ViaTalk → Re: DTMF Relay Support

I'm also using CallManager Express (CME) with 12.4(11)T. I haven't noticed a problem with DTMF relay except when calling voicemail. I have 'dtmf-relay rtp-nte' set. I went back and forth with support for a while and didn't really get anywhere. I've just been ignoring the problem for a while because I get the voicemail sent to email and I can log into the website to delete it. I realize that's not a pretty solution. Once I get my hands on a Cisco Unity Express (CUE) module, I'm probably just going to stop using their voicemail anyways unless my router is down.

If you want to share part or all of your configuration with me, I can try to help you troubleshoot.

So are you saying that you can call an automated attendant system and use touchtone to navigate a menu?

Mine worked previously. I did not change configuration or IOS version and it stopped working one day. I would call my voicemail at my office or a conference call system and it would not register any number I punched on the phone, no matter if I used a 7960 IP phone or a regular analog handset attached to my ATA logged into my CME router via skinny.

I am using the same dial-peer configuration with 'dtmf-relay rtp-nte' set. Everything that I have heard or read states that ViaTalk switched to in-band DTMF which is not the same as rtp-nte. Cisco does not support the method that they are now using for DTMF relay.

Feel free to PM me and I can share configs.

So ViaTalk supposedly uses RFC 2833 in-band RTP signaling using the Notification Telephone Event payload type. This is supposed to match up to Cisco's 'dtmf-relay rtp-nte' option on the dial-peer configuration. The gotcha is what protocol your phones or devices are using to talk to CME locally.

»fonosip.com/english/call ··· ger.html

They specifically mention here that SCCP based phones do not use in-band DTMF relay which is a given. So Skinny phones require that you use sip-notify on the dial-peer for DTMF relay. Since ViaTalk only supports RFC 2833 it essentially looks like you are forced to run a SIP image on all your CME devices no matter what if you want DTMF relay to work. Arrgghh!! You'd think Cisco would translate automatically. Of course not... That is a feature to not allow that to occur.

Go down to the section titled "DTMF Relay for SIP Applications and Voice Mail". So all my IP phones are running a Skinny image and even my ATA 188 adapter is
running a Skinny image. I guess I need to convert a phone to SIP and test it out.

If this is the issue it will be one more big caveat to add to my CME/ViaTalk document.

I will post a follow up when I have a chance to test it out.

The link you provided is for CME 3.0 which only supports sip-notify for dtmf-relay. With new versions after 3.2 and specifically 4.0, "dtmf-relay rtp-nte" should work just fine. The current documentation can be found at:

»www.cisco.com/en/US/prod ··· 03b.html

That being said, trying a phone with a SIP load has been on my list for a while now. Let me know if you find anything interesting.

So, I finally was able to convert one of my 7960 phones to a SIP image and register it under CME. I set DTMF relay to use rtp-nte on the associated voice register pool. I dialed out using ViaTalk, which the associated dial-peers already have 'dtmf-relay rtp-nte' set, and I was not able to navigate my work voicemail system. It did not register any digit I dialed. I dialed out using my FXO POTS line and it worked without any issues. Exact same result as when I use the SCCP Skinny image on my phones. So back to square one...

My personal feeling is that something changed within Asterix and its support for RFC 2833 RTP Named Telephone Events payload. DTMF relay worked fine from May until sometime around August and one day it just stopped working. I made no changes on my end. I have to assume that an Asterix upgrade occured that caused interoperability with Cisco's RFC 2833 implementation to stop working.