Conversant Questions 1

[smaslin]

I basically inherited a phone system at my company about a year ago when the phone admin was canned. I've been slowly working my way through all aspects of our Definity G3, as well as Audix and CAS. Now on to the Conversant.

Some of our vectors actually use the "converse-on" command. I'm able to figure out how to set up new recordings, and actually record them, and able to use them in a vector.

But what I'm looking for is any documentation or assistance on setting up the "serious" functions of the Conversant. Such as speech recognition, connection into databases for retreival to callers, etc.

Anyone with any information?

 

[FoneFun]

What version of the Conversant do you have?
Do you know what options you have purchased?

Most of the "serious" functions of the Conversant are achieved through the Conversant IVR's scripts.

Recording announcements and using them in a vector is part of your PBX, not the Conversant (which is an IVR). Once your do the "Converse-On", you have sent the call to the IVR and the IVR controls it until the end of the script, at which point, the PBX takes the call back and continues along the vector.

A nice GUI that you can use to generate the IVR scripts is Avaya IVR Designer (formerly Voice@Works). AID or V@W will depend on when you purchased it. Otherwise I suppose you could just code it the hard way though Script Builder or TAS, but I wouldn't be able to help you much there since I've never done it that way.

Speech recognition is an optional feature of the Conversant, and if I remember correctly, a fairly expensive option too. Do you know if its an option you purchased?
Some of the options you may have would be:
- Text to Speech
- Flexword Recognition
- Whole Word Recognition
- Fax
- ASAI

As for the database, the Conversant IVR should have it's own Oracle database. Most of the functionality here would have to be built, depending on your requirements. If you have apps in place already, then you'd have to figure out what they do, where the data goes/comes from, etc.

As for documentation, lots of it can be found on Avaya's site, in the support section. Search in the Conversant or Avaya Interactive Response or something like that. The Voice@works program has some help in the help menu too.

Since your posting was a bit general, I don't know if this answers your question. Otherwise let me know if you are looking to achieve something specific.

 

[Eliav]

Hi FoneFun.
I'm using this thread for another problem that i have with my Conversant system.
We have Conversant Map 40 that we worked with for several months (we created script with the Voice@Work and upload them to the Conversant).
A few month ago one of our Tech guys disconnect the switch from the Conversant system and when we tried to reconnect it we had a lot of problems.
After numerous attempts to connect the Conversant to work (when we tried to call the Conversant system we where hearing a dial tone and receiving the following trace from the selected channel :
CH001: TSM[001]: reset chan 1 (state 1)
CDH: CH001: Processing SERVICE record:
starttime=2005/03/30-10:07:27,endtime=2005/03/30-10:07:27,callid=25,serv
id=37,name=temp
CDH: CH001: Inserting CALL record:
starttime=2005/03/30-10:07:27,endtime=2005/03/30-10:07:27,cid=25
CH001: AD: main: irInit the channel.
CH001: TWIP: NEWCALL 0 digs
CH001: AD: dispatch_call: ENTER: cid=08070ab0 chan=1 dispmode=IRD_AD_STARTUP
CH001: AD: dispatch_call: Dispatching process=TSM service=temp type=PERMANENT
TSM[001]: CH001: Received IRE_EXEC IREM_EXEC
TSM[001]: CH001: iri_reserve_resource: IRER_NORESOURCES: resource allocation fai
lure, capability=4 ret=-1
TSM[001]: CH001: iri_reserve_resource: IRER_NORESOURCES: resource allocation fai
lure, capability=6 ret=-1
TSM[001]: CH001: ERROR TSM_SPUNAVAIL: tsm.c(347) chtype: -- func: "VOICE or TTS
"

We decided to reboot the Conversant machine - our support guy told us he doesn't familiar with this problem and he hope the reboot will solve it, but after rebooting the Conversant machine the same is happing.
Any ideas what is the problem?
Thanks in advance.
Eliav

 

[dmouge]

Hi Eliav,Smaslin


Regarding your issue Eliav:
check the SSP board of your system. maybe this board is a brokien state or out of service.... I do not have nay idea about the trace but i get this kind of issue where the system was rebootin and not coming up properly becasue this board was put in borken and it was not.... even if you have the state borken try to put it in manual out of service then in service...

Smaslin: on conversant you can used vonetix to connect to other database tahn the local Oracle DB for instance. i'm using it via web interface (the query is execute on a web server giving me the result between 2 specific tag on the web page). I 'm using it on conversant ucs1000 a quite old system now but runing fine.

