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  1. Senior Member JeanM's Avatar
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    #26
    Confirmed, signaling and now rtp is working both ways.

    Here is a good debug

    CME_A_#debug ccsip all
    This may severely impact system performance. Continue? [confirm]
    All SIP Call tracing is enabled
    CME_A_#
    CME_A_#
    CME_A_#
    CME_A_#
    CME_A_#
    *Jan 4 05:22:11.180: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 20.20.20.2:59570
    *Jan 4 05:22:11.180: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
    *Jan 4 05:22:11.180: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    INVITE sip:6601@192.168.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 20.20.20.2:5060;branch=z9hG4bK919B5
    From: "Goldie" <sip:7701@20.20.20.2>;tag=6C27D4-751
    To: <sip:6601@192.168.1.200>
    Date: Fri, 01 Mar 2002 12:58:35 GMT
    Call-ID: E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    Supported: 100rel,timer,replaces
    Min-SE: 1800
    Cisco-Guid: 3688424679-743051734-2148715475-4247906904
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Remote-Party-ID: "Goldie" <sip:7701@20.20.20.2>;party=calling;screen=no;priv acy=off
    Timestamp: 1014987515
    Contact: <sip:7701@20.20.20.2:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Length: 189


    v=0
    o=CiscoSystemsSIP-GW-UserAgent 4151 6864 IN IP4 65.65.65.200
    s=SIP Call
    c=IN IP4 65.65.65.200
    t=0 0
    m=audio 19012 RTP/AVP 8
    c=IN IP4 20.20.20.2
    a=rtpmap:8 PCMA/8000
    a=ptime:20


    *Jan 4 05:22:11.188: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 20.20.20.2,Port 5060, Transport 1, SentBy Port 5060
    *Jan 4 05:22:11.188: //-1/DBD8E4E78012/SIP/State/sipSPIChangeState: 0x8561974C : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
    *Jan 4 05:22:11.188: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 20.20.20.2,Port 59570, Transport 1, SentBy Port 5060
    *Jan 4 05:22:11.188: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone PST to SIP default timezone = GMT
    *Jan 4 05:22:11.192: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 20.20.20.2,Port 59570, Transport 1, SentBy Port 5060
    *Jan 4 05:22:11.196: //-1/DBD8E4E78012/SIP/Info/sipSPISetInfoFromRpid: Received current remote name: Goldie, current remote number: 7701
    *Jan 4 05:22:11.196: //-1/DBD8E4E78012/SIP/Info/sipSPISetInfoFromRpid: Received ;screen=no ;privacy=off -> Setting Octet3A 0x80, extended_privacy 0x00
    *Jan 4 05:22:11.200: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetGtdBody: No valid GTD body found.
    *Jan 4 05:22:11.200: //-1/DBD8E4E78012/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
    *Jan 4 05:22:11.200: //-1/DBD8E4E78012/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x8561974C key=E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.2006601
    *Jan 4 05:22:11.200: //-1/DBD8E4E78012/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id
    *Jan 4 05:22:11.200: //-1/DBD8E4E78012/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: 6601
    *Jan 4 05:22:11.200: //-1/DBD8E4E78012/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: 7701
    *Jan 4 05:22:11.204: //-1/DBD8E4E78012/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name Goldie, number 7701, Calling oct3 0x00, oct_3a 0x80, Called number 6601
    *Jan 4 05:22:11.204: //-1/DBD8E4E78012/SIP/Info/sipSPIGetCallConfig: Peer tag 1 matched for incoming call
    *Jan 4 05:22:11.208: //-1/DBD8E4E78012/SIP/Info/sipSPIGetCallConfig: Not using Voice Class Codec
    *Jan 4 05:22:11.208: //-1/DBD8E4E78012/SIP/Info/sipSPIContinueNewMsgInvite: Calling name Goldie, number 7701, Calling oct3 0x00, oct_3a 0x80, ext_priv 0x00, Called number 6601, oct3 0x00
    *Jan 4 05:22:11.208: //-1/DBD8E4E78012/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE
    *Jan 4 05:22:11.208: //-1/DBD8E4E78012/SIP/Info/sipSPIContinueNewMsgInvite: Requires reliable-provisional support
    *Jan 4 05:22:11.208: //-1/DBD8E4E78012/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
    *Jan 4 05:22:11.208: //39/DBD8E4E78012/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
    *Jan 4 05:22:11.212: //39/DBD8E4E78012/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1
    *Jan 4 05:22:11.212: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :20, codecbytes: 160
    *Jan 4 05:22:11.212: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
    *Jan 4 05:22:11.212: //39/DBD8E4E78012/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec g711alaw
    *Jan 4 05:22:11.212: //39/DBD8E4E78012/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
    *Jan 4 05:22:11.212: //39/DBD8E4E78012/SIP/Info/sipSPIDoDTMFRelayNegotiation: Requested DTMF-RELAY option(s) not found in Preferred DTMF-RELAY option list!
    *Jan 4 05:22:11.212: //39/DBD8E4E78012/SIP/Info/sipSPIStreamTypeAndDtmfRelay: DTMF Relay mode: Inband Voice
    *Jan 4 05:22:11.216: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0
    *Jan 4 05:22:11.216: //39/DBD8E4E78012/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
    *Jan 4 05:22:11.216: //39/DBD8E4E78012/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
    *Jan 4 05:22:11.216: //39/DBD8E4E78012/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
    payload_type=8, codec_bytes=160, codec=g711alaw, dtmf_relay=inband-voice
    stream_type=voice-only (0), dest_ip_address=20.20.20.2, dest_port=19012
    *Jan 4 05:22:11.216: //39/DBD8E4E78012/SIP/Media/sipSPIUpdCallWithSdpInfo:
    Preferred Codec : g711alaw, bytes :160
    Preferred DTMF relay : h245-alphanumeric
    Preferred NTE payload : 101
    Early Media : No
    Delayed Media : No
    Bridge Done : No
    New Media : No
    DSP DNLD Reqd : No


