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Panasonic TDA600 + H323 + Asterisk

непроходят звонки от Panasonic на Asterisk
Сообщений: 8

Panasonic TDA600 + H323 + Asterisk

Здравствуйте,

на Panasonic TDA600 стоит плата IP-GW16 и смотрит в Астериск, на астериске настроено пара внутренних номеров и соединение с панасоником через h323. Проблема в том, что звонки от астериска на панасоник уходят, а вот обратно нет.

Asterisk - 172.20.0.25
Panasonic - 172.20.0.27

h323.conf
[general]
bindaddr=172.20.0.25
port=1720

[panasonic]
faststart=yes
context=office
type=friend
host=172.20.0.27
port=1720
disallow=all
allow=alaw
canreinvite=no
dtmfmode=rfc2833
h245tunneling=yes


extension.conf
[office]
include => confserv
exten => 2000,1,Dial(SIP/2000,20)
exten => 2001,1,Dial(SIP/2001,20)
exten => _1XXX,1,Dial(H323/${EXTEN}@panasonic)
exten => _9.,1,Dial(H323/${EXTEN}@panasonic)
exten => _35.,1,Dial(H323/${EXTEN}@panasonic)
exten => 500,1,Dial(H323/500@panasonic)

exten => 2100,1,Answer()
exten => 2100,n,Wait(1)
exten => 2100,n,MeetMe(100,dmq)
exten => 2100,n,Hangup

exten => i,1,Hangup
exten => t,1,Hangup
;exten =>

[mm-announce]
exten => 9999,1,Set(CALLERID(name)="MMGETOUT")
exten => 9999,n,Answer
exten => 9999,n,Playback(conf-will-end-in)
exten => 9999,n,Playback(digits/5)
exten => 9999,n,Playback(minutes)
exten => 9999,n,Hangup

[default]

include => office

;Used by cbEnd script to play end of conference warning
exten => _mmplay.,1,Answer
exten => _mmplay.,2,MeetMe(${EXTEN:6}|dq)
exten => _mmplay.,3,Hangup

[confserv]
;Make sure you change 1199 to your conference bridge extension(s)
;more information on this can be found at the asterisk web site.
exten => 7777,1,Answer
exten => 7777,n,Wait(3)
exten => 7777,n,CBMysql()
exten => 7777,n,Hangup


sip.conf
[2000]
type=friend
host=dynamic
username=2000
secret=2000
nat=no
canreinvite=no
context=office
callerid="Viktor" <2000>
allow=alaw
qualify=yes


при включении h323 set debug в консоли астериска стал звонить X-Lite (2000), до включения звонок не проходил, прилагаю дебаг на момент звонка.

Q.931 message data is 69 octets {
08 01 30 05 04 03 80 90 a3 18 01 81 1c 28 91 aa ..0..........(..
06 80 01 00 82 01 00 8b 01 00 a1 1a 02 01 40 06 ..............@.
04 2b 0c 09 00 80 0f c0 e7 e3 e0 eb fc e4 ee e2 .+..............
20 c0 f0 f1 e5 ed 6c 06 09 80 31 36 31 39 70 05 .....l...1619p.
89 32 30 30 30 .2000
}ocalhost*CLI>
Error while decoding Q.931 message
localhos--Received SETUP message
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:2060 setup_incoming_call: Setting up incoming call for ip$172.20.0.27:1236/12
-- Setting up Call
-- t*CLI> Call token: [ip$172.20.0.27:1236/12]
-- t*CLI> Calling party name: []
-- t*CLI> Calling party number: [1619]
-- t*CLI> Called party name: [2000]
-- t*CLI> Called party number: [2000]
-- t*CLI> Calling party IP: [172.20.0.27]
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1564 find_user: Could not find user by name 1619 or address 172.20.0.27
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:2125 setup_incoming_call: Sending 1619@172.20.0.27 to context [default] extension 2000
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:2425 set_local_capabilities: Setting capabilities for connection ip$172.20.0.27:1236/12
Setting capabilities to 0x4010f (g723|gsm|ulaw|alaw|g729|h261)
Capabilities in preference order is ()
Allowed Codecs:
localhos Table:
G.723.1A <1>
G.723.1 <2>
GSM-06.10 <3>
G.711-uLaw-64k <4>
G.711-ALaw-64k <5>
G.729A <6>>
G.729 <7>I>
UserInput/hookflash <8>
UserInput/RFC2833 <9>
UserInput/dtmf <10>
Set:host*CLI>
0:host*CLI>
0:st*CLI>
G.723.1A <1>
G.723.1 <2>
GSM-06.10 <3>
G.711-uLaw-64k <4>
G.711-ALaw-64k <5>
G.729A <6>
G.729 <7>
1:st*CLI>
UserInput/hookflash <8>
2:st*CLI>
UserInput/RFC2833 <9>
UserInput/dtmf <10>
localhost*CLI>
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:2438 set_local_capabilities: Capabilities for connection ip$172.20.0.27:1236/12 is set
Using 172.20.0.25 for outbound H.245 transport
localhos-- Transmitting RFC2833 on payload 101
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:972 __oh323_rtp_create: Created RTP channel
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:977 __oh323_rtp_create: Setting NAT on RTP to 0
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1876 external_rtp_create: Sending RTP 'US' 172.20.0.25:28686
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1876 external_rtp_create: Sending RTP 'US' 172.20.0.25:28686
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1876 external_rtp_create: Sending RTP 'US' 172.20.0.25:28686
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1876 external_rtp_create: Sending RTP 'US' 172.20.0.25:28686
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1876 external_rtp_create: Sending RTP 'US' 172.20.0.25:28686
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1876 external_rtp_create: Sending RTP 'US' 172.20.0.25:28686
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1876 external_rtp_create: Sending RTP 'US' 172.20.0.25:28686
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1876 external_rtp_create: Sending RTP 'US' 172.20.0.25:28686
localhos=-= In OnAnswerCall for call 12
localhost*CLI> - Progress Indicator: 0
localhost*CLI> - Inserting PI of 0 into ALERTING message
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:2181 answer_call: Preparing Asterisk to answer for ip$172.20.0.27:1236/12
[Feb 8 21:13:32] DEBUG[1854]: pbx.c:1843 pbx_extension_helper: Launching 'Dial'
-- Executing [2000@default:1] Dial("H323/ip$172.20.0.27:1236/12", "SIP/2000|20") in new stack
[Feb 8 21:13:32] DEBUG[1854]: chan_sip.c:16543 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw)
[Feb 8 21:13:32] DEBUG[1854]: chan_sip.c:4614 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Feb 8 21:13:32] DEBUG[1854]: chan_sip.c:2779 do_setnat: Setting NAT on RTP to On
localhos-- Started logical channel: sending G.711-ALaw-64k
[Feb 8 21:13:32] DEBUG[1854]: channel.c:3468 ast_channel_inherit_variables: Not copying variable DIALEDTIME.
