Здравствуйте не могу настроить Trixbox на выход в город через FXO порт устройсва AudioCodes MP-114

вот текст при наборе городского номера
-- Executing [222222@from-internal:1] Macro("SIP/100-b7812040", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/100-b7812040", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/100-b7812040", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/100-b7812040", "1|Set|REALCALLERIDNUM=100") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/100-b7812040", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/100-b7812040", "AMPUSERCIDNAME=100") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/100-b7812040", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/100-b7812040", "AMPUSERCID=100") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/100-b7812040", "CALLERID(all)="100" <100>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/100-b7812040", "REALCALLERIDNUM=100") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/100-b7812040", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/100-b7812040", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/100-b7812040", "Using CallerID "100" <100>") in new stack
-- Executing [222222@from-internal:2] Set("SIP/100-b7812040", "_NODEST=") in new stack
-- Executing [222222@from-internal:3] Macro("SIP/100-b7812040", "record-enable|100|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/100-b7812040", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/100-b7812040", "recordingcheck|20090427-122811|1240795691.21") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090427-122811|1240795691.21: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/100-b7812040", "") in new stack
-- Executing [222222@from-internal:4] Macro("SIP/100-b7812040", "dialout-trunk|2|222222||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/100-b7812040", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/100-b7812040", "0?sub-pincheck|s|1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/100-b7812040", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/100-b7812040", "DIAL_NUMBER=222222") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/100-b7812040", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/100-b7812040", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/100-b7812040", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/100-b7812040", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/100-b7812040", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/100-b7812040", "outbound-callerid|2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/100-b7812040", "0|SetCallerPres|") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/100-b7812040", "0|Set|REALCALLERIDNUM=100") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/100-b7812040", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/100-b7812040", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/100-b7812040", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/100-b7812040", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/100-b7812040", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/100-b7812040", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/100-b7812040", "1?exit") in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/100-b7812040", "") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/100-b7812040", "0|AGI|fixlocalprefix") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/100-b7812040", "OUTNUM=222222") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/100-b7812040", "custom=SIP/pstn") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-b7812040", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/100-b7812040", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-b7812040", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/100-b7812040", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-b7812040", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-b7812040", "SIP/pstn/222222|300|") in new stack
-- Called pstn/222222
-- SIP/pstn-093c1a48 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/100-b7812040", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/100-b7812040", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/100-b7812040", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [222222@from-internal:5] Macro("SIP/100-b7812040", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/100-b7812040", "all-circuits-busy-now|noanswer") in new stack
-- <SIP/100-b7812040> Playing 'all-circuits-busy-now' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/100-b7812040", "pls-try-call-later|noanswer") in new stack
-- <SIP/100-b7812040> Playing 'pls-try-call-later' (language 'en')
-- Executing [s@macro-outisbusy:3] Macro("SIP/100-b7812040", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/100-b7812040", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/100-b7812040", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/100-b7812040", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/100-b7812040", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/100-b7812040", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/100-b7812040", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/100-b7812040' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/100-b7812040' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/100-b7812040'

вы этим логом только констатируете факт, что не можете позвонить:
-- Called pstn/222222
-- SIP/pstn-093c1a48 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

никаких идей, почему это может происходить, лог не дает.

а что надо? настройки взяты с сайта
http://trixbox.org/forums/trixbox-forums/trunks/trixbox-2-2-and-audio-codes-mp-114-fxo-setup

Outgoing Settings
Trunk Name: PSTN


PEER Details:
allow=ulaw
context=from-trunk
dtmfmode=inband
host=192.168.0.7
nat=no
qualify=no
type=peer

Incoming Settings
USER context: MP114
User details:
canreinvite=no
context=from-trunk
dtmfmode=inband
host=192.168.0.7
nat=never
type=user
Submit changes

Outbound routes->add route
Route name : PSTN
Dial patterns:NNXXXX
Trunk Sequence: The Trunk Sequence controls the order of trunks that will be used when the above Dial Patterns are matched.

