Please check the attached

LAJITH POTHU VARAKKAL · ‎11-12-2014

Hi All,

I have an issue here. The DTMF is not recognized by the Unity when user wants to do remote login to voicemail box by pressing *

Call Flow : T1 --> AS5400 --> SIP Trunk --> CUCM 9.1.2 --> SCCP --> CUC 9.1.2

Time : Nov 12 20:06:56.417 UTC

Calling Party Number i = 0x1183, '914466553077'
Called Party Number i = 0xA1, '2067677' - 99992067677

I can see in CCAPI, * being pressed and NOTIFY message is sent to CUCM, and I get 403 Forbidden as response.

The dial-peer configuration point to CUCM is below

dial-peer voice 4320 voip
tone ringback alert-no-PI
description --- PSTN to XXX 9999.XXXXXXX ---
preference 1
destination-pattern 9999.......$
no modem passthrough
session protocol sipv2
session target ipv4:XXXXX
voice-class codec 1
voice-class sip early-offer forced
voice-class sip options-keepalive
dtmf-relay sip-notify rtp-nte
fax rate 7200
ip qos dscp cs3 signaling
no vad

Logs are attached. Please help me to find out the issue.

Tagir Temirgaliyev · ‎11-12-2014

collect from as5400

debug ccsip mess

LAJITH POTHU VARAKKAL · ‎11-12-2014

Please check the attached traces and debug.

Thank You

Jayanth Velkuri · ‎11-12-2014

Enable “Accept Unsolicited Notification” under SIP Security Profile and test

LAJITH POTHU VARAKKAL · ‎11-12-2014

Hi,

We are not disabling it. It's always checked and enabled.

Can we see the digit passing to the CUCM by the gateway in notify messages when UN is chosen?

Jayanth Velkuri · ‎11-12-2014

remove rtp-nte in dial-peer

check mtp if not

LAJITH POTHU VARAKKAL · ‎11-12-2014

Kindly let me know why we need to remove from dial-peer. Per my understanding from the traces the CUCM negotiated to UN with the GW. How can we ensure * is passed to CUCM?

Thank You

Jayanth Velkuri · ‎11-12-2014

Consume mask is not set. Relaying Digit * to dstCallId 0x2B36E

Ayodeji Okanlawon · ‎11-13-2014

Why are you using sip notify? I suggest you remove sip notify and use only rtp-nte on that dial-peer. Ensure the sip trunk to gateway on cucm is set to use no preference for dtmf.

If you still have issues, consider using sip intergration between cuc and cucm. But sccp should work.

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LAJITH POTHU VARAKKAL · ‎11-13-2014

Hi,

I have configured only rtp-nte on the dial-peer, trunk is already set to no preference. But it also did not work. I am not looking forward to change the integration to SIP between CUCM and Unity.

Did you get chance to go through the traces? Can we know if the digit (*) has forwarded to CUCM by the gateway in Notify?

Thank You.

Ayodeji Okanlawon · ‎11-14-2014

I have looked at your traces and here is my break down of what is going on.

++++Here is the INVITE sent by the gateway. The gateway advertises sip notify to cucm and rtp-nte+++

Sent:
INVITE sip:99992067677@19.106.214.6:5060 SIP/2.0
Via: SIP/2.0/UDP 19.212.0.103:5060;branch=z9hG4bKB6665B1
From: <sip:914466553077@19.212.0.103>;tag=14F300AC-1C72
To: <sip:99992067677@19.106.214.6>
Date: Wed, 12 Nov 2014 20:06:56 GMT
Call-ID: 49794196-69DE11E4-BEFBBF6C-A1033357@19.212.0.103

Call-Info: <sip:19.212.0.103:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 60
Allow-Events: telephone-event
P-Asserted-Identity: <sip:914466553077@19.212.0.103>
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 798

-
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2. +++CUCM determines the DTMF caps of both (unity connection and sip trunk, so it can send it in its SDP answer+++

NB: mLocalDtmfCaps...UNSOL=1, KPML=1, Inband=0(0) (this is DTMF supported by sccp unity integration)

mEndppointsDtmfCaps...UNSOL=1, KPML=0, Inband=0(101) (this is the DTMF supported by your gateway based on dial-peer config)

From this, we can see that CUCM has disabled rtp-nte (Inband=0(101)) and its going to do unsolicited notify since both endpoint support it.

