[cisco-voip] dtmf from cucm to 2821 cube to sip trunk

Tue Oct 27 08:43:32 EDT 2009

In the 183 Session Progress they're not advertising DTMF:

m=audio 45846 RTP/AVP 0

There should be a 100 or 101 there.  Although, 183 is just ringback.
You would want to pick up on the other side and they should send a 200
OK with a new SDP.  If the other side did pick up, you need to tell
the provider that they need to send a 200 OK, because they're not.

-nick

On Tue, Oct 27, 2009 at 7:36 AM, Dane Newman <dane.newman at gmail.com> wrote:
> Nick
>
> I removed  voice-class sip asymmetric payload dtmf and added in the other
> line
>
> Just to state incoming dtmf works but not outbound the ITSP has told me they
> are using two different sip servers/vendors for processing inbound and
> outbound
> How does this translate into what I should sent the following too?
>
> rtp payload-type nse
> rtp payload-type nte
>
> In the debug trhe following where set
>
> rtp payload-type nse 101
>  rtp payload-type nte 100
>
> In the debug of ccsip If I am looking at it correctly I see me sending this
>
> *Oct 27 12:34:09.128: //846/8094E28C1800/SIP/Media/sipSPIAddSDPMediaPayload:
> Preferred method of dtmf relay is: 6, with payload: 100
> *Oct 27 12:34:09.128:
> //846/8094E28C1800/SIP/Info/sipSPIAddSDPPayloadAttributes:
>  max_event 15
>
> and
>
>
> *Oct 27 12:34:10.836:
> //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload
> from X-cap = 0
> *Oct 27 12:34:10.836:
> //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not present
> in SDP. Disable modem relay
>
>
> Sent:
> INVITE sip:18774675464 at 64.154.41.200:5060 SIP/2.0
> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A01ECD
> Remote-Party-ID:
> <sip:6782282221 at 173.14.220.57>;party=calling;screen=yes;privacy=off
> From: <sip:6782282221 at sip.talkinip.net>;tag=2EDA9C8-25D6
> To: <sip:18774675464 at 64.154.41.200>
> Date: Tue, 27 Oct 2009 12:34:09 GMT
> Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
> Min-SE:  1800
> Cisco-Guid: 2157240972-3604177326-402682881-167847941
> User-Agent: Cisco-SIPGateway/IOS-12.x
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
> NOTIFY, INFO, REGISTER
> CSeq: 101 INVITE
> Max-Forwards: 70
> Timestamp: 1256646849
> Contact: <sip:6782282221 at 173.14.220.57:5060>
> Expires: 180
> Allow-Events: telephone-event
> Content-Type: application/sdp
> Content-Disposition: session;handling=required
> Content-Length: 250
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 7043 4703 IN IP4 173.14.220.57
> s=SIP Call
> c=IN IP4 173.14.220.57
> t=0 0
> m=audio 16462 RTP/AVP 0 100
> c=IN IP4 173.14.220.57
> a=rtpmap:0 PCMU/8000
> a=rtpmap:100 telephone-event/8000
> a=fmtp:100 0-15
> a=ptime:20
>
>
> Then when I do a search for fmtp again further down I see
>
> Sent:
> INVITE sip:18774675464 at 64.154.41.200:5060 SIP/2.0
> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
> Remote-Party-ID:
> <sip:6782282221 at 173.14.220.57>;party=calling;screen=yes;privacy=off
> From: <sip:6782282221 at sip.talkinip.net>;tag=2EDA9C8-25D6
> To: <sip:18774675464 at 64.154.41.200>
> Date: Tue, 27 Oct 2009 12:34:09 GMT
> Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
> Min-SE:  1800
> Cisco-Guid: 2157240972-3604177326-402682881-167847941
> User-Agent: Cisco-SIPGateway/IOS-12.x
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
> NOTIFY, INFO, REGISTER
> CSeq: 102 INVITE
> Max-Forwards: 70
> Timestamp: 1256646849
> Contact: <sip:6782282221 at 173.14.220.57:5060>
> Expires: 180
> Allow-Events: telephone-event
> Proxy-Authorization: Digest
> username="1648245954",realm="64.154.41.110",uri="sip:18774675464 at 64.154.41.200:5060",response="ab63d4755ff4182631ad2db0f9ed0e44",nonce="12901115532:303fa5d884d6d0a5a0328a838545395b",algorithm=MD5
> Content-Type: application/sdp
> Content-Disposition: session;handling=required
> Content-Length: 250
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 7043 4703 IN IP4 173.14.220.57
> s=SIP Call
> c=IN IP4 173.14.220.57
> t=0 0
> m=audio 16462 RTP/AVP 0 100
> c=IN IP4 173.14.220.57
> a=rtpmap:0 PCMU/8000
> a=rtpmap:100 telephone-event/8000
> a=fmtp:100 0-15
> a=ptime:20
> *Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
> *Oct 27 12:34:09.332:
> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
> ccsip_spi_get_msg_type returned: 2 for event 1
> *Oct 27 12:34:09.332:
> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
> context=0x00000000
> *Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
> Checking Invite Dialog
> *Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
> From: <sip:6782282221 at sip.talkinip.net>;tag=2EDA9C8-25D6
> To: <sip:18774675464 at 64.154.41.200>
> Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
> CSeq: 102 INVITE
> Content-Length: 0
> *Oct 27 12:34:09.332: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:
> INVITE response with no RSEQ - disable IS_REL1XX
> *Oct 27 12:34:09.332: //846/8094E28C1800/SIP/State/sipSPIChangeState:
> 0x4A357FCC : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to
> (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
> *Oct 27 12:34:10.832: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
> *Oct 27 12:34:10.832:
> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
> ccsip_spi_get_msg_type returned: 2 for event 1
> *Oct 27 12:34:10.832:
> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
> context=0x00000000
> *Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
> Checking Invite Dialog
> *Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 183 Session Progress
> To: <sip:18774675464 at 64.154.41.200>;tag=3465630735-938664
> From: <sip:6782282221 at sip.talkinip.net>;tag=2EDA9C8-25D6
> Contact: <sip:18774675464 at 64.154.41.200:5060>
> Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
> CSeq: 102 INVITE
> Content-Type: application/sdp
> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
> Content-Length: 146
> v=0
> o=msx71 490 6110 IN IP4 64.154.41.200
> s=sip call
> c=IN IP4 64.154.41.101
> t=0 0
> m=audio 45846 RTP/AVP 0
> a=ptime:20
> a=rtpmap:0 PCMU/8000
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:
> INVITE response with no RSEQ - disable IS_REL1XX
> *Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD
> found in inbound container
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoMediaNegotiation:
> Number of m-lines = 1
> SIP: Attribute mid, level 1 instance 1 not found.
> *Oct 27 12:34:10.836:
> //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already
> bound, use existing source_media_ip_addr
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:
> Media src addr for stream 1 = 173.14.220.57
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:
> Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoPtimeNegotiation:
> One ptime attribute found - value:20
> *Oct 27 12:34:10.836:
> //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec:
> g711ulaw ptime :20, codecbytes: 160
> *Oct 27 12:34:10.836:
> //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec:
> g711ulaw codecbytes :160, ptime: 20
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPIDoPtimeNegotiation:
> Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec
> g711ulaw
> *Oct 27 12:34:10.836:
> //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPICheckDynPayloadUse:
> Dynamic payload(100) could not be reserved.
> *Oct 27 12:34:10.836:
> //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full named
> event(NE) match in fmtp list of events.
