Добрый день,
Есть офисная АТС Samsung OS500 (к ней подключен поток Е1), циска 2811 (172.16.16.11) и Asterisk 1.4.21.2-2 (172.16.17.49)

При простом звонке с самсунга на Asterisk и обратно все работает нормально.
При создании конференции с ip-телефона подключенного к Asterisk конференция стартует нормально.
Если позвонить с номера подключенного к офисной АТС то происходит срыв звонка... :(

кусок конфига циски циски
dial-peer voice 6900 voip
description Asterisk
destination-pattern 69..
session protocol sipv2
session target ipv4:172.16.17.49
session transport udp
dtmf-relay rtp-nte h245-alphanumeric
no vad

лог циски
полный лог циски могу выложить но очень он уж большой

Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=30860, current_seq_num=0x2C5
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=30860, current_seq_num=0x244C
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g729r8, Bytes=20
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=0, from CLI config=0
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap list
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled
Sep 4 09:11:15.125: //30860/02B26F31322C/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
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo: 1 Active Streams
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo: Adding stream type (voice+dtmf) from media line 1
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=1,caps.stream[0].stream_type=0x2, caps.stream_list.xmitFunc=
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo: 0x44E75E40 (gccb)
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/ccsip_caps_ind: Load DSP with codec : g729r8, Bytes=20
Sep 4 09:11:15.125: //30860/02B26F31322C/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->flags_ipip = 0x340F
Sep 4 09:11:15.129: //30860/02B26F31322C/SIP/Info/ccsip_caps_ack: Set forking flag to 0x0
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=30860, current_seq_num=0x244C
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=30860, current_seq_num=0x244C
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g729r8, Bytes=20
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=0, from CLI config=0
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap list
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled
Sep 4 09:11:15.165: //30860/02B26F31322C/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
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo: 1 Active Streams
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo: Adding stream type (voice+dtmf) from media line 1
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=1,caps.stream[0].stream_type=0x2, caps.stream_list.xmitFunc=
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Media/sipSPISetStreamInfo: 0x44E75E40 (gccb)
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/ccsip_caps_ind: Load DSP with codec : g729r8, Bytes=20
Sep 4 09:11:15.165: //30860/02B26F31322C/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->flags_ipip = 0x340F
Sep 4 09:11:15.993: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 172.16.17.49:5060
Sep 4 09:11:15.993: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
Sep 4 09:11:15.993: //30860/02B26F31322C/SIP/Info/ccsip_new_msg_preprocessor: ****Found CCB in UAC table

Sep 4 09:11:15.993: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:2224@172.16.17.11:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.17.49:5060;branch=z9hG4bK39fdfe32;rport
From: <sip:6910@172.16.17.49>;tag=as45c2573e
To: <sip:2224@172.16.17.11>;tag=C0907344-12BE
Call-ID: BB55532D-986911DE-A6D6C2EF-8CB055C@172.16.17.11
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0



Sep 4 09:11:15.993: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 172.16.17.49,Port 5060, Transport 1, SentBy Port 5060
Sep 4 09:11:15.993: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone MSK to SIP default timezone = GMT
Sep 4 09:11:15.993: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 172.16.17.49,Port 5060, Transport 1, SentBy Port 5060
Sep 4 09:11:15.993: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_set_release_source_for_peer: ownCallId[30860], src[4]
Sep 4 09:11:15.993: //30860/02B26F31322C/SIP/Info/sipSPIStopHoldTimer: Stopping hold timer
Sep 4 09:11:15.993: //30860/02B26F31322C/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for outgoing call
Sep 4 09:11:15.993: //30860/02B26F31322C/SIP/State/sipSPIChangeState: 0x46285ABC : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
Sep 4 09:11:15.993: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/Transport/sipSPISendByeResponse: Sending BYE Response to the transport layer
Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/Transport/sipSPITransportSendMessage: msg=0x46646BA4, addr=172.16.17.49, port=5060, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x41087578
Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Sep 4 09:11:15.997: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x46646BA4, addr=172.16.17.49, port=5060, connId=0 for UDP
Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/Info/sentByeResponse: Sent 200ok to the BYE, tearing down the call
Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/Info/sipSPIIcpifUpdate: CallState: 4 Playout: 0 DiscTime:323069320 ConnTime 323069218
Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/State/sipSPIChangeState: 0x46285ABC : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)
Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x46285ABC
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 2224
Called Number : 6910
Source IP Address (Sig ): 172.16.17.11
Destn SIP Req Addr:Port : 172.16.17.49:5060
Destn SIP Resp Addr:Port : 172.16.17.49:5060
Destination Name : 172.16.17.49

Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729r8
Negotiated Codec Bytes : 20
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101
Source IP Address (Media): 172.16.17.11
Source IP Port (Media): 19540
Destn IP Address (Media): 172.16.17.49
Destn IP Port (Media): 17244
Orig Destn IP Address:Port (Media): 0.0.0.0:0

Sep 4 09:11:15.997: //30860/02B26F31322C/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 200



логи астериска

astercc1*CLI> sip debug
SIP Debugging enabled
astercc1*CLI>
<--- SIP read from 172.16.17.11:55260 --->
INVITE sip:6910@172.16.17.49:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.17.11:5060;branch=z9hG4bK23864BE
From: <sip:2224@172.16.17.11>;tag=CF5C382C-2054
To: <sip:6910@172.16.17.49>
Date: Mon, 07 Sep 2009 06:08:30 GMT
Call-ID: B30E5F4A-9AAB11DE-AA58C2EF-8CB055C@172.16.17.11
Supported: 100rel,timer,replaces
Min-SE: 1800
Cisco-Guid: 45249921-2733852990-864048692-875836655
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:2224@172.16.17.11>;party=calling;screen=no;privacy=off
Timestamp: 1252303710
Contact: <sip:2224@172.16.17.11:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 270

v=0
o=CiscoSystemsSIP-GW-UserAgent 7426 5272 IN IP4 172.16.17.11
s=SIP Call
c=IN IP4 172.16.17.11
t=0 0
m=audio 16902 RTP/AVP 18 101
c=IN IP4 172.16.17.11
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (20 headers 12 lines) ---
Sending to 172.16.17.11 : 5060 (no NAT)
Using INVITE request as basis request - B30E5F4A-9AAB11DE-AA58C2EF-8CB055C@172.16.17.11
Found no matching peer or user for '172.16.17.11:55260'
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 172.16.17.11:16902
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.17.11:16902
Looking for 6910 in from-sip-external (domain 172.16.17.49)
list_route: hop: <sip:2224@172.16.17.11:5060>

<--- Transmitting (no NAT) to 172.16.17.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.17.11:5060;branch=z9hG4bK23864BE;received=172.16.17.11
From: <sip:2224@172.16.17.11>;tag=CF5C382C-2054
To: <sip:6910@172.16.17.49>
Call-ID: B30E5F4A-9AAB11DE-AA58C2EF-8CB055C@172.16.17.11
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6910@172.16.17.49>
Content-Length: 0