3. +++Next we see the 200 OK that CUCM sends to your gateway, advertising sip notify as dtmf method+++

Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 19.212.0.103:5060;branch=z9hG4bKB6665B1
From: <sip:914466553077@19.212.0.103>;tag=14F300AC-1C72
To: <sip:99992067677@19.106.214.6>;tag=576759~67704c44-6827-40a1-8fb8-a01f1cab7131-242230140
Date: Wed, 12 Nov 2014 20:06:56 GMT

--

Call-Info: <sip:19.106.214.6:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

--

Content-Type: application/sdp
Content-Length: 181

v=0
o=CiscoSystemsCCM-SIP 576759 1 IN IP4 19.106.214.6
s=SIP Call
c=IN IP4 19.106.182.4
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25136 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

4. +++Next we see cucm creates a subscription for sip notify+++

17948853.012 |15:06:56.572 |AppInfo |SIPCdpc(26591) - subscribeForUnsolDtmf: creating implicit subscription.

5. +++Next cucm receives notify from your gateway+++

17949975.001 |15:07:02.581 |AppInfo |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 632 from 19.212.0.103:[62100]:
[1401438,NET]
NOTIFY sip:99992067677@19.106.214.6:5060 SIP/2.0
Via: SIP/2.0/UDP 19.212.0.103:5060;branch=z9hG4bKB66B6DA
From: <sip:914466553077@19.212.0.103>;tag=14F300AC-1C72
To: <sip:99992067677@19.106.214.6>;tag=576759~67704c44-6827-40a1-8fb8-a01f1cab7131-242230140
Call-ID: 49794196-69DE11E4-BEFBBF6C-A1033357@19.212.0.103
CSeq: 103 NOTIFY
Max-Forwards: 70
Date: Wed, 12 Nov 2014 20:07:02 GMT
User-Agent: Cisco-SIPGateway/IOS-12.x
Event: telephone-event
Subscription-State: active
Contact: <sip:19.212.0.103:5060>
P-Asserted-Identity: <sip:914466553077@19.212.0.103>
Content-Type: audio/telephone-event
Content-Length: 4

6..However for some reason cucm terminates the sip notify subscription and sends 403 forbidden++

17949979.000 |15:07:02.582 |SdlSig |SIPSubTerminated |wait

17949979.001 |15:07:02.582 |AppInfo |//SIP/SIPHandler/ccbId=576759/scbId=0/wait_SIPSubTerminated: inDialogSubReferCounter=0 scbid=576760

Suggestions:

1. As suggested earlier ensure that the SIP Trunk Security Profile is configured to accept unsolicited notification. After you have made this change, reset the sip trunk to the gateway

2. If the above still doesn't work, you can use sip-kpml. From the cucm logs we can see that the sccp integration supports kpml, hence configure the following on your gateway dial-peer

dial-peer voice 4320 voip

dtmf-relay sip-kpml rtp-nte

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LAJITH POTHU VARAKKAL · ‎11-19-2014

Hi Ayodeji,

Accept Unsolicited Notification is already checked on the SIP Trunk Security Profile. I have configured rtp-nte only as DTMF method in dial-peer and reset the SIP trunk from CUCM. Now the DTMF works.

Thanks a lot for your help. I really appreciate it :)

Regards,

Lajith

Ayodeji Okanlawon · ‎11-19-2014

Wao Lajith, you appreciate my help and you rate the post 2 stars!!! I spared my precious time to look at your logs and all you can say is my post is rubbish by giving it two stars!!! Since 2006 that I joined the community, you will be the first to do this..I don't think its nice.

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Joseph Rivera-Santiago · ‎12-10-2014

Hey Ayodeji,

It looks like I am having a similar issue as well. The previous "owner" of our SIP/CUCM environment had individual DialPeers for each # we own, instead of a catch all. I have cleaned it up as best as I could and have attached a sanitized config.

DTMF is working fine on our main AA in Unity ( a 252-XXX-XXXX ) #. However we are trying to add additional Call Handlers for other locations (customers can hear store hours, location etc.)

I am looking for a best practice approach, possibly having 1 Outgoing DP, 2 Incoming DP etc.

Whenever I fix DTMF from external #s, it breaks our users ability to dial DTMF out (join conference, dial in meetings etc.)

Could you possibly take a look, and see where the issue may lie?

Thank you

Ayodeji Okanlawon · ‎12-10-2014

First of all you need to reconfigure your dial-peers. You shouldn't use alpha-numeric or signal on sip dial-peers. So please reconfigure as shown below.

dial-peer voice 110 voip
no dtmf-relay h245-alphanumeric rtp-nte h245-signal
dtmf-relay rtp-nte

dial-peer voice 100 voip
no dtmf-relay h245-alphanumeric rtp-nte h245-signal
dtmf-relay rtp-nte

Next you need to describe which part of the call flow DTMF is failing. You need to let us know your detailed call flow as well..eg

pstn---sip---cube--sip--cucm--sip--unity connection

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DTMF Issue in SIP