DTMF for Incoming over SIP trunk is not working

[Anas Abueideh]

Dear Experts

Kindly find the above topology.I have issues with DTMF calls when they hit the IVR. the calls codec is G729, we have configured IOS transcoder to be used when the calls hits the UCCX.

The IVR is working but unfortunately the DTMF. it always give an error OOBand RFC2833 mismatch which is required MTP.

the cube dial-peer towards CUCM 10.5 has dtmf-relay rtp-nte. 

the transcoder configuration is

dspfarm profile 1 transcode  
 codec g729abr8
 codec g729br8
 codec g729ar8
 codec g729r8
 codec g711alaw
 codec g711ulaw
 maximum sessions 14
 associate application SCCP

the transcoder is registered and associated with the SIP trunk and CTI route point.

Thank you in advance

Anas

[GHARIP MOHAMED MARZOK]

i have the output of debug voip ccapi inout  for now 

[Ayodeji Okanlawon]

I am confused. The dialled number is an outbound number matching dial-peer 5. Is this not an inbound call? How is the call routed to voicemail?

[GHARIP MOHAMED MARZOK]

t

[Ayodeji Okanlawon]

Okay from the logs, the call is coming from extension 600 and you are trying to transfer to the same extension..Can you transfer to another extension and send me also debug ccsip messages.

[GHARIP MOHAMED MARZOK]

ا

[Ayodeji Okanlawon]

Gharip,

I have looked at the sip messages and It appears the digits pressed is not sent over to CUE by the gateway. To confirm this we need to test the working internal calls. Can you please do the following:

Conf t:

service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit

Then..

<Enable debugs, then test internal working call again.>

debug ccsip messages

debug ephone detail

<Enable session capture to txt file in terminal program.> (such as Putty)

then do the ff:

terminal length 0
show logging

[GHARIP MOHAMED MARZOK]

can you tell please what you thinking about ? what the commands you need to apply will affect ? because it is live environment , also it was working fine before and suddenly stopped without any changes 

many thanks for your interest 

[Ayodeji Okanlawon]

I want to establish why the internal call is working by looking at the logs for internal calls. This debug are light and wouldn't break anything.

[GHARIP MOHAMED MARZOK]

i am asking about the commands not the debugs ,

thanks .

[Ayodeji Okanlawon]

The commands just increase the size of the buffer and how the debugs are stored. This will log the debugs into the router buffer instead of the console.

[GHARIP MOHAMED MARZOK]

can we ignore it and get the logs you need ?

could this issue cause from unity side ?

[Ayodeji Okanlawon]

No we cant. These commands are harmless and you should always have them in your configuration. They help a lot in troubleshooting. This issue looks like the gateway is not sending the digits to unity. This is why we need to do this test.

[GHARIP MOHAMED MARZOK]

find attached needed logs ,

calling number is 600

called 888 which is the IVR 

the extension he dialed is 601 and the call works .

[Ayodeji Okanlawon]

Okay thanks for the logs. After looking at it, this definitely look like a gateway/dsp problem..

Here is my analysis.

+++After you press digit 6 on th ephone we see this on the logs+++
014111: Feb  9 16:16:55.692: ephone-101[100/71][SEP203A0722D9AD]:KeypadButtonMessage 6
++Next is that the gateway sends the digit to CUE via sip notify+++

Here we see the actual digit that is sent to CUE (digit 6  = 06 80 00 64)

014118: Feb  9 16:16:55.692: //-1/xxxxxxxxxxxx/SIP/Msg/sipDisplayBinaryData:
 Sending: Binary Message Body
014119: Feb  9 16:16:55.692: Content-Type: audio/telephone-event
06 00 07 D0
Sent:
NOTIFY sip:888@192.168.3.6:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.3.5:5060;branch=z9hG4bK418391260
From: "Hisham Samir" <sip:600@192.168.3.5>;tag=1AF9F55C-1897
To: <sip:888@192.168.3.6>;tag=ds133ba612
Call-ID: 2186D2F4-AF9D11E4-88E9C821-9D78A823@192.168.3.5
CSeq: 102 NOTIFY
Max-Forwards: 70
Date: Mon, 09 Feb 2015 14:16:55 GMT
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Event: telephone-event
Subscription-State: active
Contact: <sip:192.168.3.5:5060>
Content-Type: audio/telephone-event
Content-Length: 4

---
014122: Feb  9 16:16:55.696: //-1/xxxxxxxxxxxx/SIP/Msg/sipDisplayBinaryData:
 Sending: Binary Message Body
014123: Feb  9 16:16:55.696: Content-Type: audio/telephone-event
06 80 00 64

And after digit 0 was pressed on the phone we see this..

014141: Feb  9 16:16:56.032: //-1/xxxxxxxxxxxx/SIP/Msg/sipDisplayBinaryData:
 Sending: Binary Message Body
014142: Feb  9 16:16:56.032: Content-Type: audio/telephone-event
00 80 00 64 ( this is 0)

In the logs for external calls we do not see the actual digit been sent over SIP to CUE even though its received on the gateway

I suggest you open a TAC case with cisco. This might be dsp related.