    *Jan 4 05:22:11.216: //39/DBD8E4E78012/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.200
    *Jan 4 05:22:11.220: //39/DBD8E4E78012/SIP/Info/sipSPI_ipip_report_media_to_peer:
    callId 39 peer 0 flags 0x201
    *Jan 4 05:22:11.220: //39/DBD8E4E78012/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    CallID 39, sdp 0x884BECCC channels 0x8561B2FC
    *Jan 4 05:22:11.220: //39/DBD8E4E78012/SIP/Info/copy_channels:
    callId 39 size 0 ptr 0x8816A250)
    *Jan 4 05:22:11.220: //39/DBD8E4E78012/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Hndl ptype 8 mline 1
    *Jan 4 05:22:11.220: //39/DBD8E4E78012/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711alaw
    *Jan 4 05:22:11.220: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
    *Jan 4 05:22:11.220: //39/DBD8E4E78012/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
    *Jan 4 05:22:11.220: //39/DBD8E4E78012/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
    Adding codec 6 ptype 8 time 20, bytes 160 as channel 0 mline 1 ss 0 20.20.20.2:19012
    *Jan 4 05:22:11.224: //39/DBD8E4E78012/SIP/Info/sipSPI_ipip_report_media_to_peer:
    Report initial call media
    *Jan 4 05:22:11.224: //39/DBD8E4E78012/SIP/Info/copy_channels:
    callId 39 size 80 ptr 0x84AB59F0)
    *Jan 4 05:22:11.224: //39/DBD8E4E78012/SIP/Info/sipSPI_ipip_report_media_to_peer:
    CCSIP: Unable to report channel ind
    *Jan 4 05:22:11.224: //39/DBD8E4E78012/SIP/Media/sipSPIUpdCallWithSdpInfo:
    Stream type : voice-only
    Media line : 1
    State : STREAM_ADDING (2)
    Callid : -1
    Negotiated Codec : g711alaw, bytes :160
    Negotiated DTMF relay : inband-voice
    Negotiated NTE payload : 0
    Negotiated CN payload : 0
    Media Srce Addr/Port : 192.168.1.200:0
    Media Dest Addr/Port : 20.20.20.2:19012


    *Jan 4 05:22:11.224: //39/DBD8E4E78012/SIP/Info/sipSPIHandleInviteMedia:
    Negotiated Codec : g711alaw, bytes :160
    Preferred Codec : g711alaw, bytes :160
    Preferred DTMF relay 1 : 3
    Preferred DTMF relay 2 : 0
    Negotiated DTMF relay : 0
    Preferred and Negotiated NTE payloads: 101 0
    Preferred and Negotiated NSE payloads: 100 0
    Preferred and Negotiated Modem Relay: 0 0
    Preferred and Negotiated Modem Relay GwXid: 1 0