 

[Eliav]

Hi dmouge
Thanks for your replay.
The board is in INSERV status, and you can see from the trace that the call reaches the channel but face a "resource allocation" problem.
any other ideas?
Eliav

 

[FoneFun]

Eliav, have you tried the dmouge's suggestion of putting the SSP board manually out of service, then back? I have also had this problem with the SSP (I think this is the resource that does the speech), and rebooting didn't always fix it. Sometimes, I would have to wait for the system to come back up, then busy out the SSP resource (put it out of service), and release it (put it back into service).

After upgrading to the last SP available for our Conversant 8, this has not re-occurred yet, so you might want to look for that if you have a service contract.

 

[Eliav]

Hi FoneFun
That was the problem, after restarting the SSP card there is nor ressurce allocation problem.
But now the calls dosen't reach the card at all.
any idea's
thanks
Eliav

 

[FoneFun]

Does the IVR actually answer the call?
If you use the Converse-On, are your IVR port agents logged in? You should be able to check that in the IVR or using CMS. Look at the Split/Skill that you converse to.

 

[Eliav]

Hi FoneFun
This the problem - the IVR doesn't answer the call - when i monitor the relevant channel there is no "action" on the relevant channel.

 

[FoneFun]

Can you see if there are any agents logged in with the skill used in the converse-on?
The IVR answers calls using agents that it logs in on the channels. So if you do a Converse-on skill 10, use CMS and you should see agents (1 for each port you have), logged in and available.

You can also check if in the IVR, under Voice Services, Number services, if you are monitoring the VDN (DNIS).
Something to note though, if your PBX is version 8 or less, then it's the first VDN, not the last one.
In my version 11 PBX, I have an option in the VDNs called Allow VDN override for ASAI messaging. This is like the Allow VDN override that controls Skills/VOA, but specifically for ASAI messages, which is how the IVR gets the DNIS. If you are set up to answer based on DNIS, and a call is passed to the IVR with an unrecognized DNIS, it will not answer the call, even though it rings on the port.

In the end, what I ended up doing is creating an IVR app that I set to answer on ANY - ANY dnis. When I converse-on to the IVR skill, I pass the VDN (which is the current VDN) to the IVR. Based on the VDN that is passed to the script, it will execute the next script (IE: VDN 1000 launches app A, VDN 1001 launches app B). I put this info in a table to avoid having to recompile this app each time I have a new script.

Hope that helps.

 

[Eliav]

Hi FoneFun
First I would like to thank you for your detailed replays - I found them very helpful.
The thing is I don't have a CMS - we are using Genesys CTI and just to make things more understandable we are just checking the IVR for future use.
Currently the IVR is connected to the switch through a DS1 card in the PBX (V.9), i just defined a station in the switch (1601 - type DS1FD) and in the Conversant i assign this station to the channel that "holds" the voice@work script.
Currently the problem is that when I call the system I continue to hear a ring back tone - and nothing else but when I monitor the relevant channel I see that the call has entered the IVR system.
I attached a trace of the channel:
CH001: TWIP: NEWCALL 0 digs
CH001: AD: dispatch_call: ENTER: cid=080710c8 chan=1 dispmode=IRD_AD_STARTUP
CH001: AD: dispatch_call: Dispatching process=TSM service=test_reb type=PERMANEN
T
TSM[001]: CH001: Received IRE_EXEC IREM_EXEC
CH001: TSM[001]: tsmSetTalkoff(1)
CH001: TSM[001]: -- Program "test_reb" (level 0) Begins:
CH001: TSM[001]: extend instruction Start function:
CH001: TSM[001]: setcca instruction Start function:
CH001: TSM[001]: tttime instruction Start function:
CH001: TSM[001]: vctime instruction Start function:
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: No attachments for prog: 0x8075604
CH001: TSM[001]: No attachments for prog: 0x8075604
CH001: TSM[001]: Added attachment 0x8071c48 (number 1) for prog: 0x8075604 (atde
s: 3) size: 28
CH001: TSM[001]: Found attachment 0x8071c48 for prog: 0x8075604 (atdes: 3)
TSM[001]: CH001: ERROR: Bad DIP Name: dbdip2
TSM[001]: CH001: ERROR: tsmFetchDipno() failed. arg: 0
CH001: TSM[001]: Found attachment 0x8071c48 for prog: 0x8075604 (atdes: 3)
TSM[001]: CH001: ERROR: Bad DIP Name: dbdip3
TSM[001]: CH001: ERROR: tsmFetchDipno() failed. arg: 0
CH001: TSM[001]: Found attachment 0x8071c48 for prog: 0x8075604 (atdes: 3)
TSM[001]: CH001: ERROR: Bad DIP Name: dbdip4
TSM[001]: CH001: ERROR: tsmFetchDipno() failed. arg: 0
CH001: TSM[001]: Found attachment 0x8071c48 for prog: 0x8075604 (atdes: 3)
TSM[001]: CH001: ERROR: Bad DIP Name: dbdip5
TSM[001]: CH001: ERROR: tsmFetchDipno() failed. arg: 0
CH001: TSM[001]: test_reb: STEP: 0 VALUE: "Application Name: test_reb"
CH001: TSM[001]: test_reb: STEP: 0 VALUE: "Application Version: 4/6/2005 3:19:59
PM"
CH001: TSM[001]: test_reb: STEP: 0 VALUE: "Avaya Inc. Voice@Work Version: 4.5.14
"
CH001: TSM[001]: test_reb: STEP: 0 VALUE: "Code Generator Version: 4.5.14"
CH001: TSM[001]: test_reb: STEP: 0 VALUE: "Platform: GBCS"
CH001: TSM[001]: test_reb: STEP: 0 VALUE: "@(#)CVISCMDS.PAS 21.1.2.1"
CH001: TSM[001]: Pushed stack, level 1
CH001: TSM[001]: Set PC to 7964
CH001: TSM[001]: Popped stack (rts), level 0
CH001: TSM[001]: test_reb: STEP: 1 VALUE: "Start"
CH001: TSM[001]: tsmSetTimer(45000)
TSM[001]: CH001: Ignoring unexpected event: IRE_NEWCALL
CH001: TWIP: T1_OFFHK, who=55
CH001: TSM[001]: EVENT id: IRE_ANSWER_DONE
cid: 0x806d5c0 tag: 134642624 text_len: 0 text: 0x0
mods: IREM_COMPLETE (6) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: tic instruction Eventhdlr:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: Timer cancelled. 45000 msec remaining
CH001: TSM[001]: Set PC to 572
CH001: TSM[001]: test_reb: STEP: 2 VALUE: "Hello"
CH001: TSM[001]: play(talkfile: 102 phrase: 5001 style: 50)
CH001: TSM[001]: play(talkfile: 102 phrase: 5002 style: 50)
CH001: TSM[001]: tsmSetTalkoff(1)
CH001: TSM[001]: tsmSetTalkoff(1)
CH001: TSM[001]: VOICE Playback STARTED (WAIT irEflag: 0)
CH001: TSM[001]: tsmSetTimer(0)
CH001: TWIP: Command Error: Received Unexpected Request (type 0x5) (NOTE: expect
ed for first on-hook after reset)
CH001: TSM[001]: EVENT id: IRE_PLAY_DONE
cid: 0x806d5c0 tag: 0 text_len: 0 text: 0x0
mods: IREM_COMPLETE (6) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: tflush instruction Eventhdlr:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: Set PC to 656
CH001: TSM[001]: test_reb: STEP: 3 VALUE: "MainMenu"
CH001: TSM[001]: play(talkfile: 102 phrase: 5000 style: 50)
CH001: TSM[001]: play(talkfile: 102 phrase: 5003 style: 50)
CH001: TSM[001]: tsmSetTalkoff(1)
CH001: TSM[001]: VOICE Playback STARTED (WAIT irEflag: 0)
CH001: TSM[001]: tsmSetTimer(0)
CH001: TWIP: Command Error: Received Unexpected Request (type 0x5) (NOTE: expect
ed for first on-hook after reset)
CH001: TSM[001]: EVENT id: IRE_PLAY_DONE
cid: 0x806d5c0 tag: 0 text_len: 0 text: 0x0
mods: IREM_COMPLETE (6) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: tsmFlushEventhdlr for getinput:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: tsmSetTimer(45000)
CH001: TWIP: Command Error: Received Unexpected Request (type 0x5) (NOTE: expect
ed for first on-hook after reset)
