cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1948
Views
10
Helpful
7
Replies

DTMF issue in outgoing call over SIP Trunk

hassanalirazi
Level 1
Level 1

Hi Guys,

I am facing an issue in which DTMF tones are not being forwarded when calling outbound inbound is working fine.

I have an Alcatel PBX connected to a h323 gateway and then forwarded to the h323 gateway with the SIP trunk.

The PBX is sending the dtmf using h245 signal and is being recieved by our gateway. The SIP gateway on the other hand seems to be getting the digits but the events seem to be wrong.

this is the dial-peer config on the client gateway and the h245 asn1 debugs from the client gateway

dial-peer voice 9 voip

translation-profile outgoing ANI

destination-pattern 00T

session target ipv4:10.64.120.11

dtmf-relay rtp-nte

no vad

Feb 22 06:38:12.296: H245 MSC INCOMING PDU ::=

value MultimediaSystemControlMessage ::= indication : userInput : signal :
    {
      signalType "0"
      duration 60
      rtp
      {
        logicalChannelNumber 1
      }
    }

Feb 22 06:38:13.008: H245 MSC INCOMING ENCODE BUFFER::= 6D810766C0003B000000
Feb 22 06:38:13.008:
Feb 22 06:38:13.008: H245 MSC INCOMING PDU ::=

value MultimediaSystemControlMessage ::= indication : userInput : signal :
    {
      signalType "6"
      duration 60
      rtp
      {
        logicalChannelNumber 1
      }
    }

Help Please thanks

1 Accepted Solution

Accepted Solutions

Hassan...From your config the call flow does not  look correct..

You have customer gateway, and sip gateway

Please describe the call flow correctly...There are two gateways here

Alcatel---h233--CUBE(CE)--------inbound dp=h323----------->sipgw--------outbounddp=sip----------->ITSP

You have only showed the outbound sip dial-peer. There must be an inbound dial-peer from the CE router to the SIP gateway.

From the logs..

The inbound dial-peer on CE router is dial-peer voice 2 voip. Do you have dtmf-relay h245 on that?

On the SIPGW

inbound dial-peer=1 (do you have dtmf relay h245 on this?

outbound dialpeer=100

I can also see that you pressed digit 29014( this were sent to the ITSP)

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

View solution in original post

7 Replies 7

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

Hi,

First of all you need to draw your call flow for us..So we can get a better picture..eg

alcatel--h323----cube---sip---ITSP  is this correct..if it is not then draw up the correct flow

Second the config on your dial-peer is wrong. You cant use dtmf-relay rtp-nte on a h323 dial-peer...

Here are my suggestions...

Configure your inbound h323 dial-peer from alcaltel with dtmf-relay h245

!#inbound dial-peer

dial-peer voice xxx voip

dtmf-relay h245-alphanumeric

and configure outbound sip dial-peer to sip trunk to use digit-drop

//outbound to ITSP SIP dial-peer

!#NEW CONFIG

dial-peer voice yyy voip

dtmf-relay rtp-nte digit-drop

if it doesnt work...

please send us

debug voip rtp session named-event

debug ccsip messages

debug voip ccapi inout

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

Hi,

Yes the Flow is as above. I have tried it using the digit drop command but it didnt work

The Dial-peer on the SIP router is

dial-peer voice 100 voip

translation-profile outgoing ANI

destination-pattern 00441743742310

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte digit-drop

no vad

dial-peer on the customer gateway

dial-peer voice 9 voip
translation-profile outgoing ANI
destination-pattern 00T
session target ipv4:10.64.120.11
dtmf-relay h245-alphanumeric
no vad

dial-peer voice 1 voip
destination-pattern 2310
session target ipv4:10.116.32.12
dtmf-relay h245-alphanumeric
no vad

Thanks

Hassan...From your config the call flow does not  look correct..

You have customer gateway, and sip gateway

Please describe the call flow correctly...There are two gateways here

Alcatel---h233--CUBE(CE)--------inbound dp=h323----------->sipgw--------outbounddp=sip----------->ITSP

You have only showed the outbound sip dial-peer. There must be an inbound dial-peer from the CE router to the SIP gateway.

From the logs..

The inbound dial-peer on CE router is dial-peer voice 2 voip. Do you have dtmf-relay h245 on that?

On the SIPGW

inbound dial-peer=1 (do you have dtmf relay h245 on this?

outbound dialpeer=100

I can also see that you pressed digit 29014( this were sent to the ITSP)

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

HI,

yes there is a dial-peer 2 on the CE router and it was missing the h245 command do you think that could be the issue?

the dial-peer now is

dial-peer voice 2 voip

incoming called-number .

dtmf-relay h245-alphanumeric

no vad ial-peer voice 2 voip
incoming called-number .
dtmf-relay h245-alphanumeric
no vad

The inbound dial-peer is as such on the SIP router

dial-peer voice 1 voip
translation-profile incoming SIP-INBOUND

media flow-around

signaling update from gtd  calling

session protocol sipv2

session target sip-server

incoming called-number .

voice-class codec 1 

dtmf-relay rtp-nte sip-kpml

no vad

Thanks

Yes that can be an issue. If your inbound dial-peer on the sip gateway is a sip dial-peer then your config is wrong

Your outbound dial-peer from CE router is a h323 dial-peer, your inbound dial-peer that receives that call has to be a h323 dial-peer....

So your dial-peer voice 1 voip need to be changed to h323 dial-peer..You should then remove dtmf-relay rtp-nte sip-kpml and add only h245-lapha

May I ask why you are not doing a complete sip integration...

If your pbx does not support sip..

You can do inbound h323 from pbx to h323 and then outboud from CE router to SIPGW to be sip, then inbound to SIPGW to be sip and outbound to ITSP to be sip

Alcatel--inboundp=h323------->CERouter----outboundp=sip--------->SIPGW--------inbounddp=sip---SIPGW-----outbounddp=sip---------------->ITSP

In this call flow, you will then configure dtmf relay rtp-nte digit drop on sip dial-peers and h245 only on the inbound dp from alcatel to CE router

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

Hi,

Its a mixed sort of a network and thats how it got handed over. The issue got resolved i set the outgoing dial-peer to rtp-nte as our SP was accepting that and the incoming dial-peer to h245.

Thanks for all your support i learned a lot from this discussion thank you

Regards

Hassan

Glad I could help. But you have not rated any of my posts. You also have not marked the thread as answered. it is useful for others who have similar problems. It is also good to encourage people like me

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts
Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: