Подскажите советом: при попытке позвонить на выключеный моб. тел идут три длинных гудка и обрыв.
debian 5.0 asterisk 1.6.1.20

[sip.conf]
[general]
context=default-sip ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; Specifying a port in a SIP peer definition or
; when dialing outbound calls will supress SRV
; lookups for that peer or call.

externip={IP сервера asterisk}
localnet=10.0.0.0/255.255.255.0
dtmfmode=rfc2833
dtmfcodec=97



[authentication]


[voipprov]
type=friend
host={IP провайдера}
context=voipprov_sip
disallow=all
nat=no
restrictcid=no
;dtmfmode=inband
insecure=invite
qualify=no
canreinvite=yes
;allow=g729
allow=alaw
allow=ulaw


[200] ; xlite phone
type=friend
nat=yes
host=dynamic
username=200
secret=yt ctujlyz
dtmfmode=rfc2833
canreinvite=no
context=office
callerid=<мой ном. тел.>

;allow=g729
allow=alaw
allow=ulaw

[201] ; xlite phone
type=friend
nat=yes
host=dynamic
username=200
secret=yt ctujlyz
dtmfmode=rfc2833
canreinvite=no
context=office
callerid=<мой ном. тел.>
;allow=g729
allow=alaw
allow=ulaw


[extensions.conf]
[general]
static=yes
writeprotect=no
clearglobalvars=no


[default-sip]
include => voipprov_sip
include => office

[office]
exten => 200,1,Dial(SIP/200)
exten => 201,1,Dial(SIP/201)

exten => 500,1,Verbose(1, VoIP Test)
exten => 500,n,Answer()
exten => 500,n,Echo()
exten => 500,n,Hangup()

include=>nabor

[nabor]
exten => _X.,1,Dial(SIP/voipprov/${EXTEN})

[voipprov_sip]
;exten => {мой ном. тел.},1,Answer
exten => {мой ном. тел.},1,Dial(SIP/200,10)
exten => {мой ном. тел.},n,GotoIf($[${DIALSTATUS} = BUSY])
exten => {мой ном. тел.},n,Dial(SIP/200&SIP/201,20)
exten => {мой ном. тел.},n,GotoIf($[${DIALSTATUS} = BUSY])
exten => {мой ном. тел.},n,Hangup



Вот логи:


== Using SIP RTP CoS mark 5
-- Executing [0678450025@office:1] Dial("SIP/200-00000000", "SIP/voipprov/0678450025") in new stack
== Using SIP RTP CoS mark 5
-- Called voipprov/0678450025
-- SIP/voipprov-00000001 is making progress passing it to SIP/200-00000000
-- Got SIP response 480 "Temporarily Unavailable" back from {IP провайдера}
-- SIP/voipprov-00000001 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/200-00000000' status is 'CONGESTION'






[Aug 13 12:00:11] NOTICE[6398] loader.c: 1 modules will be loaded.
[Aug 13 12:00:11] NOTICE[6398] cdr.c: CDR simple logging enabled.
[Aug 13 12:00:11] NOTICE[6398] loader.c: 162 modules will be loaded.
[Aug 13 12:00:11] NOTICE[6398] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Aug 13 12:00:11] WARNING[6398] utils.c: trying to reset empty pool
[Aug 13 12:00:11] WARNING[6398] utils.c: trying to reset empty pool
[Aug 13 12:00:11] WARNING[6398] utils.c: trying to reset empty pool
[Aug 13 12:00:11] VERBOSE[6398] chan_sip.c: SIP channel loading...
[Aug 13 12:00:11] NOTICE[6398] chan_sip.c: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser'
[Aug 13 12:00:11] NOTICE[6398] pbx_ael.c: Starting AEL load process.
[Aug 13 12:00:11] NOTICE[6398] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Aug 13 12:00:11] NOTICE[6398] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Aug 13 12:00:11] NOTICE[6398] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Aug 13 12:00:11] NOTICE[6398] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Aug 13 12:00:11] NOTICE[6398] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
[Aug 13 12:00:11] NOTICE[6398] chan_ooh323.c: ---------------------------------------------------------------------------------
--- ******* IMPORTANT NOTE ***********
---
--- This module is currently unsupported. Use it at your own risk.
---
---------------------------------------------------------------------------------
[Aug 13 12:00:11] WARNING[6398] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23.
[Aug 13 12:00:11] WARNING[6398] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Aug 13 12:00:11] WARNING[6398] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35.
[Aug 13 12:00:11] WARNING[6398] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39.
[Aug 13 12:00:11] WARNING[6398] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47.
[Aug 13 12:00:11] WARNING[6398] translate.c: plc_samples 160 format f
[Aug 13 12:00:19] DEBUG[6426] acl.c: Found IP address for this socket
[Aug 13 12:00:19] DEBUG[6426] chan_sip.c: Target address {IP провайдера} is not local, substituting externip
[Aug 13 12:00:19] DEBUG[6426] chan_sip.c: Setting SIP_TRANSPORT_UDP with address {IP сервера asterisk}:5060
[Aug 13 12:00:19] DEBUG[6426] chan_sip.c: Allocating new SIP dialog for bvenlvnhcvclhvxhjelclkcyobullcvj@SE2000 - OPTIONS (No RTP)
[Aug 13 12:00:19] DEBUG[6426] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[Aug 13 12:00:19] DEBUG[6426] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for {IP провайдера}:5060
[Aug 13 12:00:19] DEBUG[6426] chan_sip.c: SIP message could not be handled, bad request: bvenlvnhcvclhvxhjelclkcyobullcvj@SE2000 $
[Aug 13 12:00:24] DEBUG[6426] acl.c: Found IP address for this socket