CH001: TSM[001]: EVENT id: IRE_INPUT_DONE
cid: 0x806d5c0 tag: 0 text_len: 0 text: 0x0
mods: IREM_TT_PRE (42) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: getinput instruction Eventhdlr:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: Timer cancelled. 39460 msec remaining
CH001: TSM[001]: Set PC to 1228
CH001: TSM[001]: Set PC to 1420
CH001: TSM[001]: play(talkfile: 102 phrase: 5006 style: 50)
CH001: TSM[001]: play(talkfile: 102 phrase: 5005 style: 50)
CH001: TSM[001]: Set PC to 712
CH001: TSM[001]: play(talkfile: 102 phrase: 5000 style: 50)
CH001: TSM[001]: play(talkfile: 102 phrase: 5003 style: 50)
CH001: TSM[001]: tsmSetTalkoff(1)
CH001: TSM[001]: VOICE Playback STARTED (WAIT irEflag: 0)
CH001: TSM[001]: tsmSetTimer(0)
CH001: TWIP: Command Error: Received Unexpected Request (type 0x5) (NOTE: expect
ed for first on-hook after reset)
CH001: TWIP: Command Error: Received Unexpected Request (type 0x5) (NOTE: expect
ed for first on-hook after reset)
CH001: TSM[001]: EVENT id: IRE_PLAY_DONE
cid: 0x806d5c0 tag: 0 text_len: 0 text: 0x0
mods: IREM_COMPLETE (6) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: tsmFlushEventhdlr for getinput:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: tsmSetTimer(45000)
CH001: TSM[001]: EVENT id: IRE_INPUT_DONE
cid: 0x806d5c0 tag: 0 text_len: 0 text: 0x0
mods: IREM_TT_PRE (42) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: getinput instruction Eventhdlr:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: Timer cancelled. 39750 msec remaining
CH001: TSM[001]: Set PC to 1228
CH001: TSM[001]: Set PC to 1420
CH001: TSM[001]: play(talkfile: 102 phrase: 5006 style: 50)
CH001: TSM[001]: play(talkfile: 102 phrase: 5005 style: 50)
CH001: TSM[001]: Set PC to 712
CH001: TSM[001]: play(talkfile: 102 phrase: 5000 style: 50)
CH001: TSM[001]: play(talkfile: 102 phrase: 5003 style: 50)
CH001: TSM[001]: tsmSetTalkoff(1)
CH001: TSM[001]: VOICE Playback STARTED (WAIT irEflag: 0)
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: EVENT id: IRE_PLAY_DONE
cid: 0x806d5c0 tag: 0 text_len: 0 text: 0x0
mods: IREM_COMPLETE (6) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: tsmFlushEventhdlr for getinput:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: tsmSetTimer(45000)
CH001: TSM[001]: EVENT id: IRE_INPUT_DONE
cid: 0x806d5c0 tag: 0 text_len: 0 text: 0x0
mods: IREM_TT_PRE (42) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: getinput instruction Eventhdlr:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: Timer cancelled. 39750 msec remaining
CH001: TSM[001]: Set PC to 1228
CH001: TSM[001]: Set PC to 1420
CH001: TSM[001]: Set PC to 2380
CH001: TSM[001]: test_reb: STEP: 6 VALUE: "End"
CH001: TSM[001]: play(talkfile: 102 phrase: 5014 style: 50)
CH001: TSM[001]: tsmSetTalkoff(1)
CH001: TSM[001]: tsmSetTalkoff(1)
CH001: TSM[001]: VOICE Playback STARTED (WAIT irEflag: 0)
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: EVENT id: IRE_PLAY_DONE
cid: 0x806d5c0 tag: 0 text_len: 0 text: 0x0
mods: IREM_COMPLETE (6) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: tflush instruction Eventhdlr:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: Set PC to 2452
CH001: TSM[001]: test_reb: STEP: 7 VALUE: "END1"
CH001: TSM[001]: tsmSetTimer(45000)
CH001: TWIP: T1_ONHK, who=55
CH001: TSM[001]: EVENT id: IRE_DISCONNECT_DONE
cid: 0x806d5c0 tag: 0 text_len: 0 text: 0x0
mods: IREM_COMPLETE (6) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: tic instruction Eventhdlr:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: Timer cancelled. 43000 msec remaining
CH001: TSM[001]: Set PC to 2496
CH001: TSM[001]: test_reb: STEP: 8 VALUE: "Stop"
CH001: TSM[001]: Set PC to 5352
CH001: TSM[001]: test_reb: STEP: 0 VALUE: "end:"
CH001: TSM[001]: Set PC to -1
CH001: TSM[001]: Script Terminates
CH001: TSM[001]: -- Program "test_reb" (level 0) Ends
CH001: TSM[001]: dipterm instruction Exit function:
CH001: TSM[001]: Found attachment 0x8071c48 for prog: 0x8075604 (atdes: 3)
CH001: TSM[001]: Sending DIP message: 340 (seq: 0 size: 16) to DIP1 (NO WAIT)
CH001: TSM[001]: Found attachment 0x8071c48 for prog: 0x8075604 (atdes: 3)
CH001: TSM[001]: Deleted attachment 0x8071c48 for prog: 0x8075604 atdes: 3 (0 at
tachments left)
CH001: TSM[001]: extend instruction Exit function:
CH001: TSM[001]: getdig instruction Exit function:
CH001: TSM[001]: phreserve instruction Exit function:
CDH: CH001: Processing SERVICE record:
starttime=2005/04/06-11:20:40,endtime=2005/04/06-11:21:44,callid=38,serv
id=50,name=test_reb
CDH: CH001: Inserting CALL record:
starttime=2005/04/06-11:20:40,endtime=2005/04/06-11:21:44,cid=38
CH001: AD: main: irInit the channel.
CH001: TSM[001]: reset chan 1 (state 0)