    *Jan 4 05:22:11.228: //39/DBD8E4E78012/SIP/Info/sipSPIDoQoSNegotiation: SDP body with media description
    *Jan 4 05:22:11.228: //39/DBD8E4E78012/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
    *Jan 4 05:22:11.228: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17184 for stream 1
    *Jan 4 05:22:11.228: //39/DBD8E4E78012/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=17184
    *Jan 4 05:22:11.232: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
    *Jan 4 05:22:11.232: //39/DBD8E4E78012/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 17184
    *Jan 4 05:22:11.232: //39/DBD8E4E78012/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    *Jan 4 05:22:11.232: //39/DBD8E4E78012/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
    *Jan 4 05:22:11.232: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateRawMsg: No GTD passed.
    *Jan 4 05:22:11.236: //39/DBD8E4E78012/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS
    *Jan 4 05:22:11.236: //39/DBD8E4E78012/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
    *Jan 4 05:22:11.236: //39/DBD8E4E78012/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x8561974C key=E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200C51530-236B
    *Jan 4 05:22:11.236: //39/DBD8E4E78012/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 27 to table
    *Jan 4 05:22:11.240: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: msg=0x887481D0, addr=20.20.20.2, port=59570, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x00000000
    *Jan 4 05:22:11.240: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
    *Jan 4 05:22:11.240: //39/DBD8E4E78012/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
    *Jan 4 05:22:11.240: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x887481D0, addr=20.20.20.2, port=5060, connId=0 for UDP
    *Jan 4 05:22:11.244: //39/DBD8E4E78012/SIP/State/sipSPIChangeState: 0x8561974C : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_RECD_INVITE, SUBSTATE_NONE)
    *Jan 4 05:22:11.244: //39/DBD8E4E78012/SIP/Info/sipSPIProcessContactInfo: Previous Hop 20.20.20.2:5060
    *Jan 4 05:22:11.248: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
    *Jan 4 05:22:11.252: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler:
    *Jan 4 05:22:11.256: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 13
    *Jan 4 05:22:11.256: //39/DBD8E4E78012/SIP/Info/ccsip_event_handler:
    ccsip_event_handler: peer ID 40 chans 0x8578F670 event 138 flags 0x12020038 0x601 data 0x8578F670
    *Jan 4 05:22:11.256: //39/DBD8E4E78012/SIP/Info/ccsip_event_handler:
    ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 40 chans 0x8578F670 event 138 flags 0x12020038 0x601 data 0x8578F670
    *Jan 4 05:22:11.256: //39/DBD8E4E78012/SIP/Info/ccsip_event_handler:
    ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 40 chans 0x8578F670 event 138 flags 0x12020038 0x601 data 0x8578F670, type = 6
    *Jan 4 05:22:11.256: //39/DBD8E4E78012/SIP/Info/ccsip_event_handler:
    ccsip_event_handler: !!!!!CC_EV_H245_SET_MODE: not clearing VTSP flag
    *Jan 4 05:22:11.256: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED
    *Jan 4 05:22:11.268: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 20.20.20.2:5060;branch=z9hG4bK919B5
    From: "Goldie" <sip:7701@20.20.20.2>;tag=6C27D4-751
    To: <sip:6601@192.168.1.200>;tag=C51530-236B
    Date: Sun, 04 Jan 2015 05:22:11 GMT
    Call-ID: E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    Timestamp: 1014987515
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 101 INVITE
    Allow-Events: telephone-event
    Content-Length: 0






    *Jan 4 05:22:11.276: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROGRESS
    *Jan 4 05:22:11.276: //39/DBD8E4E78012/SIP/Info/ccsip_bridge: confID = 9, srcCallID = 39, dstCallID = 40
    *Jan 4 05:22:11.276: //39/DBD8E4E78012/SIP/Info/sipSPIUupdateCcCallIds: Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 39/40
    *Jan 4 05:22:11.276: //39/DBD8E4E78012/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=-1, new streamcallid=39
    *Jan 4 05:22:11.276: //39/DBD8E4E78012/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
    *Jan 4 05:22:11.276: //39/DBD8E4E78012/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 39) to the VOIP RTP library
    *Jan 4 05:22:11.280: //39/DBD8E4E78012/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.200
    *Jan 4 05:22:11.280: //39/DBD8E4E78012/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
    *Jan 4 05:22:11.280: //39/DBD8E4E78012/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
    laddr = 192.168.1.200, lport = 17184, raddr = 20.20.20.2, rport=19012, do_rtcp=TRUE
    src_callid = 39, dest_callid = 40, stream type = voice-only, stream direction = SENDRECV
    media_ip_addr = 20.20.20.2
    *Jan 4 05:22:11.280: //39/DBD8E4E78012/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
    *Jan 4 05:22:11.280: //39/DBD8E4E78012/SIP/Media/sipSPIGetNewLocalMediaDirection:
    New Remote Media Direction = SENDRECV
    Present Local Media Direction = SENDRECV
    New Local Media Direction = SENDRECV
    retVal = 0