[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Target address {IP клиента} is not local, substituting externip
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Setting SIP_TRANSPORT_UDP with address {IP сервера asterisk}:5060
[Aug 13 12:00:24] VERBOSE[6426] netsock.c: == Using SIP RTP CoS mark 5
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Setting NAT on RTP to Off
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Allocating new SIP dialog for 1281681162-1232-%D0%9C%D0%AB-%D0%9F%D0%9A@192.168.0.100 - INVITE (With RTP)
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Setting NAT on RTP to On
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for {IP клиента}:5070
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Stopping retransmission on '1281681162-1232-%D0%9C%D0%AB-%D0%9F%D0%9A@192.168.0.100' of Response 67: Match Found
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Setting NAT on RTP to On
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP o=NCHSoftware-Talk 1281681134 1281681165 IN IP4 192.168.0.100... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP s=Express Talk Call... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.100... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-32/8000... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:13 CN/8000... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=direction:active... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw)
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Checking SIP call limits for device 200
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Updating call counter for incoming call
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: *** Our native formats are 0x8 (alaw)
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw)
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263)
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)

[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: This channel will not be able to handle video.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: build_route: Contact hop: <sip:200@192.168.0.100:5070>
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: SIP/200-00000000: New call is still down.... Trying...
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for {IP клиента}:5070
[Aug 13 12:00:24] DEBUG[6403] devicestate.c: No provider found, checking channel drivers for SIP - 200
[Aug 13 12:00:24] DEBUG[6403] chan_sip.c: Checking device state for peer 200
[Aug 13 12:00:24] DEBUG[6403] devicestate.c: Changing state for SIP/200 - state 1 (Not in use)
[Aug 13 12:00:24] DEBUG[6403] devicestate.c: device 'SIP/200' state '1'
[Aug 13 12:00:24] DEBUG[6433] app_queue.c: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Aug 13 12:00:24] DEBUG[6437] pbx.c: Launching 'Dial'
[Aug 13 12:00:24] VERBOSE[6437] pbx.c: -- Executing [0678450025@office:1] Dial("SIP/200-00000000", "SIP/voipprov/0678450025") in new stack
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw)
[Aug 13 12:00:24] VERBOSE[6437] netsock.c: == Using SIP RTP CoS mark 5
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Allocating new SIP dialog for 374a21286b0c5e3023df884b053a9dd3@10.0.0.253 - INVITE (With RTP)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Setting NAT on RTP to Off
[Aug 13 12:00:24] DEBUG[6437] acl.c: Found IP address for this socket
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Target address {IP провайдера} is not local, substituting externip
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Setting SIP_TRANSPORT_UDP with address {IP сервера asterisk}:5060
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: *** Our native formats are 0x8 (alaw)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: *** Joint capabilities are 0x8 (alaw)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: This channel will not be able to handle video.
[Aug 13 12:00:24] DEBUG[6437] channel.c: Not copying variable DIALEDTIME.
[Aug 13 12:00:24] DEBUG[6437] channel.c: Not copying variable ANSWEREDTIME.
[Aug 13 12:00:24] DEBUG[6437] channel.c: Not copying variable DIALEDPEERNAME.
[Aug 13 12:00:24] DEBUG[6437] channel.c: Not copying variable DIALEDPEERNUMBER.
[Aug 13 12:00:24] DEBUG[6437] channel.c: Not copying variable DIALSTATUS.
[Aug 13 12:00:24] DEBUG[6437] channel.c: Not copying variable SIPCALLID.
[Aug 13 12:00:24] DEBUG[6437] channel.c: Not copying variable SIPDOMAIN.
[Aug 13 12:00:24] DEBUG[6437] channel.c: Not copying variable SIPURI.
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Outgoing Call for 0678450025
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Updating call counter for outgoing call

[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: ** Our prefcodec: 0x8 (alaw)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: -- Done with adding codecs to SDP
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Initializing initreq for method INVITE - callid 05546ddf4b3f38d81899955509c38bbf@{IP сервера asterisk}
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for {IP провайдера}:5060
[Aug 13 12:00:24] VERBOSE[6437] app_dial.c: -- Called voipprov/0678450025
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '05546ddf4b3f38d81899955509c38bbf@{IP сервера

asterisk}$
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: SIP response 100 to standard invite
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '05546ddf4b3f38d81899955509c38bbf@{IP сервера

asterisk}$
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: SIP response 183 to standard invite
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP o=HuaweiSoftX3000 216243 216243 IN IP4 {IP провайдера}... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP s=Sip Call... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP c=IN IP4 {IP провайдера}... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: We're settling with these formats: 0x8 (alaw)
[Aug 13 12:00:24] DEBUG[6426] chan_sip.c: We have an owner, now see if we need to change this call
[Aug 13 12:00:24] VERBOSE[6437] app_dial.c: -- SIP/voipprov-00000001 is making progress passing it to SIP/200-00000000
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Setting framing from config on incoming call
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Text flag: True
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: -- Done with adding codecs to SDP
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw)
[Aug 13 12:00:24] DEBUG[6437] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for {IP клиента}:5070
[Aug 13 12:00:24] DEBUG[6437] rtp.c: Ooh, format changed from unknown to alaw
[Aug 13 12:00:24] DEBUG[6437] rtp.c: Created smoother: format: 8 ms: 20 len: 160
[Aug 13 12:00:28] DEBUG[6437] rtp.c: Got RTCP report of 104 bytes
[Aug 13 12:00:28] DEBUG[6437] rtp.c: Unknown RTCP packet (pt=207) received from {IP провайдера}:31109
[Aug 13 12:00:29] DEBUG[6426] acl.c: Found IP address for this socket
[Aug 13 12:00:29] DEBUG[6426] chan_sip.c: Target address {IP провайдера} is not local, substituting externip
[Aug 13 12:00:29] DEBUG[6426] chan_sip.c: Setting SIP_TRANSPORT_UDP with address {IP сервера asterisk}:5060

