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DTMF tones from CUCUM 9 thru H323 GW out SIP trunk not working

  This is the setup.  Currently in lab environment for a client, but needs to go into production

IP Phone -> CUCM 9 -> H323 GW -> SIP Trunk -> Proprietary device -> Analog phone

Calls complete both ways with no issues.  Proprietary devices only uses G711ulaw, so I have configured a xcoder on the H323 GW to transcode to G729 across the WAN link (between the CUCM cluster and the H323 GW).

Pressing keys/sending DTMF tones from the IP phone are not heard in the analog phone

Running a debug voice ccpai inout at the H323 gateway shows me that the DTMF tones are being received the GW and are being sent along.  See below:

Seaport#

Seaport#

Seaport#! Pressing digit "9" on VoIP phone

Seaport#

Seaport#

Seaport#

Seaport#

*Nov  5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:

   Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E

*Nov  5 15:41:57.637: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:

   Check DTMF relay digit begin for 3way conf

*Nov  5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:

   Consume mask is not set. Relaying Digit 9 to dstCallId 0x49E

*Nov  5 15:41:57.713: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:

   Check DTMF relay digit end for 3way conf

Seaport#

Seaport#! Pressing digit "9" on VoIP phone                " on VoIP phone                 5" on VoIP phone              

Seaport#

Seaport#

Seaport#

*Nov  5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:

   Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E

*Nov  5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:

   Check DTMF relay digit begin for 3way conf

*Nov  5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:

   Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E

*Nov  5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:

   Check DTMF relay digit end for 3way conf

Seaport#

Seaport#! Pressing digit "       5" on VoIP phone              
Seaport#
Seaport#
Seaport#
*Nov  5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov  5 15:42:14.913: //1181/00B99D4F0500/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
*Nov  5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 5 to dstCallId 0x49E
*Nov  5 15:42:14.989: //1181/00B99D4F0500/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
Seaport#
Seaport#

However, debug ccsip does not give me any indications that the DTMF tone is being sent out the SIP trunk.  Debug ccsip all attached.

Relevant portions of the H323 configuration are below

voice service voip

no ip address trusted authenticate

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

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

sip

  bind control source-interface Loopback0

  bind media source-interface Loopback0

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

codec preference 3 g729br8

!

interface Loopback0

ip address 172.16.88.254 255.255.255.255

ip pim sparse-dense-mode

h323-gateway voip interface

h323-gateway voip bind srcaddr 172.16.88.254

!

interface GigabitEthernet0/1

ip address 192.168.200.254 255.255.255.0

duplex auto

speed auto

!

          

interface Loopback0
ip address 172.16.88.254 255.255.255.255
ip pim sparse-dense-mode
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.88.254
!

interface GigabitEthernet0/1                                   <- interface to proprietary device

ip address 192.168.200.254 255.255.255.0

duplex auto

speed auto

!

interface GigabitEthernet0/2                                  <-interface to Local LAN supporting IP Phones

ip address 10.10.10.254 255.255.255.0

duplex auto

speed auto

!

sccp local GigabitEthernet0/2

sccp ccm 10.10.10.254 identifier 1 priority 1 version 3.1

!

sccp ccm group 1

bind interface GigabitEthernet0/2

associate ccm 1 priority 1

associate profile 10 register xcoder_1

!

dspfarm profile 10 transcode 

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 10

associate application SCCP

!

dial-peer voice 2 voip

          description Default Incoming Dial Peer

incoming called-number .

voice-class codec 1 

dtmf-relay h245-alphanumeric h245-signal rtp-nte

!

dial-peer voice 6 voip

destination-pattern 90052..                      <- DN of analog phone

session protocol sipv2

session target ipv4:192.168.200.1            <- IP of proprietary device

codec g711ulaw

no vad

!

sip-ua

registrar ipv4:172.16.88.254 expires 3600

no transport tcp

!

telephony-service

sdspfarm units 4

sdspfarm transcode sessions 2

sdspfarm tag 1 xcoder_1

I also ran the debug voip rtp session named-event all but nothing was displayed when I pressed the digits on the IP Phone.

Jeff

1 REPLY

DTMF tones from CUCUM 9 thru H323 GW out SIP trunk not working

Please configure "dtmf-relay rtp-nte" command under SIP dial-peers.

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.
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