11-08-2015 11:41 PM - edited 03-18-2019 11:42 AM
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
11-09-2015 01:46 AM
11-09-2015 11:02 PM
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.
11-10-2015 12:37 AM
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
11-10-2015 11:15 PM
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
11-11-2015 12:28 AM
try using h323 and check what dial-peers are matched. show call acti voi comp
11-09-2015 03:26 AM
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
11-09-2015 06:34 PM
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