I have the following issues with DTMF......
If I dial from my cell phone or a land line to my menu in UCCE/CVP I do not get the option. It is like the system is not getting it. If I dial the DN internally it works and if I dial 9XXXXXXXXXX from a IP Comm it works.
External Phone (Cell, Land Line) PSTN>CUBE>CUSP>CVP/VXML>Menu (I hear the prompt press 1 nothing happens) then call times out.
Internal IP Comm dial 9+ CUCM>CUSP>CUBE>PSTN>CUBE>CUSP>CVP/VXML (Basically trombone the call)
I have MTP checked on the SIP Trunk to CUSP and using SW MTP..
dial-peer voice 104 voip
description *** Straight to CUSP for UCCX/UCM CC Demo Lab 480-XXX-XXXX
session protocol sipv2
session target ipv4:172.28.36.125:5060 -------This sends it back to the SIP proxy CUSP
voice-class codec 1
no voice-class sip localhost
no voice-class sip asserted-id
no voice-class sip privacy-policy passthru
voice-class sip options-keepalive up-interval 30 down-interval 15 retry 2
dtmf-relay rtp-nte sip-notify
ip qos dscp cs3 signaling
I have tried running the CCSIP debug all and it over whelms my gateway..... What do you suggest since I can not use that and my call path is end to end SIP from the Telco to my solution.
If your gateway is particularly busy I would suggest using voice call filtering on your SIP debugs.
It allows you to filter only the debug output for the calls you are interested. For example, you could build a filter list to only capture debugs for calls originating from your cell phone number.
I would recommend taking a "debug ccsip messages" with a filter list, and seeing what DTMF relay method is negotiated on both legs of the call. You can compare the method negotiated with the call that works against the call that does not work.
Some methods of DTMF relay require a transcoder to be invoked on the CUBE directly (MTP will not work).
See "table 3 RTP-RTP Flow-Through"
There are also some useful commands for verifying DTMF relay on CUBE here as well: