Cisco CME i dolazni pozivi

U pokušaju da namestim SIP trunk na poslu krenuo sam da rekonfigurišem Cisco 2811 ruter. Do sada sam koristio ISDN BRI pa je ruter ve
bio delimično konfigurisan. Prilagodio sam sve što sam smatrao da treba i uz malu pomoć provajdera uspostavio odlazne pozive. Međutim, sa dolaznim pozivima situacija se zapetljala: pozivi stižu do mog rutera ali se ne rutiraju prema telefonima. Probao sam i sa toroute skriptom koja mi je na Telekomovom SIP trunku i Cisco 2911 ruteru sa IOS 15.1 rešila problem. Međutim kod mene ne funkcioniše. Ja imam stari IOS 12.4 i CCME 4.1 (12.4(15)T / CME 4.1(0))

Kada sa mobilnog pozovem svoj broj iz seta za SIP dobijem ovaj debug:

Feb 7 07:31:31.413: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.135.1.17:5060
*Feb 7 07:31:31.413: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
*Feb 7 07:31:31.413: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
*Feb 7 07:31:31.413: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected];user=phone SIP/2.0
Max-Forwards: 68
Via: SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq
To: <sip:[email protected];user=phone>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9d52abc7155773
Call-ID: [email protected]
CSeq: 456411359 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:10.135.1.17;transport=udp;lr>
Min-Se: 900
P-Asserted-Identity: <sip:[email protected];user=phone>
Session-Expires: 7200
Supported: histinfo
Content-Type: application/sdp
Content-Length: 231
Allow: INVITE, CANCEL, ACK, BYE
Accept: application/sdp

v=0
o=VpnImsGw 645000 645000 IN IP4 10.135.1.17
s=VpnImsGw_Session
c=IN IP4 10.135.1.113
t=0 0
m=audio 64244 RTP/AVP 8 99 18 96
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=rtpmap:96 AMR/8000
a=fmtp:96 octet-align=1

*Feb 7 07:31:31.413: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
*Feb 7 07:31:31.413: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x4726EDE4) with key=[152] to table
*Feb 7 07:31:31.417: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.135.1.17,Port 5060, Tran
sport 1, SentBy Port 5060
*Feb 7 07:31:31.417: //-1/3A0C10F68F0E/SIP/State/sipSPIChangeState: 0x4726EDE4 : State change from (STATE_NONE, SUBSTATE_NONE) to (S
TATE_IDLE, SUBSTATE_NONE)
*Feb 7 07:31:31.417: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.135.1.17,Port 5060, Tran
sport 1, SentBy Port 5060
*Feb 7 07:31:31.417: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.135.1.17,Port 5060, Tran
sport 1, SentBy Port 5060
*Feb 7 07:31:31.417: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container
*Feb 7 07:31:31.417: //-1/3A0C10F68F0E/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
*Feb 7 07:31:31.417: //-1/3A0C10F68F0E/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4726EDE4 key=ERICSSONBTK_TERM_b2e1d347359
[email protected]+38121abcd811
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: +38121abcd811
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: +38164abcd690
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name , number +38164abcd690, Calli
ng oct3 0x00, oct_3a 0x00, Called number +38121abcd811
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIGetCallConfig: Non dial peer leg - using RTP Supported Codecs
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 18
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 0
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 8
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 4
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 2
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 15
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 3
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIContinueNewMsgInvite: Calling name , number +38164abcd690, Calling oct3 0x00, o
ct_3a 0x80, ext_priv 0x00, Called number +38121abcd811, oct3 0x00
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp
NONE, next_tgrp NONE
*Feb 7 07:31:31.421: //-1/3A0C10F68F0E/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPINegotiateSessionExpires:
Session-Expires value: 7200 refresher: 3
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPINegotiateSessionExpires: Min-SE Header: 900
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Error/sipSPIProcessCallInfoHeader: Call-Info header with for Unsolicited Notify Absent,D
isabling Unsolicited Notifies
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload
for m-line 1
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPIDoPtimeNegotiation: No ptime present or multiple ptime attributes that can't
be handled
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(99) reserved.
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPIReserveRtpNtePayload: Reserved the new NTE payload type 99
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF relay option
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of partial named event(NE) match in fmtp list of
events.
*Feb 7 07:31:31.425: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple directio
n attributes that can't be handled for m-line:1 and num-a-lines:0
*Feb 7 07:31:31.425: //1579/3A0C10F68F0E/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
payload_type=8, codec_bytes=160, codec=g711alaw, dtmf_relay=rtp-nte
stream_type=voice+dtmf (1), dest_ip_address=10.135.1.113, dest_port=64244
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/State/sipSPIChangeStreamState: Stream (callid = -1) State changed from (STREAM_DEAD) t
o (STREAM_ADDING)
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g729r8, bytes :20
Preferred DTMF relay : rtp-nte
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No

