cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
2461
Views
0
Helpful
9
Replies

CUCM -> Flowroute outbound calls working but dial tones are not regonized

Signups
Level 1
Level 1

Hi, I've noticed a strange issue with my CUCM deployment.


Its a fairly simple setup with a cisco 2901 as the CUBE connected to flowroute with their default setting and the Max Audio Bit Rate is set as

64 kbps (G.722, G.711)

 

However when I dial multiple known working numbers that use dial tone recognition (banks, phone meetings etc) the dial tones are ignored by these systems.

 

Ive tried on multiple devices with the same result (IPCommunicator, IP Phones and even xlite).

 

Can anyone suggest what may be causing this to happen ?

 

 

 

1 Accepted Solution

Accepted Solutions

R0g22
Cisco Employee
Cisco Employee
Add the following -

conf t
voice class uri IN sip
host ipv4:X.X.X.X >> IP of CUCM. 1 node
host ipv4:Y.Y.Y.Y >> IP of CUCM. 2 node

dial-peer voice 100 voip
description CUCM INBOUND
session protocol sipv2
incoming uri via IN
voice-class sip bind control source-interface <LAN interface>
voice-class sip bind media source-interface <LAN interface>
dtmf-relay sip-kpml rtp-nte
codec g711ulaw
no vad

Add the voice-class binds on your flowroute facing DP's. The interface would be your external/ITSP facing interface.

View solution in original post

9 Replies 9

Georgios Fotiadis
VIP Alumni
VIP Alumni

This is a DTMF negotiation issue. What is the dtmf-relay configured under your dial-peers (incoming from eg. CUCM and outgoing to Flowroute)?

Georgios
Please rate if you find this helpful.

R0g22
Cisco Employee
Cisco Employee
For starters, add "dtmf-relay sip-kpml rtp-nte" to your incoming and outgoing dial-peers and then test.

Yes I had dtmf-relay rtp-nte ive run

no dtmf-relay
followed by dtmf-relay sip-kpml rtp-nte

Saved and reloaded the cube but the issue remains on every dial tone regonition system i call.

 

Here are my two dial peers

 

dial-peer voice 101 voip
 description *** Outgoing Flowroute ***
 translation-profile outgoing addlocal
 destination-pattern 1[2-9]..[2-9]......
 session protocol sipv2
 session target dns:sip.flowroute.com
 voice-class codec 1  
 voice-class sip asserted-id pai
 voice-class sip profiles 1
 media-class 3
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 102 voip
 description *** Outgoing International ***
 destination-pattern 011T
 session protocol sipv2
 session target dns:sip.flowroute.com
 voice-class codec 1  
 no voice-class sip asserted-id
 voice-class sip profiles 1
 media-class 3
 dtmf-relay rtp-nte
 no vad

R0g22
Cisco Employee
Cisco Employee
Enable the following debugs -

debug ccsip message
debug ccsip error
debug voice ccapi inout
debug voip rtp session named-event

Reproduce the issue and copy the logs in a text file.

Ok here are the results. Again no matter how many numbers I pressed I could not get it to read a single one. The remote system acts as if nothing but silence was played back

R0g22
Cisco Employee
Cisco Employee
IOS is selected DP=0 due to which dtmf negotiation is failing -
Mar 9 19:59:52.072: //-1/6CC2F1800000/CCAPI/cc_api_call_setup_ind_common:
Interface=0x41C17004, Call Info(
Calling Number=4206969696,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=17127704160(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=41

Attach a "show run" from the CUBE please.

Sure it is also hosting 4 flexnet access points on the POE module hence the trucking and there is also a FXO card setup

 

 

R0g22
Cisco Employee
Cisco Employee
Add the following -

conf t
voice class uri IN sip
host ipv4:X.X.X.X >> IP of CUCM. 1 node
host ipv4:Y.Y.Y.Y >> IP of CUCM. 2 node

dial-peer voice 100 voip
description CUCM INBOUND
session protocol sipv2
incoming uri via IN
voice-class sip bind control source-interface <LAN interface>
voice-class sip bind media source-interface <LAN interface>
dtmf-relay sip-kpml rtp-nte
codec g711ulaw
no vad

Add the voice-class binds on your flowroute facing DP's. The interface would be your external/ITSP facing interface.

Perfect thanks a million that fixed it. I hope this helps someone else too.