[cisco-voip] DTMF Issue with one external number

Wed Oct 17 02:16:34 EDT 2012

When an inbound call comes in requiring DTMF (to voicemail or something like that), what does that look like?

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From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of george.hendrix at l-3com.com
Sent: Wednesday, October 17, 2012 3:18 AM
To: Derek Wyss
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] DTMF Issue with one external number

The RTP NTE negotiated is indeed 101.  However, when I enter the commands to set it to 100, I get the following on the debug.

Preferred Codec        : g711ulaw, bytes :160
        Preferred  DTMF relay  : rtp-nte
        Preferred NTE payload  : 100
        Early Media            : No
        Delayed Media          : Yes
        Bridge Done            : No
        New Media              : No
        DSP DNLD Reqd          : No

Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : -1
          Negotiated Codec       : g711ulaw, bytes :160
          Nego. Codec payload    : 0 (tx), 0 (rx)
          Negotiated DTMF relay  : inband-voice
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload  : 0

Bill Hendrix  |  Network/VOIP Engineer
L3 STRATIS  POWERED BY EXCELLENCE

From: Derek Wyss [mailto:wyss34 at gmail.com]<mailto:[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<mailto: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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