    *Jan 4 05:22:11.288: //39/DBD8E4E78012/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=39, current_seq_num=0x1049
    *Jan 4 05:22:11.288: //39/DBD8E4E78012/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=39, current_seq_num=0x14E4
    *Jan 4 05:22:11.288: //39/DBD8E4E78012/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g711alaw, Bytes=160
    *Jan 4 05:22:11.288: //39/DBD8E4E78012/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0
    *Jan 4 05:22:11.292: //39/DBD8E4E78012/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE_AND_OOB
    *Jan 4 05:22:11.292: //39/DBD8E4E78012/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=0, from CLI config=0
    *Jan 4 05:22:11.292: //39/DBD8E4E78012/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...
    *Jan 4 05:22:11.292: //39/DBD8E4E78012/SIP/Info/sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed
    *Jan 4 05:22:11.292: //39/DBD8E4E78012/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap list
    *Jan 4 05:22:11.292: //39/DBD8E4E78012/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled
    *Jan 4 05:22:11.292: //39/DBD8E4E78012/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
    *Jan 4 05:22:11.292: //39/DBD8E4E78012/SIP/Media/sipSPISetStreamInfo: 1 Active Streams
    *Jan 4 05:22:11.296: //39/DBD8E4E78012/SIP/Media/sipSPISetStreamInfo: Adding stream type (voice-only) from media line 1
    *Jan 4 05:22:11.296: //39/DBD8E4E78012/SIP/Media/sipSPISetStreamInfo:
    caps.stream_count=1,caps.stream[0].stream_type=0x1, caps.stream_list.xmitFunc=
    *Jan 4 05:22:11.296: //39/DBD8E4E78012/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=
    *Jan 4 05:22:11.296: //39/DBD8E4E78012/SIP/Media/sipSPISetStreamInfo: 0x887A6CA0 (gccb)
    *Jan 4 05:22:11.296: //39/DBD8E4E78012/SIP/Info/ccsip_caps_ind: Load DSP with codec : g711alaw, Bytes=160
    *Jan 4 05:22:11.296: //39/DBD8E4E78012/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->flags_ipip = 0x603
    *Jan 4 05:22:11.300: //39/DBD8E4E78012/SIP/Info/ccsip_caps_ack: Set forking flag to 0x0
    *Jan 4 05:22:11.300: //39/DBD8E4E78012/SIP/Info/sipSPIValidateGtd: No rawMsg from CCAPI
    *Jan 4 05:22:11.300: //39/DBD8E4E78012/SIP/Info/sipSPISendInviteResponse183: Session Type is Media/Qos/Security/RTR SDP body is attached
    *Jan 4 05:22:11.308: //39/DBD8E4E78012/SIP/Transport/sipSPISendInviteResponse: Sending 183 Response to the Transport Layer
    *Jan 4 05:22:11.308: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: msg=0x887482F0, addr=20.20.20.2, port=59570, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x8105FF1C
    *Jan 4 05:22:11.308: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
    *Jan 4 05:22:11.308: //39/DBD8E4E78012/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
    *Jan 4 05:22:11.308: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x887482F0, addr=20.20.20.2, port=5060, connId=0 for UDP
    *Jan 4 05:22:11.308: //39/DBD8E4E78012/SIP/Info/sentInviteResponse18x: Sent a 18x Response
    *Jan 4 05:22:11.308: //39/DBD8E4E78012/SIP/Info/setPrackLockIfNeeded: Transaction active. Facilities will be queued.
    *Jan 4 05:22:11.316: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 20.20.20.2:5060;branch=z9hG4bK919B5
    From: "Goldie" <sip:7701@20.20.20.2>;tag=6C27D4-751
    To: <sip:6601@192.168.1.200>;tag=C51530-236B
    Date: Sun, 04 Jan 2015 05:22:11 GMT
    Call-ID: E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    Timestamp: 1014987515
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 101 INVITE
    Require: 100rel
    RSeq: 7694
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
    Allow-Events: telephone-event
    Contact: <sip:6601@192.168.1.200:5060>
    Content-Disposition: session;handling=required
    Content-Type: application/sdp
    Content-Length: 194


    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3190 4529 IN IP4 192.168.1.200
    s=SIP Call
    c=IN IP4 192.168.1.200
    t=0 0
    m=audio 17184 RTP/AVP 8
    c=IN IP4 192.168.1.200
    a=rtpmap:8 PCMA/8000
    a=ptime:20


    *Jan 4 05:22:11.368: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 20.20.20.2:59570
    *Jan 4 05:22:11.368: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
    *Jan 4 05:22:11.368: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    PRACK sip:6601@192.168.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 20.20.20.2:5060;branch=z9hG4bKA2409
    From: "Goldie" <sip:7701@20.20.20.2>;tag=6C27D4-751
    To: <sip:6601@192.168.1.200>;tag=C51530-236B
    Date: Fri, 01 Mar 2002 12:58:35 GMT
    Call-ID: E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    CSeq: 102 PRACK
    RAck: 7694 101 INVITE
    Max-Forwards: 70
    Content-Length: 0






