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...
