DTMF FROM PSTN TO A SIP TRUNK

[zacariasnhancune007]

I dear im facing a big issue here.

I have a CCME 7.1 running IOS c2900-universalk9-mz.SPA.150-1.M6.bin. This CCME is connected to PSTN using BRI interfaces for outbound and inbound public phone calls. The company also needs to make some calls to a SIP server ( a kind of IVR) over a sip trunk that is beyond the wan VSAT link.

When I make the call using some SCCP and SIP phones registered on the CCME, I can send DTMF to the IVR system but when the call comes from PSTN via BRI the call gets connected but i cant send DTMF to the IVR system.

On the first attempt neither phones (PSTN or internal) were able to call this IVR but  I had to configure software MTP and the internal phones could call the IVR system over WAN VSAT link.

This is the configuration of the CCME. Check also the attachment

sh running-config

Building configuration...

network-clock-participate wic 0

network-clock-participate wic 1

isdn switch-type basic-net3

!

!

trunk group AAA_MOBILE_BRI

hunt-scheme longest-idle

!

voice-card 0

dspfarm

dsp services dspfarm

!

!

voice call send-alert

voice rtp send-recv

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none

h323

  no telephony-service ccm-compatible

sip

  registrar server

!

voice class codec 2

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

!

!

voice register global

mode cme

source-address 172.16.X.250 port 5060

timeouts interdigit 4

max-dn 40

max-pool 20

timezone 29

time-format 24

date-format D/M/Y

tftp-path flash:

create profile sync 000342051580142A

!

voice register dn  6

number 255

name 255

label 255

!

!

voice register pool  6

id mac 0000.0000.0005

number 1 dn 6

dtmf-relay rtp-nte

username 255 password 255

codec g711ulaw

!

voice hunt-group 1 sequential

final 3000

list 254,253,252,251,250

pilot 3000

!

!

interface GigabitEthernet0/0

description LINK-TO-SW-AAA-SEDE-01

ip address 172.16.X.250 255.255.255.0

duplex full

speed 1000

h323-gateway voip interface

h323-gateway voip bind srcaddr 172.16.X.250

!

interface BRI0/0/0

no ip address

isdn switch-type basic-net3

isdn point-to-point-setup

isdn incoming-voice voice

isdn static-tei 0

trunk-group AAA_MOBILE_BRI

!

interface BRI0/0/1

no ip address

isdn switch-type basic-net3

isdn point-to-point-setup

isdn incoming-voice voice

isdn static-tei 0

!

interface BRI0/1/0

no ip address

isdn switch-type basic-net3

isdn point-to-point-setup

isdn incoming-voice voice

isdn static-tei 0

!

interface BRI0/1/1

no ip address

isdn switch-type basic-net3

isdn point-to-point-setup

isdn incoming-voice voice

isdn static-tei 0

!

ip route 0.0.0.0 0.0.0.0 172.16.X.254

!

!

voice-port 0/0/0

compand-type a-law

cptone PT

!

voice-port 0/0/1

compand-type a-law

cptone PT

!

voice-port 0/1/0

compand-type a-law

cptone PT

!

voice-port 0/1/1

compand-type a-law

cptone PT

!

!

!

sccp local GigabitEthernet0/0

sccp ccm 172.16.X.250 identifier 1 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate profile 1 register MTPAAA

keepalive retries 1

keepalive timeout 10

switchover method immediate

switchback method immediate

!

dspfarm profile 1 mtp 

description MTPAAA

codec g711ulaw

maximum sessions software 100

associate application SCCP

!

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 300 voip

description ----- SIPTrunk to OneContact -----

destination-pattern 6...

session protocol sipv2

session target ipv4:A.B.C.19

voice-class codec 2

dtmf-relay rtp-nte sip-notify

no vad

!

telephony-service

sdspfarm units 1

sdspfarm tag 1 MTPAAA

max-ephones 25

max-dn 50

ip source-address 172.16.X.250 port 2000

timeouts interdigit 4

system message AAAA

cnf-file location flash:

cnf-file perphone

user-locale PT

network-locale PT

load 7937 apps37sccp.1-2-1-0.bin

time-zone 29

time-format 24

date-format dd-mm-yy

max-conferences 8 gain -6

call-forward pattern 2..

moh music-on-hold.au

transfer-system full-consult

transfer-pattern 2..

create cnf-files version-stamp Jan 01 2002 00:00:00

!

!

ephone-template  1

softkeys hold  Resume Newcall

softkeys idle  Redial Newcall Dnd Cfwdall Pickup

softkeys alerting  Endcall Callback

softkeys connected  Endcall Hold Trnsfer Park Confrn

softkeys ringing  Answer Dnd

!

!

ephone-dn  1  dual-line

number 200

description YYYY

name YYYY

corlist incoming Class-30

!

!

ephone-dn  2  dual-line

number 201

pickup-group 298

description KKKK

name KKKK

corlist incoming Class-40

!

!

ephone  1

mac-address 6C50.4DDA.5528

ephone-template 1

type 7962

button  1:1 2w2 3w3 4w4

button  5w5

!

!

