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DTMF to remote CUE

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

9 REPLIES
Hall of Fame Super Silver

DTMF to remote CUE

Add dtmf-relay to dial-peers on site A.

HTH,

Chris

DTMF to remote CUE

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

New Member

Re: DTMF to remote CUE

Seems to be g711 end to end, so normally there is no transcoding involved here.

Re: DTMF to remote CUE

use the following command on your dial-peer

dtmf-relay rtp-nte

this will allow the dtmf signal to travel on top of rtp

DTMF to remote CUE

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

Bronze

Re: DTMF to remote CUE

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

With best regards... Ashok ----------- Pls kindly rate if helpful or answered your question.

DTMF to remote CUE

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

New Member

Re: DTMF to remote CUE


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

Regards, Avinash

Re: DTMF to remote CUE

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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