Thanks for all the help

Eliav

 

[BIS]

Eliav,

You defined the station.

Did you add a hunt group with AAS set to yes, and assigned this to an agent ID with a Port Extension of your DS1FD station?

 

[dmouge]

Hi,

Verify that you get an answer phone in your script otherwise you can have this kind of behaviour.. Or it can ba another bug of the voice @ work ... I prefer to works trought the old script builder even if it is not nice to see i found it easier and more reliable than the V@W.

 

[Eliav]

Hi BIS and dmouge
dmouge - I have an answer phone in my script - I tested this script in the past (before all the cards problems) and it worked fine
BIS - I changed the hunt (that was already defined) to AAS - yes (it was AAS - no) but I didn't understand the part of the agent ID - if you can explain more what you mean
Thanks
Eliav

p.s this is the hunt group seetings


Group Number: 1 ACD? y
Group Name: IVR Skill Queue? y
Group Extension: 1501 Vector? y
Group Type: ead-mia
TN: 1
COR: 1 MM Early Answer? n
Security Code:
ISDN Caller Display:

Queue Length: 99
Calls Warning Threshold: Port:
Time Warning Threshold: Port:


HUNT GROUP

Skill? y Expected Call Handling Time (sec): 180
AAS? y Acceptable Service Level (sec): 20
Measured: internal
Supervisor Extension:


Controlling Adjunct: none


VuStats Objective:
Timed ACW Interval (sec):
Multiple Call Handling: none

Redirect on No Answer (rings):
Redirect to VDN:
Forced Entry of Stroke Counts or Call Work Codes? n


HUNT GROUP

Message Center: none






LWC Reception: none

 

[FoneFun]

Hm, I've never spent much time looking at traces so I'm not sure what I'm looking at.

It looks like you are doing things with a table. Do the tables exist? When you uploaded the V@W script the first time, on the tab called Code Generation, did you specify to include in the Target Files to Generate:
- Script files (*.h, *.t and *.D
- Database files (*.sql)
- Phrase list files (*.pl)
- Phrase sound files (*.vis)

Assuming you have recorded the phases you are trying to play, you need the sound files and include them in your upload. Otherwise, all you will hear is ring back.

On the Application Transfer tab, you need to select all the files if this is the first time. I generally ommit the .sql files if I've already done it once, otherwise it will squish the existing tables and recreate it.

And finally, the application install tab, you need to do it once with the Overrite option, to restart the database DIPs,

To be sure that the tables exist, you can telnet to the ivr and run sqlplus. Do a "select * from

If you are having problems with recordings, another thing to try is the built in phrases. In your Phases window (in voice @ works), right click and select Standard Speech. This should give you a bunch of pre-recorded words from the IVR.
Go to your Hello Announcement, edit node, double click the Initial Prompt, and in the prompt text you can drag some Standard Speech into it. I often use the phrase 1123 which says "thank you", and 1044 "goodbye" when I'm testing.

