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не работают звонки с asterisk на cisco 5350

0

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

спросил Jul 12 '12

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обновил Jul 12 '12

zzuz Gravatar zzuz flag of Russian Federation
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http://line24.ru/

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вербос с астериска можете показать?

telefonist (Jul 13 '12)edit

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 (Jul 13 '12)edit

Очень на это похоже!

telefonist (Jul 13 '12)edit

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 (Sep 23 '12)edit

====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 (Sep 23 '12)edit

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 (Sep 23 '12)edit

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

evers (Sep 23 '12)edit

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

brost (Mar 15 '13)edit

всего то 6 месяцев прошло.

zzuz (Mar 15 '13)edit

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

brost (Mar 18 '13)edit

Будьте первым, кто ответит на этот вопрос!

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Задан: Jul 12 '12

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Обновлен: Jul 12 '12

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