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не работают звонки с asterisk на cisco 5350

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С АТС-cisco-* звонок проходит , а вот в обратном направлении к сожалению нет. Товарищи , уважаемые , помогите разобрать дебаг с cisco:

**GW#debug ccsip calls** SIP Call statistics tracing is enabled GW# Jul 12 13:16:49: //297806/86E4F67583B7/SIP/Call/sipSPICallInfo: The Call Setup Information is: Call Control Block (CCB) : 0x6588B2E4 State of The Call : STATE_DEAD TCP Sockets Used : NO Calling Number : 4010111 Called Number : astral Source IP Address (Sig ): 10.241.2.240 Destn SIP Req Addr:Port : 10.241.1.215:5060 Destn SIP Resp Addr:Port : 10.241.1.215:5060 Destination Name : 10.241.1.215 GW# Jul 12 13:16:49: //297806/86E4F67583B7/SIP/Call/sipSPIMediaCallInfo: Number of Media Streams: 1 Media Stream : 1 Negotiated Codec : g711alaw Negotiated Codec Bytes : 160 Negotiated Dtmf-relay : 6 Dtmf-relay Payload : 101 Source IP Address (Media): 10.241.2.240 Source IP Port (Media): 16866 Destn IP Address (Media): 10.241.1.215 Destn IP Port (Media): 19060 Orig Destn IP Address:Port (Media): 0.0.0.0:0 Jul 12 13:16:49: //297806/86E4F67583B7/SIP/Call/sipSPICallInfo: Disconnect Cause (CC) : 16 Disconnect Cause (SIP) : 500 ____________________________________________________________________ **GW#debug ccsip all** INVITE sip:astral@10.241.2.240 SIP/2.0 Via: SIP/2.0/UDP 10.241.1.215:5060;branch=z9hG4bK1a90413a;rport Max-Forwards: 70 From: "4010111" <sip:4010111@10.241.1.215>;tag=as072bf4c5 To: <sip:astral@10.241.2.240> Contact: <sip:4010111@10.241.1.215:5060> Call-ID: 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.13.0 Date: Thu, 12 Jul 2012 09:06:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 235 v=0 o=root 729095750 729095750 IN IP4 10.241.1.215 s=Asterisk PBX 1.8.13.0 c=IN IP4 10.241.1.215 t=0 0 m=audio 11818 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/State/sipSPIChangeState: 0x634D9914 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE) Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.241.1.215,Port 5060, Transport 1, SentBy Port 5060 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone UA to SIP default timezone = GMT Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.241.1.215,Port 5060, Transport 1, SentBy Port 5060 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetGtdBody: No valid GTD body found. Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table. Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x634D9914 key=08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060astral Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: astral Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: 4010111 Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name 4010111, number 4010111, Calling oct3 0x00, oct_3a 0x80, Called number astral Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIGetCallConfig: Peer tag 4010 matched for incoming call Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIGetCallConfig: Not using Voice Class Codec Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPICopyPeerDataToCCB: From CLI: Modem NSE payload = 100, Passthrough = 0, Modem relay = 0, Gw-Xid = 1 SPRT latency 200, SPRT Retries = 12, Dict Size = 1024 String Len = 32, Compress dir = 3 Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIContinueNewMsgInvite: Calling name 4010111, number 4010111, Calling oct3 0x00, oct_3a 0x80, ext_priv 0x00, Called number astral, oct3 0x00 Jul 12 12:06:19: //-1/ADB7E5DBB0E0/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:20 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :20, codecbytes: 160 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved. Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoDTMFRelayNegotiation: Payload type (101) is reserved for requested dtmf relay mode. Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full named event(NE) match in fmtp list of events. Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sip_do_nse_negotiation: SDP not present. Use local NSE payload 100. Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/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=10.241.1.215, dest_port=11818 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Media/sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :160 Preferred DTMF relay : sip-notify Preferred NTE payload : 101 Early Media : No Delayed Media : No Bridge Done : No New Media : No DSP DNLD Reqd : No Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.241.2.240 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_report_media_to_peer: callId 287855 peer 0 flags 0x201 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: CallID 287855, sdp 0x6724C3A4 channels 0x634DB1D4 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Hndl ptype 8 mline 1 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecing codec g711alaw Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :20, codecbytes: 160 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Adding codec 6 ptype 8 time 20, bytes 160 as channel 0 mline 1 ss 0 10.241.1.215:11818 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Hndl ptype 101 mline 1 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_report_media_to_peer: Report initial call media Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/copy_channels: callId 287855 size 80 ptr 0x631EC078) Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_report_media_to_peer: CCSIP: Unable to report channel ind Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : voice+dtmf Media line : 1 State : STREAM_ADDING (2) Callid : -1 Negotiated Codec : g711alaw, bytes :160 Negotiated DTMF relay : rtp-nte Negotiated NTE payload : 101 Negotiated CN payload : 0 Media Srce Addr/Port : 10.241.2.240:0 Media Dest Addr/Port : 10.241.1.215:11818 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIHandleInviteMedia: Negotiated Codec : g711alaw, bytes :160 Preferred Codec : g711alaw, bytes :160 Preferred DTMF relay 1 : 8 Preferred DTMF relay 2 : 6 Negotiated DTMF relay : 6 Preferred and Negotiated NTE payloads: 101 101 Preferred and Negotiated NSE payloads: 100 100 Preferred and Negotiated Modem Relay: 0 0 Preferred and Negotiated Modem Relay GwXid: 1 0 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIDoQoSNegotiation: SDP body with media description Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17558 for stream 1 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=17558 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 17558 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Negotiated method of dtmf relayand pyld: 6 101 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateRawMsg: No GTD passed. Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table. Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x634D9914 key=08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060F7BF92C-5AA Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 4646F to table Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipSPITransportSendMessage: msg=0x65447538, addr=10.241.1.215, port=5060, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x00000000 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x65447538 to default port=5060 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: Connection obtained...sending msg=0x65447538 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x65447538, addr=10.241.1.215, port=5060, connId=1 for UDP Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/State/sipSPIChangeState: 0x634D9914 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_RECD_INVITE, SUBSTATE_NONE) Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIProcessContactInfo: Previous Hop 10.241.1.215:5060 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/act_recdinvite_disconnect: Performing disconnect Jul 12 12 GW_ADPSU#:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIValidateGtd: No rawMsg from CCAPI Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipSPITransportSendMessage: msg=0x65445C78, addr=10.241.1.215, port=5060, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x6151A678 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x65445C78 to default port=5060 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Transport/sipTransportLogicSendMsg: Connection obtained...sending msg=0x65445C78 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x65445C78, addr=10.241.1.215, port=5060, connId=1 for UDP Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sentErrResDisconnecting: Sent an 3456XX Error Response Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/State/sipSPIChangeState: 0x634D9914 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE) Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.241.1.215:5060;branch=z9hG4bK1a90413a;rport From: "4010111" <sip:4010111@10.241.1.215>;tag=as072bf4c5 To: <sip:astral@10.241.2.240>;tag=F7BF92C-5AA Date: Thu, 12 Jul 2012 09:06:19 GMT Call-ID: 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 10.241.1.215:5060;branch=z9hG4bK1a90413a;rport From: "4010111" <sip:4010111@10.241.1.215>;tag=as072bf4c5 To: <sip:astral@10.241.2.240>;tag=F7BF92C-5AA Date: Thu, 12 Jul 2012 09:06:19 GMT Call-ID: 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Reason: Q.850;cause=16 Content-Length: 0 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 10.241.1.215:5060 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x655522E0 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIFindCcbUASReqTable: *****CCB found in UAS Request table. ccb=0x634D9914 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: ACK sip:astral@10.241.2.240 SIP/2.0 Via: SIP/2.0/UDP 10.241.1.215:5060;branch=z9hG4bK1a90413a;rport Max-Forwards: 70 From: "4010111" <sip:4010111@10.241.1.215>;tag=as072bf4c5 To: <sip:astral@10.241.2.240>;tag=F7BF92C-5AA Contact: <sip:4010111@10.241.1.215:5060> Call-ID: 08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.13.0 Content-Length: 0 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.241.1.215,Port 5060, Transport 1, SentBy Port 5060 Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone UA to SIP default timezone = GMT Jul 12 12:06:19: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 10.241.1.215,Port 5060, Transport 1, SentBy Port 5060 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIIcpifUpdate: CallState: 2 Playout: 0 DiscTime:25978296 ConnTime 0 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/State/sipSPIChangeState: 0x634D9914 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE) Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Call/sipSPICallInfo: The Call Setup Information is: Call Control Block (CCB) : 0x634D9914 State of The Call : STATE_DEAD TCP Sockets Used : NO Calling Number : 4010111 Called Number : astral Source IP Address (Sig ): 10.241.2.240 Destn SIP Req Addr:Port : 10.241.1.215:5060 Destn SIP Resp Addr:Port : 10.241.1.215:5060 Destination Name : 10.241.1.215 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Call/sipSPIMediaCallInfo: Number of Media Streams: 1 Media Stream : 1 Negotiated Codec : g711alaw Negotiated Codec Bytes : 160 Negotiated Dtmf-relay : 6 Dtmf-relay Payload : 101 Source IP Address (Media): 10.241.2.240 Source IP Port (Media): 17558 Destn IP Address (Media): 10.241.1.215 Destn IP Port (Media): 11818 Orig Destn IP Address:Port (Media): 0.0.0.0:0 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Call/sipSPICallInfo: Disconnect Cause (CC) : 16 Disconnect Cause (SIP) : 500 Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteCcbFromUASReqTable: ****Deleting from UAS Request table. Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x634D9914 key=08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060astral Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteCcbFromUASRespTable: ****Deleting from UAS Response table. Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x634D9914 key=08bbc173115f2a3013bc10d134182be8@10.241.1.215:5060F7BF92C-5AA Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 4646F Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd Jul 12 12:06:19: //287855/ADB7E5DBB0E0/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 634D9914 
удалить закрыть спам изменить тег редактировать

спросил 2012-07-12 14:22:19 +0400

evers Gravatar evers
1 ● 2 ● 2

обновил 2012-07-12 15:16:54 +0400

zzuz Gravatar zzuz flag of Russian Federation
7174 ● 2 ● 6 ● 75
http://line24.ru/

Comments

вербос с астериска можете показать?

telefonist ( 2012-07-13 14:48:55 +0400 )редактировать

достаточно

SIP/2.0 500 Internal Server Error

там видимо просто кривая настройка. Автор просто забил на чтение мануала, как это чаще всего бывает.

zzuz ( 2012-07-13 14:52:26 +0400 )редактировать

Очень на это похоже!

telefonist ( 2012-07-13 15:32:51 +0400 )редактировать

мануала ? а есть нормальный по сопряжению с cisco ? вот привожу конфиг
on ASTERISK
===sip.conf===
[astral]
type = friend
host = 10.241.2.240
username = astral
secret = astral!
qualify = yes
context = internal
disallow = all
allow = ulaw
insecure = port,invite
canreinvite = no
dtmfmode = rfc2833
language = ru

evers ( 2012-09-24 02:31:58 +0400 )редактировать

====extensions.conf===

[globals]
cisco = SIP/astral

[internal]
exten => _63XX,1,Dial(${cisco}) внутренние номера атс и циско
exten => _63XX,n,Hangup()

=====на CISCO====
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server
no call service stop

sip-ua
authentication username astral password astral!
registrar ipv4:10.241.1.215:5060 expires 3600
sip-server ipv4:10.241.1.215

evers ( 2012-09-24 02:33:51 +0400 )редактировать

ну и диалпир с циски на астериск

dial-peer voice 4010 voip
tone ringback alert-no-PI
destination-pattern 4010...
progress_ind setup enable 3
voice-class sip rel1xx disable
session protocol sipv2
session target ipv4:10.241.1.215:5060
session transport udp
dtmf-relay sip-notify rtp-nte h245-signal h245-alphanumeric
codec g711ulaw
no vad

evers ( 2012-09-24 02:34:19 +0400 )редактировать

ткните в мануал пожалуйста если он нормальный существует , а то гугл много знает , чересчур много )))

evers ( 2012-09-24 02:36:12 +0400 )редактировать

маршрутизация на AS5350 у вас настроена? можете выложить описание pots пиров?

brost ( 2013-03-15 15:20:36 +0400 )редактировать

всего то 6 месяцев прошло.

zzuz ( 2013-03-15 15:29:28 +0400 )редактировать

да. я уже понял. когда написал. вредно на ночь читать форум ))

brost ( 2013-03-18 10:00:00 +0400 )редактировать

Будьте первым, кто ответит на этот вопрос!

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Задан: 2012-07-12 14:22:19 +0400

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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.