<------------>
-- Executing [6910@from-sip-external:1] NoOp("SIP/172.16.17.11-09974c68", "Received incoming SIP connection from unknown peer to 6910") in new stack
-- Executing [6910@from-sip-external:2] Set("SIP/172.16.17.11-09974c68", "DID=6910") in new stack
-- Executing [6910@from-sip-external:3] Goto("SIP/172.16.17.11-09974c68", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/172.16.17.11-09974c68", "1?from-trunk|6910|1") in new stack
-- Goto (from-trunk,6910,1)
-- Executing [6910@from-trunk:1] NoOp("SIP/172.16.17.11-09974c68", "Catch-All DID Match - Found 6910 - You probably want a DID for this.") in new stack
-- Executing [6910@from-trunk:2] Goto("SIP/172.16.17.11-09974c68", "ext-did|s|1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/172.16.17.11-09974c68", "__FROM_DID=s") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/172.16.17.11-09974c68", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/172.16.17.11-09974c68", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/172.16.17.11-09974c68", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/172.16.17.11-09974c68", "") in new stack
-- Executing [s@ext-did:3] GotoIf("SIP/172.16.17.11-09974c68", "0 ?cidok") in new stack
-- Executing [s@ext-did:4] Set("SIP/172.16.17.11-09974c68", "CALLERID(name)=2224") in new stack
-- Executing [s@ext-did:5] NoOp("SIP/172.16.17.11-09974c68", "CallerID is "2224" <2224>") in new stack
-- Executing [s@ext-did:6] Set("SIP/172.16.17.11-09974c68", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@ext-did:7] SetCallerPres("SIP/172.16.17.11-09974c68", "allowed_not_screened") in new stack
-- Executing [s@ext-did:8] Goto("SIP/172.16.17.11-09974c68", "ext-meetme|6910|1") in new stack
-- Goto (ext-meetme,6910,1)
-- Executing [6910@ext-meetme:1] Macro("SIP/172.16.17.11-09974c68", "user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/172.16.17.11-09974c68", "AMPUSER=2224") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/172.16.17.11-09974c68", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/172.16.17.11-09974c68", "1|Set|REALCALLERIDNUM=2224") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/172.16.17.11-09974c68", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/172.16.17.11-09974c68", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/172.16.17.11-09974c68", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/172.16.17.11-09974c68", "0?continue") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/172.16.17.11-09974c68", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/172.16.17.11-09974c68", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/172.16.17.11-09974c68", "Using CallerID "2224" <2224>") in new stack
-- Executing [6910@ext-meetme:2] Set("SIP/172.16.17.11-09974c68", "MEETME_ROOMNUM=6910") in new stack
-- Executing [6910@ext-meetme:3] GotoIf("SIP/172.16.17.11-09974c68", "0?USER") in new stack
-- Executing [6910@ext-meetme:4] Answer("SIP/172.16.17.11-09974c68", "") in new stack
Audio is at 172.16.17.49 port 14120
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.16.17.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.17.11:5060;branch=z9hG4bK23864BE;received=172.16.17.11
From: <sip:2224@172.16.17.11>;tag=CF5C382C-2054
To: <sip:6910@172.16.17.49>;tag=as5e93457a
Call-ID: B30E5F4A-9AAB11DE-AA58C2EF-8CB055C@172.16.17.11
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6910@172.16.17.49>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 2670 2670 IN IP4 172.16.17.49
s=session
c=IN IP4 172.16.17.49
t=0 0
m=audio 14120 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
-- Executing [6910@ext-meetme:5] Wait("SIP/172.16.17.11-09974c68", "1") in new stack
astercc1*CLI>
<--- SIP read from 172.16.17.11:55260 --->
ACK sip:6910@172.16.17.49:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.17.11:5060;branch=z9hG4bK2387265
From: <sip:2224@172.16.17.11>;tag=CF5C382C-2054
To: <sip:6910@172.16.17.49>;tag=as5e93457a
Date: Mon, 07 Sep 2009 06:08:30 GMT
Call-ID: B30E5F4A-9AAB11DE-AA58C2EF-8CB055C@172.16.17.11
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
-- Executing [6910@ext-meetme:6] Set("SIP/172.16.17.11-09974c68", "MEETME_OPTS=M") in new stack
-- Executing [6910@ext-meetme:7] Goto("SIP/172.16.17.11-09974c68", "STARTMEETME|1") in new stack
-- Goto (ext-meetme,STARTMEETME,1)
-- Executing [STARTMEETME@ext-meetme:1] MeetMe("SIP/172.16.17.11-09974c68", "6910|M|") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
-- Created MeetMe conference 1023 for conference '6910'
== Spawn extension (ext-meetme, STARTMEETME, 1) exited non-zero on 'SIP/172.16.17.11-09974c68'
-- Executing [h@ext-meetme:1] Hangup("SIP/172.16.17.11-09974c68", "") in new stack
== Spawn extension (ext-meetme, h, 1) exited non-zero on 'SIP/172.16.17.11-09974c68'
Scheduling destruction of SIP dialog 'B30E5F4A-9AAB11DE-AA58C2EF-8CB055C@172.16.17.11' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:2224@172.16.17.11:5060> for address/port to send to
set_destination: set destination to 172.16.17.11, port 5060
Reliably Transmitting (no NAT) to 172.16.17.11:5060:
BYE sip:2224@172.16.17.11:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.17.49:5060;branch=z9hG4bK4fe52ede;rport
From: <sip:6910@172.16.17.49>;tag=as5e93457a
To: <sip:2224@172.16.17.11>;tag=CF5C382C-2054
Call-ID: B30E5F4A-9AAB11DE-AA58C2EF-8CB055C@172.16.17.11
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
astercc1*CLI>
<--- SIP read from 172.16.17.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.17.49:5060;branch=z9hG4bK4fe52ede;rport
From: <sip:6910@172.16.17.49>;tag=as5e93457a
To: <sip:2224@172.16.17.11>;tag=CF5C382C-2054
Date: Mon, 07 Sep 2009 06:08:31 GMT
Call-ID: B30E5F4A-9AAB11DE-AA58C2EF-8CB055C@172.16.17.11
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 102 BYE


<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'B30E5F4A-9AAB11DE-AA58C2EF-8CB055C@172.16.17.11' Method: ACK
astercc1*CLI> sip no debug
SIP Debugging Disabled



Пробовал связывать и по E1 и по H323 везде тоже самое.
Кто нибудь сталкивался с данной проблемой?
Я уже голову себе сломал всю почему идет разъединение звонка.

