Call Menu (IVR tones) not detected at times

[AlexR]

Hello Matt,Everyone!

In a project im working on I've set up a Call Menu for a incoming DID, it works just as expected most of the time (kudos on that option!), but we have had several clients arguing that sometimes they press an option and the system plays the recording for selecting a choice again.

The weird thing it does not happen for all calls... or even most calls, only with some clients at times. I have tried placing the relaxdtmf option to yes, and still I'm getting this problem, the DID is on an E1 line, so there should not be much problem here.

Should i place a "hard coded" IVR as suggested in the manual instead? Is the call menu feature still in beta?

Any ideas are welcome!


Greetings,


Alejandro

[mflorell]

We have had a couple clients with this issue, and the only thing that fixed it was moving to a Sangoma T1/E1 card with HWEC(Hardware Echo Cancellation). The HWEC chipset is able to more reliably handle DTMF and we have seen it fix sporadic issues like this.

[williamconley]

when you post, please post your entire configuration including (but not limited to) your installation method, vicidial version and build, asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)
________

in case this issue is resolved through a hardware replacement, it is very handy to have the hardware listed.

also, listing all your specs could find your solution immediately if something is out of place!

[AlexR]

Ok thanks William, i was not aware of that new rule, I've been posting for a while... but that sure makes sense:

VERSION: 2.4-261
BUILD: 100703-1322
asterisk-1.4.27.1-vici
Single Server/Fedora 13 X86_64
Digium TE122B With Hardware Echo Cancel

I think i might have resolved this changind the tone duration in chan_dahdi.conf like this

Code: Select all

toneduration=300

also as i mentioned i have the relaxdtmf setting to yes.

Thank you all for your help as always!


Alejandro

[williamconley]

excellent and thanks for posting a possible solution. please post back if it works or not to let us know!

[AlexR]

William, I just wanted to followup: This is working like a charm now. I hear that they might be problems when a female speaker voice tone might be recognized as DTMF (this is true, based on reports), but we still have not encountered such an annoying voice.

Cheers,


Alejandro

[VFRDavid]

This post was about dahdi channels - I have tried a few options in sip.conf for the dtmfmode= setting - and - I cannot seem to get consistent recognition of keys pressed for Call Menus or Survey campaign choices. I've tried calling from several phones (Cisco SIP, Zoiper/Soft, land lines) - it seems to be something with the way that ViCi/Asterisk is (not) recognizing them...

Versions:
Asterisk: 11.21.2-vici
Vici v2.12b0.5
SVN 2534
DB Schema 1459

not sure what other versions you need - sorry if the above list is not complete / enough to assist...

David

[williamconley]

This post was about dahdi channels - I have tried a few options in sip.conf for the dtmfmode= setting - and - I cannot seem to get consistent recognition of keys pressed for Call Menus or Survey campaign choices. I've tried calling from several phones (Cisco SIP, Zoiper/Soft, land lines) - it seems to be something with the way that ViCi/Asterisk is (not) recognizing them...

Basics:

WHAT settings have you used for DTMF?

Who is pusthing the buttons (Agent, Prospect, 3rd party caller?). Usually it's the prospect, but I believe in checking. No assumptions.

Have you verified the DTMF settings from the source path through which the tone proceeds? (IE: If agent, verify from the phone to the servr. If prospect, verify from prospect to carrier to server ... entire path you have access to).

Check the asterisk logs of the arriving signal and see what's stopping the DTMF. Is asterisk receiving it, but somehow it's getting lost?

What settings does the Carrier recommend/specify for DTMF?

Have you tested a different carrier?

Is this for inbound or outbound? (If it's only for one of those, have you thoroughly tested the other? I've had instances where ONLY ONE CID would fail for outbound DTMF through a small group of carriers (no kidding! one cid, outbound only, a just a few carriers)