[Feb 8 21:13:32] DEBUG[1854]: channel.c:3468 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME.
[Feb 8 21:13:32] DEBUG[1854]: channel.c:3468 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME.
[Feb 8 21:13:32] DEBUG[1854]: channel.c:3468 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER.
[Feb 8 21:13:32] DEBUG[1854]: channel.c:3468 ast_channel_inherit_variables: Not copying variable DIALSTATUS.
[Feb 8 21:13:32] DEBUG[1854]: chan_sip.c:3042 sip_call: Outgoing Call for 2000
localhost*CLI> -- channelsOpen = 1
localhost*CLI> External RTP Session Starting
localhost*CLI> RTP channel id 1 parameters:
localhost*CLI> -- remoteIpAddress: 172.20.0.27
localhost*CLI> -- remotePort: 5010
localhost*CLI> -- ExternalIpAddress: 172.20.0.25
localhost*CLI> -- ExternalPort: 28686
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1902 setup_rtp_connection: Setting up RTP connection for ip$172.20.0.27:1236/12
-- Called 2000
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1942 setup_rtp_connection: Native format is set to 8 from 8 by RTP payload type 8
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1989 setup_rtp_connection: RTP connection prepared for ip$172.20.0.27:1236/12
localhos-- Started logical channel: receiving G.711-ALaw-64k
[Feb 8 21:13:32] DEBUG[1854]: channel.c:2960 set_format: Set channel SIP/2000-08236a28 to read format alaw
[Feb 8 21:13:32] DEBUG[1854]: channel.c:2960 set_format: Set channel H323/ip$172.20.0.27:1236/12 to read format ulaw
localhost*CLI> -- channelsOpen = 2
localhost*CLI> External RTP Session Starting
localhost*CLI> RTP channel id 1 parameters:
localhost*CLI> -- remoteIpAddress: 172.20.0.27
localhost*CLI> -- remotePort: 5010
localhost*CLI> -- ExternalIpAddress: 172.20.0.25
localhost*CLI> -- ExternalPort: 28686
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1902 setup_rtp_connection: Setting up RTP connection for ip$172.20.0.27:1236/12
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1942 setup_rtp_connection: Native format is set to 8 from 8 by RTP payload type 8
[Feb 8 21:13:32] DEBUG[1852]: chan_h323.c:1989 setup_rtp_connection: RTP connection prepared for ip$172.20.0.27:1236/12
localhosExternalRTPChannel Destroyed
localhosExternalRTPChannel Destroyed
localhosExternalRTPChannel Destroyed
localhosExternalRTPChannel Destroyed
localhosExternalRTPChannel Destroyed
localhosExternalRTPChannel Destroyed
Peer capability is G.711-ALaw-64k <1>
Found peer capability G.711-ALaw-64k <1>, Asterisk code is 8, frame size (in ms) is 20
Peer capability is G.711-uLaw-64k <2>
Found peer capability G.711-uLaw-64k <2>, Asterisk code is 4, frame size (in ms) is 20
Peer capability is G.729A <3>
Found peer capability G.729A <3>, Asterisk code is 256, frame size (in ms) is 20
Peer capability is G.729 <4>
Found peer capability G.729 <4>, Asterisk code is 256, frame size (in ms) is 0
Peer capability is UserInput/dtmf <5>
Peer capabilities = 0x10c (ulaw|alaw|g729), ordered list is (alaw|ulaw|g729)
[Feb 8 21:13:32] DEBUG[1853]: chan_h323.c:2395 set_peer_capabilities: Got remote capabilities from connection ip$172.20.0.27:1236/12
[Feb 8 21:13:32] DEBUG[1853]: chan_h323.c:2409 set_peer_capabilities: prefs[0]=alaw:20
[Feb 8 21:13:32] DEBUG[1853]: chan_h323.c:2409 set_peer_capabilities: prefs[1]=ulaw:20
[Feb 8 21:13:32] DEBUG[1853]: chan_h323.c:2409 set_peer_capabilities: prefs[2]=g729:20
[Feb 8 21:13:32] DEBUG[1819]: chan_sip.c:2255 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '1542accf7cb51df7436487b737d3356d@172.20.0.25' Request 102: Found
-- SIP/2000-08236a28 is ringing
[Feb 8 21:13:32] DEBUG[1854]: chan_h323.c:879 oh323_indicate: OH323: Indicating 3 on ip$172.20.0.27:1236/12
localhosSending alerting
[Feb 8 21:13:32] DEBUG[1854]: chan_h323.c:931 oh323_indicate: OH323: Indicated 3 on ip$172.20.0.27:1236/12, res=-1
[Feb 8 21:13:32] DEBUG[1854]: channel.c:2559 ast_indicate_data: Driver for channel 'H323/ip$172.20.0.27:1236/12' does not support indication 3, emulating it
