07-23-2012 02:55 PM - edited 03-16-2019 12:20 PM
Dear All,
I'm trying integration with SBC PSTN with SIP trunk (inband-g711aluw). All calls is always success ring and getting connection or answered, but the problem are when sending dtmf always send double digit and the destination does not receive tone. any idea about this ?
I have trying :
CME - CUBE (SIP trunk - inband G711aluw) - SBC -> DTMF Fail
CUCM - CUBE (SIP trunk - inband G711aluw) - SBC -> DTMF Fail
Regards
RD
07-23-2012 02:59 PM
Can you post your cube config?
Sent from Cisco Technical Support iPhone App
07-24-2012 05:27 AM
Hi Cris,
Below my configuration
CISCO#term le 0
CISCO#sh run
Building configuration...
Current configuration : 6250 bytes
!
! Last configuration change at 09:52:39 UTC Mon Jul 23 2012 by bicisco
! NVRAM config last updated at 09:53:50 UTC Mon Jul 23 2012 by bicisco
! NVRAM config last updated at 09:53:50 UTC Mon Jul 23 2012 by bicisco
version 15.2
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
service password-encryption
service sequence-numbers
!
hostname CISCO
!
boot-start-marker
boot system flash:c3900-universalk9-mz.SPA.152-3.T.bin
boot-end-marker
!
!
! card type command needed for slot/vwic-slot 0/1
logging buffered 10000000
no logging console
enable secret 5 $1$mqjI$MtrUawubK5sp8b1P2gpeh1
enable password 7 030752180500
!
aaa new-model
!
!
aaa authentication login h323 local
aaa authorization exec h323 local
!
!
!
!
!
aaa session-id common
no network-clock-participate wic 0
!
no ipv6 cef
!
!
!
!
!
no ip domain lookup
no ip cef
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice-card 0
dspfarm
dsp services dspfarm
!
!
!
voice service voip
address-hiding
mode border-element license capacity 500
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
registrar server
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
!
!
!
voice translation-rule 1
rule 1 /^0\(.+\)/ /\1/
!
voice translation-rule 2
rule 1 /.+/ /2125525860/
!
!
voice translation-profile TO-TELKOM
translate calling 2
translate called 1
!
!
!
license udi pid C3900-SPE100/K9 sn FOC14176RVM
license accept end user agreement
license boot module c3900 technology-package uck9
hw-module pvdm 0/1
!
hw-module sm 1
!
!
!
username bicisco privilege 15 secret 5 $1$TDcE$SOAyL1syyplCmgtYsC.gF.
!
redundancy
!
!
!
!
!
!
interface Loopback0
no ip address
shutdown
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
description JKT-TIP-01-MGMT-3550-01 Fa0/10
ip address 10.222.1.40 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
!
interface GigabitEthernet0/2
description METRO-TELKOM Port 3
ip address 10.14.0.6 255.255.255.252
duplex auto
speed auto
no cdp enable
!
interface Serial0/0/0
no ip address
shutdown
clock rate 2016000
!
interface Serial0/0/1
no ip address
shutdown
clock rate 2016000
!
interface Serial0/0/2
no ip address
shutdown
clock rate 2016000
!
interface Serial0/0/3
no ip address
shutdown
clock rate 2016000
!
interface GigabitEthernet1/0
no ip address
shutdown
!
interface GigabitEthernet1/1
description Internal switch interface connected to EtherSwitch Service Module
no ip address
!
interface Vlan1
no ip address
shutdown
!
!
ip default-gateway 10.222.1.1
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 10.222.1.1
!
ip access-list extended INSIDE
permit udp host 10.204.206.252 any
deny udp any any eq 5060
permit ip any any
ip access-list extended OUTSIDE
permit udp host 10.14.0.5 any
deny udp any any eq 5060
deny udp any any eq tftp
deny tcp any any eq www
deny tcp any any eq telnet
deny tcp any any eq ftp
deny tcp any any eq 22
permit ip any any
!