> *Oct 27 12:34:10.836:
> //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload
> from X-cap = 0
> *Oct 27 12:34:10.836:
> //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not present
> in SDP. Disable modem relay
> *Oct 27 12:34:10.836:
> //846/8094E28C1800/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction
> attribute present or multiple direction attributes that can't be handled for
> m-line:1 and num-a-lines:0
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:
> Codec negotiation successful for media line 1
>         payload_type=0, codec_bytes=160, codec=g711ulaw, dtmf_relay=rtp-nte
>         stream_type=voice+dtmf (1), dest_ip_address=64.154.41.101,
> dest_port=45846
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/State/sipSPIChangeStreamState:
> Stream (callid =  -1)  State changed from (STREAM_DEAD) to (STREAM_ADDING)
> *Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:
>         Preferred Codec        : g711ulaw, bytes :160
>         Preferred  DTMF relay  : rtp-nte
>         Preferred NTE payload  : 100
>         Early Media            : No
>         Delayed Media          : No
>         Bridge Done            : No
>         New Media              : No
>         DSP DNLD Reqd          : No
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already
> bound, use existing source_media_ip_addr
> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:
> Media src addr for stream 1 = 173.14.220.57
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:
>  callId 846 peer 845 flags 0x200005 state STATE_RECD_PROCEEDING
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
> CallID 846, sdp 0x497E29C0 channels 0x4A35926C
> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:
>  callId 846 size 240 ptr 0x4A170B28)
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
> Hndl ptype 0 mline 1
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting
> codec g711ulaw
> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/codec_found:
> Codec to be matched: 5
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO
> CODEC 5
> *Oct 27 12:34:10.840:
> //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec:
> g711ulaw codecbytes :160, ptime: 20
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media
> negotiation done:
> stream->negotiated_ptime=20,stream->negotiated_codec_bytes=160, coverted
> ptime=20 stream->mline_index=1, media_ndx=1
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
> Adding codec 5 ptype 0 time 20, bytes 160  as channel 0 mline 1 ss 1
> 64.154.41.101:45846
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to
> channel- AFTER CODEC FILTERING:
> ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = 5
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to
> channel- AFTER CODEC FILTERING:
> ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = -1
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:
>  callId 846 flags 0x100 state STATE_RECD_PROCEEDING
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:
> Report initial call media
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags
> 0x400018, ccb->pld.flags_ipip 0x200005
> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:
>  callId 846 size 240 ptr 0x4DEC000C)
> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/ccsip_update_srtp_caps:
> 5030: Posting Remote SRTP caps to other callleg.
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: do
> cc_api_caps_ind()
> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:
>           Stream type            : voice+dtmf
>           Media line             : 1
>           State                  : STREAM_ADDING (2)
>           Stream address type    : 1
>           Callid                 : 846
>           Negotiated Codec       : g711ulaw, bytes :160
>           Nego. Codec payload    : 0 (tx), 0 (rx)
>           Negotiated DTMF relay  : rtp-nte
>           Negotiated NTE payload : 100 (tx), 100 (rx)
>           Negotiated CN payload  : 0
>           Media Srce Addr/Port   : [173.14.220.57]:16462
>           Media Dest Addr/Port   : [64.154.41.101]:45846
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers
> recvd from app container
> *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No
> QSIG Body found in inbound container
> *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No
> RawMsg Body found in inbound container
> *Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No
> Data to form The Raw Message
> *Oct 27 12:34:10.840:
> //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress:
> ccsip_api_call_cut_progress returned: SIP_SUCCESS
> *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/State/sipSPIChangeState:
> 0x4A357FCC : State change from (STATE_RECD_PROCEEDING,
> SUBSTATE_PROCEEDING_PROCEEDING)  to (STATE_RECD_PROCEEDING, SUBSTATE_NONE)
> *Oct 27 12:34:10.844:
> //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress: Transaction
> Complete. Lock on Facilities released.
> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge: confID = 6,
> srcCallID = 846, dstCallID = 845
> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds:
> Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 846/845
> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds:
> Old streamcallid=846, new streamcallid=846
> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_gw_set_sipspi_mode:
> Setting SPI mode to SIP-H323
> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge:
> xcoder_attached = 0, xmitFunc = 1131891908, ccb xmitFunc = 1131891908
> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIProcessRtpSessions:
> sipSPIProcessRtpSessions
> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIAddStream: Adding
> stream 1 of type voice+dtmf (callid 846) to the VOIP RTP library
> *Oct 27 12:34:10.844:
> //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already
> bound, use existing source_media_ip_addr
> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:
> Media src addr for stream 1 = 173.14.220.57
> *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:
> sipSPIUpdateRtcpSession for m-line 1
> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:
> rtcp_session info
>         laddr = 173.14.220.57, lport = 16462, raddr = 64.154.41.101,
> rport=45846, do_rtcp=TRUE
>         src_callid = 846, dest_callid = 845, stream type = voice+dtmf,
> stream direction = SENDRECV
>         media_ip_addr = 64.154.41.101, vrf tableid = 0 media_addr_type = 1
> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:
> RTP session already created - update
> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession:
> stun is disabled for stream:4A1709F8
> *Oct 27 12:34:10.848:
> //846/8094E28C1800/SIP/Media/sipSPIGetNewLocalMediaDirection:
>         New Remote Media Direction = SENDRECV
>         Present Local Media Direction = SENDRECV
>         New Local Media Direction = SENDRECV
>         retVal = 0
> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/State/sipSPIChangeStreamState:
> Stream (callid =  846)  State changed from (STREAM_ADDING) to
> (STREAM_ACTIVE)
> *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Info/ccsip_bridge: really can't
> find peer_stream for
>                                                 dtmf-relay interworking
> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Entry
> *Oct 27 12:34:11.140:
> //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT
> VALUES: stream_callid=846, current_seq_num=0x23ED
> *Oct 27 12:34:11.140:
> //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES:
> stream_callid=846, current_seq_num=0x11D9
> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP
> with negotiated codec: g711ulaw, Bytes=160
> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Set
> forking flag to 0x0
> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode:
> Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx payload =
> 100, tx payload = 100
> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
> Preferred (or the one that came from DSM) modem relay=0, from CLI config=0
> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
> Disabling Modem Relay...
> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
> Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap
> list
> *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Modem
> Relay & Passthru both disabled
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: nse
> payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0,
> sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 1
> Active Streams
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
> Adding stream type (voice+dtmf) from media
> line 1 codec g711ulaw
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
> caps.stream_count=1,caps.stream[0].stream_type=0x3,
> caps.stream_list.xmitFunc=
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
> voip_rtp_xmit, caps.stream_list.context=
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
> 0x497E0B60 (gccb)
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP
> with codec : g711ulaw, Bytes=160, payload = 0
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
> ccsip_caps_ind: ccb->pld.flags_ipip = 0x200405
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No video
> caps detected in the caps posted by peer leg
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Setting
> CAPS_RECEIVED flag
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Calling
> cc_api_caps_ack()
> *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ack: Set
> forking flag to 0x0
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Entry
> *Oct 27 12:34:11.168:
> //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT
> VALUES: stream_callid=846, current_seq_num=0x11D9
> *Oct 27 12:34:11.168:
> //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES:
> stream_callid=846, current_seq_num=0x11D9
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP
> with negotiated codec: g711ulaw, Bytes=160
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Set
> forking flag to 0x0
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode:
> Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx payload =
> 100, tx payload = 100
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
> Preferred (or the one that came from DSM) modem relay=0, from CLI config=0
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
> Disabling Modem Relay...
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
> Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap
> list
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: Modem
> Relay & Passthru both disabled
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/sip_set_modem_caps: nse
> payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0,
> sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo: 1
> Active Streams
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
> Adding stream type (voice+dtmf) from media
> line 1 codec g711ulaw
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
> caps.stream_count=1,caps.stream[0].stream_type=0x3,
> caps.stream_list.xmitFunc=
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
> voip_rtp_xmit, caps.stream_list.context=
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
> 0x497E0B60 (gccb)
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Load DSP
> with codec : g711ulaw, Bytes=160, payload = 0
> *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
> ccsip_caps_ind: ccb->pld.flags_ipip = 0x200425
> *Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No video
> caps detected in the caps posted by peer leg
> *Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: Second TCS
> received for transfers across trunk - set CAPS2_RECEIVED
> *Oct 27 12:34:15.876: //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession:
> stun is disabled for stream:4A1709F8
> *Oct 27 12:34:15.876: //846/8094E28C1800/SIP/Info/ccsip_call_statistics:
> Stats are not supported for IPIP call.
