cannot send dtmf into pstn using h323 SIP Trunk

[Fachroky Wiratama]

Hello,

I have a problem that my IP Phone cannot pass the dtmf, when make a call and need to press 0 1 2 or something like that it doesnt work.

here is the configuration for the dial-peer

dial-peer voice 200 voip
 description ## outgoing all ##
 translation-profile outgoing outgoing-called
 destination-pattern 9T
 session target ipv4:172.16.34.100
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay  rtp-nte h245-alphanumeric
 no vad

and when i debug it appear like this

Nov  3 12:32:15.835: //59/B7CEDD9E8088/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x3C
Nov  3 12:32:15.835: //59/B7CEDD9E8088/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
Nov  3 12:32:15.835: //59/B7CEDD9E8088/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x3C
Nov  3 12:32:15.835: //59/B7CEDD9E8088/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
Nov  3 12:32:19.579: //59/B7CEDD9E8088/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x3C
Nov  3 12:32:19.579: //59/B7CEDD9E8088/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
Nov  3 12:32:19.579: //59/B7CEDD9E8088/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x3C
Nov  3 12:32:19.579: //59/B7CEDD9E8088/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
Nov  3 12:32:22.251: //59/B7CEDD9E8088/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x3C
Nov  3 12:32:22.251: //59/B7CEDD9E8088/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf

oh yaa.. it ever worked before but i dont know it doesnt work now even though i did not touch the configuration for the dtmf

please help :(

here is the config if needed

Thanks You,

Rocky

[Ayodeji Okanlawon]

What is the call flow. Which destination is the IP phone calling?

[Fachroky Wiratama]

From my Router to the building router here use sip trunk and then from building router to the ISP

172.16.34.100 is building router ip, then i dont know how the building router connect to ISP

Usually use sip trunk.

[lorenz84gDD]

the dial-peer you posted is h323, I wouldn't make "transcoding" between h323 and SIP. The one you call building-router is most probably a router with a sip trunk to your provider. Here are multiple factors that may cause your issue:

- transcoding between h323 and SIP

- dtmf relay methods mismatch i.e on one router you have h245-alpha and on another rtp-nte

- check what dial-peer are matched for a call when dtmf aren't working.

You should have on both your routers SIP in/out dial-peers like the following.

dial-peer voice 200 voip
 description ## outgoing all ##
 translation-profile outgoing outgoing-called
 incoming called-n 9T
 destination-pattern 9T
 session target ipv4:172.16.34.100
 ses pro sipv2 
 voice-class codec 1  
! voice-class h323 1
 dtmf-relay  sip-kpml sip-notify
 no vad

[Fachroky Wiratama]

the protocol between my router and the provider is h323 if i changed it into sip i cannot make a call.

is there any possibility ?

thanks,

Rocky

[lorenz84gDD]

try using h323 and check what dial-peers are matched. show call acti voi comp

[lorenz84gDD]

it's not clear what you have on the other side of this dial-peer. is it your isp? Maybe something changed on that side...

dial-peer voice 200 voip
 description ## outgoing all ##
 translation-profile outgoing outgoing-called
 destination-pattern 9T
 session target ipv4:172.16.34.100
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay  rtp-nte h245-alphanumeric
 no vad

[Fachroky Wiratama]

so the problem maybe in the other side ?