Asterisk IVR Issues

Asterisk Asterisk Support
1

Hi All,

I have been posting on here for a while so i am not a first time poster, sadly though since the change over to the new Forums i have lost my old user name and also my old Nic, nothing i do gets be back into the Forums using my previous login details.

The Problem:

Firstly i have to point out, that the system i am referring to below is a Live production system that is currently handling many calls per minute and is working relatively fine.

Currently we have an IVR setup using our own voice prompts, they are in .WAV format and have been recorded in house, the files were recorded as per what * supports, so i can illuminate this as being a problem, and the fact that they are working fine as well.

What the issue is, when someone calls into our server (E-1) which is then passed onto the Main server (SIP Interconnection Over the CAT5-E Cable through a CISCO Switch) which is where the IVR system is located, they are presented with the IVR and the Voice prompts.

However i am being faced with a large number of calls from customers advising me of the following;

  1. They are unable to interrupt the voice prompt properly, sometimes there is a delay in the Interruption, sometimes it just wont allow for the Interruption.

  2. Sometimes it does not pick up on all the digits that are being pressed and therefore is unable to carry out the required task cause the number is either invalid or not correct.

This is becoming a huge problem for me as we have tried just about everything we can think of at this stage, somethings work in a small way whilst other things don’t.

I am at the point where i just cant trust the * box to handle such a task and thinking of just purchasing an entirely different system to do the IVR system, however this just does not make sense, this is too trivial of a problem for it to not be solved, i mean withe everything * can handle i cant see why this it cant :open_mouth:

I would dearly appreciate anyones advise or assistance on this issue, we are thinking of upgrading to Version 1.4 but i wont do that until i am satisfied that major problems are not going to destroy a deployment of the upgrade and bring our systems down, however though i am not even sure if Version 1.4 of * has any marginal improvements on this problem.

I look forward to your replies and i wish to thank you in advance for those who do.

Cheers,

David.

2

Many people with cell phones can not use the ivr, unless they press and hold the buttons for a little longer.
You will that those from ohter Voip system will have issues as well.
my nokia 5150 phone has no issue yet one of my partners Fancy 400.00 PDA is useless with the IVR.

You need to test from a good line.
there is a DTMF setting called relaxed have you tested that??

and be sure that the failover for the IVR is to a live exten if you want to have happy (well not as ticked off) customers.

3

[quote=“bubba”]Many people with cell phones can not use the ivr, unless they press and hold the buttons for a little longer.
You will that those from ohter Voip system will have issues as well.
my nokia 5150 phone has no issue yet one of my partners Fancy 400.00 PDA is useless with the IVR.

You need to test from a good line.
there is a DTMF setting called relaxed have you tested that??

and be sure that the failover for the IVR is to a live exten if you want to have happy (well not as ticked off) customers.[/quote]

OK, based on what your saying is that when i pass the call from the E-1 Asterisk Server (Using Digium cards for this) to the SIP server, this is the point of problem, so if i put the IVR system onto the E-1 Server and have it process all the calls i wont have this issue right?

I was thinking also to move the Server Interconnection from SIP to IAX2 but the question then is why would IAX be better then SIP.

I understand that Version 1.4 is supposed to have a better SIP stack and also better control over the DTMF, but i am too scared to move over to it since there just isn’t enough feed back from the community yet on it.

I know there should be a work around, i mean i have been using commercial VoIP systems for some years now and this seems to be one of the major flaws that other VoIP system don’t have, however don’t get me wrong i love Asterisk and what it can do, but i am a little Miffed as to why it has such a problem, VoIP and DTMF have been able to work hand-in-hand for some time now.

Any way i appreciate your post.

Cheers,

David.