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.
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