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cisco 1750 + asterisk

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

natrox
Сообщения: 37
Зарегистрирован: 20 янв 2014, 04:58
Откуда: Иркутская область, г. Усть-Илимск
Контактная информация:

Re: cisco 1750 + asterisk

Сообщение natrox »

Позвонил на АТС, Caller ID у них не используется. На попытку выяснить какое у них оборудование стоит и почему не используется Caller ID ничего внятного не ответили, партизаны блин. Это значит все? Сушить весла? Интересно тогда как на Авайе номера определялись? Странно все это...

И еще, мой конфиг оказался не совсем правильным. Проблема в том, что после звонка линия остается занята на циске. Астериск, судя по логу, говорит о том что звонок завершен, а вот циска линию не освобождает. При попытке позвонить на этот номер еще раз, линия занята.
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: debug ccsip all
1d03h: 0x826C32AC : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
1d03h: adding call id 35 to table

1d03h: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
1d03h: CCSIP-SPI-CONTROL: act_idle_call_setup
1d03h: act_idle_call_setup:Not using Voice Class Codec

1d03h: act_idle_call_setup: preferred_codec set[0] type :g711alaw bytes: 160
1d03h: 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
1d03h: sipSPIValidateGtd: Signal Forward disabled
1d03h: sipSPICanSetFallbackFlag - Local Fallback is not active
1d03h: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
1d03h: 0x826C32AC : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_CONNECTING)
1d03h: 0x826C32AC : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING)
1d03h: sipSPIUsetBillingProfile: sipCallId for billing records =
1d03h: ****Adding to UAC table. ccb=0x826C32AC key=19511C1D-831411E3-804EA446-7FC98526@192.168.128.73
1d03h: CCSIP-SPI-CONTROL: act_idle_connection_created
1d03h: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 91.219.137.26:5060, local_port 50680
1d03h: CCSIP-SPI-CONTROL: sipSPIOutgoingCallSDP
1d03h: sipSPISetMediaSrcAddr: media src addr for stream 1 = 192.168.128.73
1d03h: sipSPIReserveRtpPort: reserved port 16514 for stream 1
1d03h: Preferred method of dtmf relay is: 6, with payload : 101

1d03h: convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, p
time: 20

1d03h: sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed
1d03h: CCSIP-SPI-CONTROL: Converting TimeZone IRKT to SIP default timezone = GMT
1d03h: Received Octet3A=0x00 -> Setting ;screen=no ;privacy=off
1d03h: sipSPIAddLocalContact
1d03h: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
1d03h: sip_stats_method
1d03h: sipSPIProcessRtpSessions
1d03h: sipSPIAddStream: Adding stream 1 (callid 53) to the VOIP RTP library
1d03h: sipSPISetMediaSrcAddr: media src addr for stream 1 = 192.168.128.73
1d03h: sipSPIUpdateRtcpSession: for m-line 1
1d03h: sipSPIUpdateRtcpSession: rtcp_session info
laddr = 192.168.128.73, lport = 16514, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 53, dest_callid = -1, stream type = 1

1d03h: sipSPIUpdateRtcpSession: No rtp session, creating a new one

1d03h: sipSPIAddStream: AddStream in idle state to open a 'recvonly' media session
1d03h: act_idle_connection_created: Transaction active. Facilities will be queued.
1d03h: 0x826C32AC : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE)
1d03h: Sent:
INVITE sip:5002@91.219.137.26:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.73:5060
From: <sip:192.168.128.73>;tag=5DF42E0-13C3
To: <sip:5002@91.219.137.26>
Date: Thu, 23 Jan 2014 03:20:09 GMT
Call-ID: 19511C1D-831411E3-804EA446-7FC98526@192.168.128.73
Supported: timer,100rel
Min-SE: 60
Cisco-Guid: 424546133-2199130595-2152440902-2143913254
User-Agent: Cisco-SIPGateway/IOS-12.x
Accept-Language: ru
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 15
Remote-Party-ID: <sip:192.168.128.73>;party=calling;screen=no;privacy=off
Timestamp: 1390447209
Contact: <sip:192.168.128.73:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 253

v=0
o=CiscoSystemsSIP-GW-UserAgent 8365 9999 IN IP4 192.168.128.73
s=SIP Call
c=IN IP4 192.168.128.73
t=0 0
m=audio 16514 RTP/AVP 8 101
c=IN IP4 192.168.128.73
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