*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.89.15.xxx
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_report_media_to_peer:
callId 1579 peer 0 flags 0x201
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CallID 1579, sdp 0x477B6AD8 channels 0x4726FEB4
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 8 mline 1
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711alaw
*Feb 7 07:31:31.429: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_pti
me=0,stream->negotiated_codec_bytes=160, coverted ptime=20 stream->mline_index=1, media_ndx=1
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 6 ptype 8 time 20, bytes 160 as channel 0 mline 1 ss 0 10.135.1.113:64244
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 99 mline 1
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: setting ipip_caps DTMF to RFC2833: callid = 15
79, dtmf = 6
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 18 mline 1
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Media/sipSPISelectCodecVersion: Codec (g729br8) is not in preferred list
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: An exact codec match not configured, using int
eroperable codec g729r8
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g729r8
SIP: (1579) Attribute ptime, level 1 instance 1 not found.
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation NOT done, get ptime from sdp
: ptime=0, media_ndx=1
*Feb 7 07:31:31.429: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g729r8 ptime :0, codecbytes: 0
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Codec bytes 0, use default packet rate 20
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
*Feb 7 07:31:31.429: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 16 ptype 18 time 0, bytes 20 as channel 1 mline 1 ss 0 10.135.1.113:64244
*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 96 mline 1
*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_report_media_to_peer:
Report initial call media
*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Info/copy_channels:
callId 1579 size 208 ptr 0x45C945B0)
*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_report_media_to_peer:
CCSIP: Unable to report channel ind
*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Callid : -1
Negotiated Codec : g711alaw, bytes :160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated DTMF relay : rtp-nte
Negotiated NTE payload : 99 (tx), 99 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : 10.89.15.xxx:0
Media Dest Addr/Port : 10.135.1.113:64244

*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Info/sipSPIHandleInviteMedia:
Negotiated Codec : g711alaw, bytes :160
Preferred Codec : g729r8, bytes :20
Preferred DTMF relay 1 : 6
Preferred DTMF relay 2 : 0
Negotiated DTMF relay : 6
Preferred and Negotiated NTE payloads: 101 99
Preferred and Negotiated NSE payloads: 100 0
Preferred and Negotiated Modem Relay: 0 0
Preferred and Negotiated Modem Relay GwXid: 1 0

*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Info/sipSPIDoQoSNegotiation: SDP body with media description
*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
*Feb 7 07:31:31.433: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17230 for stream 1
*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=17230
*Feb 7 07:31:31.433: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
*Feb 7 07:31:31.433: //1579/3A0C10F68F0E/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 17230
*Feb 7 07:31:31.437: //1579/3A0C10F68F0E/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Negotiated method of dtmf relayand pyld: 6 99