А чего не в ВЕРХНЕМ РЕГИСТРЕ?
SIP-debug вам в данном случае помогает?

поясните пожалуйста?
если вы придираетесь что выделил лог циски жирным, то сделал это намеренно, чтоб можно было легко отделить лог циски от лога астериска, если это очень раздражает кого либо, то можно легко исправить :))

SIP-debug вам в данном случае помогает?
ну из логов видно что идет обычное разъединение звонка, как будто кто то вышает трубку, почему это происходит ни как не пойму :(

:) - злонамернно ...

Лог циски - не нужен. А на * пока уберите дебаг и покажите вывод ...

И еще, покажите ваш диал.план

хмм....
может я слепой, но что то кнопки изменить пост не вижу о_О
сдается мне изменять можно только последнее сообщение...
так бы убрал конечно.

Вывод прост - во время звонка соединение устанавливается, конференция стартует, потом идет сброс звонка как будто кто то повесил трубку...
вопрос почему?

Номерной план:
офисная АТС - 2ХХХ
Asterisk - 69ХХ

покажите контекст куда попадает вызов (extensions.conf) и вывод консоли при таком вызове. (только без дебага)

у меня редактирование номерного плана и т.д. полностью через веб интерфейс, т.е. в файл extensions.conf и т.д. практически пустой, как пример:
extensions.conf (содержимое файла):
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
бла бла бла и т.д.
#include extensions_custom.conf

[from-trunk] ; just an alias since VoIP shouldn't be called PSTN
include => from-pstn

[from-pstn]
Type :quit<Enter> to exit Vim


extensions_custom.conf
[astercc-barge]
exten => _X.,1,NoOP(${EXTEN})
exten => _X.,n,meetme(${EXTEN}|pqdx)
exten => _X.,n,hangup

и т.д.

в веб интерфейсе есть соотвественно 2 тел. номера 6900 и 6901, конференция 6910

вывод консоли (звонок с номера 2224 на 6910)
-- Executing [6910@from-sip-external:1] NoOp("SIP/172.16.17.11-b7501558", "Received incoming SIP connection from unknown peer to 6910") in new stack
-- Executing [6910@from-sip-external:2] Set("SIP/172.16.17.11-b7501558", "DID=6910") in new stack
-- Executing [6910@from-sip-external:3] Goto("SIP/172.16.17.11-b7501558", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/172.16.17.11-b7501558", "1?from-trunk|6910|1") in new stack
-- Goto (from-trunk,6910,1)
-- Executing [6910@from-trunk:1] NoOp("SIP/172.16.17.11-b7501558", "Catch-All DID Match - Found 6910 - You probably want a DID for this.") in new stack
-- Executing [6910@from-trunk:2] Goto("SIP/172.16.17.11-b7501558", "ext-did|s|1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/172.16.17.11-b7501558", "__FROM_DID=s") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/172.16.17.11-b7501558", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/172.16.17.11-b7501558", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/172.16.17.11-b7501558", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/172.16.17.11-b7501558", "") in new stack
-- Executing [s@ext-did:3] GotoIf("SIP/172.16.17.11-b7501558", "0 ?cidok") in new stack
-- Executing [s@ext-did:4] Set("SIP/172.16.17.11-b7501558", "CALLERID(name)=2224") in new stack
-- Executing [s@ext-did:5] NoOp("SIP/172.16.17.11-b7501558", "CallerID is "2224" <2224>") in new stack
-- Executing [s@ext-did:6] Set("SIP/172.16.17.11-b7501558", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@ext-did:7] SetCallerPres("SIP/172.16.17.11-b7501558", "allowed_not_screened") in new stack
-- Executing [s@ext-did:8] Goto("SIP/172.16.17.11-b7501558", "ext-meetme|6910|1") in new stack
-- Goto (ext-meetme,6910,1)
-- Executing [6910@ext-meetme:1] Macro("SIP/172.16.17.11-b7501558", "user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/172.16.17.11-b7501558", "AMPUSER=2224") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/172.16.17.11-b7501558", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/172.16.17.11-b7501558", "1|Set|REALCALLERIDNUM=2224") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/172.16.17.11-b7501558", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/172.16.17.11-b7501558", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/172.16.17.11-b7501558", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/172.16.17.11-b7501558", "0?continue") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/172.16.17.11-b7501558", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/172.16.17.11-b7501558", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/172.16.17.11-b7501558", "Using CallerID "2224" <2224>") in new stack
-- Executing [6910@ext-meetme:2] Set("SIP/172.16.17.11-b7501558", "MEETME_ROOMNUM=6910") in new stack
-- Executing [6910@ext-meetme:3] GotoIf("SIP/172.16.17.11-b7501558", "0?USER") in new stack
-- Executing [6910@ext-meetme:4] Answer("SIP/172.16.17.11-b7501558", "") in new stack
-- Executing [6910@ext-meetme:5] Wait("SIP/172.16.17.11-b7501558", "1") in new stack
-- Executing [6910@ext-meetme:6] Set("SIP/172.16.17.11-b7501558", "MEETME_OPTS=Ms") in new stack
-- Executing [6910@ext-meetme:7] Goto("SIP/172.16.17.11-b7501558", "STARTMEETME|1") in new stack
-- Goto (ext-meetme,STARTMEETME,1)
-- Executing [STARTMEETME@ext-meetme:1] MeetMe("SIP/172.16.17.11-b7501558", "6910|Ms|") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
-- Created MeetMe conference 1023 for conference '6910'
== Spawn extension (ext-meetme, STARTMEETME, 1) exited non-zero on 'SIP/172.16.17.11-b7501558'
-- Executing [h@ext-meetme:1] Hangup("SIP/172.16.17.11-b7501558", "") in new stack
== Spawn extension (ext-meetme, h, 1) exited non-zero on 'SIP/172.16.17.11-b7501558'

покажите файлы /etc/asterisk/meetme.conf и etc/asterisk/meetme_additional.conf

и еще приведите вывод, при котором все это работает ...

meetme.conf
[rooms]
#include meetme_additional.conf

meetme_additional.conf
conf => 6910,

Рабочая конференция (звонок с 6901 на 6910)

-- Executing [6910@from-internal:1] Macro("SIP/6901-b7501558", "user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/6901-b7501558", "AMPUSER=6901") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/6901-b7501558", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/6901-b7501558", "1|Set|REALCALLERIDNUM=6901") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/6901-b7501558", "AMPUSER=6901") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/6901-b7501558", "AMPUSERCIDNAME=6901") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/6901-b7501558", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/6901-b7501558", "AMPUSERCID=6901") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/6901-b7501558", "CALLERID(all)="6901" <6901>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/6901-b7501558", "REALCALLERIDNUM=6901") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/6901-b7501558", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/6901-b7501558", "0?continue") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/6901-b7501558", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/6901-b7501558", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/6901-b7501558", "Using CallerID "6901" <6901>") in new stack
-- Executing [6910@from-internal:2] Set("SIP/6901-b7501558", "MEETME_ROOMNUM=6910") in new stack
-- Executing [6910@from-internal:3] GotoIf("SIP/6901-b7501558", "0?USER") in new stack
-- Executing [6910@from-internal:4] Answer("SIP/6901-b7501558", "") in new stack
-- Executing [6910@from-internal:5] Wait("SIP/6901-b7501558", "1") in new stack
-- Executing [6910@from-internal:6] Set("SIP/6901-b7501558", "MEETME_OPTS=Ms") in new stack
-- Executing [6910@from-internal:7] Goto("SIP/6901-b7501558", "STARTMEETME|1") in new stack
-- Goto (from-internal,STARTMEETME,1)
-- Executing [STARTMEETME@from-internal:1] MeetMe("SIP/6901-b7501558", "6910|Ms|") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
-- Created MeetMe conference 1023 for conference '6910'
-- <SIP/6901-b7501558> Playing 'conf-onlyperson' (language 'en')
-- Started music on hold, class 'default', on SIP/6901-b7501558
-- Stopped music on hold on SIP/6901-b7501558
== Spawn extension (from-internal, STARTMEETME, 1) exited non-zero on 'SIP/6901-b7501558'
-- Executing [h@from-internal:1] Macro("SIP/6901-b7501558", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/6901-b7501558", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/6901-b7501558", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/6901-b7501558", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/6901-b7501558", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/6901-b7501558", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/6901-b7501558", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/6901-b7501558' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/6901-b7501558'