For Dial Patterns that match long distance numbers, for example, you'd want to pick the cheapest routes for long distance (ie, VoIP trunks first) followed by more expensive routes (POTS lines).
SIP/PSTN

-AudioCodes Setup

Quick Setup:
IP configuration: If you can’t figure this one out there’s little or no chance you’ll get this working. Put on dunce cap and sit in corner.
SIP parameters:
Gateway Name: I use it’s IP address so no dns issues.
Working with Proxy = Yes
Proxy IP address= the IP of the Asterisk box.
Proxy Name= the IP of the Asterisk box.
Protocol Management:
Protocol Definition ->
General Parameters:
Channel Select Mode=Ascending
And make sure SIP ports are set for 5060
Proxy and Registration:
Proxy Name and Proxy IP Address= Asterisk Server
Enable Registration: I didn’t .
Gateway Name and Registration Name: MP-114 IP address
Subscription and Registration Mode: Per Gateway (don’t remember if this matters).
Coders:make sure ulaw’s there
DTMF & Dialing: Max digits-> put a high number like 32
Routing Tables:
Tel -> IP routing and IP-> Tel routing = I used
Dest IP/Phone Prefix =*
Source IP/Phone Prefix =*
Dest/Source IP Address = Asterisk IP Address
Endpoint Phone Numbers: Match channels to phone numbers.
Channels= 1 -4
Phone numbers = your phone numbers
Hunt Group Settings:
I used Cyclical Ascending
End Point Settings:
Automatic Dialing:Destination Phone Numbers should match the numbers you have in inbound context in extensions.conf. In our example -> exten => _ 2125551212,1,Answer()
Advanced Applications:
FXO Settings: Dialing Mode should be set to One Stage.
That should get you up and running. Although little differences in setups can cause major headaches and frustrations, I hope that this gives you a good starting reference point. We’ll be putting this and other guides on our wiki when it becomes available (with screencaps).

надо написать в консоли sip set debug и посмотреть лог

etransmitting #3 (NAT) to 192.168.0.6:5060:
OPTIONS sip:192.168.0.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK11369554;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as6fec0c7f
To: <sip:192.168.0.6>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 5385bb962fc12e622143a4b1307bf6fe@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 28 Apr 2009 03:52:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
trixbox1*CLI>
<--- SIP read from 192.168.0.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK11369554;rport
4U└*From: "Unknown"<sip:Unknown@192.168.0.1>;tag=as6fec0c7f
To: <sip:192.168.0.6>
Call-ID: 5385bb962fc12e622143a4b1307bf6fe@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Retransmitting #3 (NAT) to 192.168.0.6:5060:
OPTIONS sip:192.168.0.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7c14609a;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as55add409
To: <sip:192.168.0.6>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 4ae4c1cc6fe748194b24e6357ec0843e@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 28 Apr 2009 03:52:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
trixbox1*CLI>
<--- SIP read from 192.168.0.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7c14609a;rport
4U└*From: "Unknown"<sip:Unknown@192.168.0.1>;tag=as55add409
To: <sip:192.168.0.6>
Call-ID: 4ae4c1cc6fe748194b24e6357ec0843e@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from 192.168.0.8:1059 --->
OPTIONS sip:192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.8:1059;rport;branch=z9hG4bKc0a803720000000b49f68c80000046ed000004d5
Content-Length: 0
Call-ID: 253E60D6-533B-41DF-9899-5176F4A0B504@192.168.3.114
CSeq: 346 OPTIONS
From: <sip:501@192.168.0.8>;tag=72292812927
Max-Forwards: 70
To: <sip:192.168.0.1>


<------------->
--- (8 headers 0 lines) ---
Looking for s in others (domain 192.168.0.1)