*Feb 7 07:31:31.437: //1579/3A0C10F68F0E/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = ERICSSONBTK_TERM_b2e1d34
[email protected]
*Feb 7 07:31:31.437: //1579/3A0C10F68F0E/SIP/Info/ccsip_api_call_setup_ind: Headers from INVITE added to callInfo container
*Feb 7 07:31:31.437: ccDumpTdRequestDataSip:
*Feb 7 07:31:31.437: reqURI=sip:[email protected];user=phone
*Feb 7 07:31:31.437: calling_urip=sip:[email protected];user=phone
*Feb 7 07:31:31.437: url_dump_header_line_avpair:
*Feb 7 07:31:31.437: num_headers = 16
*Feb 7 07:31:31.437: headers[0].linep = From:<sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9d
52abc7155773, len = 101
*Feb 7 07:31:31.437: data.attr.datap = From:<sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9
d52abc7155773, len = 4
*Feb 7 07:31:31.437: data.value.datap = <sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9d52a
bc7155773, len = 96
*Feb 7 07:31:31.437: headers[1].linep = To:<sip:[email protected];user=phone>, len = 48
*Feb 7 07:31:31.437: data.attr.datap = To:<sip:[email protected];user=phone>, len = 2
*Feb 7 07:31:31.437: data.value.datap = <sip:[email protected];user=phone>, len = 45
*Feb 7 07:31:31.437: headers[2].linep = Via:SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq, len = 72
*Feb 7 07:31:31.441: data.attr.datap = Via:SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq, len = 3
*Feb 7 07:31:31.441: data.value.datap = SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq, len = 68
*Feb 7 07:31:31.441: headers[3].linep = Call-Id:[email protected], len = 69
*Feb 7 07:31:31.441: data.attr.datap = Call-Id:[email protected], len = 7
*Feb 7 07:31:31.441: data.value.datap = [email protected], len = 61
*Feb 7 07:31:31.441: headers[4].linep = Cseq:456411359 INVITE, len = 21
*Feb 7 07:31:31.441: data.attr.datap = Cseq:456411359 INVITE, len = 4
*Feb 7 07:31:31.441: data.value.datap = 456411359 INVITE, len = 16
*Feb 7 07:31:31.441: headers[5].linep = Contact:<sip:[email protected];transport=udp>, len = 45
*Feb 7 07:31:31.441: data.attr.datap = Contact:<sip:[email protected];transport=udp>, len = 7
*Feb 7 07:31:31.441: data.value.datap = <sip:[email protected];transport=udp>, len = 37
*Feb 7 07:31:31.441: headers[6].linep = Content-Length:231, len = 18
*Feb 7 07:31:31.441: data.attr.datap = Content-Length:231, len = 14
*Feb 7 07:31:31.441: data.value.datap = 231, len = 3
*Feb 7 07:31:31.441: headers[7].linep = Content-Type:application/sdp, len = 28
*Feb 7 07:31:31.441: data.attr.datap = Content-Type:application/sdp, len = 12
*Feb 7 07:31:31.441: data.value.datap = application/sdp, len = 15
*Feb 7 07:31:31.441: headers[8].linep = Record-Route:<sip:10.135.1.17;transport=udp;lr>, len = 47
*Feb 7 07:31:31.441: data.attr.datap = Record-Route:<sip:10.135.1.17;transport=udp;lr>, len = 12
*Feb 7 07:31:31.441: data.value.datap = <sip:10.135.1.17;transport=udp;lr>, len = 34
*Feb 7 07:31:31.441: headers[9].linep = Max-Forwards:68, len = 15
*Feb 7 07:31:31.441: data.attr.datap = Max-Forwards:68, len = 12
*Feb 7 07:31:31.441: data.value.datap = 68, len = 2
*Feb 7 07:31:31.441: headers[10].linep = Min-Se:900, len = 10
*Feb 7 07:31:31.441: data.attr.datap = Min-Se:900, len = 6
*Feb 7 07:31:31.441: data.value.datap = 900, len = 3
*Feb 7 07:31:31.441: headers[11].linep = P-Asserted-Identity:<sip:[email protected];user=phone>, len = 70
*Feb 7 07:31:31.441: data.attr.datap = P-Asserted-Identity:<sip:[email protected];user=phone>, len = 19
*Feb 7 07:31:31.441: data.value.datap = <sip:[email protected];user=phone>, len = 50
*Feb 7 07:31:31.441: headers[12].linep = Session-Expires:7200, len = 20
*Feb 7 07:31:31.441: data.attr.datap = Session-Expires:7200, len = 15
*Feb 7 07:31:31.441: data.value.datap = 7200, len = 4
*Feb 7 07:31:31.441: headers[13].linep = Supported:histinfo, len = 18
*Feb 7 07:31:31.441: data.attr.datap = Supported:histinfo, len = 9
*Feb 7 07:31:31.441: data.value.datap = histinfo, len = 8
*Feb 7 07:31:31.441: headers[14].linep = Allow:INVITE, CANCEL, ACK, BYE, len = 30
*Feb 7 07:31:31.441: data.attr.datap = Allow:INVITE, CANCEL, ACK, BYE, len = 5
*Feb 7 07:31:31.441: data.value.datap = INVITE, CANCEL, ACK, BYE, len = 24
*Feb 7 07:31:31.441: headers[15].linep = Accept:application/sdp, len = 22
*Feb 7 07:31:31.441: data.attr.datap = Accept:application/sdp, len = 6
*Feb 7 07:31:31.441: data.value.datap = application/sdp, len = 15num_bodies = 0
*Feb 7 07:31:31.441: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
*Feb 7 07:31:31.445: //1579/3A0C10F68F0E/SIP/Info/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing negotiated dtmf = 6,
*Feb 7 07:31:31.445: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateRawMsg: No GTD passed.
*Feb 7 07:31:31.445: //1579/3A0C10F68F0E/SIP/Info/ccsip_set_bearer_capability:
Bearer Capability: Speech (0x00)
*Feb 7 07:31:31.445: //1579/3A0C10F68F0E/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS
*Feb 7 07:31:31.445: //1579/3A0C10F68F0E/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 62B to table
*Feb 7 07:31:31.445: //1579/3A0C10F68F0E/SIP/Transport/sipSPITransportSendMessage: msg=0x47F81648, addr=10.135.1.17, port=5060, sentB
y_port=5060, is_req=0, transport=1, switch=0, callBack=0x00000000
*Feb 7 07:31:31.445: //1579/3A0C10F68F0E/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Feb 7 07:31:31.445: //1579/3A0C10F68F0E/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Feb 7 07:31:31.445: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x47F81648, addr=10.135.1.17,
port=5060, connId=0 for UDP
*Feb 7 07:31:31.445: //1579/3A0C10F68F0E/SIP/State/sipSPIChangeState: 0x4726EDE4 : State change from (STATE_IDLE, SUBSTATE_NONE) to
(STATE_RECD_INVITE, SUBSTATE_NONE)
*Feb 7 07:31:31.449: //1579/3A0C10F68F0E/SIP/Info/sipSPIProcessContactInfo: Previous Hop 10.135.1.17:5060
*Feb 7 07:31:31.453: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
*Feb 7 07:31:31.461: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler:
*Feb 7 07:31:31.461: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 147)
*Feb 7 07:31:31.461: //1579/3A0C10F68F0E/SIP/Info/ccsip_event_handler:
ccsip_event_handler: peer ID 1580 chans 0x45CAC1B8 event 147 flags 0x40001C 0x180 0x601 data 0x45CAC1B8
*Feb 7 07:31:31.461: //1579/3A0C10F68F0E/SIP/Info/ccsip_event_handler:
ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 1580 chans 0x45CAC1B8 event 147 flags 0x40001C 0x180 0x601 data 0x45CAC1B8, type =
1
*Feb 7 07:31:31.461: //1579/3A0C10F68F0E/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-TDM
*Feb 7 07:31:31.461: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED
*Feb 7 07:31:31.461: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 3
*Feb 7 07:31:31.469: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9d52abc7155773
To: <sip:[email protected];user=phone>
Date: Thu, 07 Feb 2013 07:31:31 gmt
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 456411359 INVITE
Allow-Events: telephone-event
Content-Length: 0



