[ marxer @ 09.02.2013. 12:57 ] @
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. 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