!
nls resp-timeout 1
cpd cr-id 1
!
snmp-server community %BI-Net&=+ RO
snmp-server ifindex persist
snmp-server enable traps entity-sensor threshold
!
!
!
control-plane
!
call treatment on
call threshold global cpu-avg low 68 high 75
call threshold global total-calls low 200 high 250
call spike 10 steps 6 size 250
!
!
!
!
!
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm 10.222.1.40 identifier 1 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/0
associate ccm 1 priority 1
associate profile 1 register XCODE_CUBE
registration retries 5
!
dspfarm profile 1 transcode
codec pass-through
codec g711alaw
codec g729r8
codec g711ulaw
maximum sessions 28
associate application SCCP
!
dial-peer voice 11 voip
description INBOUND-FROM-CME-CUCM
session protocol sipv2
incoming called-number 0[0-9]T
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 12 voip
description INBOUND-FROM-TELKOM
session protocol sipv2
incoming called-number 2552586.
codec g711alaw
no vad
!
dial-peer voice 22 voip
description OUTBOUND-TO-CME
destination-pattern 2552586[0-3]
session protocol sipv2
session target ipv4:10.204.206.252
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
no vad
!
dial-peer voice 23 voip
description OUTBOUND-TO-TELKOM
translation-profile outgoing TO-TELKOM
destination-pattern .T
session protocol sipv2
session target ipv4:10.14.0.5
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
codec g711alaw
no vad
!
dial-peer voice 24 voip
description OUTBOUND-TO-CUCM
destination-pattern 2552586[0-3]
session protocol sipv2
session target ipv4:10.204.27.222
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
no vad
!
!
!
!
gatekeeper
shutdown
!
!
telephony-service
sdspfarm units 2
sdspfarm transcode sessions 128
sdspfarm tag 1 XCODE_CUBE
sdspfarm tag 2 MTP_CUBE
max-ephones 1
max-dn 2
ip source-address 10.222.1.40 port 2000
max-conferences 8 gain -6
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Jul 16 2012 07:57:41
!
!
!
line con 0
logging synchronous
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
flowcontrol software
line vty 0 4
logging synchronous
transport input all
!
scheduler allocate 20000 1000
ntp source GigabitEthernet0/0
ntp server 10.240.255.1
!
end
Thanks
Regards,
RD
07-23-2012 03:04 PM
Can you also send along with your cube config, a "debug voip rtp session named-event "
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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson
07-23-2012 05:52 PM
you are most likley trying to use OOB DTMF and your trunk is using inband??
what dtmf are you using on your outbound and and inbound dial-peers on the CUBE?
If your dial-peers and CUCM trunk is configured correctly and you are trying to convert OOB and inband DTMF, build a MTP on your DSP farm on your CUBE and add it to your MGRL for the trunk. this takes about 2 minutes to config. if this doesnt work, pull some traces and post.
07-24-2012 05:31 AM
Hi Nick,
I already configured transcode at cube configuration, any idea do you have ?
