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?