Hi,

I am using the Avaya Aura BDE and i am actively working on a TSAPI project.

What softphone can i use to test calls with on my Win7 Development machine?

Thanks,

Rich.

The problemnatic shortcut was one that the installer created.

I realized that one sign of this was that the DADS created the log files for each station on the Desktop where the shortcut was ran from. Now they are created where the DADS EXE is located.

Hi John,

I have checked the /opt/configs/r16_twocabinets/user/vfm1.cfg file and this shows more than the 3 stations(40010, 40011, 40012) as being Agent1, not Standard.

I cannot seem to edit the file as it seems to be owned by root and i dont have the password for root access.

I opened the DADS console, selected the from the Phones Menu the option 8434, entered one of the 3 stations (listed above) and was able to make a phone call between 2 IP Agent softphones. So this little trick does work, but sometimes i find that it does not stay that way and need to go back into DADS to reset it.

My plan is to be able to have some stations for Developer testing and then have DADS to load test the System. So do i need to exclude those extensions from the vfm1.cfg script?

Thanks,

Rich.

Hi John,

Yes this is the path and i have checked the shortcut and it already uses that location.

...

I just found the fix. I created my own shortcut and it worked fine. Then i figured out that the Start In property in the Shortcut was not pointing to the folder you mentioned. The App must default to the IP address i was seeing as it could not see the INI file.

Thanks,

Rich.

Hi,

I am trying to configure the stations (and add more) that come with the default install of the Avaya BDE, so that they work with IP Agent and have some of the Feature buttons.

Although the stations appear to work, my first problem is that when i call from IP Agent #1 to IP Agent #2 the call lasts for a millisecond and appears as a missed call. I would like for it to ring for longer and then be answered by an agent.

My second issue is activating the feature buttons so that the station works with some of the feature buttons like release.

I have tried looking through some docs but have not a good enough guide and need to get running quickly. So i would appreciate either some direction or/and some references to good documents.

FYI - the Stations i have tried using are: 40010, 40011, 40012.

Thanks again!

Rich.

Hi,

I have just followed the documentation for installing and configuring the Avaya BDE. Everything worked which was nice.

However, when i changed the IP addresses of the Avaya BDE to mach the subnet of the network it is attached to, everything accepted it except for the DADS running outside of the VMs (on the Host).

IP Changes:
CM = 192.168.10.129 (Was CM = 192.168.15.129)
AES = 192.168.10.128 (Was AES = 192.168.15.128)

I have changed the IP for the Communication Manager in the vfmgui.ini file as per the doc, but it still uses the original IP. I presume its using one buried elsewhere?

The one line in this INI file now reads:
CmIPAddr=192.168.10.129

Can you help me configure the new IP into DADS?

Thanks!

Rich.

John,

I found that TSAPI FAQ that you mentioned, It describes how to enable sending UCID over the Trunk. So other readers dont have to find it, i will paste it here:
To receive Universal Call Identification (UCID) information in the events, the following system parameters must be correctly set.

Type the command 'change system-parameters features' on Avaya Site Administrator command window. Enable the following features on the CM in order to get UCID in service initiated event:
Create Universal Call ID (UCID) to 'y' (Setting to 'y' indicates that UCID will be generated for each call)
UCID network Node ID to 1 (or any number between 1 to 32767 and unique to the switch in the network)
Send UCID to ASAI to 'y' (Setting to 'y' enable transmission of UCID information)


Type the command 'change trunk-group N' on Avaya Site Administrator command window and enable the 'Send UCID' field from page 3 for each trunk group that is used for communication between CM systems. This field specifies whether or not the trunk should transmit Universal Call IDs. Sending UCIDs in a network of CMs allows an application server that interworks with all of them to realize it already has handled a call when it is transferred to an agent on another switch.
----------------------------------------------

Is there any way i can do this without using UCID?
I am still stuck with my previous posting, about how our queuing application running on a SIP IVR (Avaya/3rd party)can idenify each of the many active calls to our passive CTI Product?

As this part of our feature is designed to work with SIP, when i finish writing this i plan for it to work on any SIP environment.

Thanks.

Hi John,

I am asking from the perspective of the IVR because we have an application running there too. Its job is to queue calls and it communicates to tell our Product which call in the IVR to deal with, which is why we need to tie data with a particular call.

Am i correct in assuming that the IVR can find and send our Product a CallID such as "Call-ID: a84b4c76e66710@pc33.atlanta.com"?
Looking at the SIP Spec, i dont see any other way an IVR can identify more than one call simultaneously without using the generic SIP Call ID, unless its Avaya specific.

However, i am trying to find something generic because i write less code, and also because clients are not always a perfect scenario (having Avaya equipment installed throughout their whole site). As I have worked before with sites that have a mixture, such as an Avaya Switch but a non-Avaya IVR.

Thanks.

I was wondering if you could guide me on something that would help in some development. Currently I am working on a feature for our product that needs to talk to a T1 or SIP type IVR. I am able to talk to a T1 type IVR and that works, but I'm stuck on how a SIP IVR differentiates between all its active calls? An IVR with a T1 has a one-to-one mapping between its lines/ports, but this is not true with a IVR implemented as SIP.

I currently identify calls on a T1 type IVR via its Trunk Grouping and Member, but cannot apply the same logic to a SIP type IVR.

Thanks,

Rich.