07-05-2013 08:42 AM - edited 03-16-2019 06:14 PM
Hi,
I have two sites with one CME installation each, and an integrated CUE on the second site (site B).
If i´m doing a call from site A (7940 sccp phone) to the CUE via the CME of site B, i can´t get the DTMF tones through.
If i configure a dial peer on CME A which points directly to the address of the CUE in site B, the DTMF tones
are working fine.
Not-working configuration (Dial Peer points to CME on site B):
dial-peer voice 1401 voip
description To CME in Site B
destination-pattern [5][8][0-4][0-9]
session protocol sipv2
session target ipv4:10.38.0.2
codec g711ulaw
no vad
Working configuration (Dial Peer points directly to CUE on site B):
dial-peer voice 1401 voip
description To CUE in Site B
destination-pattern 5848
session protocol sipv2
session target ipv4:10.38.0.3
codec g711ulaw
no vad
CME-Configuration of Site B:
dial-peer voice 2 voip
description **Incoming Dial Peer for calls from Site A**
incoming called-number 58..
codec g711ulaw
dial-peer voice 9998 voip
description CUE PromptMgmt
destination-pattern 5848
session protocol sipv2
session target ipv4:10.38.0.3
dtmf-relay sip-notify
codec g711ulaw
no vad
!
Any hints are much appreciated!
Thanks
Heinz
07-05-2013 08:53 AM
Add dtmf-relay to dial-peers on site A.
HTH,
Chris
07-05-2013 08:56 AM
Hi Heinz,
I think it is related to dtmf negotiation, you need to configure transcoder on site B to do the dtmf negotiation.
with your non working config check debug ccsip media to see what is the negotiated dtmf.
HTH
Anas
please rate all the helpful posts
07-05-2013 09:45 AM
Seems to be g711 end to end, so normally there is no transcoding involved here.
07-05-2013 12:23 PM
use the following command on your dial-peer
dtmf-relay rtp-nte
this will allow the dtmf signal to travel on top of rtp
07-06-2013 12:13 AM
Hi,
i already tried all variants of dtmf-relays on the outgoing dial-peers of site A, but none of them worked.
It seems to be that the DTMF codes are removed from the stream on the CME router B when it forwards
the call to the CUE.
Any more ideas?
Thanks
Heinz
07-06-2013 04:37 AM
Hi Heinz,
As suggested by other friend, pls share the logs captured with "debug ccsip media" command on CME at site B while making test calls to CUE.
And, have you got any logs on CmE at site A confirming the issue with DTMF only?
Ashok.
Sent from Cisco Technical Support iPhone App
07-07-2013 04:53 AM
Hi,
below is the output of a "debug ccsip media" while connecting to the CUE to from Site A to Site B.
No matter what kind of DTMF relay I use on the Dial Peer of Site A, i can´t get the tones trough.
Again: If i´m pointing the dial-peer of Site A directly to the remote CUE, it always works.
Thanks
Heinz
002944: Jul 7 13:47:34.277 CEST: //-1/D79FCF7586DA/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
002945: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2
002946: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 fo
r codec g711ulaw
002947: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711ulaw, bytes :160
Preferred DTMF relay : inband-voice
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No
002948: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2
002949: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 157399
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.38.0.2]:0
Media Dest Addr/Port : [10.38.8.2]:16940
002950: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 157399
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.38.0.2]:0
Media Dest Addr/Port : [10.38.8.2]:16940
002951: Jul 7 13:47:34.277 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 19160 for stream 1
002952: Jul 7 13:47:34.277 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 157399
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.38.0.2]:19160
Media Dest Addr/Port : [10.38.8.2]:16940
002953: Jul 7 13:47:34.281 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
002954: Jul 7 13:47:34.281 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2
002955: Jul 7 13:47:34.281 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 19126 for stream 1
002956: Jul 7 13:47:34.281 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE
002957: Jul 7 13:47:34.281 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: No active streams.
002958: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
002959: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 157400) to the VOIP RTP library
002960: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2
002961: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
002962: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.38.0.2, lport = 19126, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 157400, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr = - , vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
002963: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
002964: Jul 7 13:47:34.285 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPICreateRtpSession: stun is disabled
002965: Jul 7 13:47:34.285 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp p
assthru enabled
002966: Jul 7 13:47:34.285 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=157400
002967: Jul 7 13:47:34.321 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2
002968: Jul 7 13:47:34.321 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around
002969: Jul 7 13:47:34.321 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711ulaw, bytes :160
Preferred DTMF relay : sip-notify
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No
002970: Jul 7 13:47:34.321 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2
002971: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 157400
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : sip-notify
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.38.0.2]:19126
Media Dest Addr/Port : [10.38.0.3]:20898
002972: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo: 0 Active Streams
002973: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo: Number of active streams is zero (0)!
002974: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc=
002975: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context=
002976: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb)
002977: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 157400
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : sip-notify
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.38.0.2]:19126
Media Dest Addr/Port : [10.38.0.3]:20898
002978: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:35CCDBD8
002979: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
002980: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 157399) to the VOIP RTP library
002981: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2
002982: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
002983: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.38.0.2, lport = 19160, raddr = 10.38.8.2, rport=16940, do_rtcp=TRUE
src_callid = 157399, dest_callid = 157400, stream type = voice-only, stream direction = SENDRECV
media_ip_addr = 10.38.8.2, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
002984: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
002985: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPICreateRtpSession: stun is disabled
002986: Jul 7 13:47:34.325 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp p
assthru enabled
002987: Jul 7 13:47:34.325 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=157399
002988: Jul 7 13:47:34.325 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
002989: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
002990: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 157400) to the VOIP RTP library
002991: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.38.0.2
002992: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
002993: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.38.0.2, lport = 19126, raddr = 10.38.0.3, rport=20898, do_rtcp=TRUE
src_callid = 157400, dest_callid = 157399, stream type = voice-only, stream direction = SENDRECV
media_ip_addr = 10.38.0.3, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
002994: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update
002995: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:35CCDBD8
002996: Jul 7 13:47:34.325 CEST: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=157400
002997: Jul 7 13:47:34.325 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
002998: Jul 7 13:47:45.077 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:367D9708
002999: Jul 7 13:47:45.077 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:35CCDBD8
003000: Jul 7 13:47:45.081 CEST: //157399/D79FCF7586DA/SIP/Media/sipSPIHandleDestroyRtpSession: stream:367D9708
003001: Jul 7 13:47:45.081 CEST: //157400/D79FCF7586DA/SIP/Media/sipSPIHandleDestroyRtpSession: stream:35CCDBD8
07-07-2013 10:51 AM
Hi Heinz,
On the inbound dial peer of CME B can you please configure session protocol sipv2 and try the dtmf?
Sent from Cisco Technical Support iPad App
07-07-2013 01:10 PM
Also I would say add the dtmf relay on the inbound dialpeer.
Remember that this dial peer is forwarding to your cue module.
I hope it helps.
Sent from Cisco Technical Support iPhone App
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