1d03h: Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.128.73:5060;received=192.168.128.73;rport=50680
From: <sip:192.168.128.73>;tag=5DF42E0-13C3
To: <sip:5002@91.219.137.26>
Call-ID: 19511C1D-831411E3-804EA446-7FC98526@192.168.128.73
CSeq: 101 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5002@91.219.137.26:5060>
Content-Length: 0

1d03h: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 91.219.137.26:5060
1d03h: Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.128.73:5060;received=192.168.128.73;rport=50680
From: <sip:192.168.128.73>;tag=5DF42E0-13C3
To: <sip:5002@91.219.137.26>;tag=as11264b86
Call-ID: 19511C1D-831411E3-804EA446-7FC98526@192.168.128.73
CSeq: 101 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:5002@91.219.137.26:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 262

v=0
o=root 134157346 134157346 IN IP4 91.219.137.26
s=Asterisk PBX 11.5.0
c=IN IP4 91.219.137.26
t=0 0
m=audio 17264 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

1d03h: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 91.219.137.26:5060
1d03h: CCSIP-SPI-CONTROL: act_sentinvite_new_message
1d03h: CCSIP-SPI-CONTROL: sipSPICheckResponse
1d03h: sip_stats_status_code
1d03h: Roundtrip delay 32 milliseconds for method INVITE

1d03h: 0x826C32AC : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (ST
ATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
1d03h: CCSIP-SPI-CONTROL: act_recdproc_new_message
1d03h: CCSIP-SPI-CONTROL: act_recdproc_new_message_response
1d03h: CCSIP-SPI-CONTROL: sipSPICheckResponse
1d03h: sip_stats_status_code
1d03h: Roundtrip delay 40 milliseconds for method INVITE

1d03h: act_recdproc_new_message_response: Transaction active. Facilities will be queued.
1d03h: CCSIP-SPI-CONTROL: sipSPIUACSessionTimer
1d03h: SE Value: 1800
1d03h: SE Header:
1d03h: Refresh param: uas
1d03h: sipSPIGetGtdBody: No valid GTD body found.
1d03h: CCSIP-SPI-CONTROL: act_recdproc_continue_200_processing
1d03h: CCSIP-SPI-CONTROL: act_recdproc_continue_200_processing: *** This ccb is the parent

1d03h: sipSPICompareRespMediaInfo
1d03h: sipSPICompareRespMediaInfo No Comparsion needed as 18x response SDP is either absent or ignored
1d03h: sipSPIDoMediaNegotiation: number of m lines is 1
1d03h: sipSPIDoAudioNegotiation: Codec (g711alaw) Negotiation Successful on Static Payload for m-line 1

1d03h: sipSPIDoPtimeNegotiation: One ptime attribute found - value:20
1d03h: convert_ptime_to_codec_bytes: Values :Codec: g711alaw ptime :20, codecbytes: 160

1d03h: convert_codec_bytes_to_ptime: Values :Codec: g711alaw codecbytes :160, ptime: 20

1d03h: sipSPIDoDTMFRelayNegotiation: m-line index 1
1d03h: sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved.
1d03h: sipSPIDoDTMFRelayNegotiation: Payload type (101) is reserved for requested dtmf relay mode.
1d03h: sipSPIDoDTMFRelayNegotiation: Case of full named event(NE) match in fmtplist of events.
1d03h: sip_sdp_get_modem_relay_cap_params:
1d03h: sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0
1d03h: sip_do_nse_negotiation: SDP not present. Use local NSE payload 100.
1d03h: sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
1d03h: sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0

1d03h: 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=91.219.137.26, dest_port=17264
1d03h: sipSPIReplaceSDP
1d03h: sipSPICopySdpInfo
1d03h: sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711alaw, bytes :160
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

1d03h: sipSPISetMediaSrcAddr: media src addr for stream 1 = 192.168.128.73
1d03h: sipSPIUpdCallWithSdpInfo:Stream Type:1
M-line Index : 1
State : STREAM_ADDING (2)
Callid : 53
Negotiated Codec : g711alaw, bytes :160
Negotiated DTMF relay : rtp-nte
Negotiated NTE payload : 101
Negotiated CN payload : 0
Media Srce Addr/Port : 192.168.128.73:16514
Media Dest Addr/Port : 91.219.137.26:17264

