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New Member

DTMF CUBE H323-to-SIP problem

Problem Details: My scheme:

cucmbe 7.1

h323 gateway 3825

sip trunk provider

CUCMBE - H323 - GATEWAY - SIP TRUNK

The incoming calls from sip trunk are routed to a unity callhandlers, the input pressed are not reconized...the user press 1 or 2 or 3.....and nothing happens...but internals ip phones

press 1 or 2 or 3...work just fine...

the outbound dtmf works fine...

the problem is inbounds...

I think is dtmf-relay problem...!!!

this is my configuration:

voice service voip

allow-connection h323 to sip

allow-connection sip to h323

h323

emptycapability

h225 id-passthru

h245 passthru tcsnonstd-passthru

!

dial-peer voice 1 voice

description cucmbe to gateway cube

answer-address .T

incoming called-number .T

dtmf-relay h245-alphanumeric

codec transparent

ip qos dscp cs5 media

ip qos dscp cs5 signaling

!

dial-peer voice 2 voice

description gateway cube to sip trunk

destination-pattern .T

session target ipv4:200.20.21.200

dtmf-relay rtp-nte

codec g729br8

clid network-number 53197010

ip qos dscp cs5 media

ip qos dscp cs5 signaling

!

dial-peer voice 3 voice

description gateway cube to cucmbe

destination-pattern 3197010

session target ipv4:172.25.51.253

dtmf-relay h245-alphanumeric

codec transparent

ip qos dscp cs5 media

ip qos dscp cs5 signaling

!

Note: the sip provider have to recive calling number : 53197010 to make calls.

that why the dial-peer 2 have the command clid network-number: 53197010, the outside world communicate with me withn the number 3197010. In the call manager have a translation rule 3197010 to voicemail 555. that why the people calls 3197010 and go to the callhandler unity.

pls help!!!!

1 REPLY
New Member

Re: DTMF CUBE H323-to-SIP problem

Try adding the following two statements to your sip dial-peers:

voice-class sip dtmf-relay force rtp-nte

dtmf-relay sip-notify rtp-nte

no promises but this is what we have in our dial-peers and dtmf works both ways.

Hope this helps!

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