10-15-2012 04:04 PM - edited 03-16-2019 01:41 PM
It has SIP trunks to the PSTN when dialing to the PSTN to an IVR or automatic answering does not collect the digits to be dialed to select menu options or the digits of the extension to which you want to dial.
The dial-peer is configured as follows:
dial-peer voice 3002 voip
description LLAMADAS 01800 POR SIP
destination-pattern 01800.......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:64.132.31.26
dtmf-relay cisco-rtp rtp-nte
clid network-number 55110713
My question is if I have to change the DTMF command for this:
dtmf-relay h245-alphanumeric
The Gateway is configured as H323.
What is your recommendation?
Thank you
regards
10-16-2012 01:52 AM
Hi, I believe it is H.323 to SIP CUBE. please confirm the dtmf method your SIP provider supports. if it is rfc2833, in sip dial-peer, you can configure "dtmf-relay rtp-nte" what is the in-leg dtmf method you have configured for h.323 to cucm?
For DTMF method in SIP: http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html
10-16-2012 03:14 AM
Please send the full config. Are you using sip-sip or h323-sip? What I mean is that what is your dial-peer to cucm configured for?
Also what DTMF preference have configured for your sip trunk?
Please rate all useful posts
"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson
10-16-2012 09:27 AM
Hi Aokanlawon, sureshsub
This gateway is configured as H323, I send the topology and full config on voice gateway:
.....
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol cisco
!
!
!
voice class codec 1
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 g723r63
codec preference 4 g711ulaw
codec preference 5 g711alaw
!
!
!
!
voice class h323 1
h225 timeout tcp establish 3
!
!
voice translation-rule 1
rule 1 /0050/ /8500/
!
!
voice translation-profile VOSS
translate called 1
!
!
voice-card 0
dsp services dspfarm
!
!
!
!
!
controller E1 0/0/0
framing NO-CRC4
ds0-group 0 timeslots 1-15,17-20 type r2-digital r2-compelled ani
cas-custom 0
country telmex
category 2
answer-signal group-b 1
caller-digits 4
dnis-digits min 4 max 4
description 20 TRONCALES A PSTN TELMEX
!
!
class-map match-any VOIP-SIGNAL
match ip dscp cs5
match ip precedence 4
match ip precedence 3
class-map match-any VOIP-RTP
match ip dscp ef
match ip precedence 5
!
!
policy-map QOS-Policy
class VOIP-SIGNAL
priority percent 5
class VOIP-RTP
priority percent 70
class class-default
fair-queue
random-detect
policy-map PARENT
class class-default
shape average 1700000
service-policy QOS-Policy
!
!
!
!
!
interface GigabitEthernet0/0
description TO LAN
no ip address
ip virtual-reassembly
duplex auto
speed auto
!
interface GigabitEthernet0/0.10
description INTERFAZ DE DATOS
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
ip nat inside
ip virtual-reassembly
!
interface GigabitEthernet0/0.20
description INTERFAZ DE VOZ
encapsulation dot1Q 20
ip address 10.10.20.1 255.255.255.0
h323-gateway voip interface
h323-gateway voip h323-id GW-VOSS
h323-gateway voip bind srcaddr 10.10.20.1
!
interface GigabitEthernet0/0.30
description SIP
encapsulation dot1Q 30
ip address 10.10.30.1 255.255.255.0
ip access-group securevoipsip in
no cdp enable
!
!
interface GigabitEthernet0/0.90
description INTERFAZ DE ADMIN
encapsulation dot1Q 90 native
ip address 10.10.90.1 255.255.255.0
ip nat inside
ip virtual-reassembly
!
interface GigabitEthernet0/1
ip address dhcp
ip flow ingress
ip nat outside
no ip virtual-reassembly
duplex auto
speed auto
no cdp enable
!
ip local pool ippool 10.10.254.10 10.10.254.12
ip forward-protocol nd
ip route 64.132.31.27 255.255.255.255 64.132.31.26
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
!
ip nat inside source static tcp 10.10.10.11 80 interface GigabitEthernet0/1 8080
ip nat inside source list 103 interface GigabitEthernet0/1 overload
ip nat inside source static tcp 10.10.10.21 3306 interface GigabitEthernet0/1 3306
!
