BAZA IP не проходят исходящие и входящие звонки

Приветствую!

Помогите разобраться в чем проблема, не могу никак настроить в Trixbox Исходящие и Входящие звонки с BAZA Офис 495.

Конфигурация:

Транк:

Trunk - Baza (sip) Trunk Name: Baza Outbound Caller ID: 8495XXXXXXX Dialed Number Manipulation Rules: . Trunk Name: Baza *Peer Details:* disallow=all allow=alaw&ulaw&gsm canredirect=no canreinvite=no host=qwerty.cnt.ru insecure=port,invite secret=PASSWD type=peer username=8495XXXXXXX fromuser=8495XXXXXXX fromdomain=qwerty.cnt.ru context=from-trunk User Context: 8495XXXXXXX *User Details:* secret=PASSWD type=user context=from-trunk Register String: 8495XXXXXXX:PASSWD@qwerty.cnt.ru/8495XXXXXXX 

Исходящий маршрут:

Outband Routes Route Name: BazaOut Dial Patterns: 8916XXXXXXX Trunk Sequence 0 baza 

Входящий маршрут:

Description: BazaOut DID Number: 8495XXXXXXX Extensions: <101> Alex 

Логи:

trixbox1*CLI> sip show registry Host Username Refresh State Reg.Time qwerty.cnt.ru:5060 84956693455 120 Request Sent 1 SIP registrations. 

Исходящий звонок на мобильный 8916XXXXXXX:

 -- Called Baza/89161111111 -- ast_get_srv: SRV lookup for '_sip._UDP.qwerty.cnt.ru' mapped to host sip.qwerty.cnt.ru, port 5060 REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 213.85.168.52:5060: REGISTER sip:qwerty.cnt.ru SIP/2.0 Via: SIP/2.0/UDP 89.XXX.XXX.XXX:5060;branch=z9hG4bK3dcfada6;rport Max-Forwards: 70 From: <sip:84956693455@qwerty.cnt.ru>;tag=as392d8517 To: <sip:84956693455@qwerty.cnt.ru> Call-ID: 4152177e200c66e670fbc08503521803@127.0.0.1 CSeq: 195 REGISTER User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78 Expires: 120 Contact: <sip:84956693455@89.XXX.XXX.XXX> Event: registration Content-Length: 0 --- Really destroying SIP dialog '4152177e200c66e670fbc08503521803@127.0.0.1' Method: REGISTER trixbox1*CLI> <--- SIP read from UDP://192.168.1.4:53060 ---> <-------------> Scheduling destruction of SIP dialog '4525185d746db5a4367db8686cdbdb79@qwerty.cnt.ru' in 32000 ms (Method: INVITE) -- SIP/Baza-00000007 is circuit-busy Scheduling destruction of SIP dialog '4525185d746db5a4367db8686cdbdb79@qwerty.cnt.ru' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:20] Goto("SIP/101-00000006", "s-CONGESTION,1") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/101-00000006", "1?noreport") in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,3) -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/101-00000006", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack -- Executing [89161111111@from-internal:5] Macro("SIP/101-00000006", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Playback("SIP/101-00000006", "all-circuits-busy-now,noanswer") in new stack Audio is at 192.168.1.197 port 13088 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP trixbox1*CLI> <--- Transmitting (no NAT) to 192.168.1.4:53060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.4:53060;branch=z9hG4bK-d8754z-51dfe2673b88bc76-1---d8754z-;received=192.168.1.4;rport=53060 From: "п-п¦п¦п¦я¦п¦п¦"<sip:101@192.168.1.197:5060>;tag=4f0d8891 To: <sip:89161111111@192.168.1.197:5060>;tag=as475dc30c Call-ID: MmY4MGVmNmIxZDZlNDM2NDg3ZWMzNjBkZjdkOTg2Mjc. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:89161111111@192.168.1.197> Content-Type: application/sdp Content-Length: 302 v=0 o=root 1952683458 1952683458 IN IP4 192.168.1.197 s=Asterisk PBX 1.6.0.26-FONCORE-r78 c=IN IP4 192.168.1.197 t=0 0 m=audio 13088 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- <SIP/101-00000006> Playing 'all-circuits-busy-now.ulaw' (language 'en') -- Executing [s@macro-outisbusy:2] Playback("SIP/101-00000006", "pls-try-call-later,noanswer") in new stack -- <SIP/101-00000006> Playing 'pls-try-call-later.ulaw' (language 'en') -- Executing [s@macro-outisbusy:3] Macro("SIP/101-00000006", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000006", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/101-00000006", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/101-00000006", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/101-00000006", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/101-00000006' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/101-00000006' in macro 'outisbusy' == Spawn extension (from-internal, 89161111111, 5) exited non-zero on 'SIP/101-00000006' -- Executing [h@from-internal:1] Macro("SIP/101-00000006", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000006", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/101-00000006", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/101-00000006", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/101-00000006", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/101-00000006' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000006' Scheduling destruction of SIP dialog 'MmY4MGVmNmIxZDZlNDM2NDg3ZWMzNjBkZjdkOTg2Mjc.' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 192.168.1.4:53060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.4:53060;branch=z9hG4bK-d8754z-51dfe2673b88bc76-1---d8754z-;received=192.168.1.4;rport=53060 From: "п-п¦п¦п¦я¦п¦п¦"<sip:101@192.168.1.197:5060>;tag=4f0d8891 To: <sip:89161111111@192.168.1.197:5060>;tag=as475dc30c Call-ID: MmY4MGVmNmIxZDZlNDM2NDg3ZWMzNjBkZjdkOTg2Mjc. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> trixbox1*CLI> <--- SIP read from UDP://192.168.1.4:53060 ---> ACK sip:89161111111@192.168.1.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.4:53060;branch=z9hG4bK-d8754z-51dfe2673b88bc76-1---d8754z-;rport Max-Forwards: 70 To: <sip:89161111111@192.168.1.197:5060>;tag=as475dc30c From: "п-п¦п¦п¦я¦п¦п¦"<sip:101@192.168.1.197:5060>;tag=4f0d8891 Call-ID: MmY4MGVmNmIxZDZlNDM2NDg3ZWMzNjBkZjdkOTg2Mjc. CSeq: 2 ACK Content-Length: 0 

Соответственно всем время выдает сигнал занято. При входящем звонке в соответствии с правилами выставлена переадресация на номер 880308495XXXXXXX (по правилам Базы), однако никакой реакции на входящий звонок нет.

При настройке софтфона напрямую через Baza IP и исходящие и входящие звонки работают нормально.