> *Oct 27 12:34:15.876: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued
> event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
> *Oct 27 12:34:15.880:
> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
> ccsip_spi_get_msg_type returned: 3 for event 7
> *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Info/sipSPISendCancel:
> Associated container=0x4E310C1C to Cancel
> *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Transport/sipSPISendCancel:
> Sending CANCEL to the transport layer
> *Oct 27 12:34:15.880:
> //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage: msg=0x4DF0D994,
> addr=64.154.41.200, port=5060, sentBy_port=0, is_req=1, transport=1,
> switch=0, callBack=0x419703BC
> *Oct 27 12:34:15.880:
> //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage: Proceedable for
> sending msg immediately
> *Oct 27 12:34:15.880:
> //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: switch transport
> is 0
> *Oct 27 12:34:15.880:
> //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: Set to send the
> msg=0x4DF0D994
> *Oct 27 12:34:15.880:
> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send
> for msg=0x4DF0D994, addr=64.154.41.200, port=5060, connId=2 for UDP
> *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Info/sentCancelDisconnecting:
> Sent Cancel Request, starting CancelWaitResponseTimer
> *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/State/sipSPIChangeState:
> 0x4A357FCC : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE)  to
> (STATE_DISCONNECTING, SUBSTATE_NONE)
> *Oct 27 12:34:15.888: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Sent:
> CANCEL sip:18774675464 at 64.154.41.200:5060 SIP/2.0
> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
> From: <sip:6782282221 at sip.talkinip.net>;tag=2EDA9C8-25D6
> To: <sip:18774675464 at 64.154.41.200>
> Date: Tue, 27 Oct 2009 12:34:09 GMT
> Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
> CSeq: 102 CANCEL
> Max-Forwards: 70
> Timestamp: 1256646855
> Reason: Q.850;cause=16
> Content-Length: 0
> *Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
> *Oct 27 12:34:15.900:
> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
> ccsip_spi_get_msg_type returned: 2 for event 1
> *Oct 27 12:34:15.900:
> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
> context=0x00000000
> *Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
> Checking Invite Dialog
> *Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
> From: <sip:6782282221 at sip.talkinip.net>;tag=2EDA9C8-25D6
> To: <sip:18774675464 at 64.154.41.200>
> Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
> CSeq: 102 CANCEL
> Content-Length: 0
> *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:
> non-INVITE response with no RSEQ - do not disable IS_REL1XX
> *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPIIcpifUpdate:
> CallState: 3 Playout: 0 DiscTime:4913670 ConnTime 0
> *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
> Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
> *Oct 27 12:34:15.912:
> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
> ccsip_spi_get_msg_type returned: 2 for event 1
> *Oct 27 12:34:15.912:
> //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
> context=0x00000000
> *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
> Checking Invite Dialog
> *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>
> On Mon, Oct 26, 2009 at 7:36 PM, Nick Matthews <matthnick at gmail.com> wrote:
>>
>> You would want to check the SDP of 200 OK the provider sends for your
>> outgoing call.  It will list the payload type for the dtmf in the
>> format a=fmtp 101 1-16, or something similar.  You want to find out
>> what payload type they are advertising (or if they are at all).  It
>> would be worth checking the incoming INVITE from them to see what
>> they're using when they send the first SDP.
>>
>> On that note, I would also remove the asymmetric payload command - to
>> my knowledge it doesn't do what you're expecting it to.  You may want
>> to try this command:
>> voice-class sip dtmf-relay force rtp-nte
>>
>>
>> -nick
>>
>> On Mon, Oct 26, 2009 at 5:16 PM, Dane Newman <dane.newman at gmail.com>
>> wrote:
>> > Hello,
>> >
>> > I am having an issue with dtmf working outbound.  Inbound dtmf works
>> > fine.
>> > It took some playing around with it.  At first it didnt work till the
>> > payload was ajusted.    I am now trying to get outbound dtmf working
>> > properly.
>> >
>> > On my 2821 I debugged voip rtp session named-events and then made a call
>> > to
>> > a 1800 number and hit digits.  I didn't see any dtmf output on the
>> > router
>> > nothing showed up in the debug.  Does this mean I can safely asume that
>> > the
>> > problem for right now is not on the ITSP side but on my side since dtmf
>> > is
>> > not being sent down the sip trunk?
>> >
>> > I have my cuc 7.x configured to talk to my 2821 via h323.  The
>> > configuration
>> > of the cisco 2821 is shown below.  Does anyone have any ideas what I can
>> > do
>> > so dtmf digits process properly outbound?