!
!
route-map To-Prodigy permit 10
match ip address 23
set ip default next-hop 200.38.193.226
!
!
!
control-plane
!
!
!
call filter match-list 13 voice
outgoing dialpeer 3003
!
voice-port 0/0/0:0
description TRONCALES DIGITALES TELMEX (444) 8340050
!
voice-port 0/1/0
supervisory disconnect dualtone mid-call
cptone MX
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx 5000
description TRONCAL ANALOGA (444) 1232306
caller-id enable
!
voice-port 0/1/1
supervisory disconnect dualtone mid-call
cptone MX
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx 5000
description TRONCAL ANALOGA 444 1232305
caller-id enable
!
voice-port 0/2/0
!
voice-port 0/2/1
!
!
mgcp fax t38 ecm
!
sccp local GigabitEthernet0/0.20
sccp ccm 10.10.20.2 identifier 1 version 7.0
sccp ip precedence 3
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register XCODER
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 15
associate application SCCP
!
!
dial-peer voice 1000 pots
description TDM-IN
incoming called-number .
direct-inward-dial
port 0/0/0:0
!
dial-peer voice 2000 voip
description MARCACION A CALL MANAGER PUBLISHER
destination-pattern [1-8]...
voice-class codec 1
voice-class h323 1
session target ipv4:10.10.20.2
incoming called-number .
dtmf-relay cisco-rtp rtp-nte h245-alphanumeric h245-signal
fax-relay ecm disable
fax rate 9600
no vad
!
dial-peer voice 9000 pots
description LLAMADAS LOCALES SLP A PSTN DIGITAL PTO 0/0/0:0
preference 2
destination-pattern #9[1-9]......
port 0/0/0:0
forward-digits 7
!
dial-peer voice 3000 voip
description LLAMADAS EU-CANADA POR SIP
destination-pattern 001.............
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:64.132.31.26
dtmf-relay cisco-rtp rtp-nte
clid network-number 55110713
!
dial-peer voice 3001 voip
description LLAMADAS INTERNACIONALES POR SIP
destination-pattern 00T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:64.132.31.26
dtmf-relay cisco-rtp rtp-nte
clid network-number 55110713
!
dial-peer voice 3002 voip
description LLAMADAS 01800 POR SIP
destination-pattern 01800.......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:64.132.31.26
dtmf-relay cisco-rtp rtp-nte
clid network-number 55110713
!
....
It's the first time I must configure voice gateways that have SIP trunks to the PSTN, if you need any other information they do get.
Thank you for your support.
regards
10-16-2012 09:56 AM
Can you try this..
Configure only dtmf relay rtp-nte on your sip dial-peer
eg..
dial-peer voice 3002 voip
dtmf-relay rtp-nte
then configure the following on your h323 dial-peer (make sure you configure this on all the h323 dial-peer to cucm)
Also ensure you remove the old dtmf config.
dial-peer voice 2000 voip
dtmf-relay rtp-nte digit-drop sip-kpml
Do a test call and send the ff debugs if it doesnt work
debug ccsip messages
debug voip rtp session named-event
Please rate all useful posts
"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson
10-17-2012 02:11 AM
I think he should try "dtmf-relay rtp-nte" on both the incoming (h323 dial peer) and outgoing (sip dialpeer) call legs.
please remove "dtmf-relay cisco-rtp rtp-nte h245-alphanumeric h245-signal" from the dial-peer voice 2000 voip and configure only rtp-nte with the command "dtmf-relay rtp-nte".
In dial-peer voice 3002 voip also, remove "dtmf-relay cisco-rtp rtp-nte" and add only "dtmf-relay rtp-nte".
if that doesn't work, as aokanlawon said, collect debug ccsip messages & debug voip rtp session named-event.
please provide, calling party number, called party number and time of call.
thanks
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide