Здравствуйте. Установил gentoo x64 и настроил на нее asterisk 1.6.2.13 (единственно его собирал еще с dahdi) . астер подключен к ldap из него берет пользователей и номера. вот такая загвоздка если в extensions приписать switch. то при любом звонке пользователя астериск выключается, при чем в консоль не выдает никакой ошибки. До этого стоял gentoo x86 и asterisk 1.6.2.8 все работало, хотя и вылетали ошибки, но все работало. а щас все время выключается, скажите, пожалуйста в чем может быть причина, и посоветуйте как это устранить.

"У меня поломался сервер , скажите, пожалуйста , что не так? как починить?"

понятно... какие конфиги или логи выложить... просто написал что вылетает из-за того когда я пишу строчку в extensions.conf

switch => Realtime/123@extensions

затем в extconfig.conf указал откуда брать это
extensions=>ldap,"ou=...,dc=...,dc=...",extensions

и соответсвенно в res_ldap.conf
указал
[extensions]
context=AstAccountContext
exten=AstExten
priority=AstPriority
app=AstApplication
appdata=AstApplicationData
additionalFilter=(objectClass=AsteriskSIPUser)

как только закоментируешь строку switch => Realtime/123@extensions, сразу все нормально не вылетает, но нельзя позвонить пользователю а исходящий работает вызов.

а случаем не
switch => Realtime/123@AstAccountContext ?

так мы пишем
switch => Realtime/(контекст)@(здесь пишем константу которую берем из exconfig.conf)

а в exconfig.conf указал откуда брать это
realtime_ext=>ldap,"ou=...,dc=...,dc=...",extensions

и получилось так
switch => Realtime/job@realtime_ext

или я ошибаюсь в написании? ну по крайней мере раньше на 1.6.2.8 работала такая комбинация, а щас на 1.6.2.13 вылетает asterisk, от такого написания

Включите дебаг. Думаю так быстрей станет ясно , что ему не нравится.

включил дебаг, после того как кладу трубку он сразу отключается.. понять не могу почему.. подскажите, кто сведующий в этом...

Dec 3 13:37:31] DEBUG[29621] chan_sip.c: We're settling with these formats: 0x8 (alaw)
[Dec 3 13:37:31] VERBOSE[29641] app_dial.c: -- SIP/100-00000001 is making progress passing it to SIP/ast-00000000
[Dec 3 13:37:31] DEBUG[29641] chan_sip.c: Setting framing from config on incoming call
[Dec 3 13:37:31] DEBUG[29641] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False Text flag: True
[Dec 3 13:37:31] DEBUG[29641] chan_sip.c: ** Our prefcodec: 0x0 (nothing).
[Dec 3 13:37:31] DEBUG[29641] chan_sip.c: -- Done with adding codecs to SDP
[Dec 3 13:37:31] DEBUG[29641] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw)
[Dec 3 13:37:31] DEBUG[29641] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 192.168.100.11:5065
[Dec 3 13:37:31] DEBUG[29641] rtp.c: Ooh, format changed from unknown to alaw
[Dec 3 13:37:31] DEBUG[29641] rtp.c: Created smoother: format: 8 ms: 20 len: 160
[Dec 3 13:37:31] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:31] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:31] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:31] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:31] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:31] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:32] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Ooh, format changed from unknown to alaw
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Created smoother: format: 8 ms: 20 len: 160
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Acked pending invite 102
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Stopping retransmission on '1a1f31674901d6e05ba422860469f822@192.168.100.4' of Request 102: Match Found
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing session-level SDP o=9338 1291391013 1291391013 IN IP4 192.168.100.6... UNSUPPORTED.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing session-level SDP s=AddPac Gateway SDP... UNSUPPORTED.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.100.6... OK.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing session-level SDP t=1291391013 0... UNSUPPORTED.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000/1... OK.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: We're settling with these formats: 0x8 (alaw)
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Updating call counter for outgoing call
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: build_route: Contact hop: sip:9338@192.168.100.6
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Trying to put 'ACK sip:933' onto UDP socket destined for 192.168.100.6:5060
[Dec 3 13:37:33] VERBOSE[29641] app_dial.c: -- SIP/100-00000001 answered SIP/ast-00000000
[Dec 3 13:37:33] DEBUG[29594] chan_sip.c: Checking device state for peer 100
[Dec 3 13:37:33] DEBUG[29594] devicestate.c: Changing state for SIP/100 - state 1 (Not in use)
[Dec 3 13:37:33] DEBUG[29594] devicestate.c: device 'SIP/100' state '1'
[Dec 3 13:37:33] DEBUG[29604] app_queue.c: Device 'SIP/100' changed to state '1' (Not in use) but we don't care because they're not a member of any que
[Dec 3 13:37:33] DEBUG[29641] chan_sip.c: SIP answering channel: SIP/ast-00000000
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Setting the marker bit due to a source update
[Dec 3 13:37:33] DEBUG[29641] chan_sip.c: Setting framing from config on incoming call
[Dec 3 13:37:33] DEBUG[29641] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False Text flag: True
[Dec 3 13:37:33] DEBUG[29641] chan_sip.c: ** Our prefcodec: 0x0 (nothing).
[Dec 3 13:37:33] DEBUG[29641] chan_sip.c: -- Done with adding codecs to SDP
[Dec 3 13:37:33] DEBUG[29641] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw)
[Dec 3 13:37:33] DEBUG[29641] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.100.11:5065
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Changing ssrc from 2023162710 to 1025821808 due to a source change
[Dec 3 13:37:33] DEBUG[29594] chan_sip.c: Checking device state for peer ast
[Dec 3 13:37:33] DEBUG[29594] devicestate.c: Changing state for SIP/ast - state 1 (Not in use)
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Changing ssrc from 2026748427 to 223470155 due to a source change
[Dec 3 13:37:33] DEBUG[29594] devicestate.c: device 'SIP/ast' state '1'
[Dec 3 13:37:33] DEBUG[29604] app_queue.c: Device 'SIP/ast' changed to state '1' (Not in use) but we don't care because they're not a member of any que
[Dec 3 13:37:33] VERBOSE[29641] rtp.c: -- Packet2Packet bridging SIP/ast-00000000 and SIP/100-00000001
[Dec 3 13:37:33] DEBUG[29621] chan_sip.c: Stopping retransmission on '5238d175-e202-1910-9f2a-002354bd54cb@ap-torgi' of Response 2: Match Found
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Forcing Marker bit, because SSRC has changed
[Dec 3 13:37:33] DEBUG[29641] rtp.c: p2p-rtp-bridge: Got a FRAME_CONTROL (25) frame on channel SIP/ast-00000000
[Dec 3 13:37:33] DEBUG[29641] channel.c: Returning from native bridge, channels: SIP/ast-00000000, SIP/100-00000001
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Changing ssrc from 1025821808 to 467100014 due to a source change
[Dec 3 13:37:33] DEBUG[29641] rtp.c: Changing ssrc from 223470155 to 1724575733 due to a source change
[Dec 3 13:37:33] VERBOSE[29641] rtp.c: -- Packet2Packet bridging SIP/ast-00000000 and SIP/100-00000001
[Dec 3 13:37:35] DEBUG[29641] rtp.c: Got RTCP report of 84 bytes
[Dec 3 13:37:41] DEBUG[29621] acl.c: Found IP address for this socket
[Dec 3 13:37:41] DEBUG[29621] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.100.4:5060
[Dec 3 13:37:41] DEBUG[29621] chan_sip.c: Allocating new SIP dialog for 84fff44c-a76f-7a17-8000-0002a40480e0@192.168.100.6 - REGISTER (No RTP)
[Dec 3 13:37:41] DEBUG[29621] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.100.6:5060
[Dec 3 13:37:41] DEBUG[29621] chan_sip.c: Store REGISTER's Contact header for call routing.
[Dec 3 13:37:41] DEBUG[29621] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.100.6:5060
[Dec 3 13:37:41] DEBUG[29594] chan_sip.c: Checking device state for peer 100
[Dec 3 13:37:41] DEBUG[29594] devicestate.c: Changing state for SIP/100 - state 1 (Not in use)
[Dec 3 13:37:41] DEBUG[29594] devicestate.c: device 'SIP/100' state '1'
[Dec 3 13:37:41] DEBUG[29604] app_queue.c: Device 'SIP/100' changed to state '1' (Not in use) but we don't care because they're not a member of any que
[Dec 3 13:37:42] DEBUG[29641] rtp.c: Got RTCP report of 84 bytes
[Dec 3 13:37:43] DEBUG[29641] rtp.c: Got RTCP report of 88 bytes
[Dec 3 13:37:43] DEBUG[29641] rtp.c: Got RTCP report of 64 bytes
[Dec 3 13:37:45] DEBUG[29621] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1a1f31674901d6e05ba422860469f822@192.168.100.4
[Dec 3 13:37:45] DEBUG[29621] chan_sip.c: Reeived bye, issuing owner hangup
[Dec 3 13:37:45] DEBUG[29621] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.100.6:5060
[Dec 3 13:37:45] DEBUG[29641] rtp.c: p2p-rtp-bridge: Ooh, got a hangup
[Dec 3 13:37:45] DEBUG[29641] channel.c: Returning from native bridge, channels: SIP/ast-00000000, SIP/100-00000001
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: substituted: string: 'dc=,dc=,dc=' => 'dc=,dc=,dc='.
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: basedn: 'dc=...,dc=...,dc=...' => 'dc=...,dc=...,dc=...'.
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: Everything seems fine.
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: name='exten' value='h'
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: name='context' value='office'
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: name='priority' value='1'
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: Could not find any entry matching (&(objectClass=AsteriskSIPUser)(AstExtension=h)(AstAccountContext=of
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: substituted: string: 'dc=...,dc=...,dc=...' => 'dc=...,dc=...,dc=...'.
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: basedn: 'dc=...,dc=...,dc=...' => 'dc=...,dc=...,dc=...'.
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: Everything seems fine.
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: name='exten LIKE' value='\_%'
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: name='context' value='office'
[Dec 3 13:37:45] DEBUG[29641] res_config_ldap.c: name='priority' value='1'

Could not find any entry matching (&(objectClass=AsteriskSIPUser)(AstExtension=h)(AstAccountContext=of

странная строка контекста.контекст вообще office ищется.

спасибо всем за советы и участие. Но проблема с вылетом решил, поставил генту х86 и этот же астериск с теми же конфами. все и заработало, точнее перестал астер сам выключаться.