[Aug 13 12:00:29] DEBUG[6426] chan_sip.c: Allocating new SIP dialog for kn88kvhxcocjlbun7nexej8uyjuboxxn@SE2000 - OPTIONS (No RTP)
[Aug 13 12:00:29] DEBUG[6426] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[Aug 13 12:00:29] DEBUG[6426] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for {IP провайдера}:5060
[Aug 13 12:00:29] DEBUG[6426] chan_sip.c: SIP message could not be handled, bad request: kn88kvhxcocjlbun7nexej8uyjuboxxn@SE2000 $
[Aug 13 12:00:33] DEBUG[6437] rtp.c: Got RTCP report of 104 bytes
[Aug 13 12:00:33] DEBUG[6437] rtp.c: Unknown RTCP packet (pt=207) received from {IP провайдера}:31109
[Aug 13 12:00:36] DEBUG[6426] chan_sip.c: Acked pending invite 102
[Aug 13 12:00:36] DEBUG[6426] chan_sip.c: Stopping retransmission on '05546ddf4b3f38d81899955509c38bbf@{IP сервера asterisk}' of Request 102: Match Found
[Aug 13 12:00:36] VERBOSE[6426] chan_sip.c: -- Got SIP response 480 "Temporarily Unavailable" back from {IP провайдера}
[Aug 13 12:00:36] DEBUG[6426] chan_sip.c: Trying to put 'ACK sip:067' onto UDP socket destined for {IP провайдера}:5060
[Aug 13 12:00:36] DEBUG[6426] chan_sip.c: Setting SIP_ALREADYGONE on dialog 05546ddf4b3f38d81899955509c38bbf@{IP сервера asterisk}
[Aug 13 12:00:36] VERBOSE[6437] app_dial.c: -- SIP/voipprov-00000001 is circuit-busy
[Aug 13 12:00:36] DEBUG[6437] channel.c: Hanging up channel 'SIP/voipprov-00000001'
[Aug 13 12:00:36] DEBUG[6437] chan_sip.c: Hangup call SIP/voipprov-00000001, SIP callid 05546ddf4b3f38d81899955509c38bbf@{IP сервера asterisk}
[Aug 13 12:00:36] VERBOSE[6437] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[Aug 13 12:00:36] DEBUG[6403] devicestate.c: No provider found, checking channel drivers for SIP - voipprov
[Aug 13 12:00:36] DEBUG[6437] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
[Aug 13 12:00:36] DEBUG[6403] chan_sip.c: Checking device state for peer voipprov
[Aug 13 12:00:36] DEBUG[6403] devicestate.c: Changing state for SIP/voipprov - state 1 (Not in use)
[Aug 13 12:00:36] DEBUG[6403] devicestate.c: device 'SIP/voipprov' state '1'
[Aug 13 12:00:36] DEBUG[6437] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
[Aug 13 12:00:36] DEBUG[6433] app_queue.c: Device 'SIP/voipprov' changed to state '1' (Not in use) but we don't care because they're not a member of any qu$
[Aug 13 12:00:36] VERBOSE[6437] pbx.c: -- Auto fallthrough, channel 'SIP/200-00000000' status is 'CONGESTION'
[Aug 13 12:00:36] DEBUG[6437] chan_sip.c: Trying to put 'SIP/2.0 503' onto UDP socket destined for {IP клиента}:5070
[Aug 13 12:00:36] DEBUG[6437] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1281681162-1232-%D0%9C%D0%AB-%D0%9F%D0%9A@192.168.0.100
[Aug 13 12:00:36] DEBUG[6437] channel.c: Soft-Hanging up channel 'SIP/200-00000000'
[Aug 13 12:00:36] DEBUG[6437] channel.c: Soft-Hanging up channel 'SIP/200-00000000'
[Aug 13 12:00:36] DEBUG[6403] devicestate.c: No provider found, checking channel drivers for SIP - 200
[Aug 13 12:00:36] DEBUG[6437] channel.c: Hanging up channel 'SIP/200-00000000'
[Aug 13 12:00:36] DEBUG[6437] chan_sip.c: Hangup call SIP/200-00000000, SIP callid 1281681162-1232-%D0%9C%D0%AB-%D0%9F%D0%9A@192.168.0.100
[Aug 13 12:00:36] DEBUG[6403] chan_sip.c: Checking device state for peer 200
[Aug 13 12:00:36] DEBUG[6403] devicestate.c: Changing state for SIP/200 - state 1 (Not in use)
[Aug 13 12:00:36] DEBUG[6403] devicestate.c: device 'SIP/200' state '1'
[Aug 13 12:00:36] DEBUG[6433] app_queue.c: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Aug 13 12:00:36] DEBUG[6403] devicestate.c: No provider found, checking channel drivers for SIP - 200
[Aug 13 12:00:36] DEBUG[6403] chan_sip.c: Checking device state for peer 200
[Aug 13 12:00:36] DEBUG[6403] devicestate.c: Changing state for SIP/200 - state 1 (Not in use)
[Aug 13 12:00:36] DEBUG[6403] devicestate.c: device 'SIP/200' state '1'
[Aug 13 12:00:36] DEBUG[6433] app_queue.c: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Aug 13 12:00:36] DEBUG[6426] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Aug 13 12:00:36] DEBUG[6426] chan_sip.c: Stopping retransmission on '1281681162-1232-%D0%9C%D0%AB-%D0%9F%D0%9A@192.168.0.100' of Response 68: Match Found
[Aug 13 12:00:36] DEBUG[6426] chan_sip.c: Destroying SIP dialog 1281681162-1232-%D0%9C%D0%AB-%D0%9F%D0%9A@192.168.0.100
[Aug 13 12:00:36] DEBUG[6426] chan_sip.c: Destroying SIP dialog 05546ddf4b3f38d81899955509c38bbf@{IP сервера asterisk}
[Aug 13 12:00:39] DEBUG[6426] acl.c: Found IP address for this socket
[Aug 13 12:00:39] DEBUG[6426] chan_sip.c: Target address {IP провайдера} is not local, substituting externip
[Aug 13 12:00:39] DEBUG[6426] chan_sip.c: Setting SIP_TRANSPORT_UDP with address {IP сервера asterisk}:5060
[Aug 13 12:00:39] DEBUG[6426] chan_sip.c: Allocating new SIP dialog for 7o88chjvocnlbyicnobjeyxc78yx8iuh@SE2000 - OPTIONS (No RTP)
[Aug 13 12:00:39] DEBUG[6426] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[Aug 13 12:00:39] DEBUG[6426] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for {IP провайдера}:5060
[Aug 13 12:00:39] DEBUG[6426] chan_sip.c: SIP message could not be handled, bad request: 7o88chjvocnlbyicnobjeyxc78yx8iuh@SE2000 $
[Aug 13 12:00:42] VERBOSE[6436] asterisk.c: -- Remote UNIX connection disconnected
[Aug 13 12:00:45] DEBUG[6426] chan_sip.c: Auto destroying SIP dialog 'uikbyjyl8y8yyekekie8lbchie7vu7ob@SE2000'
[Aug 13 12:00:45] DEBUG[6426] chan_sip.c: Destroying SIP dialog uikbyjyl8y8yyekekie8lbchie7vu7ob@SE2000
[Aug 13 12:00:49] DEBUG[6426] acl.c: Found IP address for this socket
[Aug 13 12:00:49] DEBUG[6426] chan_sip.c: Target address {IP провайдера} is not local, substituting externip
[Aug 13 12:00:49] DEBUG[6426] chan_sip.c: Setting SIP_TRANSPORT_UDP with address {IP сервера asterisk}:5060
[Aug 13 12:00:49] DEBUG[6426] chan_sip.c: Allocating new SIP dialog for icucohlbh7hnykiehebue7hh8ijlibli@SE2000 - OPTIONS (No RTP)
[Aug 13 12:00:49] DEBUG[6426] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[Aug 13 12:00:49] DEBUG[6426] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for {IP провайдера}:5060
[Aug 13 12:00:49] DEBUG[6426] chan_sip.c: SIP message could not be handled, bad request: icucohlbh7hnykiehebue7hh8ijlibli@SE2000

