dtmf tone issue on trunk

[kangodan]

I have a sip trunk between a Cisco call manager and an asterisk as the gateway to make voip calls. With a sip cliet registered to the asterisk gateway i am able to make calls a pass through the dtmf tone accurately however when i make a call from a phone connected to the Cisco call manager i have an issue with dtmf.
I compared the capture from both instances and could not see the difference between the tone pass through. They both we passing through the right tone.
What can i do to help resolve my issue.

 

[KozarTech]

Hi Kangodan,
I may be able to help with your issue I just need a little more information.

1) Is the call manager using asterisk on the back end?
2) What kind of call manager is it your working with?

the sip end points would be registering with aterisk or the call manager. If the call goes through on askerisk fine but not the call manager then the problem is the dtmf settings in the call manager. Can you give me more details about your Dtmf settings in the call manager

use RFC2833 for dtmf settings