Re: DTMF issue on FXS port

sdwitner1 · ‎10-13-2010

I have a Cisco 2811 that is terminating SIP trunks from my provider. From there I have FXS ports on the router that connect to a PBX. When caller call in and get the voicemail gretting they have the option to press 9 to leave a message. when they press 9 the digit is not being recongnized when coming from the VoIP network.

dksingh · ‎10-13-2010

Hi there,

would need to see the dtmf relay being negotiated and digit 9 coming in..

u probably want to run following debugs and check....:

deb ccsip mess

deb voip ccapi inout

deb vpm sig

ADAM CRISP · ‎10-13-2010

yeah,

Most SP's will want to send you DTMF information 'out of band' eg using SIP INFO, SIP NOTIFY or RTP-NTE (technically inband but as good as)

RTP-NTE is very common, so make sure you have this one enabled in your incoming dial peer

eg

dial-peer voice 999

incoming called number

dtmf-relay rtp-nte !etc

!

To check on whether your SP is using RTP-NTE look at an incoming SIP INVITE messge

here's an example: - look for telephone-event

UC520#debug ccsip messages
SIP Call messages tracing is enabled

UC520#term mon
UC520#
Oct 13 15:30:24.945: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:AAA1869222509@10.149.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.199.1.2:5060;branch=z9hG4bK-sYn-0-5d5c38313b18
Max-Forwards: 4
Contact: <10.199.1.2:5060>
From: <>01869222500@voip.co.uk;user=dialstring>;tag=5e5c6344fa
To: <>01869222509@voip.co.uk;user=dialstring>
Call-ID: cc52219c-d6de-11df-91e7-4726eb401cbd
CSeq: 677119416 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL
Expires: 180
Content-Type: application/sdp
Content-Length: 552

v=0
o=CiscoSystemsSIP-GW-UserAgent 1646 6861 IN IP4 10.200.254.1
s=SIP Call
c=IN IP4 10.199.1.2
t=0 0
m=audio 40402 RTP/AVP 8 0 18 4 98 99 2 102 3 125 121 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:98 G726-16/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:102 GSM-EFR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:125 X-CCD/8000
a=rtpmap:121 frf-dialed-digit/8000
a=fmtp:121 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Your router will eventually send a 200 OK, with the negotiated media terms - and hopefully include some commonality

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.199.1.2:5060;branch=z9hG4bK-sYn-0-5d5c38313b18
From: <>01869222500@voip.co.uk;user=dialstring>;tag=5e5c6344fa
To: <>01869222509@voip.co.uk;user=dialstring>;tag=3CCFE0-6EF
Date: Wed, 13 Oct 2010 15:30:24 GMT
Call-ID: cc52219c-d6de-11df-91e7-4726eb401cbd
CSeq: 677119416 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "Cisco cfiveohfourgee" <2002>;party=called;screen=no;privacy=off
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 229

v=0
o=CiscoSystemsSIP-GW-UserAgent 8460 2059 IN IP4 10.149.1.7
s=SIP Call
c=IN IP4 10.149.1.7
t=0 0
m=audio 18218 RTP/AVP 0 101
c=IN IP4 10.149.1.7
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Adam

sdwitner1 · ‎10-13-2010

dial-peer voice 9999999 voip

voice-class codec 1

session protocol sipv2

incoming called-number .

dtmf-relay rtp-nte

no vad

ADAM CRISP · ‎10-13-2010

Well OK,

Can you debug ccsip messages, place a call into your PBX and press dome digits, and then paste the SIp trace?

thanks

sdwitner1 · ‎10-13-2010

I will do the debugs tonight and post the results. Also I have had to add the following commands to dial peers in the past in order to get DTMF to be recognized by voicemail systems...

rtp payload-type nse 105

rtp payload-type nte 100

dksingh · ‎10-13-2010

So it seems your SIP provider is using 100 for RTP NTE.

IOS uses 101 by default and reserves 100 for NSE.

Run deb ccsip mess to confirm and then add those

two commands accordingly to negotiate the right payload

type and it shd work.

ADAM CRISP · ‎10-13-2010

Yeah that's OK. We can deal with that when you get your SIP capture.

We're going to look at the SDP within the Invite, Your 200 OK and also to see whether there are any other SIP message types present when you keys.

cheers

sdwitner1 · ‎10-14-2010

I was unable to duplicate the issue last night. As of right now everything is working without any config changes made. Thanks everyone for the help and pointing me in a good direction for troubleshooting.