*Feb 7 07:31:32.353: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISC_PROG_IND
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Info/ccsip_bridge: confID = 579, srcCallID = 1579, dstCallID = 1580
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Info/sipSPIUupdateCcCallIds: Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 1579
/1580
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=-1, new streamcallid=1579
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Info/ccsip_bridge: xcoder_attached = 0, xmitFunc = 1097540320, ccb xmitFunc = 1097540320
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 1579) to the VOIP RTP
library
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.89.15.xxx
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.89.15.xxx, lport = 17230, raddr = 10.135.1.113, rport=64244, do_rtcp=TRUE
src_callid = 1579, dest_callid = 1580, stream type = voice+dtmf, stream direction = SENDRECV
media_ip_addr = 10.135.1.113, vrf tableid = 0
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0

*Feb 7 07:31:32.353: //1579/3A0C10F68F0E/SIP/State/sipSPIChangeStreamState: Stream (callid = 1579) State changed from (STREAM_ADDIN
G) to (STREAM_ACTIVE)
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=1579, current_seq_
num=0xC27
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=1579, current_seq_num=
0x85
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g711alaw, Bytes=160
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with
rx payload = 99, tx payload = 99
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=1, from C
LI config=0
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generat
e SDP Xcap list
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/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
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Media/sipSPISetStreamInfo: 1 Active Streams
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Media/sipSPISetStreamInfo: Adding stream type (voice+dtmf) from media line 1
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=1,caps.stream[0].stream_type=0x2, caps.stream_list.xmitFunc=
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Media/sipSPISetStreamInfo: 0x48214720 (gccb)
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/ccsip_caps_ind: Load DSP with codec : g711alaw, Bytes=160, payload = 8
*Feb 7 07:31:32.357: //1579/3A0C10F68F0E/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->pld.flags_ipip = 0x603
*Feb 7 07:31:32.361: //1579/3A0C10F68F0E/SIP/Info/ccsip_caps_ack: Set forking flag to 0x7
*Feb 7 07:31:32.361: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 16
*Feb 7 07:31:32.361: //1579/3A0C10F68F0E/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
*Feb 7 07:31:32.361: //1579/3A0C10F68F0E/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4726EDE4 key=ERICSSONBTK_TERM_b2e1d3473
[email protected]
*Feb 7 07:31:32.361: //1579/3A0C10F68F0E/SIP/Info/sipSPISendInviteResponse: Associated container=0x47C43A5C to Invite Response 183
*Feb 7 07:31:32.361: //1579/3A0C10F68F0E/SIP/Transport/sipSPISendInviteResponse: Sending 183 Response to the Transport Layer
*Feb 7 07:31:32.361: //1579/3A0C10F68F0E/SIP/Transport/sipSPITransportSendMessage: msg=0x479D63DC, addr=10.135.1.17, port=5060, sentB
y_port=5060, is_req=0, transport=1, switch=0, callBack=0x413DC80C
*Feb 7 07:31:32.361: //1579/3A0C10F68F0E/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Feb 7 07:31:32.365: //1579/3A0C10F68F0E/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Feb 7 07:31:32.365: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x479D63DC, addr=10.135.1.17,
port=5060, connId=0 for UDP
*Feb 7 07:31:32.365: //1579/3A0C10F68F0E/SIP/Info/sentInviteResponse18x: Sent a 18x Response
*Feb 7 07:31:32.365: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9d52abc7155773
To: <sip:[email protected];user=phone>;tag=A761F90-36E
Date: Thu, 07 Feb 2013 07:31:31 gmt
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 456411359 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Record-Route: <sip:10.135.1.17;transport=udp;lr>
Reason: Q.850;cause=1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 232

v=0
o=CiscoSystemsSIP-GW-UserAgent 1213 2055 IN IP4 10.89.15.xxx
s=SIP Call
c=IN IP4 10.89.15.xxx
t=0 0
m=audio 17230 RTP/AVP 8 99
c=IN IP4 10.89.15.xxx
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-16

*Feb 7 07:31:39.273: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.135.1.17:5060
*Feb 7 07:31:39.273: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
*Feb 7 07:31:39.273: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
*Feb 7 07:31:39.273: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:[email protected];user=phone SIP/2.0
Max-Forwards: 68
Via: SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq
To: <sip:[email protected];user=phone>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9d52abc7155773
Call-ID: [email protected]
CSeq: 456411359 CANCEL
Record-Route: <sip:10.135.1.17;transport=udp;lr>
Supported: histinfo
Content-Length: 0



*Feb 7 07:31:39.273: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
*Feb 7 07:31:39.273: //1579/3A0C10F68F0E/SIP/Info/sipSPIFindCcbUASReqTable: *****CCB found in UAS Request table. ccb=0x4726EDE4
*Feb 7 07:31:39.277: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.135.1.17,Port 5060, Trans
port 1, SentBy Port 5060
*Feb 7 07:31:39.277: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.135.1.17,Port 5060, Trans
port 1, SentBy Port 5060
*Feb 7 07:31:39.277: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[1579], src[2]
*Feb 7 07:31:39.277: //1579/3A0C10F68F0E/SIP/State/sipSPIChangeState: 0x4726EDE4 : State change from (STATE_RECD_INVITE, SUBSTATE_NON
E) to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Feb 7 07:31:39.277: //1579/3A0C10F68F0E/SIP/Transport/sipSPISendCancelResponse: Sending CANCEL Response to the transport layer
*Feb 7 07:31:39.277: //1579/3A0C10F68F0E/SIP/Transport/sipSPITransportSendMessage: msg=0x48210090, addr=10.135.1.17, port=5060, sentB
y_port=5060, is_req=0, transport=1, switch=0, callBack=0x00000000
*Feb 7 07:31:39.277: //1579/3A0C10F68F0E/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Feb 7 07:31:39.277: //1579/3A0C10F68F0E/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Feb 7 07:31:39.277: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x48210090, addr=10.135.1.17,
port=5060, connId=0 for UDP
*Feb 7 07:31:39.277: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container
*Feb 7 07:31:39.277: //1579/3A0C10F68F0E/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for incoming call
*Feb 7 07:31:39.277: //1579/3A0C10F68F0E/SIP/State/sipSPIChangeState: 0x4726EDE4 : State change from (STATE_DISCONNECTING, SUBSTATE_N
ONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Feb 7 07:31:39.281: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9d52abc7155773
To: <sip:[email protected];user=phone>
Date: Thu, 07 Feb 2013 07:31:39 gmt
Call-ID: [email protected]
CSeq: 456411359 CANCEL
Content-Length: 0



*Feb 7 07:31:39.285: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
*Feb 7 07:31:39.285: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 7
*Feb 7 07:31:39.289: //1579/3A0C10F68F0E/SIP/Info/sipSPISendInviteResponse: Associated container=0x47C42FD0 to Invite Response 487
*Feb 7 07:31:39.289: //1579/3A0C10F68F0E/SIP/Transport/sipSPITransportSendMessage: msg=0x48210090, addr=10.135.1.17, port=5060, sentB
y_port=5060, is_req=0, transport=1, switch=0, callBack=0x413DC6C8
*Feb 7 07:31:39.289: //1579/3A0C10F68F0E/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Feb 7 07:31:39.289: //1579/3A0C10F68F0E/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Feb 7 07:31:39.289: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x48210090, addr=10.135.1.17,
port=5060, connId=0 for UDP
*Feb 7 07:31:39.289: //1579/3A0C10F68F0E/SIP/Info/sentRequestCancelDisconnecting: Sent a 487 Request Cancelled
*Feb 7 07:31:39.289: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9d52abc7155773
To: <sip:[email protected];user=phone>;tag=A761F90-36E
Date: Thu, 07 Feb 2013 07:31:39 gmt
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 456411359 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=16
Content-Length: 0


*Feb 7 07:31:39.313: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.135.1.17:5060
*Feb 7 07:31:39.313: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
*Feb 7 07:31:39.317: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
*Feb 7 07:31:39.317: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 10.135.1.17:5060;branch=z9hG4bK8j573f46bhzhgj2xzx58xg4cq
To: <sip:[email protected];user=phone>;tag=A761F90-36E
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_e2bb263d7edf29647c9d52abc7155773
Call-ID: [email protected]
CSeq: 456411359 ACK
Content-Length: 0



*Feb 7 07:31:39.317: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
*Feb 7 07:31:39.317: //1579/3A0C10F68F0E/SIP/Info/sipSPIFindCcbUASRespTable: *****CCB found in UAS Response table. ccb=0x4726EDE4
*Feb 7 07:31:39.317: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.135.1.17,Port 5060, Trans
port 1, SentBy Port 5060
*Feb 7 07:31:39.317: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.135.1.17,Port 5060, Trans
port 1, SentBy Port 5060
*Feb 7 07:31:39.317: //1579/3A0C10F68F0E/SIP/Info/sipSPIIcpifUpdate: CallState: 2 Playout: 0 DiscTime:17552044 ConnTime 0
*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/State/sipSPIChangeState: 0x4726EDE4 : State change from (STATE_DISCONNECTING, SUBSTATE_N
ONE) to (STATE_DEAD, SUBSTATE_NONE)
*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4726EDE4
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : +38164abcd690
Called Number : +38121abcd811
Source IP Address (Sig ): 10.89.15.xxx
Destn SIP Req Addr:Port : 10.135.1.17:5060
Destn SIP Resp Addr:Port : 10.135.1.17:5060
Destination Name : 10.135.1.17