1d03h: sipSPIProcessMediaChanges
1d03h: ccsip_process_response_contact_record_route
1d03h: 0x826C32AC : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING)
1d03h: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
1d03h: 0x826C32AC : State change from (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING) to (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING)
1d03h: CCSIP-SPI-CONTROL: act_recdproc_connection_created
1d03h: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection: Connid(2) created to 91.219.137.26:5060, local_port 55353
1d03h: 0x826C32AC : State change from (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING) to (STATE_RECD_PROCEEDING, SUBSTATE_NONE)
1d03h: CCSIP-SPI-CONTROL: sipSPIProcess200OKforinvite
1d03h: sipSPICreateRawMsg: No GTD passed.
1d03h: RequestCloseConnection: Closing connid 1 Local Port 50680
1d03h: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
1d03h: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
1d03h: sip_stats_method
1d03h: 0x826C32AC : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_NONE)
1d03h: The Call Setup Information is :
Call Control Block (CCB) : 0x826C32AC
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number :
Called Number : 5002
Number of Media Streams : 1

1d03h: Media Stream 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101
Source IP Address (Media): 192.168.128.73
Source IP Port (Media): 16514
Destn IP Address (Media): 91.219.137.26
Destn IP Port (Media): 17264

1d03h: Orig Destn IP Address:Port (Media): 0.0.0.0:0

1d03h:
Source IP Address (Sig ): 192.168.128.73
Destn SIP Req Addr:Port : 91.219.137.26:5060
Destn SIP Resp Addr:Port : 0.0.0.0:0
Destination Name : 91.219.137.26

1d03h: sipSPIProcess200OKforinvite: Transaction Complete. Lock on Facilities released.
1d03h: sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
1d03h: CCSIP-SPI-CONTROL: ccsip_bridge: confID = 3, srcCallID = 53, dstCallID = 52
1d03h: sipSPIUupdateCcCallIds: old src/dest ccCallids: -1/-1, new src/dest ccCallids: 53/52
1d03h: sipSPIUupdateCcCallIds: old streamcallid=53, new streamcallid=53
1d03h: sipSPIProcessRtpSessions
1d03h: sipSPIAddStream: Adding stream 1 (callid 53) to the VOIP RTP library
1d03h: sipSPISetMediaSrcAddr: media src addr for stream 1 = 192.168.128.73
1d03h: sipSPIUpdateRtcpSession: for m-line 1
1d03h: sipSPIUpdateRtcpSession: rtcp_session info
laddr = 192.168.128.73, lport = 16514, raddr = 91.219.137.26, rport=17264, do_rtcp=TRUE
src_callid = 53, dest_callid = 52, stream type = 1

