I using : - Call Manager 18.104.22.16800-2
- Router Cisco CISCO2911/K9 ,Version 15.0(1r)M16 , System image file is "flash0:c2900-universalk9-mz.SPA.152-1.T3.bin".
CUCM -> Trunk -> Voice Gateway -> Trunk -> SIP Server ITSP.
I integrate my call manager with voice gateway with sip trunk to ITSP SIP Server, I have some problem with receiving DTMF in incoming call, my ITSP patner said that they send DTMF in RFC4733 inband. When I "debug voip rtp named-event" nothing comes.
But for outgoing call the DTMF work fine.
My question is does cisco router support for this DTMF RFC2733 type? If yes, is there any document from cisco that say it?
Please help me with this case.
Yes it is supported on Cisco IOS Release 15.2(2)T1. Make you put the dtmf-relay rtp-nte under the incoming and outgoing dial-peers.
New standard and system keywords are added to the existing dtmf-interworking command under voice-service and dial-peer configuration modes.
Old Behavior: SIP INFO DTMF digit to RFC-4733 DTMF interworking is not supported.
New Behavior: The newly added standard keyword generates RTP NTE packets that are RFC-4733 compliant.
I already put the dtmf-relay rtp nte to the dial-peer.
here some my dial-peer confguration
dial-peer voice 102 voip
description dial to Mobile Phone
translation-profile outgoing 1
session protocol sipv2
session target ipv4:10.253.156.205
voice-class sip dtmf-relay force rtp-nte
dial-peer voice 202 voip
session protocol sipv2
session target ipv4:172.21.3.2
incoming called-number .
voice-class codec 1
I also try to put the dtmf-internetworking with
voice service voip:
- dtmf internetworking rtp-nte combine with diall-peer incoming (rtp-nte, standard, system)
- dtmf internetworking standard combine with diall-peer incoming (rtp-nte, standard, system)
the result still empty.
Can you do a test call and send us
debug ccsip messages..
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
I already open TAC this issue but still no clue, so I decide to change to E1.
Now after I change my SIP Trunk to E1 and the problem is solved.
Thank you for all your response.
Thanks and Regards,
I used the commands:"dtmf-interworking standard" under the two dial-peers (cucm/cube & cube/SP) and the DTMF tones were transmited/received fine from then on. The SIP SP is accepting only In-band tones for telephony events.
The "dtmf rtp-nte" is sending telephony event out-of-band but within the RTP session, which is not in reallity an in-band method.