А Вам что надо?
И что делает эта строчка
exten => {мой ном. тел.},n,GotoIf($[${DIALSTATUS} = BUSY]) ?
статус нужно отлавивать 'CONGESTION'

Проблема в том , что вместо гудков и отбоя , должен идти автоответ , что абонент недоступен , а у меня гудки+отбой. Оператор говорит , мол наш сервак не понимает , что ему посылают session stage о том , что включ. автоответ (гудки), и через 10 сек. провайдер , не дождавшись ответа, отбивает по тайм-ауту (отбой).

Может нужно через Hangupcause , только как ума не приложу. Все примеры в инете перепробовал.

Лог неполный , там должен быть вывод Hangupcause. А лучше еще раз подумать , что же Вам все таки нужно.
Еще , я вроде как спросил про мифический диаплан. Сядьте и подумайте над своими строками, они логически неправильные.

В том то и дело , что я не использую Hangupcause. А в диалплане написано , что если секретарь неберёт трубку 10 сек или занята , начинает звонить и телефон секретаря и телефон менеджера. Что делать ума не приложу...

exten => {мой ном. тел.},n,GotoIf($[${DIALSTATUS} = BUSY])
Зачем эта строчка?

Не проходит early media.

Пробовали sip.conf: [general]
progressinband=yes ?

Спасибо за участие. Добавил - тоже самое ((( Есть ещё варианты?

Скажите , что происходит у Вас если позвонить на мобильный который отключен?

Неужели тут у всех такая история?

А что за оборудование у провайдера стоит? Какая схема подключения?

С одной стороны провайдер с HuaweiSoftX3000, с другой я -asterisk 1.6.1.20. Клиенты - софтфоны express talk и x-lite