>> >
>> > The settings in my cuc 7.x to add the gateway h323 are
>> >
>> > h323 cucm gateway configuratration
>> > Signaling Port 1720
>> > media termination point required yes
>> > retry video call as auto yes
>> > wait for far end h.245 terminal capability set yes
>> > transmit utf-8 calling party name no
>> > h.235 pass through allowed no
>> > significant digits all
>> > redirect number IT deliver - inbound no
>> > enable inbound faststart yes
>> > display IE deliver no
>> > redirect nunmber IT deliver - outbound no
>> > enable outbound faststart yes
>> >
>> >
>> > voice service voip
>> >  allow-connections h323 to h323
>> >  allow-connections h323 to sip
>> >  allow-connections sip to h323
>> >  allow-connections sip to sip
>> >  fax protocol pass-through g711ulaw
>> >  h323
>> >   emptycapability
>> >   h225 id-passthru
>> >   h245 passthru tcsnonstd-passthru
>> >  sip
>> >
>> >
>> > voice class h323 50
>> >   h225 timeout tcp establish 3
>> > !
>> > !
>> > !
>> > !
>> > !
>> > !
>> > !
>> > !
>> > !
>> > !
>> > !
>> > voice translation-rule 1
>> >  rule 1 /.*/ /190/
>> > !
>> > voice translation-rule 2
>> >  rule 1 /.*/ /1&/
>> > !
>> > !
>> > voice translation-profile aa
>> >  translate called 1
>> > !
>> > voice translation-profile addone
>> >  translate called 2
>> > !
>> > !
>> > voice-card 0
>> >  dspfarm
>> >  dsp services dspfarm
>> > !
>> > !
>> > sccp local GigabitEthernet0/1
>> > sccp ccm 10.1.80.11 identifier 2 version 7.0
>> > sccp ccm 10.1.80.10 identifier 1 version 7.0
>> > sccp
>> > !
>> > sccp ccm group 1
>> >  associate ccm 1 priority 1
>> >  associate ccm 2 priority 2
>> >  associate profile 1 register 2821transcode
>> > !
>> > dspfarm profile 1 transcode
>> >  codec g711ulaw
>> >  codec g711alaw
>> >  codec g729ar8
>> >  codec g729abr8
>> >  codec g729r8
>> >  maximum sessions 4
>> >  associate application SCCP
>> > !
>> > !
>> > dial-peer voice 100 voip
>> >  description AA Publisher
>> >  preference 1
>> >  destination-pattern 1..
>> >  voice-class h323 50
>> >  session target ipv4:10.1.80.10
>> >  dtmf-relay h245-alphanumeric
>> >  codec g711ulaw
>> >  no vad
>> > !
>> > dial-peer voice 1000 voip
>> >  description incoming Call
>> >  translation-profile incoming aa
>> >  preference 1
>> >  rtp payload-type nse 101
>> >  rtp payload-type nte 100
>> >  incoming called-number 6782282221
>> >  dtmf-relay rtp-nte
>> >  codec g711ulaw
>> >  ip qos dscp cs5 media
>> >  ip qos dscp cs5 signaling
>> >  no vad
>> > !
>> > dial-peer voice 101 voip
>> >  description AA Subscriber
>> >  preference 2
>> >  destination-pattern 1..
>> >  voice-class h323 50
>> >  session target ipv4:10.1.80.11
>> >  dtmf-relay h245-alphanumeric
>> >  codec g711ulaw
>> >  no vad
>> > !
>> > dial-peer voice 2000 voip
>> >  description outbound
>> >  translation-profile outgoing addone
>> >  preference 1
>> >  destination-pattern .T
>> >  rtp payload-type nse 101
>> >  rtp payload-type nte 100
>> >  voice-class sip asymmetric payload dtmf
>> >  session protocol sipv2
>> >  session target ipv4:64.154.41.200
>> >  dtmf-relay rtp-nte
>> >  codec g711ulaw
>> >  no vad
>> > !
>> > !
>> > sip-ua
>> >  credentials username ***** password 7  *****  realm sip.talkinip.net
>> >  authentication username  *****  password 7  *****
>> >  authentication username  ***** password 7  *****  realm
>> > sip.talkinip.net
>> >  set pstn-cause 3 sip-status 486
>> >  set pstn-cause 34 sip-status 486
>> >  set pstn-cause 47 sip-status 486
>> >  registrar dns:sip.talkinip.net expires 60
>> >  sip-server dns:sip.talkinip.net:5060
>> > _______________________________________________
>> > cisco-voip mailing list
>> > cisco-voip at puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>> >
>> >
>
>