<--- Transmitting (no NAT) to 192.168.0.8:1059 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.8:1059;branch=z9hG4bKc0a803720000000b49f68c80000046ed000004d5;received=192.168.0.8;rport=1059
From: <sip:501@192.168.0.8>;tag=72292812927
To: <sip:192.168.0.1>;tag=as13a4f235
Call-ID: 253E60D6-533B-41DF-9899-5176F4A0B504@192.168.3.114
CSeq: 346 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '253E60D6-533B-41DF-9899-5176F4A0B504@192.168.3.114' in 32000 ms (Method: OPTIONS)
Retransmitting #4 (NAT) to 192.168.0.6:5060:
OPTIONS sip:192.168.0.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK16afaa51;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as76e5d0cf
To: <sip:192.168.0.6>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 16cb892f186ef0371be9af962ae6a7c9@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 28 Apr 2009 03:52:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '16cb892f186ef0371be9af962ae6a7c9@192.168.0.1' Method: OPTIONS
trixbox1*CLI>
<--- SIP read from 192.168.0.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK16afaa51;rport
4U└*From: "Unknown"<sip:Unknown@192.168.0.1>;tag=as76e5d0cf
To: <sip:192.168.0.6>
Call-ID: 16cb892f186ef0371be9af962ae6a7c9@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Retransmitting #4 (NAT) to 192.168.0.6:5060:
OPTIONS sip:192.168.0.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK182394bb;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as2045ac47
To: <sip:192.168.0.6>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 2be1a1927175ef6f349c29bc23b73162@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 28 Apr 2009 03:52:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '2be1a1927175ef6f349c29bc23b73162@192.168.0.1' Method: OPTIONS
trixbox1*CLI>
<--- SIP read from 192.168.0.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK182394bb;rport
4U└*From: "Unknown"<sip:Unknown@192.168.0.1>;tag=as2045ac47
To: <sip:192.168.0.6>
Call-ID: 2be1a1927175ef6f349c29bc23b73162@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Retransmitting #4 (NAT) to 192.168.0.6:5060:
OPTIONS sip:192.168.0.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK11369554;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as6fec0c7f
To: <sip:192.168.0.6>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 5385bb962fc12e622143a4b1307bf6fe@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 28 Apr 2009 03:52:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '5385bb962fc12e622143a4b1307bf6fe@192.168.0.1' Method: OPTIONS
trixbox1*CLI>
<--- SIP read from 192.168.0.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK11369554;rport
4U└*From: "Unknown"<sip:Unknown@192.168.0.1>;tag=as6fec0c7f
To: <sip:192.168.0.6>
Call-ID: 5385bb962fc12e622143a4b1307bf6fe@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Retransmitting #4 (NAT) to 192.168.0.6:5060:
OPTIONS sip:192.168.0.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7c14609a;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as55add409
To: <sip:192.168.0.6>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 4ae4c1cc6fe748194b24e6357ec0843e@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 28 Apr 2009 03:52:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '4ae4c1cc6fe748194b24e6357ec0843e@192.168.0.1' Method: OPTIONS
trixbox1*CLI>
<--- SIP read from 192.168.0.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7c14609a;rport
4U└*From: "Unknown"<sip:Unknown@192.168.0.1>;tag=as55add409
To: <sip:192.168.0.6>
Call-ID: 4ae4c1cc6fe748194b24e6357ec0843e@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

что выдернуть успел..

В приведенном логе нет ни одного INVITE (SURPISE!) в сторону шлюза,
значит неверно настроено еще на стороне астериска.
Кто-то похоже забыл что unix-like ОС case _sensitivity_
Рекомендую внимательно прочитать свое-же сообщение где цитируется какая-то документация.

unix-like ОС case _sensitivity_ - что это означает?

а под документацией подразумевается сайт с настройками???

подразумевалось ровно то, что написано - ваше же сообщение от 2009-04-28 07:40

настройки что там написанны скопированны с *.. что там не так может есть варианты?