New system - issue with dial prompt

I just setup a new system for a customer and everything is working great. Except when the billing lady is entering in information at prompts for insurance claims 25% of the time it doesn’t recognize what she is typing properly. For instance it will think a large chunk was all zero’s.

Setup:
Freepbx Stable-3.211.63-10 was installed, I’ve done all the updates.
Lenovo Thinkserver ts130
Sangoma A200D Analog Series w/ Echo Cancellation

I’m assuming it’s the hardware Echo Cancellation on the Sangoma causing the issues but I’m unsure of how to adjust how aggressive the echo cancellation is.

If you click on support and purchase support credits with some of the money you made selling the system you will get some assistance.

There is also a ton of information on DTMF issues with Asterisk and on Sangoma’s support site.

I wouldnt assume the DTMF issues are with the Sangoma first. I have dozens of them installed with PRI’s with at least 4 different providers and never had to mess with Sangoma’s DTMF settings. (All my cards had the Echo Cancel Option)

The most common problems in my experience are with Analog phones or with SIP phones and a mismatch in the DTMF handling, or relying solely on RFC2833 with lossy codecs like g729 or gsm.

If your SIP device (Phone or analog adapter) support SIP INFO mode for DTMF, I have found that DTMF=INFO has been more reliable than DTMF=RFC2833 especially if you dial DTMF’s fast.

I had a fax server hanging off a Cisco IAD’s FXS ports and found that with the IAD in INFO mode, and the FreePBX extensions for each IAD port in Info mode, long inband DTMF signalling (10 digit dialed numbers) to send calls to the fax mailboxes that matched the DID’s were much more reliable.

skyking
I did buy support for another system when I was having issues with the paid extension routing module. However that was March and they’ve already expired. I figured the free support forum was worth a try again before paying up again. I wish their expiration policy was a little more lenient, in 3 months it’s not likely I’ll run into another issue needing their support, 6 months or a year would be a lot better. I used about 10 minutes of my hour purchased last time.

I started reading about DTMF troubleshooting yesterday on sangoma’s site, I’ll look into it some more.

Peter
Sorry, I forgot to put the phones in use! They are all Polycom’s either 335 or 650. Thanks for the info, I’d also thought about codec issues as that has been an issue in the past but it was a problem with a sip trunk config and I wasn’t sure what too be checking for the sip config between the phone and server as far as codec’s go.

I understand what you are saying. Realize I take a hard line, and it is my personal opinion. Despite the fact that I have gotten to know the team members we pay for support just like anyone else. I spend 1000’s of dollars a month with Schmooze. If you are making your living off an open source package and don’t know how to fix it then you should be paying. It’s not reasonable to expect myself or any other forum member to work for free with you.

If you asked a question that had a direct answer you would get help, but you didn’t even call this a DTMF issue. The fact you don’t understand, based on your response to perter, that RFC2833 and DTMF info are out of band methods and not CODEC dependent.

I understand what you’re saying and I’m not expecting anyone to fix issues for me. On any other forum I’ve been involved with as the owner, host, or user all that’s expected is someone to point them in the right direction but not to be spoon fed info.

For those finding this thread in the future here is an excellent article to help with out-of-band dtmf which i was unaware of before.

another article on dtmf
http://wiki.sangoma.com/asterisk-debugging-dtmf

Just wanted to update with the solution, I did a couple items to get it fixed so I’ll list them.

-Set the extensions to SIP INFO dtmf in Freepbx and added a line to the config from the polycom website (link1) to switch the phone to SIP INFO. At this point the dtmf issue was considerably better but still would have issues with one IVR in particular (MET life).
-I had run through the sangoma setup (link2) but had never enabled the DAHDI Module so I did that following the instructions from the wiki (link3). After doing that it was flawless.

link1 - http://community.polycom.com/t5/VoIP/FAQ-Phone-unable-to-send-DTMF-to-an-IVR-system/td-p/4237
link2 - http://www.freepbx.org/support/documentation/installation/first-steps-after-installation/installing-and-configuring-a-sango
link3 - http://wiki.freepbx.org/display/F2/DAHDI+Configs

Thanks for pointing me in the correct direction!

Thanks for replying with the resolution and resources! Could certainly help others in the future.