*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 99 (tx), 99 (rx)
Source IP Address (Media): 10.89.15.xxx
Source IP Port (Media): 17230
Destn IP Address (Media): 10.135.1.113
Destn IP Port (Media): 64244
Orig Destn IP Address:Port (Media): 0.0.0.0:0

*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487

*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 62B
*Feb 7 07:31:39.321: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[152] removed.
*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/Info/sipSPIUdeleteCcbFromUASReqTable: ****Deleting from UAS Request table.
*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x4726EDE4 key=ERICSSONBTK_TERM
[email protected]+38121abcd811
*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/Info/sipSPIUdeleteCcbFromUASRespTable: ****Deleting from UAS Response table.
*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x4726EDE4 key=ERICSSONBTK_TERM
[email protected]
*Feb 7 07:31:39.321: //1579/3A0C10F68F0E/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are goin
g to be free'd
*Feb 7 07:31:39.325: //1579/3A0C10F68F0E/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 4726EDE4
*Feb 7 07:31:39.325: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[152]


Ovo bi trebalo da su bitni delovi konfigiracije za dolazne pozive:

voice translation-rule 3
rule 1 /38121abc801/ /111/

voice translation-rule 5
rule 1 /111/ /38121abcd801/

voice translation-profile OUT
translate calling 5
translate called 3

application
service toroute flash:toroute.tcl

dial-peer voice 50 voip
description Dolazni pozivi
translation-profile incoming OUT
service toroute (probano sa i bez ovoga)
voice-class codec 1
session protocol sipv2
session target sip-server
incoming called-number 38121abcd8..
dtmf-relay sip-notify rtp-nte

Napominjem još jednom da mi odlazni pozivi kroz trunk rade OK kao i dolazni kroz ISDN. IMa li neko nekakvu ideju?

Potpis

Iskustvo je srazmerno količini uništene opreme ...

Pozdrav

Vjerovatno se javljam poprilicno kasno, i nadam se da si do sada i sam rijesio problem, ali ako nisi, evo ti par hintova:

1. primjetices u ovom debugu da imas sljedeci info : Reason: Q.850;cause=1 . To ti oficijelno znaci "This cause indicates that the called party cannot be reached because, although the called party number is in a valid format, it is not currently allocated (assigned)" . U prevodu , dobio sam poziv, i pojma nemam kuda da ga uputim. Sto nas dovodi do hinta broj 2

2. ako pogledas svoj dial-peer, on u principu rutira pozive kojima je pozivani broj 38121abcd8.. . Dakle, kad poziv s tim stringom dodje na CME, CME koristi taj DP da forwarduje poziv. E sada, taj tvoj DP sadrzi i odredjene translacije, od kojih je jako bitna za ovaj slucaj voice translation-rule 3 , koji zapravo sve sto dodje sa stringom 38121abc801 pretvara u 111 i salje na end usera ( tvoj tel ). I tu je po mom misljenju kvaka. Naime, ovaj tvoj translation-rule je napravljen kao one-to-one, sto znaci da samo ako dodje kompletan string 38121abc801 , on ce ga proslijediti kao 111 . Medjutim, tebi ne dolazi string 38121abc801 , tebi dolazi string 38121abcd801 ( sip:[email protected];user=phone ) . Obrati paznju na D .

Probaj to promijeniti u konfiguraciji, po mom misljenju to bi onda trebalo da radi, jer ce onda CME znati sta da radi sa dolaznim brojem.

Nisi zakasnio sa odgovorom

Na žalost ono "d" koje fali sam tek sada primetio - greška u kucanju (tako je to kada smo fobični i sakrivamo informacije). Na ruteru je pravilno podešeno ali i dalje ne zna gde da pošalje poziv. Telekomova skripta ne radi pošto Telenor i Telekom ne koriste jednaku signalizaciju (očekivano)

Potpis

Iskustvo je srazmerno količini uništene opreme ...

Ajde onda vidi da li si greskom u ovaj output sa rutera upisao i 38121abc801 , iako dobijas od Telekoma To: <sip:[email protected];user=phone> ( 811 umjesto 801 ). Ja koliko vidim iz ovog debuga koji si postavio, sve se lijepo ispregovara do trenutka kada CME treba da proslijedi poziv ka end useru, sto iz nekog razloga ne moze da uradi, a ja pretpostavljam da je negdje greska u konfiguraciji. Ako je ovaj DP jedini koji ima veze sa prosljedjivanjem poziva u privatnu mrezu, pregledaj malo ovo. Takodje, i ja cu danas - sutra da probam malo detaljnije pregledat ovaj tvoj debug, pa ako ista dodatno "skontam" obavezno javljam.

