeviceId devRequest = new GetDeviceId();

		devRequest.setSwitchName(Hs1ServiceProvider.SWITCH_NAME);

		devRequest.setExtension(extension);

		devRequest.setControllableByOtherSessions(true);

		GetDeviceIdResponse devResp = this.deviceServices.getDeviceID(devRequest);

And when I make the call I Have a SubscribedResourceAvailabilityException in the client side.
Thanks a lot

SubscribedResourceAvailabilityException normally means no licenses, but some of the responses are overloaded. If you have checked the items in the FAQ and are sure you have met those pre-conditions, then check WebLM to make sure you have TSAPI Basic User licenses available. MakeCall() requires TSAPI Basic User licensing.

You can create a call control monitor on the SIP extension... if you get another SubscribedResourceAvailabilityException it probably means you do not have licensing in place. Such a monitor is not required, but it will help you know what is happening on the device (call answered, call disconnected, call offered, (un)held, transferred and conferenced) which can be useful. One very useful case is knowing if there is a call already present before trying to launch another.

We follow this list

https://www.devconnectprogram.com/site/global/products_resources/avaya_aura_application_enablement_services/support/faq/index.gsp#880

With the AES admin and it worked perfectly. Thanks a lot John.