For H323
debug h245 asn1
For SIP
debug voip rtp session named-event
Sample output
H323
Aug 21 21:07:37.902: h245_decode_one_pdu: more_pdus = 0, bytesLeftToDecode = 7
Aug 21 21:07:37.902: H245 MSC INCOMING ENCODE BUFFER::= 6D810447200063
Aug 21 21:07:37.902:
Aug 21 21:07:37.902: H245 MSC INCOMING PDU ::=
value MultimediaSystemControlMessage ::= indication : userInput : signal :
{
signalType “9″
duration 100
}
SIP
Aug 21 21:07:37.902: Pt:101 Evt:9 Pkt:04 00 00 <Snd>>>
Aug 21 21:07:37.902: Pt:101 Evt:9 Pkt:04 00 00 <Snd>>>
Aug 21 21:07:37.902: Pt:101 Evt:9 Pkt:04 00 00 <Snd>>>
Aug 21 21:07:37.902: Pt:101 Evt:9 Pkt:04 01 90 <Snd>>>
Aug 21 21:07:37.902: Pt:101 Evt:9 Pkt:84 03 20 <Snd>>>
Aug 21 21:07:37.902: Pt:101 Evt:9 Pkt:84 03 20 <Snd>>>
Aug 21 21:07:37.902: Pt:101 Evt:9 Pkt:84 03 20 <Snd>>>
command “debug voip rtp session named-event” doesn’t support Cisco proprietary sip-notify DTMF relay. I have tried
VOIP RTP Session Generic Events
VOIP RTP ALL Named Events
VOIP RTP DTMF-RELAY Events
All of these don’t work.
Dear All!
If anyone can help me with this issue, I would be very thankful.
This is the scenerio:
PSTN Phone->A SoftSwitch->Cisco AS5300->ISDN Switch(PBX) where user is asked to enter his account/pin number which should DTMF tones to authenticate him.
What happens is ISDN switch can recogniz only first digit entered. Later digits are not received.
In order to figure out the problem I performed following test:
I let the user callthought ISDN switch without authentication. The call is connected in following manner
Phone->A SoftSwitch->Cisco AS5300->ISDN Switch=>Phone2
Now in order to test the DTMF Pass-Thru. I press some digits on Phone1 and hear an on-going tone on phone2.This on-going tone(peeeeeeeeeeeeeeeep) , later digit presses are not heard on phone2. This on-going tone gets lost only if i hang-up.
On the other hand if i perform the same process in reverse order i.e. from phone2 to phone1 things work just fine. i hear multiple tones, each time i press a new digit.
The ios version which I am using is the latest available for this model as5300 c5300-is-mz.123-24a.bin which does include DTMF Feature supports over SIP.
I correct configuration as far as i know. I have searched the web for 2 days and all references told me to use this config. Here it is:
dial-peer voice 49 voip
incoming called-number 49T
voice-class codec 1
session protocol sipv2
session transport udp
dtmf-relay rtp-nte
dial-peer voice 491 pots
destination-pattern 49T
progress_ind setup enable 3
progress_ind alert enable 8
translate-outgoing called 1
direct-inward-dial
port 1:D
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711ulaw
codec preference 4 g711alaw
codec preference 5 g723r63
codec preference 6 g723r53
codec preference 7 gsmfr
codec preference 8 gsmefr
the output from debug voip rtp session named-event is as under:
Jan 24 13:50:30.655: << Pt:101 Evt:1 Pkt:0A 00 A0
Jan 24 13:50:30.683: << Pt:101 Evt:1 Pkt:0A 00 A0
Jan 24 13:50:30.683: << Pt:101 Evt:1 Pkt:0A 00 A0
Jan 24 13:50:30.683: << Pt:101 Evt:1 Pkt:0A 01 40
Jan 24 13:50:30.727: << Pt:101 Evt:1 Pkt:0A 01 E0
Jan 24 13:50:30.755: << Pt:101 Evt:1 Pkt:0A 02 80
Jan 24 13:50:30.763
wantel-1#term: << Pt:101 Evt:1 Pkt:8A 03 20
Jan 24 13:50:30.763: << Pt:101 Evt:1 Pkt:8A 03 20