Thanks
Regards
RD
07-24-2012 05:29 AM
Hi aokanlawon,
Below capture during calling and I trying to send digit dtmf:
CISCO#
044142: Jul 24 06:17:40.447: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x122 timestamp 0xB3B721B7
044143: Jul 24 06:17:40.447: <<
044144: Jul 24 06:17:40.447: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x122 timestamp 0xB3B721B7
044145: Jul 24 06:17:40.447: <<
044146: Jul 24 06:17:40.467: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x124 timestamp 0xB3B721B7
044147: Jul 24 06:17:40.467: <<
044148: Jul 24 06:17:40.467: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x124 timestamp 0xB3B721B7
044149: Jul 24 06:17:40.467: <<
044150: Jul 24 06:17:40.479: s=DSP d=VoIP payload 0x65 ssrc 0x60 sequence 0x123 timestamp 0x12295B97
044151: Jul 24 06:17:40.479: Pt:101 Evt:0 Pkt:03 00 00
044152: Jul 24 06:17:40.479: s=DSP d=VoIP payload 0x65 ssrc 0x60 sequence 0x124 timestamp 0x12295B97
044153: Jul 24 06:17:40.479: Pt:101 Evt:0 Pkt:03 00 00
044154: Jul 24 06:17:40.479: s=DSP d=VoIP payload 0x65 ssrc 0x3B90128 sequence 0x123 timestamp 0x12295B97
044155: Jul 24 06:17:40.479: Pt:101 Evt:0 Pkt:03 00 00
044156: Jul 24 06:17:40.479: s=DSP d=VoIP payload 0x65 ssrc 0x3B90128 sequence 0x124 timestamp 0x12295B97
044157: Jul 24 06:17:40.479: Pt:101 Evt:0 Pkt:03 00 00
044158: Jul 24 06:17:40.487: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x126 timestamp 0xB3B721B7
044159: Jul 24 06:17:40.487: <<
044160: Jul 24 06:17:40.487: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x126 timestamp 0xB3B721B7
044161: Jul 24 06:17:40.487: <<
044162: Jul 24 06:17:40.499: s=DSP d=VoIP payload 0x65 ssrc 0x60 sequence 0x125 timestamp 0x12295B97
044163: Jul 24 06:17:40.499: Pt:101 Evt:0 Pkt:03 00 00
044164: Jul 24 06:17:40.499: s=DSP d=VoIP payload 0x65 ssrc 0x3B90128 sequence 0x125 timestamp 0x12295B97
044165: Jul 24 06:17:40.499: Pt:101 Evt:0 Pkt:03 00 00
044166: Jul 24 06:17:40.507: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x128 timestamp 0xB3B721B7
044167: Jul 24 06:17:40.507: <<
044168: Jul 24 06:17:40.507: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x128 timestamp 0xB3B721B7
044169: Jul 24 06:17:40.507: <<
044170: Jul 24 06:17:40.519: s=DSP d=VoIP payload 0x65 ssrc 0x60 sequence 0x126 timestamp 0x12295B97
044171: Jul 24 06:17:40.519: Pt:101 Evt:0 Pkt:03 01 90
044172: Jul 24 06:17:40.519: s=DSP d=VoIP payload 0x65 ssrc 0x3B90128 sequence 0x126 timestamp 0x12295B97
044173: Jul 24 06:17:40.519: Pt:101 Evt:0 Pkt:03 01 90
044174: Jul 24 06:17:40.527: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x12A timestamp 0xB3B721B7
044175: Jul 24 06:17:40.527: <<
044176: Jul 24 06:17:40.527: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x12A timestamp 0xB3B721B7
044177: Jul 24 06:17:40.527: <<
044178: Jul 24 06:17:40.547: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x12C timestamp 0xB3B721B7
044179: Jul 24 06:17:40.547: <<
044180: Jul 24 06:17:40.547: s=VoIP d=DSP payload 0x65 ssrc 0x522CEFC sequence 0x12C timestamp 0xB3B721B7
Thanks
Regards,
RD
07-24-2012 05:39 AM
Can you do a test call and send the ff: "debug ccsip messages"
I can see you are using and sending pt 101. We need to see if your provider is doing the same
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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson
07-24-2012 05:42 AM
I can also see that you have not enabled dtmf relay on your outbound and inbound dial-peer to your sip provider...
dial-peer voice 12 voip
dial-peer voice 23 voip
description OUTBOUND-TO-TELKOM
translation-profile outgoing TO-TELKOM
destination-pattern .T
Can you add dtmf-relay rtp-nte to this dial-peers.
Please rate all useful posts
"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson
07-24-2012 05:48 AM
Hi aokanlawon,
yes, I'm not user dtmf-relay at dial-peer configuration becouse the connection used inband-SIP not out-of-band for rtp-nte.
Thanks
Regards,
RD
07-24-2012 05:53 AM
Rahman,
rtp-nte is not out of band. In trp-nte dtmf tones are carried inband. You need to specify the tyoe of dtmf to use. None is configured thats why its is not working. If you send me your debug ccsip messages, I will know what your provider is using. Most of then use rtp-nte
Please rate all useful posts
"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson
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