[Feb 8 21:13:32] DEBUG[1854]: channel.c:2702 ast_prod: Prodding channel 'H323/ip$172.20.0.27:1236/12'
[Feb 8 21:13:32] DEBUG[1854]: channel.c:2960 set_format: Set channel H323/ip$172.20.0.27:1236/12 to write format slin
[Feb 8 21:13:35] DEBUG[1854]: rtp.c:874 ast_rtcp_read: RTCP NAT: Got RTCP from other end. Now sending to address 172.20.0.65:55215
[Feb 8 21:13:35] DEBUG[1854]: rtp.c:879 ast_rtcp_read: Got RTCP report of 132 bytes
[Feb 8 21:13:35] DEBUG[1854]: rtp.c:1187 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 172.20.0.65:55214
[Feb 8 21:13:35] DEBUG[1819]: chan_sip.c:2180 __sip_ack: Acked pending invite 102
[Feb 8 21:13:35] DEBUG[1819]: chan_sip.c:2214 __sip_ack: Stopping retransmission on '1542accf7cb51df7436487b737d3356d@172.20.0.25' of Request 102: Match Found
-- SIP/2000-08236a28 answered H323/ip$172.20.0.27:1236/12
[Feb 8 21:13:35] DEBUG[1854]: chan_h323.c:879 oh323_indicate: OH323: Indicating -1 on ip$172.20.0.27:1236/12
[Feb 8 21:13:35] DEBUG[1854]: chan_h323.c:931 oh323_indicate: OH323: Indicated -1 on ip$172.20.0.27:1236/12, res=-1
[Feb 8 21:13:35] WARNING[1854]: channel.c:2568 ast_indicate_data: Unable to handle indication -1 for 'H323/ip$172.20.0.27:1236/12'
[Feb 8 21:13:35] DEBUG[1854]: channel.c:2960 set_format: Set channel H323/ip$172.20.0.27:1236/12 to write format alaw
[Feb 8 21:13:35] DEBUG[1854]: channel.c:2559 ast_indicate_data: Driver for channel 'SIP/2000-08236a28' does not support indication 3, emulating it
[Feb 8 21:13:35] DEBUG[1854]: channel.c:2960 set_format: Set channel SIP/2000-08236a28 to write format slin
[Feb 8 21:13:35] DEBUG[1854]: chan_h323.c:670 oh323_answer: Answering on H323/ip$172.20.0.27:1236/12
localhosAnswering call ip$172.20.0.27:1236/12
localhos=-= In OnConnectionEstablished for call 12
localhost*CLI> -- Connection Established with "1619 [172.20.0.27]"
[Feb 8 21:13:35] DEBUG[1854]: chan_h323.c:2003 connection_made: Call ip$172.20.0.27:1236/12 answered
[Feb 8 21:13:35] DEBUG[1854]: chan_h323.c:879 oh323_indicate: OH323: Indicating 20 on ip$172.20.0.27:1236/12
[Feb 8 21:13:35] DEBUG[1854]: chan_h323.c:931 oh323_indicate: OH323: Indicated 20 on ip$172.20.0.27:1236/12, res=0
[Feb 8 21:13:35] DEBUG[1854]: rtp.c:2797 ast_rtp_write: Ooh, format changed from unknown to ulaw
[Feb 8 21:13:35] DEBUG[1854]: rtp.c:2814 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160
[Feb 8 21:13:35] DEBUG[1854]: rtp.c:2797 ast_rtp_write: Ooh, format changed from unknown to alaw
[Feb 8 21:13:35] DEBUG[1854]: rtp.c:2814 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160
localhos-- Received RELEASE COMPLETE message...
[Feb 8 21:13:35] DEBUG[1852]: chan_h323.c:2345 hangup_connection: Hanging up connection to ip$172.20.0.27:1236/12 with cause 111
[Feb 8 21:13:35] DEBUG[1854]: channel.c:3967 ast_generic_bridge: Didn't get a frame from channel: H323/ip$172.20.0.27:1236/12
[Feb 8 21:13:35] DEBUG[1854]: channel.c:4310 ast_channel_bridge: Bridge stops bridging channels H323/ip$172.20.0.27:1236/12 and SIP/2000-08236a28
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '1619'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '1619'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '2000'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is 'default'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is 'H323/ip$172.20.0.27:1236/12'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is 'SIP/2000-08236a28'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is 'Dial'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is 'SIP/2000|20'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '2009-02-08 21:13:32'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '2009-02-08 21:13:35'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '2009-02-08 21:13:35'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '3'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '0'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is 'ANSWERED'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '(null)'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '1234116812.8'