1d03h: sipSPIUpdateRtcpSession rtp session already created - update

1d03h: CCSIP-SPI-CONTROL: ccsip_caps_ind
1d03h: ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=53, current_seq_num=0x1CD
1d03h: ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=53, current_seq_num=0x2187
1d03h: ccsip_caps_ind: Load DSP with negotiated codec : g711alaw, Bytes=160
1d03h: ccsip_caps_ind: set forking flag to 0x0
1d03h: sipSPISetDTMFRelayMode: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB
1d03h: sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=0, from CLI config=0
1d03h: sip_set_modem_caps: Disabling Modem Relay...
1d03h: sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps
1d03h: sip_set_modem_caps: Modem Relay & Passthru both disabled
1d03h: sip_set_modem_caps: nse payload = 100, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0, sprt-retry=12, latecncy=200, compres-dir=3, dict=1024,strnlen=32
1d03h: sipSPISetStreamInfo: num_streams = 1
1d03h: sipSPISetStreamInfo: adding stream type 1 from mline 1
1d03h: sipSPISetStreamInfo: caps.stream_count=1, caps.stream[0].stream_type=0x2, caps.stream_list.xmitFunc=voip_rtp_xmit, caps.stream_list.context=0x814FDDBC (gccb)
1d03h: ccsip_caps_ind: Load DSP with codec : g711alaw, Bytes=160
1d03h: CCSIP-SPI-CONTROL: ccsip_caps_ack
1d03h: ccsip_caps_ack: set forking flag to 0x0
1d03h: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060
1d03h: Sent:
ACK sip:5002@91.219.137.26:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.73:5060
From: <sip:192.168.128.73>;tag=5DF42E0-13C3
To: <sip:5002@91.219.137.26>;tag=as11264b86
Date: Thu, 23 Jan 2014 03:20:09 GMT
Call-ID: 19511C1D-831411E3-804EA446-7FC98526@192.168.128.73
Max-Forwards: 15
Content-Length: 0
CSeq: 101 ACK
Если выдернуть линию из циски, в лог падает это
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: debug ccsip all
1d03h: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
1d03h: CCSIP-SPI-CONTROL: act_active_disconnect
1d03h: sipSPIValidateGtd: No rawMsg from CCAPI
1d03h: RequestCloseConnection: Closing connid 2 Local Port 55353
1d03h: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
1d03h: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
1d03h: 0x826C32AC : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_CONNECTING)
1d03h: 0x826C32AC : State change from (STATE_ACTIVE, SUBSTATE_CONNECTING) to (STATE_ACTIVE, SUBSTATE_CONNECTING)
1d03h: udpsock_close_connect: Socket fd: 2 closed for connid 2 with remote port: 5060
1d03h: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection: Connid(1) created to 91.219.137.26:5060, local_port 50304
1d03h: 0x826C32AC : State change from (STATE_ACTIVE, SUBSTATE_CONNECTING) to (STATE_ACTIVE, SUBSTATE_NONE)
1d03h: sipSPIStopHoldTimer: Stopping hold timer
1d03h: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
1d03h: sip_stats_method
1d03h: 0x826C32AC : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
1d03h: Sent:
BYE sip:5002@91.219.137.26:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.73:5060
From: <sip:192.168.128.73>;tag=5DF42E0-13C3
To: <sip:5002@91.219.137.26>;tag=as11264b86
Date: Thu, 23 Jan 2014 03:20:09 GMT
Call-ID: 19511C1D-831411E3-804EA446-7FC98526@192.168.128.73
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 15
Timestamp: 1390447773
CSeq: 102 BYE
Content-Length: 0

1d03h: Received:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.128.73:5060;received=192.168.128.73;rport=50304
From: <sip:192.168.128.73>;tag=5DF42E0-13C3
To: <sip:5002@91.219.137.26>;tag=as11264b86
Call-ID: 19511C1D-831411E3-804EA446-7FC98526@192.168.128.73
CSeq: 102 BYE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

1d03h: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 91.219.137.26:5060
1d03h: CCSIP-SPI-CONTROL: act_disconnecting_new_message
1d03h: CCSIP-SPI-CONTROL: sact_disconnecting_new_message_response
1d03h: CCSIP-SPI-CONTROL: sipSPICheckResponse
1d03h: sip_stats_status_code
1d03h: Roundtrip delay 16 milliseconds for method BYE

1d03h: CCSIP-SPI-CONTROL: sipSPICallCleanup
1d03h: sipSPIIcpifUpdate :CallState: 4 Playout: 21420 DiscTime:9908130 ConnTime
9851779

1d03h: 0x826C32AC : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)
1d03h: The Call Setup Information is :
Call Control Block (CCB) : 0x826C32AC
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number :
Called Number : 5002
Number of Media Streams : 1

1d03h: Media Stream 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101
Source IP Address (Media): 192.168.128.73
Source IP Port (Media): 16514
Destn IP Address (Media): 91.219.137.26
Destn IP Port (Media): 17264

1d03h: Orig Destn IP Address:Port (Media): 0.0.0.0:0

1d03h:
Source IP Address (Sig ): 192.168.128.73
Destn SIP Req Addr:Port : 91.219.137.26:5060
Destn SIP Resp Addr:Port : 0.0.0.0:0
Destination Name : 91.219.137.26

1d03h:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 481

1d03h: ****Deleting from UAC table. ccb=0x826C32AC key=19511C1D-831411E3-804EA446-7FC98526@192.168.128.73
1d03h: Removing call id 35

1d03h: RequestCloseConnection: Closing connid 1 Local Port 50304
1d03h: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
1d03h: sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
1d03h: freeing ccb 826C32AC

1d03h: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060
Я теряюсь в догадках. Вроде в лог ничего ненормального не показывает, или чего-то не вижу?
ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: cisco 1750 + asterisk

Сообщение ded »

natrox
Сообщения: 37
Зарегистрирован: 20 янв 2014, 04:58
Откуда: Иркутская область, г. Усть-Илимск
Контактная информация:

Re: cisco 1750 + asterisk

Сообщение natrox »

ded, спасибо за статью. Я так и думал что в этом дело. Пол дня сегодня пытаюсь подобрать интервалы, но пока не успешно. Где или как узнать какие нужны именно мне, я так и не нашел. Сколько конфигов видел, у всех разные интервалы. Неужели нет стандарта для Русских аналоговых АТС?
ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: cisco 1750 + asterisk

Сообщение ded »

Вы просто не умеете / не хотите искать. И стандарты, и методы определения - просто записать гудки и открыть в звуковом редакторе.
Всё есть в интернете.
На память - 425 герц, 350 мсек гудок / 350 тишина, это каденция (повторяемая).
natrox
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Re: cisco 1750 + asterisk

Сообщение natrox »

Вы просто не умеете / не хотите искать.
Я сегодня весь день искал, вариантов 10 перепробовал. Результат такой же. Я конечно буду еще пытаться, но пока результат меня не радует.
ded
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Re: cisco 1750 + asterisk

Сообщение ded »

ded писал(а):и методы определения - просто записать гудки в файл WAV и открыть в звуковом редакторе..
http://www.cisco.com/en/US/tech/tk652/t ... e2d1.shtml
natrox
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Re: cisco 1750 + asterisk

Сообщение natrox »

ded, для одной линии помогло. У нас тут 2 линии, с разных АТС идут. На второй видимо сигнал отбоя не стандартный, попробую записать как Вы советовали. Большое спасибо за помощь :)
Кому интересно, сейчас конфиг вот такой:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: cisco conf
version 12.3
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname VGate
!
boot-start-marker
boot-end-marker
!
logging cns-events debugging
!
memory-size iomem 25
clock timezone IRKT 9
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
aaa new-model
!
!
aaa session-id common
ip subnet-zero
ip cef
!
!
!
ip name-server 91.219.136.4
ip name-server 91.219.137.4
no ftp-server write-enable
!
voice echo-canceller extended
voice rtp send-recv
!
voice service pots
supported-language ru
!
voice service voip
sip
bind all source-interface FastEthernet0
min-se 60
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
!
!
!
!
voice class busyout 1
!
!
voice class dualtone-detect-params 1
cadence-variation 50
!
!
voice class custom-cptone Russia
dualtone disconnect
frequency 425
cadence 350 350
!
!
!
!
!
!
!
gw-accounting aaa
!
!
!
!
interface FastEthernet0
ip address 192.168.128.73 255.255.255.0
speed auto
!
interface Serial0
no ip address
shutdown
!
interface Serial1
no ip address
shutdown
!
ip default-gateway 192.168.128.1
ip classless
ip route 0.0.0.0 0.0.0.0 192.168.128.1
no ip http server
!
!
voice-port 1/0
no comfort-noise
cptone RU
timeouts call-disconnect 3
connection plar opx 5001
description 5001
supervisory disconnect dualtone mid-call
supervisory custom-cptone Russia
supervisory dualtone-detect-params 1
no battery-reversal
!
voice-port 1/1
no comfort-noise
cptone RU
timeouts call-disconnect 3
connection plar opx 5002
description 5002
supervisory disconnect dualtone mid-call
supervisory custom-cptone Russia
supervisory dualtone-detect-params 1
no battery-reversal
!
mgcp
!
!
dial-peer voice 1 voip
description to_asterisk
preference 2
destination-pattern 500[12]
session protocol sipv2
session target ipv4:91.219.137.26:5060
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 6001 pots
description to_fxo_pstn
preference 7
destination-pattern .T
progress_ind disconnect enable 8
port 1/0
!
dial-peer voice 6002 pots
description to_fxo_pstn
preference 7
destination-pattern .T
progress_ind disconnect enable 8
port 1/1
!
sip-ua
calling-info sip-to-pstn name set SIPTOPSTN
max-forwards 15
sip-server ipv4:91.219.137.26:5060
!
!
line con 0
line aux 0
line vty 0 4
exec-timeout 60 0
password password
absolute-timeout 90
logging synchronous
transport input telnet
line vty 5 15
absolute-timeout 90
logging synchronous
!
ntp clock-period 17180042
ntp master 3
ntp server 192.168.128.6 source FastEthernet0
!
end
natrox
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Re: cisco 1750 + asterisk