Jan 24 13:50:30.763: << Pt:101 Evt:1 Pkt:8A 03 20
Jan 24 13:50:31.235: << Pt:101 Evt:2 Pkt:0A 00 A0
Jan 24 13:50:31.247: << Pt:101 Evt:2 Pkt:0A 00 A0
Jan 24 13:50:31.247: << Pt:101 Evt:2 Pkt:0A 00 A0
Jan 24 13:50:31.247: << Pt:101 Evt:2 Pkt:0A 01 40
Jan 24 13:50:31.267: << Pt:101 Evt:2 Pkt:0A 01 E0
Jan 24 13:50:31.283: << Pt:101 Evt:2 Pkt:0A 02 80
Jan 24 13:50:31.315: << Pt:101 Evt:2 Pkt:8A 03 20
Jan 24 13:50:31.343: << Pt:101 Evt:2 Pkt:8A 03 20
Jan 24 13:50:31.347: << Pt:101 Evt:2 Pkt:8A 03 20
Jan 24 13:50:32.135: << Pt:101 Evt:5 Pkt:0A 00 A0
Jan 24 13:50:32.135: << Pt:101 Evt:5 Pkt:0A 00 A0
Jan 24 13:50:32.155: << Pt:101 Evt:5 Pkt:0A 00 A0
Jan 24 13:50:32.155: << Pt:101 Evt:5 Pkt:0A 01 40
Jan 24 13:50:32.155: << Pt:101 Evt:5 Pkt:0A 01 E0
Jan 24 13:50:32.155: << Pt:101 Evt:5 Pkt:8A 02 80
Jan 24 13:50:32.227: << Pt:101 Evt:5 Pkt:8A 02 80
Jan 24 13:50:32.315: << Pt:101 Evt:5 Pkt:8A 02 80
Jan 24 13:50:32.659: << Pt:101 Evt:4 Pkt:0A 00 A0
Jan 24 13:50:32.659: << Pt:101 Evt:4 Pkt:0A 00 A0
Jan 24 13:50:32.659: << Pt:101 Evt:4 Pkt:0A 00 A0
Jan 24 13:50:32.659: << Pt:101 Evt:4 Pkt:0A 01 40
Jan 24 13:50:32.659
wantel-1#term: << Pt:101 Evt:4 Pkt:0A 01 E0
Jan 24 13:50:32.715: << Pt:101 Evt:4 Pkt:8A 02 80
Jan 24 13:50:32.715: << Pt:101 Evt:4 Pkt:8A 02 80
Jan 24 13:50:32.715: << Pt:101 Evt:4 Pkt:8A 02 80
Jan 24 13:50:33.355: << Pt:101 Evt:7 Pkt:0A 00 A0
Jan 24 13:50:33.355: << Pt:101 Evt:7 Pkt:0A 00 A0
Jan 24 13:50:33.355: << Pt:101 Evt:7 Pkt:0A 00 A0
Jan 24 13:50:33.355: << Pt:101 Evt:7 Pkt:0A 01 40
Jan 24 13:50:33.375: << Pt:101 Evt:7 Pkt:0A 01 E0
Jan 24 13:50:33.375: << Pt:101 Evt:7 Pkt:8A 02 80
Jan 24 13:50:33.415: << Pt:101 Evt:7 Pkt:8A 02 80
Jan 24 13:50:33.415: << Pt:101 Evt:7 Pkt:8A 02 80
Please advise.
Hi Mazhar,
I have the same issue with SIP, h323 is fine, what version of dspware / vcware are your VFC(s) running? It will tell you if the version is lower than required, eg:
vgw01#show vfc 2 version dspware
DSPWare version in VFC slot 2 is 3.4.51L
Firmware version mismatch for bundle AS5300 VCWare
– version found (7.40) is lower than minimum required (11.41)
Firmware version mismatch for bundle AS5300 C542
– version found (3.4.51L) is lower than minimum required (4.1.41)
vgw01#show vfc 2 version vcware
Voice Feature Card in Slot 2:
VCWare Version : 7.40
ROM Monitor Version: 1.3
DSPWare Version : 3.4.51L
Technology : C542
Firmware version mismatch for bundle AS5300 VCWare
– version found (7.40) is lower than minimum required (11.41)
Firmware version mismatch for bundle AS5300 C542
– version found (3.4.51L) is lower than minimum required (4.1.41)
Will upgrade mine at some point but they’re production boxes so I can’t do it during the day. Let us know if you come up with anything and i’ll do the same. I am running IOS 12.3(15a).
Regards,
Jono
[...] Copied from CCIE Voice [...]
How to configure…
In the VoIP dial-peers set:
dtmf-relay rtp-nte
This will use payload type 101 for the DTMF tones. If you want to redefine it, use “rtp payload-type” command in the dial-peer that requires this change.
There is a doc on the Cisco web site that covers these commands in details: “Dual Tone Multifrequency Relay for SIP calls Using Named Telephone Events” (http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guide09186a0080087edb.html#58571)