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:1692 pbx_substitute_variables_helper_full: Function result is '(null)'
[Feb 8 21:13:35] DEBUG[1854]: cdr_radius.c:221 radius_log: Unable to create RADIUS record. CDR not recorded!
[Feb 8 21:13:35] DEBUG[1854]: channel.c:2960 set_format: Set channel SIP/2000-08236a28 to write format ulaw
[Feb 8 21:13:35] DEBUG[1854]: channel.c:1522 ast_hangup: Hanging up channel 'SIP/2000-08236a28'
[Feb 8 21:13:35] DEBUG[1854]: chan_sip.c:3555 sip_hangup: Hangup call SIP/2000-08236a28, SIP callid 1542accf7cb51df7436487b737d3356d@172.20.0.25)
[Feb 8 21:13:35] DEBUG[1854]: rtp.c:1527 ast_rtp_early_bridge: Channel '<unspecified>' has no RTP, not doing anything
[Feb 8 21:13:35] DEBUG[1854]: app_dial.c:1867 dial_exec_full: Exiting with DIALSTATUS=ANSWER.
[Feb 8 21:13:35] DEBUG[1854]: pbx.c:2426 __ast_pbx_run: Spawn extension (default,2000,1) exited non-zero on 'H323/ip$172.20.0.27:1236/12'
== Spawn extension (default, 2000, 1) exited non-zero on 'H323/ip$172.20.0.27:1236/12'
[Feb 8 21:13:35] DEBUG[1854]: channel.c:1429 ast_softhangup_nolock: Soft-Hanging up channel 'H323/ip$172.20.0.27:1236/12'
[Feb 8 21:13:35] DEBUG[1854]: channel.c:1522 ast_hangup: Hanging up channel 'H323/ip$172.20.0.27:1236/12'
[Feb 8 21:13:35] DEBUG[1854]: chan_h323.c:694 oh323_hangup: Hanging up and scheduling destroy of call H323/ip$172.20.0.27:1236/12
localhos-- ClearCall: Request to clear call with token ip$172.20.0.27:1236/12, cause EndedByRemoteUser
localhos-- Sending RELEASE COMPLETE
localhos-- ClearCall: Request to clear call with token ip$172.20.0.27:1236/12, cause EndedByRemoteUser
channelsOpen = 1
channelsOpen = 0
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
-- ClearCall: Request to clear call with token ip$172.20.0.27:1236/12, cause EndedByTransportFail
18:05.614CLI> H323 Cleaner h323.cxx(897) H323 Connection ip$172.20.0.27:1236/12 terminated.
-- 1619 [172.20.0.27] has cleared the call
[Feb 8 21:13:35] DEBUG[1815]: chan_h323.c:2295 cleanup_connection: Cleaning connection to ip$172.20.0.27:1236/12
[Feb 8 21:13:35] DEBUG[1815]: chan_h323.c:2301 cleanup_connection: No connection for ip$172.20.0.27:1236/12
== H.323 Connection deleted.
[Feb 8 21:13:35] DEBUG[1819]: chan_sip.c:2214 __sip_ack: Stopping retransmission on '1542accf7cb51df7436487b737d3356d@172.20.0.25' of Request 103: Match Found
[Feb 8 21:13:38] DEBUG[1819]: chan_sip.c:4614 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
[Feb 8 21:13:38] DEBUG[1819]: chan_sip.c:2214 __sip_ack: Stopping retransmission on '28089ad30887bfbd7399ad4d08333ca6@172.20.0.25' of Request 102: Match Found
localhost*CLI> h323 set debug
H.323 debug enabled
localhost*CLI> h323 set debug off
H.323 debug disabled
localhost*CLI> h323 set trace off
H.323 trace disabled
[Feb 8 21:14:38] DEBUG[1819]: chan_sip.c:4614 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
[Feb 8 21:14:38] DEBUG[1819]: chan_sip.c:2214 __sip_ack: Stopping retransmission on '5c7618ed65f605ae40ab2133038f04bb@172.20.0.25' of Request 102: Match Found
[Feb 8 21:15:30] DEBUG[1819]: chan_sip.c:4614 sip_alloc: Allocating new SIP dialog for ZGE3ZmZiZWQ3ZmJlNmY1MjA4NGI5ZDk2YzBjYTFlMjg. - SUBSCRIBE (No RTP)
[Feb 8 21:15:38] DEBUG[1819]: chan_sip.c:4614 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
[Feb 8 21:15:38] DEBUG[1819]: chan_sip.c:2214 __sip_ack: Stopping retransmission on '0d05c4e570175ebf2f4563ec25f4837c@172.20.0.25' of Request 102: Match Found
[Feb 8 21:15:42] DEBUG[1819]: chan_sip.c:4614 sip_alloc: Allocating new SIP dialog for ZDQ1NTkyNjdiZTE3MWJjM2E3N2FjMzUyODUyMTMwMWQ. - REGISTER (No RTP)
-- Unregistered SIP '2000'
[Feb 8 21:16:15] DEBUG[1819]: chan_sip.c:2117 __sip_autodestruct: Auto destroying SIP dialog 'ZDQ1NTkyNjdiZTE3MWJjM2E3N2FjMzUyODUyMTMwMWQ.'
localhost*CLI>