Сообщение natrox »

Звонил на АТС, там сказали что частота 425, сигнал и тишина по 0,4 сек. Не помогло.
Пробовал звонить и записывать звук, сигнал получился примерно 260 и тишина примерно 250 мс. Ставил такие значения и ставил значения немного меньше, судя по тому что написано тут http://asterisk.ru/knowledgebase/anal. Не помогает.
На частоте 425 и временем 350 350, пару раз отбой срабатывал. Уже все перепробовал, неужели это из-за АТС или могут быть еще причины?
natrox
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Re: cisco 1750 + asterisk

Сообщение natrox »

Товарищи, нашел тут вот такую штуку http://remicon.ru/prod/aon/12/. С ее помощью, как я понимаю, можно преобразовывать древний Русский Caller ID в нормальный Caller ID, который будет понимать циска. Это так или я что-то не так понял?
natrox
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Re: cisco 1750 + asterisk

Сообщение natrox »

Таки настроил я циску под свои линии FXO. Намучался ужасно, но за то хорошо разобрался с настройкой :) Всем спасибо кто помогал. Заказал конвертеры Caller ID->CallerID, есть мнение что они способны преобразовать до 7 последних цифр русского АОНа и передать его как CallerID(num). Не уверен что со шлюзом будет работать, как придет узнаю :)

Кому интересно, вот мой рабочий конфиг циски:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: cisco conf
version 12.3
service timestamps debug datetime
service timestamps log datetime
no service password-encryption
!
hostname WGate
!
boot-start-marker
boot-end-marker
!
no logging cns-events
!
memory-size iomem 25
clock timezone IRKT 9
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
aaa new-model
!
!
aaa session-id common
ip subnet-zero
ip cef
!
!
!
ip name-server 91.219.136.4
ip name-server 91.219.137.4
no ftp-server write-enable
!
voice echo-canceller extended
voice rtp send-recv
!
voice service pots
supported-language ru
!
voice service voip
sip
bind all source-interface FastEthernet0
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
!
!
!
!
!
voice class dualtone-detect-params 1
cadence-variation 20
!
!
voice class custom-cptone SIB
dualtone disconnect
frequency 415
cadence 450 450
!
!
voice class custom-cptone DIS
dualtone disconnect
frequency 425
cadence 500 250
!
!
!
!
!
!
!
gw-accounting aaa
!
!
!
!
interface FastEthernet0
ip address 192.168.128.73 255.255.255.0
speed auto
!
interface Serial0
no ip address
shutdown
!
interface Serial1
no ip address
shutdown
!
ip default-gateway 192.168.128.1
ip classless
ip route 0.0.0.0 0.0.0.0 192.168.128.1
no ip http server
!
!
voice-port 1/0
no comfort-noise
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 5001
supervisory disconnect dualtone mid-call
supervisory custom-cptone SIB
supervisory dualtone-detect-params 1
no battery-reversal
!
voice-port 1/1
no comfort-noise
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 5002
supervisory disconnect dualtone mid-call
supervisory custom-cptone DIS
supervisory dualtone-detect-params 1
no battery-reversal
!
mgcp
!
!
dial-peer voice 6002 pots
preference 7
destination-pattern .T
port 1/1
!
dial-peer voice 1 voip
preference 2
destination-pattern 500[12]
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
no vad
!
sip-ua
max-forwards 5
retry bye 2
sip-server ipv4:91.219.137.26:5060
!
!
line con 0
line aux 0
line vty 0 4
password password
!
ntp clock-period 17180042
ntp master 3
ntp server 192.168.128.6 source FastEthernet0
!
end
Сигналы отбоя получал так: подключал линию к обычному телефону. Потом звонил на него с другого, сбрасывал вызов и записывал звук через микрофон. Потом с помощью WaveLab измерял частоту и примерную длину сигнала и тишины. :)
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