[cisco-voip] dtmf from cucm to 2821 cube to sip trunk
Dane Newman
dane.newman at gmail.com
Tue Oct 27 07:36:55 EDT 2009
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<sip%3A6782282221 at 173.14.220.57>
>;party=calling;screen=yes;privacy=off
From: <sip:6782282221 at sip.talkinip.net <sip%3A6782282221 at sip.talkinip.net>
>;tag=2EDA9C8-25D6
To: <sip:18774675464 at 64.154.41.200 <sip%3A18774675464 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<sip%3A6782282221 at 173.14.220.57>
>;party=calling;screen=yes;privacy=off
From: <sip:6782282221 at sip.talkinip.net <sip%3A6782282221 at sip.talkinip.net>
>;tag=2EDA9C8-25D6
To: <sip:18774675464 at 64.154.41.200 <sip%3A18774675464 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 <sip%3A6782282221 at sip.talkinip.net>
>;tag=2EDA9C8-25D6
To: <sip:18774675464 at 64.154.41.200 <sip%3A18774675464 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 <sip%3A18774675464 at 64.154.41.200>
>;tag=3465630735-938664
From: <sip:6782282221 at sip.talkinip.net <sip%3A6782282221 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 <sip%3A6782282221 at sip.talkinip.net>
>;tag=2EDA9C8-25D6
To: <sip:18774675464 at 64.154.41.200 <sip%3A18774675464 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 <sip%3A6782282221 at sip.talkinip.net>
>;tag=2EDA9C8-25D6
To: <sip:18774675464 at 64.154.41.200 <sip%3A18774675464 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
> >
> >
>
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