Cisco CUE with MS Exchange 2010

harsh.rohit · ‎03-20-2013

Hi

We have a client which has integrated MS Exchange 2010 as Unified Messaging. However we have a problem passing the DTMF Digits. The call flow is as follows:

Calling from PSTN --> Board Number --> Cisco IPIVR (To Enter the Desired Extension) --> IP Phone Rings -- > Call gets forwarded to MS Exchange

Once the call is on the MS Exchange we are able to hear the prompts but cannot pass the DTMF Tones.

The Dial-Peer configuration for VM Pilot is:

dial-peer voice 10 voip

description Voice_mail_dialling_internal

destination-pattern 9998

session protocol sipv2

session target ipv4:xxx.xxx.xxx.xxx

session transport tcp

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

I would really appreciate the help in this regard.

Regards,

Rohit

Anas Abueideh · ‎03-23-2013

Hi,

Exchange 2010 DTMF is the RFC2833 which is rtp-nte. I think your issue is at exchange side. kindly find the below link

http://technet.microsoft.com/en-us/library/bb232158(v=exchg.141).aspx

Try to configure transcoder, it might help

HTH

Anas

please rate if it is helpful

daniel.bloom · ‎03-23-2013

You could also use the following debugs to determine what DTMF relay method is negotiated.

Debug ccsip messages
Debug ccsip call

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giorgiosalsiri · ‎03-23-2013

If you are integrating to it using SIP, you may need to check what type of DTMF relay is required. You may need to use SIP-KPML or SIP-NOTIFY out of band type of DTMF relay.

HTH

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Hythim Ali El Hadad · ‎03-24-2013

try to force rtp-nte

dtmf-relay force rtp-nte

also try to get these debugs:

debug voip ccapi inout

debug ccsip all

debug voip rtp sess name

Regards

Haitham

harsh.rohit · ‎03-25-2013

Here are the logs:

Received:

SIP/2.0 200 OK

FROM: ;tag=D21B25AC-18A9

TO: <9998>;epid=A9915E2CC8;tag=48dfa3e116

CSEQ: 102 INVITE

CALL-ID: 62529C75-945F11E2-9C3CE62E-1818A020@xxx.xxx.xxx.xxx

VIA: SIP/2.0/TCP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK89111082

CONTACT: ;automata

CONTENT-LENGTH: 190

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

SERVER: RTCC/3.1.0.0

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

v=0

o=- 0 0 IN IP4 xxx.xxx.xxx.xxx

s=Microsoft Exchange Speech Engine

c=IN IP4 xxx.xxx.xxx.xxx

t=0 0

m=audio 6272 RTP/AVP 0 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Mar 25 08:46:57.944: //224559/6251FFB59C38/SIP/Info/sipSPICheckResponseExt: INVITE response with no RSEQ - disable IS_REL1XX

Mar 25 08:46:57.944: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container

Mar 25 08:46:57.948: //224559/6251FFB59C38/SIP/Info/ccsip_update_srtp_caps: 7398: Not Sending NULL SRTP CAPS to SIP LEG

Mar 25 08:46:57.948: //224559/6251FFB59C38/SIP/Media/sipSPIUpdCallWithSdpInfo:

Stream type : voice+dtmf

Media line : 1

State : STREAM_ADDING (2)

Stream address type : 1

Callid : 224559

Negotiated Codec : g711ulaw, bytes :160

Nego. Codec payload : 0 (tx), 0 (rx)

Negotiated DTMF relay : rtp-nte

Negotiated NTE payload : 101 (tx), 101 (rx)

Negotiated CN payload : 0

Media Srce Addr/Port : [xx.xxx.xxx.xxx]:30578

Media Dest Addr/Port : [xxx.xxx.xxx.xxx]:6272

08:47:16.072: //-1/xxxxxxxxxxxx/SIP/Msg/sipDisplayBinaryData:

Receiving: Binary Message Body

Mar 25 08:47:16.072: Content-Type: audio/telephone-event

0B 00 01 F4

Mar 25 08:47:16.072: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_spi_unsolicited_notify_parse_dtmf: The NOTIFY message body is 0x0B0001F4

Mar 25 08:47:16.072: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_spi_unsolicited_notify_parse_dtmf: parsed digit=#, duration=500ms, end_flag=0

Mar 25 08:47:16.072: //224553/7D6540000002/CCAPI/cc_api_call_digit_begin:

Consume mask is not set. Relaying Digit # to dstCallId 0x36D2A

Mar 25 08:47:16.072: //224553/7D6540000002/CCAPI/cc_relay_digit_begin_for_3way_conference:

Check DTMF relay digit begin for 3way conf

Mar 25 08:47:16.072: //-1/xxxxxxxxxxxx/SIP/Event/ccsip_call_notify_response: Queued event from SIP SPI : SIPSPI_EV_CC_NOTIFY_RESP

Mar 25 08:47:16.072: //0/000000000000/SIP/Info/sipSPIHandleNotifyOnExistingDialog: ccsip_api_notify_ind returned: SIP_SUCCESS

Please help me in this regard.

Regards,

Rohit

Anas Abueideh · ‎03-25-2013

Hi,

I check the logs. it starts a conference after the dtmf steps. what is this conference ?

it uses rtp-nte as a DTMF, and codec g711ulaw.

run also debug voip ccapi inout to check what the digit you enter, to check exactly in what version you are sending the DTMF tone.

did you try the transcoder ? it may help

HTH

Anas

please rate if it is helpful