    *Jan 4 05:22:11.372: //39/DBD8E4E78012/SIP/Info/sipSPIFindCcbUASReqTable: *****CCB found in UAS Request table. ccb=0x8561974C
    *Jan 4 05:22:11.376: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 20.20.20.2,Port 59570, Transport 1, SentBy Port 5060
    *Jan 4 05:22:11.376: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone PST to SIP default timezone = GMT
    *Jan 4 05:22:11.380: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 20.20.20.2,Port 59570, Transport 1, SentBy Port 5060
    *Jan 4 05:22:11.380: //39/DBD8E4E78012/SIP/Info/act_recdinvite_new_message_request: Transaction Complete. Lock on Facilities released.
    *Jan 4 05:22:11.384: //39/DBD8E4E78012/SIP/Transport/sipSPISendPrackResponse: Sending PRACK Response to the transport layer
    *Jan 4 05:22:11.384: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: msg=0x887486E0, addr=20.20.20.2, port=59570, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x00000000
    *Jan 4 05:22:11.384: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
    *Jan 4 05:22:11.384: //39/DBD8E4E78012/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
    *Jan 4 05:22:11.384: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x887486E0, addr=20.20.20.2, port=5060, connId=0 for UDP
    *Jan 4 05:22:11.388: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 20.20.20.2:5060;branch=z9hG4bKA2409
    From: "Goldie" <sip:7701@20.20.20.2>;tag=6C27D4-751
    To: <sip:6601@192.168.1.200>;tag=C51530-236B
    Date: Sun, 04 Jan 2015 05:22:11 GMT
    Call-ID: E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 102 PRACK
    Content-Length: 0






    *Jan 4 05:22:17.096: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_CONNECT
    *Jan 4 05:22:17.100: //39/DBD8E4E78012/SIP/Info/sipSPIValidateGtd: No rawMsg from CCAPI
    *Jan 4 05:22:17.108: //39/DBD8E4E78012/SIP/Transport/sipSPISendInviteResponse: Sending 200OK Response to the Transport Layer
    *Jan 4 05:22:17.108: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: msg=0x887486E0, addr=20.20.20.2, port=59570, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x81060204
    *Jan 4 05:22:17.108: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
    *Jan 4 05:22:17.108: //39/DBD8E4E78012/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
    *Jan 4 05:22:17.108: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x887486E0, addr=20.20.20.2, port=5060, connId=0 for UDP
    *Jan 4 05:22:17.112: //39/DBD8E4E78012/SIP/Info/sentInviteResponse200: Sent 200Ok for Invite in state STATE_RECD_INVITE
    *Jan 4 05:22:17.112: //-1/xxxxxxxxxxxx/SIP/Info/sentInviteResponse200: Transaction active. Facilities will be queued.
    *Jan 4 05:22:17.112: //39/DBD8E4E78012/SIP/State/sipSPIChangeState: 0x8561974C : State change from (STATE_RECD_INVITE, SUBSTATE_NONE) to (STATE_SENT_SUCCESS, SUBSTATE_NONE)
    *Jan 4 05:22:17.112: //39/DBD8E4E78012/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
    *Jan 4 05:22:17.116: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 20.20.20.2:5060;branch=z9hG4bK919B5
    From: "Goldie" <sip:7701@20.20.20.2>;tag=6C27D4-751
    To: <sip:6601@192.168.1.200>;tag=C51530-236B
    Date: Sun, 04 Jan 2015 05:22:11 GMT
    Call-ID: E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    Timestamp: 1014987515
    Server: Cisco-SIPGateway/IOS-12.x
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
    Supported: replaces
    Allow-Events: telephone-event
    Contact: <sip:6601@192.168.1.200:5060>
    Content-Type: application/sdp
    Content-Length: 194


    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3190 4529 IN IP4 192.168.1.200
    s=SIP Call
    c=IN IP4 192.168.1.200
    t=0 0
    m=audio 17184 RTP/AVP 8
    c=IN IP4 192.168.1.200
    a=rtpmap:8 PCMA/8000
    a=ptime:20


    *Jan 4 05:22:17.160: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 20.20.20.2:59570
    *Jan 4 05:22:17.160: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
    *Jan 4 05:22:17.160: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:6601@192.168.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 20.20.20.2:5060;branch=z9hG4bKB2144
    From: "Goldie" <sip:7701@20.20.20.2>;tag=6C27D4-751
    To: <sip:6601@192.168.1.200>;tag=C51530-236B
    Date: Fri, 01 Mar 2002 12:58:35 GMT
    Call-ID: E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    Max-Forwards: 70
    CSeq: 101 ACK
    Content-Length: 0






