[cisco-voip] DTMF Issue with one external number

Tue Oct 16 09:49:03 EDT 2012

Thanks for the info.  I will do a debug on this and see what I get.  Also, even on the numbers I call that work, it appears the router is sending double digits.  If you look below, it sent * twice even though I only pressed it once.

Oct 16 07:36:35.343: //137015/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit * to dstCallId 0x21738
Oct 16 07:36:35.343: //137015/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
Oct 16 07:36:35.419: //137015/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end
   Consume mask is not set. Relaying Digit * to dstCallId 0x21738
Oct 16 07:36:35.419: //137015/xxxxxxxxxxxx/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf

-Bill

From: Derek Wyss [mailto:wyss34 at gmail.com]
Sent: Tuesday, October 16, 2012 8:40 AM
To: Hendrix, George (Bill) @ NSS - STRATIS
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] DTMF Issue with one external number

What does your SDP show using debug ccsip all?

I've ran into this before where the provider had a different RTP map to their IP customers vs their ISDN customers.  What I had to do was create separate dialpeers for those numbers with a different RTP map.  See example below:
v=0
o=CiscoSystemsSIP-GW-UserAgent 9601 2828 IN IP4 X.X.X.X
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=audio 16384 RTP/AVP 18 0 101
c=IN IP4 10.8.2.4
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:101 X-NSE/8000
a=fmtp:101 192-194

v=0
o=Sonus_UAC 16372 5325 IN IP4 X.X.X.X
s=SIP Media Capabilities
c=IN IP4 X.X.X.X
t=0 0
m=audio 23002 RTP/AVP 18 0 100
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20

As mentioned above the yellow highlighted portion of the SDP's is where we can see a mismatch in our payload type.  You can see the telco is sending 100 and we are sending 101.  To resolve this issue you have to remap the rtp payload type for signaling and telephony events.  Here is the commands I had to run to send 100 as our NTE to match what the provider was expecting...

Lincoln-VG01(config-dial-peer)#rtp payload-type nse 98
Lincoln-VG01(config-dial-peer)#rtp payload-type nte 100
Lincoln-VG01(config-dial-peer)#rtp payload-type nse 101

Because the signaling is defaulted at nte 101 and nse 100 you have to remove 100 from nse by assigning it a random unused value before you can assign 100 to nte.  See this image for the default reserved values: https://communities.cisco.com/servlet/JiveServlet/downloadImage/2-5295-2243/450-185/defaultpayloadtype.png

Hope this helps,

Derek

On Tue, Oct 16, 2012 at 7:25 AM, <george.hendrix at l-3com.com<mailto:george.hendrix at l-3com.com>> wrote:
Hey guys,

  We have an odd issue going on with DTMF.  Below is the path to the PSTN.

CUCM <-> h.323 Gateway <-> SIP Provider.

The problem is that when we dial into this one external number and press 1 to select option 1, it doesn't seem to accept the digit.  If we dial into that same number from anywhere else, it works fine.  Having said that, I can dial into other numbers from the system and have no issue at all with dtmf.  Any ideas of what this issue is?

Thanks,
Bill

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