Poz

ako imaš bilo kakve ideje (a tako mi deluje) pošalji mi na koji mail da ti prosledim neizmenjeni debug umesto ovog gde sam očigledno loše sakrivao neke informacije

Potpis

Iskustvo je srazmerno količini uništene opreme ...

Mozes li da mi posaljes sadrzaj tcl skripte koju koristis za rutiranje poziva?
Znam da je ima po netu ali hocu da vidim kako si je iskonfigurisao...

Pozdrav,
Dragan

Potpis

Dragan

###################################################
proc setup { } {
# local settings
set areaCode 1
set leadNum 24
# end local settings
leg proceeding leg_incoming
set To [infotag get leg_proto_headers "To"]
set numero $To
regexp {sip:[0-9]+([0-9]{2})@} $To w numero
set numero $areaCode$numero
if { $numero == 801 } { set numero 111 }
if { $numero == 802 } { set numero 112 }
if { $numero == 803 } { set numero 113 }
if { $numero == 804 } { set numero 114 }
if { $numero == 805 } { set numero 115 }
if { $numero == 806 } { set numero 116 }
if { $numero == 807 } { set numero 117 }
if { $numero == 808 } { set numero 118 }
if { $numero == 809 } { set numero 119 }
if { $numero == 810 } { set numero 120 }
if { $numero == 381213409811 } { set numero 121 }
if { $numero == 812 } { set numero 122 }
if { $numero == 813 } { set numero 123 }
if { $numero == 814 } { set numero 124 }
if { $numero == 815 } { set numero 125 }
if { $numero == 816 } { set numero 126 }
if { $numero == 817 } { set numero 127 }
if { $numero == 818 } { set numero 128 }
if { $numero == 819 } { set numero 129 }
if { [regexp {tel:\+[0-9]+} $To telBroj] } { set numero $areaCode$leadNum }
leg setup $numero callInfo leg_incoming
}

proc setup_done { } {
# Handle SETUP DONE.
}


proc cleanup { } {
call close
}


requiredversion 2.0


#----------------------------------
# State Machine
#----------------------------------

set fsm(any_state,ev_disconnected) "cleanup same_state"
set fsm(CALL_INIT,ev_setup_indication) "setup GETDEST"
set fsm(GETDEST,ev_setup_done) "setup_done CALLACTIVE"
set fsm(CALLACTIVE,ev_disconnected) "cleanup CALLDISCONNECT"
set fsm(CALLDISCONNECT,ev_disconnected) "cleanup same_state"
set fsm(CALLDISCONNECT,ev_disconnect_done) "cleanup same_state"

fsm define fsm CALL_INIT
######################################################


A evo i objašnjenja iz Telenora koje mi (nažalost) nije rešilo problem:

"Skript vrsi manipulaciju nad SIP URI-jem tako da se 3 poslednje cifre numeracije koju su dodeljeneu pretvaraju u broj lokala. Ista takva obrada bi trebala da se izvrsi i nad tel URI-jem, za slucaj kada poziv dolazi van naše IMS mreže, medjutim, toga u skriptu nema.

Telekom nije koristio tel URI te je ova skripta kod njih ispravno funkcionisala.
Prilikom testnog poziva koji smo napravili sa SIP telefona (kada se koristi se SIP URI), dobija se ton zvonjave, dok u slucaju poziva sa mobilnog telefona ili nekog Telekomovig fiksnog broja (kada se koristi tel URI) zvonjave nema, sto ide u prilog ovoj tvrdnji."

Potpis

Iskustvo je srazmerno količini uništene opreme ...

Osnovni broj na SIP-u je? Pretpostavljam 38121abcd800? Ili nesto drugo?

Potpis

Dragan

Tako je, broj koji se završava na 800 je noseći broj za numeraciju

Potpis

Iskustvo je srazmerno količini uništene opreme ...

Probaj sa ovom promenom u TCL skripti:
set areaCode 8
set leadNum 00

Ostalo ostavi kako I jeste...pa testiraj neki broj.

Potpis

Dragan

Probao. U terminalu sada nemam ni da poziv stiže do mog rutera ... moraću se kasnije pozabaviti ovim novim momentom ...

Potpis

Iskustvo je srazmerno količini uništene opreme ...

OK kad stignes baci mi samo I konfiguraciju svog dolaznog dial-peer-a.

Potpis

Dragan

I samo ova izmena uneo sam gresku...ides sa ovim parametrima u TCL-u:

set areaCode 1
set leadNum 00


Uzimas 2 cifre iz dobijenog broja I posto je kod tebe numeracija na 1 dodajemo 2 dobijene cifre I prosledjujemo.

Znaci ovo je final varijanta :)

Nakon toga debug I ostalo...I obavezno na dolaznom dial-peer-u sa "service [ime skripte]"...I ne treba ti nikakva translacija na dolaznom dial-peer-u...skripta ce to odraditi za tebe. Znaci pozovi samo servis na tom mestu...