    *Jan 4 05:22:17.164: //39/DBD8E4E78012/SIP/Info/sipSPIFindCcbUASReqTable: *****CCB found in UAS Request table. ccb=0x8561974C
    *Jan 4 05:22:17.164: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 20.20.20.2,Port 59570, Transport 1, SentBy Port 5060
    *Jan 4 05:22:17.168: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone PST to SIP default timezone = GMT
    *Jan 4 05:22:17.168: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 20.20.20.2,Port 59570, Transport 1, SentBy Port 5060
    *Jan 4 05:22:17.168: //39/DBD8E4E78012/SIP/Info/act_sentsucc_new_message_request: Transaction Complete. Lock on Facilities released.
    *Jan 4 05:22:17.168: //39/DBD8E4E78012/SIP/State/sipSPIChangeState: 0x8561974C : State change from (STATE_SENT_SUCCESS, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_NONE)
    *Jan 4 05:22:17.168: //39/DBD8E4E78012/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x8561974C
    State of The Call : STATE_ACTIVE
    TCP Sockets Used : NO
    Calling Number : 7701
    Called Number : 6601
    Source IP Address (Sig ): 192.168.1.200
    Destn SIP Req Addr:Port : 20.20.20.2:5060
    Destn SIP Resp Addr:Port : 20.20.20.2:59570
    Destination Name : 20.20.20.2


    *Jan 4 05:22:17.172: //39/DBD8E4E78012/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream : 1
    Negotiated Codec : g711alaw
    Negotiated Codec Bytes : 160
    Negotiated Dtmf-relay : 0
    Dtmf-relay Payload : 0
    Source IP Address (Media): 192.168.1.200
    Source IP Port (Media): 17184
    Destn IP Address (Media): 20.20.20.2
    Destn IP Port (Media): 19012
    Orig Destn IP Address:Port (Media): 0.0.0.0:0


    *Jan 4 05:22:17.172: //39/DBD8E4E78012/SIP/Info/sipSPICreateAndStartRtpTimer:
    *Jan 4 05:22:17.172: //39/DBD8E4E78012/SIP/Info/sipSPICreateAndStartRtpTimer: Media Inactivity Timer is disabled.
    *Jan 4 05:22:17.172: //39/DBD8E4E78012/SIP/Info/sipSPIProcessHoldTimerForCall: Media IP Addr 20.20.20.2, RTCP Type 3
    *Jan 4 05:22:17.172: //39/DBD8E4E78012/SIP/Info/sipSPIStopHoldTimer: Stopping hold timer
    CME_A_#
    CME_A_#
    CME_A_#
    *Jan 4 05:22:57.592: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 20.20.20.2:59570
    *Jan 4 05:22:57.596: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
    *Jan 4 05:22:57.596: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    BYE sip:6601@192.168.1.200:5060 SIP/2.0
    Via: SIP/2.0/UDP 20.20.20.2:5060;branch=z9hG4bKC6DD
    From: "Goldie" <sip:7701@20.20.20.2>;tag=6C27D4-751
    To: <sip:6601@192.168.1.200>;tag=C51530-236B
    Date: Fri, 01 Mar 2002 12:58:35 GMT
    Call-ID: E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Max-Forwards: 70
    Timestamp: 1014987561
    CSeq: 103 BYE
    Reason: Q.850;cause=16
    Content-Length: 0






    *Jan 4 05:22:57.600: //39/DBD8E4E78012/SIP/Info/sipSPIFindCcbUASReqTable: *****CCB found in UAS Request table. ccb=0x8561974C
    *Jan 4 05:22:57.600: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 20.20.20.2,Port 59570, Transport 1, SentBy Port 5060
    *Jan 4 05:22:57.600: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone PST to SIP default timezone = GMT
    *Jan 4 05:22:57.604: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 20.20.20.2,Port 59570, Transport 1, SentBy Port 5060
    *Jan 4 05:22:57.604: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[39], src[2]
    *Jan 4 05:22:57.604: //39/DBD8E4E78012/SIP/Info/sipSPIStopHoldTimer: Stopping hold timer
    *Jan 4 05:22:57.604: //39/DBD8E4E78012/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for incoming call
    *Jan 4 05:22:57.604: //39/DBD8E4E78012/SIP/State/sipSPIChangeState: 0x8561974C : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
    *Jan 4 05:22:57.612: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
    *Jan 4 05:22:57.620: //39/DBD8E4E78012/SIP/Transport/sipSPISendByeResponse: Sending BYE Response to the transport layer
    *Jan 4 05:22:57.620: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: msg=0x88748410, addr=20.20.20.2, port=59570, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x8106070C
    *Jan 4 05:22:57.620: //39/DBD8E4E78012/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
    *Jan 4 05:22:57.620: //39/DBD8E4E78012/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
    *Jan 4 05:22:57.620: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x88748410, addr=20.20.20.2, port=5060, connId=0 for UDP
    *Jan 4 05:22:57.620: //39/DBD8E4E78012/SIP/Info/sentByeResponse: Sent 200ok to the BYE, tearing down the call
    *Jan 4 05:22:57.624: //39/DBD8E4E78012/SIP/Info/sipSPIIcpifUpdate: CallState: 4 Playout: 39210 DiscTime:1296244 ConnTime 1292193
    *Jan 4 05:22:57.624: //39/DBD8E4E78012/SIP/State/sipSPIChangeState: 0x8561974C : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)
    *Jan 4 05:22:57.628: //39/DBD8E4E78012/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x8561974C
    State of The Call : STATE_DEAD
    TCP Sockets Used : NO
    Calling Number : 7701
    Called Number : 6601
    Source IP Address (Sig ): 192.168.1.200
    Destn SIP Req Addr:Port : 20.20.20.2:5060
    Destn SIP Resp Addr:Port : 20.20.20.2:59570
    Destination Name : 20.20.20.2


