yeh. I did. Take a loot at a debug ccsip calls from IPIPGW w/ dtmf-sip-notify
and after with dtmf-relay rtp-nte
*****w/ SIP outgoing dial-peer as dtmf-relay sip-notity
HQ-RTR#debug ccsip calls
SIP Call statistics tracing is enabled
HQ-RTR#
*Apr 3 21:07:12.511: //395/8F4BBEB8835E/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x474E7F24
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 1600
Called Number : 1600
Source IP Address (Sig ): 10.2.200.3
Destn SIP Req Addr:Port : 10.2.200.21:5060
Destn SIP Resp Addr:Port : 10.2.200.21:5060
Destination Name : 10.2.200.21
*Apr 3 21:07:12.511: //395/8F4BBEB8835E/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0
Source IP Address (Media): 10.2.200.3
Source IP Port (Media): 18602
Destn IP Address (Media): 10.2.200.2
Destn IP Port (Media): 19008
Orig Destn IP Address:Port (Media): 0.0.0.0:0
***w/ SIP outgoing dial-peer as dtmf-relay rtp-nte
*Apr 3 21:08:05.599: //397/AB5CDC578361/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x474F4E9C
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 1600
Called Number : 1600
Source IP Address (Sig ): 10.2.200.3
Destn SIP Req Addr:Port : 10.2.200.21:5060
Destn SIP Resp Addr:Port : 10.2.200.21:5060
Destination Name : 10.2.200.21
*Apr 3 21:08:05.599: //397/AB5CDC578361/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101
Source IP Address (Media): 10.2.200.3
Source IP Port (Media): 19546
Destn IP Address (Media): 10.2.200.2
Destn IP Port (Media): 16540
Orig Destn IP Address:Port (Media): 0.0.0.0:0
*Apr 3 21:08:05.599: //397/AB5CDC578361/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 200
From: cl...@mcglamry.net
To: spoli...@hotmail.com; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] DTMF digits from CME calls to Unity does not
work
Date: Fri, 3 Apr 2009 16:05:12 -0400
I remember Mark Snow mentioning at one point that there
are certain transformations around DTMF relay that are not supported in a CUBE
environment. I think this was it.
Have you tried changing the sip config to SIP-Notify with
H245-Alpha on the H.323 side?
Mark Snow, if you're monitoring this thread, could you
please pipe up? What was the "allowed" DTMF relay transformation and what
wasn't supported?
Cliff
----- Original Message -----
From:
Sergio
Polizer
To: cl...@mcglamry.net ; ccie_voice@onlinestudylist.com
Sent: Friday, April 03, 2009 1:49
PM
Subject: RE: [OSL | CCIE_Voice] DTMF
digits from CME calls to Unity does not work
Yes, I do. It is transcoding the differents codecs from this
call. A simple hold/unhold works also.
The only problem is related to the
DTMF. When I call to the Unity from CME phone It does not "understand" the
DTMF digits.
CME Phone ---h323 (h245-alp) g729--->
IPIPGW----SIP(rtp-nte)g711u---->Unity
From: cl...@mcglamry.net
To: spoli...@hotmail.com; ccie_voice@onlinestudylist.com
Subject:
Re: [OSL | CCIE_Voice] DTMF digits from CME calls to Unity does not
work
Date: Fri, 3 Apr 2009 10:13:02 -0400
Do you have a transcoder available and registered on the
IPIPGW? This sounds like an issue where you may be having a
supplementary services problem. If I'm right, you probably can't hold
and unhold the call either.
-----
Original Message -----
From:
Sergio
Polizer
To:
ccie_voice@onlinestudylist.com
Sent:
Thursday, April 02, 2009 8:08 PM
Subject:
[OSL | CCIE_Voice] DTMF digits from CME calls to Unity does not work
Hi, I'm trying to make DTMF digits be
recognize by Unity from CME calls through a IPIPGW. This is the
topology:
CME Phone ---h323--->
IPIPGW----SIP---->Unity
Please, May someone give me a
suggestion? Thank You!
This is the config at
CME:
dial-peer voice 21 voip
preference
1
destination-pattern [12]...
voice-class codec
1
voice-class h323 1
session target
ipv4:10.2.200.21
dtmf-relay h245-alphanumeric
fax protocol
pass-through g711ulaw
This is the config at IPIPGW:
dial-peer
voice 12 voip
incoming called-number [12]...
dtmf-relay
h245-alphanumeric
!
dial-peer voice 13
voip
destination-pattern [12]...
session protocol
sipv2
session target ipv4:10.2.200.21
dtmf-relay rtp-nte
(***I tried rtp-nte digit-drop h245-alp but no success)
codec
g711ulaw
!
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