DTMF problem on CME router

[kiranoddiraju]

Guys,

I have a 2811 CME gateway connect to the Telco via SIP. I have no problem with inbound and outbound calls, they work fine but I have problem with DTMF. The Telco says they are not receiving any digits from our side. I have the below dial-peer config on my side, please let me know what I need to change...

dial-peer voice 9 voip

description <<< Outbound to Telco >>>

translation-profile outgoing outgoing_digits

destination-pattern 9.T

voice-class codec 1

session protocol sipv2

session target ipv4:xxx.xxx.xxx.xxx

dtmf-relay sip-notify

no vad

!

AE-BAH-2811-01(config-dial-peer)#do sh ver

Cisco IOS Software, 2800 Software (C2800NM-SPSERVICESK9-M), Version 12.4(15)T8, RELEASE SOFTWARE (fc3)

Technical Support: http://www.cisco.com/techsupport

Copyright (c) 1986-2008 by Cisco Systems, Inc.

Compiled Mon 01-Dec-08 15:28 by prod_rel_team

ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)

AE-BAH-2811-01 uptime is 2 days, 19 hours, 19 minutes

System returned to ROM by Reload Command

System restarted at 19:53:18 AE Tue Jan 20 2009

System image file is "flash:c2800nm-spservicesk9-mz.124-15.T8.bin"

Cheers,

K

[paolo bevilacqua]

No DSP. You should use g.711 as I indicated before.

[Nicholas Matthews]

Hi Kiran,

Your voice class codec has g729 and g711 defined. The provider only advertises g711, so you are using g711 already for these calls.

If you think about this - without DSPs the router is not able to insert or change voice in the RTP packets. The SCCP IP phone is only going to send a Keypad message, and it doesn't send in-band information into the stream.

So, in order for in-band information to be inserted into the stream, you will need DSPs on the router to do this.

I would look into getting DSPs for this router, or more preferably, contacting your SIP provider and begging them to support RFC 2833 (rtp-nte).

If you have to transcode, it will be a pain because you will have to worry about your sessions, and troubleshooting DSPs if you ever have voice quality problems. RFC 2833 is the far more preferred option.

-nick

[kiranoddiraju]

OK! We have just tested with rtp-nte on both ends and still doesn't work...does it mean I am sending OOB?

[Nicholas Matthews · Accepted solution]

If SIP provider has changed, send the new messages log and we can confirm.

[kiranoddiraju]

hi nick,

SIP Trace attached...

[Nicholas Matthews]

Provider isn't sending RFC 2833, capability 101:

v=0

o=2Connect-MSC4 0 0 IN IP4 22.22.222.2

s=sip call

c=IN IP4 80.88.246.2

t=0 0

m=audio 50976 RTP/AVP 0

This is ours:

m=audio 18738 RTP/AVP 0 8 18 101

c=IN IP4 11.11.111.11

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

101 is DTMF

hth,

nick

[kiranoddiraju]

thanks nick & p.bevilacqua, I have sent the traces to the provider. I will update you guys when I hear from them.

Many thanks for your time.

[kiranoddiraju]

Thanks Nick, I have provided the SIP traces to the Telco and they have finally accepted that we are sending DTMF in RFC 2833 and they are not able to respond to those. They are now in contact with there vendor (NexTone).

BR,

K

[Nicholas Matthews]

You can try putting this on your dial peer (hidden command):

voice-class sip dtmf-relay force rtp-nte

This will send it no matter what your SIP provider says.

hth,

nick

[kiranoddiraju]

Hi Nick,

voice-class sip dtmf-relay force rtp-nte

The above command has solved my issue, dtmf is working fine... Great!!!

Thanks a million for your help!!!

Cheers,

Kiran

[paolo bevilacqua]

My rating to Nick for openly providing this great info.

[paolo bevilacqua]

Sorry, I could not see the rtpmap correctly. Definitely the cisco phones don't send dtmf in-band.

[A R M Mahbubul Haque]

Hey guys,

I am facing same issue in my cisco 2811 router. I use this command  "voice-class sip dtmf-relay force rtp-nte" but no luck. My configuration is attached.

Please review the configuration and tell me what to do ...???

[Jorge Armijo]

Capture:

deb ccsip messages

deb voip rtp session name-event (I think this is the correct sintax, I don't have a CLI close to me )

You should see the NSE events payload type 101 for the DTMF

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

[A R M Mahbubul Haque]

I can call through sip trunk and voice is very good. But problem is that when I call any other call center trrough SIP trunk & she ask for dial extention number and I dial my desired extention but then the call is terminated. I observed that my extention dialing is not recevied by the CME. when I dial 110 it shows twice value like as:

//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:

   Consume mask is not set. Relaying Digit 1 to dstCallId 0x3B1AD

//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:

   Check DTMF relay digit begin for 3way conf

//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:

   Consume mask is not set. Relaying Digit 1 to dstCallId 0x3B1AD

//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_end_for_3way_conference:

   Check DTMF relay digit end for 3way conf

//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:

   Consume mask is not set. Relaying Digit 1 to dstCallId 0x3B1AD

//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:

   Check DTMF relay digit begin for 3way conf

//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:

   Consume mask is not set. Relaying Digit 1 to dstCallId 0x3B1AD

//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_end_for_3way_conference:

   Check DTMF relay digit end for 3way conf

//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:

   Consume mask is not set. Relaying Digit 0 to dstCallId 0x3B1AD

//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:

   Check DTMF relay digit begin for 3way conf

//242092/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:

   Consume mask is not set. Relaying Digit 0 to dstCallId 0x3B1AD       

//242092/xxxxxxxxxxxx/CCAPI/cc_relay_digit_end_for_3way_conference:

   Check DTMF relay digit end for 3way conf

//-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

I attach the following debuging log.

#deb voice ccapi in

#deb voice ccapi ino

#deb voice ccapi inout

Please assist me. I am in stuck about this CME configuration.