    *Jan 4 05:22:57.628: //39/DBD8E4E78012/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream : 1
    Negotiated Codec : g711alaw
    Negotiated Codec Bytes : 160
    Negotiated Dtmf-relay : 0
    Dtmf-relay Payload : 0
    Source IP Address (Media): 192.168.1.200
    Source IP Port (Media): 17184
    Destn IP Address (Media): 20.20.20.2
    Destn IP Port (Media): 19012
    Orig Destn IP Address:Port (Media): 0.0.0.0:0


    *Jan 4 05:22:57.628: //39/DBD8E4E78012/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC) : 16
    Disconnect Cause (SIP) : 200


    *Jan 4 05:22:57.628: //39/DBD8E4E78012/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 27
    *Jan 4 05:22:57.636: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 20.20.20.2:5060;branch=z9hG4bKC6DD
    From: "Goldie" <sip:7701@20.20.20.2>;tag=6C27D4-751
    To: <sip:6601@192.168.1.200>;tag=C51530-236B
    Date: Sun, 04 Jan 2015 05:22:57 GMT
    Call-ID: E01FDCD9-2C4A11D6-8015CBD3-FD31EA58@65.65.65.200
    Server: Cisco-SIPGateway/IOS-12.x
    Timestamp: 1014987561
    Content-Length: 0
    CSeq: 103 BYE


    CME_A_#
    CME_A_#
    CME_A_#
    CME_A_#
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    #27

    Default Gonna have to remember this one

    So opening UDP port 5060 on a ACL allows the voice stream to go through on the FXS analog SIP RTP connection ?

    And that is both ways?

    Ways cool!!!
    Last edited by Jollycork; 01-05-2015 at 03:10 AM.
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    #28
    So any warbling or delay on the audio over the WAN? Just curious......
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  5. Senior Member JeanM's Avatar
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    #29
    VoIP quality is fine, no issues that I can report. Now, my WAN is basically a LAB "wan" made of a couple routers. I'll play with different codecs once I have more time to compare as well. Yeah, so the 2600xm routers have the Voice modules and DSP with the FXS cards for the analog phones, from there it's VoIP on the other side of the router

    My next plan of action is to extend this lab over the internet, by finding another person who is willing to configure a CME router and dial-peer. That way it will really be over internet WAN .

    Let me know if you are interested to give it a shot.
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    #30
    So this is strictly FXS/FXO analog phone on CME where CME runs on the 2620xm & your using the single FA 0/0 as your WAN link out on the 2620xm? over SIP?

    just curious again.... I remember you labbing SIP voip a while back
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  7. Senior Member JeanM's Avatar
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    #31
    This lab scenario is just as shown in the diagram in the 2nd post . Basically it's analog phone on fxs port / voice module in 2600xm running cme connected to 1st 2621 running as edge to 1st 3640 running as "service provider" wan link to 2nd 3640 as another SP, connected to 2nd 2621 running as remote edge, and connected to another 2600xm/fxs/analog phone. I basically wanted to "simulate" two branch locations , each one behind it's own internet connection with NAT on each side...

    SIP trunk I do have configured on one of the 2811, that was another lab exercise
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    #32
    Quote Originally Posted by JeanM View Post
    My next plan of action is to extend this lab over the internet, by finding another person who is willing to configure a CME router and dial-peer. That way it will really be over internet WAN .