--------------------------------
2009-02-09 10:36

Сообщений: 6521

Re: Panasonic TDA600 + H323 + Asterisk

Вместо
exten => _1XXX,1,Dial(H323/${EXTEN}@panasonic)
попробуй синтаксис
exten => _1XXX,1,Dial(H323/panasonic/${EXTEN})

h323.conf
[panasonic]
type=h323

2009-02-09 17:26

Откуда: Kiev
Сообщений: 801

Re: Panasonic TDA600 + H323 + Asterisk

Читаем в книге AFOT:
1) Для соединение конечной точки, конструкция Dial имеет вид:

Dial(technology/user[:password]@remote_host[:port][/remote_extension])

2) Если мы говорим о транке (а это Ваш случай), тогда:

Dial(technology/[trunk_context]/[remote_extension]), собственно что ded и написал
Лучший способ предвидеть будущее - изобрести его (Алан Кей, "Apple")
2009-02-09 18:25

Сообщений: 6521

Re: Panasonic TDA600 + H323 + Asterisk

AFOT - что это?
Asterisk - f%cking over there?
2009-02-09 18:35

Откуда: Kiev
Сообщений: 801

Re: Panasonic TDA600 + H323 + Asterisk

ded:

AFOT - что это?
Asterisk - f%cking over there?
to be wide of the mark, ded good luck ;)
Лучший способ предвидеть будущее - изобрести его (Алан Кей, "Apple")
2009-02-09 18:38

Сообщений: 8

Re: Panasonic TDA600 + H323 + Asterisk

Приветствую,

увы, это не помогло, меня очень беспокоит строка в дебаге
localhos-- Received RELEASE COMPLETE message...
[Feb 8 21:13:35] DEBUG[1852]: chan_h323.c:2345 hangup_connection: Hanging up connection to ip$172.20.0.27:1236/12 with cause 111


не могу найти что это за код такой...
2009-02-10 09:05

Откуда: Izhevsk
Сообщений: 30

Re: Panasonic TDA600 + H323 + Asterisk

exten => cause-111,1,NoOp(AST_CAUSE_PROTOCOL_ERROR)
2009-02-10 09:11

Avatara of Alekz
Откуда: Санкт-Петербург
Сообщений: 931

Re: Panasonic TDA600 + H323 + Asterisk

http://wiki.freeswitch.org/wiki/Hangup_causes
protocol error. This cause is used to report a protocol error event only when no other cause in the protocol error class applies.
Создам аварийную ситуацию. Дорого. На долго =)
2009-02-10 09:16

Сообщений: 1

Re: Panasonic TDA600 + H323 + Asterisk

Причин генерирования cause 111 (ошибка протокола)может быть масса. Вот некоторые из них при переходе с H323 на ОКС7:

1) Анализируется категория абонента А (calling party category). Наиболее вероятно, что она имеет значение - unknown.
2) Истечение таймера. Сработал таймер (Т7) по приёму ACM.
3) Нестыковка ISUP на станциях.
и другие. Много информации на тему генерирования причин раъединения можно найти здесь

http://calltrace.narod.ru/flash_cause_codes.htm
2010-10-25 14:59

Avatara of switch
Откуда: Уфа
Сообщений: 5856

Re: Panasonic TDA600 + H323 + Asterisk

Полезно
спасибо!
http://www.lynks.ru - Решения телефонии, мини-АТС, VoIP на основе Trixbox и Asterisk
2010-10-25 15:04

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