Re: No DTMF on inbound SIP call from CUBE to CUCM6

sktlwaltt1 · ‎06-11-2009

Hello. I have a Cube Version 12.4(24)T, and I have DTMF tone problems.

Tones outbound to the service provider and work fine.

Tones inbound to the CUCM don't. I do not hear the tones in band (this is good). I have debugged the cube and see the rcf 2833 events signaling there are tones inbound from the provider to the cube but to from the cube to the CUCM there are no dtmf tones are not recognized.

I sniffed the inbound traffic to see RFC 2833 events to the cube, but unknown protocol packets passed to the CUCM.

I am testing the tones by forwarding my IP set to unity and trying to answer my VM.

Here is my config does anyone see any huge errors? Is this a transcoding issue. A dsp issue? or a IOS software issue? Config issue?

!

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

h323

emptycapability

no h225 h245-address facility

h225 id-passthru

h245 passthru tcsnonstd-passthru

sip

header-passing error-passthru

early-offer forced

midcall-signaling passthru

sip-profiles 1

!

voice class sip-profiles 1

request INVITE sip-header Supported remove

request INVITE sip-header Min-SE remove

request INVITE sip-header Session-Expires remove

request INVITE sip-header Unsupported modify "Unsupported:" "timer"

!

voice-card 0

dspfarm

dsp services dspfarm

sccp local FastEthernet0/0

sccp ccm 10.133.2.60 identifier 10 version 6.0

sccp

!

sccp ccm group 10

bind interface FastEthernet0/0

associate ccm 10 priority 1

associate profile 20 register xcodecube1

associate profile 10 register cube_mtp

!

dspfarm profile 20 transcode

description transcoder for CUCM6

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 16

associate application SCCP

!

dspfarm profile 10 mtp

codec g711ulaw

maximum sessions software 250

associate application SCCP

!

dial-peer voice 201 voip

destination-pattern 53..

signaling forward unconditional

session protocol sipv2

session target ipv4:10.133.2.60

dtmf-relay rtp-nte

codec g711ulaw

ip qos dscp cs5 media

ip qos dscp cs5 signaling

!

dial-peer voice 101 voip

destination-pattern .T

session protocol sipv2

session target ipv4:172.21.142.87

dtmf-relay rtp-nte

codec transparent

ip qos dscp cs5 media

ip qos dscp cs5 signaling

!

dial-peer voice 103 voip

signaling forward unconditional

session protocol sipv2

incoming called-number .

dtmf-relay rtp-nte

codec g711ulaw

ip qos dscp cs5 media

ip qos dscp cs5 signaling

Thanks for your help.

Nicholas Matthews · ‎06-11-2009

Your problem is related to incoming called number . on the 103 dial peer. Incoming called number . is the first check for an incoming dial peer, and will apply those settings to every incoming call, regardless of protocol.

Thus, you will have your SIP DTMF method applied to all incoming H323 calls as well.

I would check out this document:

http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml

And this post will explain more:

http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40^1%40%40.2cc1f0f6/2#selected_message

-nick

sktlwaltt1 · ‎06-11-2009

Nick I read the posts and reconfigured the Cube.

Here is my new dial peer config.

Still getting no dtmf to unity. Where does the dtmf-relay rtp-nte command need to go? inbound from the Sip provider or on the leg to the cucm, or both? Does CUCM only want to see dtmf-relay rtp-nte on the inbound SIP trunk?

dial-peer voice 1 voip

description INCOMMING SIP TRUNK

signaling forward unconditional

session protocol sipv2

incoming called-number 53..

dtmf-relay rtp-nte

codec g711ulaw

ip qos dscp cs5 media

ip qos dscp cs5 signaling

!

dial-peer voice 10 voip

description OUTGOING SIP TRUNK TO CUCM

destination-pattern 53..

signaling forward unconditional

session protocol sipv2

session target ipv4:10.133.2.60

dtmf-relay rtp-nte

codec g711ulaw

ip qos dscp cs5 media

ip qos dscp cs5 signaling

!

dial-peer voice 20 voip

description iNCOMMING FROM cucm OUTGOING TRUNK TO SIP PROVIDER

answer-address 3835

destination-pattern .T

session protocol sipv2

session target ipv4:172.21.142.87

dtmf-relay rtp-nte

codec transparent

ip qos dscp cs5 media

ip qos dscp cs5 signaling

Nicholas Matthews · ‎06-11-2009

I didn't notice initially that you were doing SIP-SIP.

In this case RTP-NTE is the preferred method. However, MTPs are required in CUCM for RFC 2833 to be generated under most circumstances. Try checking 'MTP Required' on the SIP trunk. As well, check the phone page to see if you can enable RFC 2833 there as well.

-nick

sktlwaltt1 · ‎06-12-2009

Nick

I have a MTP configured and have it checked on the Sip trunk. There is only a disable rfc2833 on the phone settings.

When i trace the call in CUCM i see the call come in and negoate the DTMF but when transfered to unity no luck.

Any other ideas.

Trent

Nicholas Matthews · ‎06-12-2009

Hi Trent,

I would 'debug voip rtp session named' and make sure you see it in the Rcv and Snd directions.

If you do a packet capture you can use the wireshark filter 'rtpevent' to see if they are there or not.

-nick

sktlwaltt1 · ‎06-12-2009

Nick i have debugged the voip trp session named and see the messages send and rcv.

Test 1: From a CME to the CUCM6 on a sip trunk via cube, could see messages forwarded to the endpoint and worked when forwarded to unity. payload type 101

Test 2:form sip provider to CUCM6 on sip trunk via cube, can see the messages getting here but not forwarding to the endpoint payload type 120.

I will attach a sniff of the test 2 call.

Thanks for you time.

Trent

parshah · ‎06-12-2009

Trent,

On the CUBE for the SIP dial-peer that point to CUCM, can you change the dtmf-relay from rtp-nte to sip-kpml.

On CUCM you need to uncheck the 'MTP required' checkbox and reset the trunk.

Try calling from PSTN to Unity and see if dtmf works.

Thanks,

Nicholas Matthews · ‎06-12-2009

Looks like you have a payload type mismatch.

You can try adding this on your dial peer pointing towards CUCM:

dial-peer voice x

rtp payload nte 120

sktlwaltt1 · ‎06-17-2009

Hello. Update.

Turns out the provider was giving me bad information. They said they were sending rfc 2833 but really sending RAW tones.

So I configured a transcoder on the CUBE and started a telephoney service and assocated the transcoder to the CUBE.

I also still have a MTP registered to the CUCM for the SIP trunks.

When i have both MTP to the CUCM and the Transcoder to the CUBE online i have no audio passing from the CUCM via cube to Provider.

When i shut down the transcoder on the cube i can get audio passthrough but no dtmf is being transcoded.

I'm about ready to give up on sip....

Any help would be appreciated.

Trent

Nicholas Matthews · ‎06-17-2009

Hi Trent,

It's hard to speculate what the problem could be there. If possible, you could try getting rid of the MTP completely.

For troubleshooting purposes, you can also:

-Try moving the MTP to the gateway using software or hardware resources

-Reset MTPs, SCCP, trunks, Callmanager service

-Upgrade IOS on CUBE to the latest

-Remove MTP Required from SIP trunk

It sounds like a bug of some sort, or improper use of features. Try to trim this down to as few features as possible on your CUBE.

-nick