    Let me know if you are interested to give it a shot.
    I would like to give it a shot. I'm still at the beginning of my voice studies, but I should be able to mirror your config.
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  9. Senior Member JeanM's Avatar
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    #33
    davenull - cool! PM me when your side is configured and we'll exchange IP addresses and dial-peer info.
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    #34

    Default The suspence is killing me

    So were you two able to get the CME's to talk to one another?
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  11. Senior Member JeanM's Avatar
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    #35
    Haha Yeah I am waiting for davenull to PM me once his side is ready so we can test this out. Or if someone else wants to give it a shot, just configure your CME the same way and PM to exchange the internet facing ip and dial-peer info.
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    #36
    I just set up an 1841 as my edge router and re-purposed my SOHO router to act as a WAP - can't leave folks at home without internet

    I'll set up the 2811 with CME tomorrow after work.

    Sorry to keep you guys in suspense. The work has been busy - I was sidetracked into having to study powershell.
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  13. Senior Member JeanM's Avatar
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    #37
    Sweet. I did something similar with 1841 + tplink ap .

    I'll ping you tomorrow after work.
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  14. Senior Member JeanM's Avatar
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    #38
    Hey, I must have missed you when you were up. Ping me your email address, as this forum didn't notify me when you send me an IM.
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  15. Senior Member JeanM's Avatar
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    #39
    So Update.

    Myself and davenull worked on this, and got voip to voip calls to work between two CME routers each one behind edge/nat router over the internet.

    So, interestingly, we got h323 calls to work but could not get sipv2 to work. Another thing is we could not get g729 codec to work either, but g711ulaw and alaw and also g728 worked for both of us.

    In my lab, I got sipv2 working just fine using the same cme-nat-wan-nat-cme topology.

    Anyways, so we got the calls to work (including caller-id) but would like to know why g729 didn't work with h323 or sip.

    Now that I think about this, maybe sipv2 would work if we tried g711/g728 on it like we did with h323!


    Also, did not expect that #debug ccsip calls would show h323 calls ? Example below.
    Does the h323 fall under SIP realm , since it was popular before SIP etc?




    Jan 27 04:01:50.262: //35/ECD9769F8028/SIP/Call/sipSPICallInfo:
    The Call Setup Information is:
    Call Control Block (CCB) : 0x83A9B56C
    State of The Call : STATE_DEAD
    TCP Sockets Used : NO
    Calling Number : 2001
    Called Number : 7701
    Source IP Address (Sig ): 192.168.1.99
    Destn SIP Req Addr:Port : 24.240.xx.xx:1079
    Destn SIP Resp Addr:Port : 24.240.xx.xx:55375
    Destination Name : 24.240.xx.xx


    Jan 27 04:01:50.266: //35/ECD9769F8028/SIP/Call/sipSPIMediaCallInfo:
    Number of Media Streams: 1
    Media Stream : 1
    Negotiated Codec : g711alaw
    Negotiated Codec Bytes : 160
    Negotiated Dtmf-relay : 0
    Dtmf-relay Payload : 0
    Source IP Address (Media): 192.168.1.99
    Source IP Port (Media): 17090
    Destn IP Address (Media): 24.240.xx.xx
    Destn IP Port (Media): 16620
    Orig Destn IP Address:Port (Media): 0.0.0.0:0


    Jan 27 04:01:50.266: //35/ECD9769F8028/SIP/Call/sipSPICallInfo:
    Disconnect Cause (CC) : 16
    Disconnect Cause (SIP) : 200
    Last edited by JeanM; 01-27-2015 at 04:54 AM.
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  16. Senior Member
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    #40

    Default cool it worked....

    not sure why that particular codec wouldn't work,
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  17. Senior Member
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    #41

    Default Which SIP provider are you using?

    Just curious .
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  18. Senior Member JeanM's Avatar
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    #42
    flowroute, per another forum member recommendation
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  19. Senior Member
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    #43
    Jean, did I need to sign up for a sip provider too? I never signed up for any.
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  20. Senior Member JeanM's Avatar
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    #44
    No, that's for a different lab. That way you can connect your router with external sip provider like flowroute and make calls outside to PSTN/cellular or if you purchase DID(s) can make calls into your lab.
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  21. Senior Member
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    #45
    hum maybe something to try to work on translation no's and DIDs in CUCM...

    I just hook up my CUCM to POTS and use a connection PLANAR to Unity...

    might be fun to use SIP with DIDs

    If I get SIP setup into CUCM with DIDs [maybe caller IDs] I'll let you both know, we can test calling over SIP into CUCM on DIDs [provided the wife oks leaving the lab on (electric bill goes up, up, and away if I leave it on).

    I've got 3 remote locations setup over frame to my HQ CUCM, be fun to see if you guys and call into the remote sites over SIP
    Last edited by Jollycork; 02-02-2015 at 04:15 AM.
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