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New Member

CUCM 9.x with outside IVR

OK, so we are having CUCM 9.x and 2951 router with SIP trunk to PSTN. Router is set as h323 gateway for CUCM... Pretty simple configuration, nothing fancy, and everything is working as it should, but one thing.

After we are call someone outside who is using some sort of IVR(lets say you call your ISP, and first thing you get is IVR saying press 1 if you want to talk with sales, press 2 if you want to talk with IT...), connection is established, but after it, telephone doesn't react whatever number you are pressing. So after I get the message to press 2 to talk with IT for example, I can press 2 all day long, nothing would happen, I would still have innitial dialog...

Does anyone knows what is the problem with this, and where in configuration I could change it?

4 REPLIES
Hall of Fame Super Silver

CUCM 9.x with outside IVR

Sounds like DTMF mismatch, first question why are you using H323 between CUBE and CUCM if you have a SIP trunk? Do yourself a favor and convert it to SIP and then standardize on one Dtmf-relay method.

HTH,

Chris

New Member

CUCM 9.x with outside IVR

Mostly because I'm way over my head in voice over IP, so I used some similar configuration as my guide when I started, and there was h323 between CUCM and CUBE even though they too have used SIP trunk...

Also, have looked about similar dtmf mismatches, and what frightens me is that they were also using 7942 phones... But that was 3 years ago, and should've been solved by now.... Anyhow, big thanks on your response, helped me know where to look at I will try something tomorrow(looks like I have missed dtmf relay in one dial peer while configuring), and hopefully it will work.

Hall of Fame Super Silver

CUCM 9.x with outside IVR

When you get a chance also post your configuration, one thing to keep in mind is that all call flows use 2 dial peers (inbound and outbound) if you do not have an explicit inbound dial peer configured with proper dtmf method then hidden dial-peer 0 is used which is known to mess with things like this.

HTH, please rate all useful posts!

Chris

New Member

CUCM 9.x with outside IVR

so, the story is still the same, and im out of the ideas... one important note though. after we get connected and hear "welcome to call center of....", after pressing a number, we can hear a sound of number being pressed, but we are not seeing anything on phone display ! all the phones are 7942.

this is what debug says in the moment when we press the number(i have upload of whole debug ccsip, ccapi and rtp session, but not sure how to upload it):

*Dec 10 17:21:38.442: //7015/80F4757E4600/CCAPI/cc_api_call_digit_begin:

   Consume mask is not set. Relaying Digit 1 to dstCallId 0x1B68

*Dec 10 17:21:38.442: //7015/80F4757E4600/CCAPI/cc_relay_digit_begin_for_3way_conference:

   Check DTMF relay digit begin for 3way conf

*Dec 10 17:21:38.518: //7015/80F4757E4600/CCAPI/cc_api_call_digit_end:

   Consume mask is not set. Relaying Digit 1 to dstCallId 0x1B68

*Dec 10 17:21:38.518: //7015/80F4757E4600/CCAPI/cc_relay_digit_end_for_3way_conference:

   Check DTMF relay digit end for 3way conf

i would post whole configuration, but again, i dont see where can i upload file, so here are important parts:

voice-card 0

dspfarm

dsp services dspfarm

!

!

!

voice service voip

ip address trusted list

  ipv4 192.168.168.0 255.255.255.0

  ipv4 10.0.0.0 255.0.0.0

  ipv4 10.0.0.2

  ipv4 10.27.2.0 255.255.255.0

dtmf-interworking rtp-nte

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

h323

  no h225 timeout keepalive

  h225 id-passthru

  call preserve

  h245 passthru tcsnonstd-passthru

  h245 passthru all

sip

  header-passing

  early-offer forced

  midcall-signaling passthru

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g729r8

codec preference 3 g729br8

codec preference 5 g723ar53

codec preference 6 g723ar63

--More--                            codec preference 7 g723r53

codec preference 8 g723r63

!

voice class h323 100

  h225 timeout tcp establish 2

  h225 timeout setup 2

sccp local GigabitEthernet0/1

sccp ccm 10.27.2.12 identifier 1 version 7.0

sccp

!

sccp ccm group 1

bind interface GigabitEthernet0/1

associate ccm 1 priority 1

associate profile 3 register my_trans

associate profile 2 register my_mtp

associate profile 1 register my_conf

!

dspfarm profile 3 transcode 

codec g729r8

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 3

associate application SCCP

--More--                           !

dspfarm profile 1 conference 

codec g729br8

codec g729r8

codec g729ar8

codec g729abr8

codec g722-64

codec g711ulaw

codec g711alaw

maximum sessions 3

associate application SCCP

!

dspfarm profile 2 mtp 

codec g711alaw

codec pass-through

maximum sessions software 100

associate application SCCP

!

dial-peer voice 10 voip

description TELEKOM_KA_CUBE

service skripta

session protocol sipv2

session target sip-server

incoming called-number (hidden)

voice-class codec 1 offer-all

voice-class sip profiles 1

no voice-class sip encap clear-channel standard

dtmf-relay rtp-nte sip-notify

no vad

!

dial-peer voice 20 voip

description CUBE_KA_TELEKOM

translation-profile outgoing 1

preference 1

destination-pattern .T

session protocol sipv2

session target sip-server

voice-class codec 1 offer-all

voice-class sip dtmf-relay force rtp-nte

voice-class sip profiles 1

voice-class sip encap clear-channel standard

dtmf-relay rtp-nte sip-notify

no vad

--More--                           !

dial-peer voice 30 voip

description CUBE_KA_CUCM

translation-profile outgoing 2

destination-pattern (hidden)

session target ipv4:10.27.2.12

voice-class codec 1 offer-all

dtmf-relay h245-signal h245-alphanumeric

no vad

!

dial-peer voice 40 voip

description CUCM_KA_CUBE

session protocol sipv2

session target sip-server

incoming called-number .T

voice-class codec 1 offer-all

voice-class sip profiles 1

dtmf-relay rtp-nte sip-notify

no vad

!

dial-peer voice 50 voip

description OD_CUBE_KA_CUCM

destination-pattern 75..

session target ipv4:10.27.2.12

voice-class codec 1 offer-all

dtmf-relay h245-signal h245-alphanumeric

no vad

________________________________________________________________________

i have tried different scenarios with dtfm-relay rtp-nte on dial peer 30 and 50, still nothing...

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