!

ephone  2

mac-address E8BA.70FA.7387

ephone-template 1

type 7911

button  1:2

[Carlo Poggiarelli]

Hi

Can you please send a show voice call status when call from PSTN is connected to IVR?

Please send also a show viop rtp connections.

Let me know

Regards

Carlo

[zacariasnhancune007]

I dear,

sh voice call status

CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dial-peers

0x47F7 2A1F 0x29BDEE34 0/1/1.1 0/1:2 6200 g711alaw 1/300

1 active call found

sh voip rtp connections

VoIP RTP active connections :

No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP

1 18422 -1 16384 0 172.16.X.250 0.0.0.0

2 18424 18423 16434 18894 172.16.X.250 A.B.C.11

Found 2 active RTP connections

Com as melhores saudações/With Best regards

--

Zacarias Nhancune-Netwoking Engineer-CCNP,CCNA,CCNA Voice -11183622

[Carlo Poggiarelli]

Hi.

Because you configured MTP session with G711ulaw as codec, you should try to chanche compand-type to ulaw to your Voice ports.

Hope this helps

Regards

Carlo

[zacariasnhancune007]

Dear, I have tried but didn’t work too.

The sound get distorted.

Com as melhores saudações

--

Zacarias Nhancune-Netwoking Engineer-CCNP,CCNA,CCNA Voice -11183622

[Carlo Poggiarelli]

Hi.

In this case configure back to compand type a-law your voice port an try to force codec g711ulaw to your dialpeer 300.

Let me know.

HTH

Carlo

[ADAM CRISP]

Hello.

Please capture the output of "debug ccsip messages" while you place a call to your IVR. This will allow us to see any out of band DTMF method offered by the external system.

Adam

[zacariasnhancune007]

HI dear,

Check the attachment please

Com as melhores saudações

--

Zacarias Nhancune-Netwoking Engineer-CCNP,CCNA,CCNA Voice -11183622

[shrvaran]

Hi Zacarias,

Please add below commands under "dspfarm profile 1 mtp"

codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8

Issue command

no sccp

then

sccp

Then place a call from PSTN and see if it works.

Also please check if MTP is invoked when call is placed from PSTN.

[zacariasnhancune007]

Hi dear,

It accepts only one codec at a time

Com as melhores saudações

--

Zacarias Nhancune-Netwoking Engineer-CCNP,CCNA,CCNA Voice -11183622

[zacariasnhancune007]

Hi dear,

I have another issue. Please help:

Im running a CUCM Business Edition 6.1 (2) and after a hard reset it say:

/common contains a file system with errors, check forced

/common Inodes that were part of a corrupted orphan link list found.

/common: UNEXPECTED INCONSISTENCY; RUN fsck MANUALLY (i.e. without -a or -p options)

/grub: clean, 26/66520 files, 16820/265072 blocks

/partB: clean, 11/3515680 files, 118535/7018396 blocks

• An error occured during the file system check.

• Dropping you to a shell; the system will reboot

• when you leave the shell.

Give root password for maintenance

(or type Control-D to continue):

Com as melhores saudações

--

Zacarias Nhancune-Netwoking Engineer-CCNP,CCNA,CCNA Voice -11183622

[ADAM CRISP]

Hello.

Thank you for the SIP trace.

Your trace shows  SIP to SIP call, I imagine from a SIP telephone to your IVR. The 200 OK from the IVR clearly shows it expects to receive RTP-NTP with payload type 101. I suspect this is working OK.

Please can you now gather a SIP trace originating the call from the PSTN for comparison ?

thanks

[zacariasnhancune007]

Hello Adam,

the pstn number im using to call inbound is 820887712. I mean, from 820887712 im calling 95100 which is a gsm/pstn number allocated to the company. This 9500 is a SIM card that is inserted in a ISDN COM.SAT BRI Modem and this one too is connected to my CCME( VIC2-2BRI-NT/TE card) using a BRI interface. When I call this 95100 the modem is configured to translate it to 6200 which the IVR number.

So the trace a sent you is from the outside. There is also some calls coming into the system but forget about does calls.

Com as melhores saudações

--

Zacarias Nhancune-Netwoking Engineer-CCNP,CCNA,CCNA Voice -11183622

[ADAM CRISP]

Hi,

Since you're encoding from ISDN to SIP, the DSP modules will already be detecting DTMF if present in the Audio stream

The 200 OK signals RTPNTE payload type 101 - so next job is to see whether you're sending this:

debug voip rtp packet - will create a lot of ourput so don't do this over the console; suggest logging to memory buffer and then after your test - show log

You're looking for the RTP payload changing to 101 for DTMF and then the RTP data should show the DTMF digit

thanks

[ADAM CRISP]

Hello.

I need to travel now.

If you trace shows DTMF being sent - then look for the DTMF volume - duration and check to see whether this is compatible with your IVR

If your trace shows NO DTMF, then this is not being detected by the DSP's which means that DTMF is not present in-band from the BRI port. You could try upping the input gain - but I would tink looking at the GSM gw would be the next place to check.

Again - you don't need any MTP for this. DSP's are already involved.

Good luck

Adam