03-20-2013 07:08 AM - edited 03-16-2019 04:21 PM
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
03-23-2013 02:24 PM
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
03-23-2013 03:25 PM
You could also use the following debugs to determine what DTMF relay method is negotiated.
Debug ccsip messages
Debug ccsip call
Sent from Cisco Technical Support iPhone App
03-23-2013 03:51 PM
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
Sent from Cisco Technical Support iPhone App
03-24-2013 02:26 AM
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
03-25-2013 06:19 AM
Here are the logs:
Received:
SIP/2.0 200 OK
FROM:
TO: <9998>;epid=A9915E2CC8;tag=48dfa3e1169998>
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:
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
03-25-2013 07:32 AM
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
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