I ran a trace on mine.
I see that the dbdip2 to dbdip5 errors are the same.
The place I see a difference is when you try to play a message. You get TWIP: Command Error.
In mine it looks like this:

CH001: TSM[001]: idcust: STEP: 6 VALUE: "CollectCustAccount"
CH001: TSM[001]: play(talkfile: 136 phrase: 5015 style: 50)
CH001: TSM[001]: tsmSetTalkoff(1)
CH001: TSM[001]: VOICE Playback STARTED (WAIT irEflag: 0)
CH001: TSM[001]: tsmSetTimer(0)
CH001: TSM[001]: EVENT id: IRE_PLAY_DONE
cid: 0x8089090 tag: 0 text_len: 0 text: 0x0
mods: IREM_COMPLETE (6) IREM_NULL (0) IREM_NULL (0) IREM_NULL (0)
CH001: TSM[001]: tsmFlushEventhdlr for getinput:
CH001: TSM[001]: Eventhdlr returned TSM_CONTINUE

So, when playing messages, it gives you the unexpected error. So you might want to examin the suggestions from BIS.
Maybe there is a problem with the line not picking up?

I think there are 2 ways for the channels on the IVR to answer, either set the hunt to AAS, and the agents to AAS, or set them up not as AAS, and have the IVR log them in (Depends on the options on your ivr though, you need the ASAI option to have the IVR log them in)

The reason for all this is that the IVR ports act like your digital phone, and it requires agent logins with the correct skill to answer the call, just like the normal queues. Setting the hunt and agent to AAS will cause the agents to always be logged in and available to those phones.

When I first set mine up, I tried this and it worked, but I decided not to go this route, but I forget the exact reason why. Instead I set mine up to not be AAS, and used the IVR's ASAI feature to log them in.


display hunt-group 90 Page 1 of 3 SPE A
HUNT GROUP

Group Number: 90 ACD? y
Group Name: IVR SKILL Queue? n
Group Extension: 1095 Vector? y
Group Type: ucd-mia
TN: 1
COR: 52 MM Early Answer? n
Security Code:
ISDN Caller Display:

Skill? y Acceptable Service Level (sec): 30
AAS? n Expected Call Handling Time (sec): 180
Measured: external
Supervisor Extension: Timed ACW Interval (sec):
Service Level Supervisor? n

Controlling Adjunct: none

Multiple Call Handling: none
Redirect on No Answer (rings):
Redirect to VDN:
Forced Entry of Stroke Counts or Call Work Codes? n

Message Center: none
LWC Reception: none
AUDIX Name:
Messaging Server Name:

And my agent:

display agent-loginID 2990 SPE A
AGENT LOGINID

Login ID: 2990 AAS? n
Name: IVR PORT AGENT AUDIX? n
TN: 1 LWC Reception: audix
COR: 20 LWC Log External Calls? n
Coverage Path: AUDIX Name for Messaging: AUDIX1
Security Code: Messaging Server Name for Messaging:
Direct Agent Skill: LoginID for ISDN Display? n
Call Handling Preference: greatest-need Password:
Service Objective? n Password (enter again):
Auto Answer: all
SN RL SL SN RL SL SN RL SL SN RL SL
1: 90 1 6: 11: 16:

So in my setup, using the cvis_menu, go to feature packages, ASAI Administration, Channel Administration, and map for each of the channels, one Agent Login, and set the status to LOGIN. Again, this will only work if you've purchased the ASAI feature. Otherwise you have to do it with the AAS.

You mentioned that you do not have CMS, but you must have something else to watch your call centre queues? Use the same tool and monitor the hunt/skill you use for the IVR and see if the call is actually picked up, and not just ringing on the station/port. It may be that the call gets to the IVR port, but is not answered.

 

[Eliav]

Hi FoneFune
Thanks for your replay.
I will try it tomorrow in the office, and update you.
Thanks again.
Eliav

 

[BIS]

Without ASAI, just create an agent, say XXXX. Give agent XXXX one skill - in this case 90. On the agent-loginID form, in the Port Extension field, put in the number of the station you defined as DS1FD.

On a side note, I usually have my IVR hunt groups as ucd ,not ead - not sure if it makes a difference in this case though.

 

[Eliav]

Hi All
Thanks for all your help.
The IVR is working fine now!!!
The problem was the SSP card and changing it status solved the problem.
Thanks all for your help.
Eliav