Pozdrav!

Potpis

Dragan

Posle dve godine ja sad pokušavam dolazne pozive sa TELENOR SIP Trunka da preusmerim na loklae I nikako ne uspevam. Da li je neko uspeo u međuvremenu?
Kod TELEKOMA to funkcioniše sa toroute.tcl skriptom . Ovde koliko vidim ne.

Please Help
10x in advanced

Pravio sam i sa telenorom...tamo ti ne treba TCL skripta - tacno.
Obicne translacije ce da rade posao ok sa Telenorom.
Da bih ti vise pomogao - daj mi output na PM od "debug ccsip messages" i svoju radnu konfiguraciju koja se tice voice-a - ne moras citavu!

Pozz,
Dragan

Potpis

Dragan

Da, uz pomoć Telenorove podrške. Potrebna je tcl skripta koja nije ista kao i za Telekom (naravno). Postoje problemi u komunikaciji sa određenim brojevima (pukne veza čim zazvoni) i to je od poslednje promene softvera u Telenoru. Neke stvari su uspeli da mi reše, neki brojevi idalje imaju problema. Evo kako izgleda skripta:

###################################################
proc setup { } {
# local settings
set areaCode 1
set leadNum 00
# end local settings
leg proceeding leg_incoming
set To [infotag get leg_proto_headers "To"]
puts "\n\n SIP HEADER: $To \n\n"
set numero $To
if [regexp {sip:\+[0-9]+([0-9]{2})@} $To w numero] {
set num $areaCode$numero
}
if [regexp {tel:\+[0-9]+([0-9]{2})} $To w numero] {

set num $areaCode$numero
}
if { $num == 100 } { set numero 255 }
if { $num == 101 } { set numero 210 }
if { $num == 102 } { set numero 222 }
if { $num == 103 } { set numero 255 }
if { $num == 104 } { set numero 244 }
if { $num == 105 } { set numero 115 }
if { $num == 106 } { set numero 123 }
if { $num == 107 } { set numero 277 }
if { $num == 108 } { set numero 121 }
if { $num == 109 } { set numero 125 }

puts "\n\n zovem lokal $num -> $numero \n\n"
leg setup $numero callInfo leg_incoming
}

proc setup_done { } {
# Handle SETUP DONE.
}


proc cleanup { } {
call close
}


requiredversion 2.0


#----------------------------------
# State Machine
#----------------------------------

set fsm(any_state,ev_disconnected) "cleanup same_state"
set fsm(CALL_INIT,ev_setup_indication) "setup GETDEST"
set fsm(GETDEST,ev_setup_done) "setup_done CALLACTIVE"
set fsm(CALLACTIVE,ev_disconnected) "cleanup CALLDISCONNECT"
set fsm(CALLDISCONNECT,ev_disconnected) "cleanup same_state"
set fsm(CALLDISCONNECT,ev_disconnect_done) "cleanup same_state"

fsm define fsm CALL_INIT
######################################################


Probaj da li će da ti reši problem

Potpis

Iskustvo je srazmerno količini uništene opreme ...

OK ako si isao sa skriptom ali vec 2 godine imam OK setup koji radi sa Telenorom bez skripte.
Nego ovo drugo sto si spomenuo da pozivi ka odredjenim brojevima pucaju - znas li nesto vise o tome?
Imam jedan nenormalan setup gde se pozivi sami okidaju kroz neki dialer i tu mi se sa telenor sip-om desava bas to...ako znas nesto vise o tome bilo bi super!

Pozdrav,
Dragan

Potpis

Dragan

Nakon inicijalne konfiguracije, koju sam završio uz obilatu pomoć ekipe iz Telenora sve je proradilo. Bez skripte sam imao odlazne pozive ali nije umeo da izrutira dolazne. I samu skriptu su prepravljali više puta dok nije sve proradilo kako treba. POsle nekih godinu dana smo počeli da imamo problema sa pozivima. Ispostavilo se da je Telenor menjao firmvere ili već neki drugi softver kod sebe i da moj sada već malko mator ruter sa verzijom IOS 12.3 ima problema sa time. Oni su sa svoje strane odradili neke korekcije i od tada radi bolje mada i dalje ponekad imam probleme sa pozivanjem određenih brojeva. Jedino rešenje za njih mi je da rutiram pozive kroz BRI

Potpis

Iskustvo je srazmerno količini uništene opreme ...

Nemam nikakvih problema sa njima bez TCL-a definitivno...ali ovo sa pucanjem poziva imam.
I da i kod mene je neki matoriji IOS 12.4 ali znam sigurno da nije 15...

Znaci nista vise sem tog upgrade firmware-a kod njih ne znas sta ima?

Hvala ti svakako,
Dragan

Potpis

Dragan

Ništa drugo ... verovatno sledi prelaz na Elastix tokom sledeće godine

Potpis

Iskustvo je srazmerno količini uništene opreme ...