ну скажем это прогресс, набор номера уже идёт, правда до абонента не дозваниваюсь..
вот лог:

v=0
o=701 24205404 24205404 IN IP4 192.168.0.6
s=ACT G1104S 01.02
c=IN IP4 192.168.0.6
t=0 0
m=audio 10000 RTP/AVP 0 8 18 4
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:4 G723/8000/1
a=rtcp:10001
a=direction:both
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Sending to 192.168.0.6 : 5060 (no NAT)
Using INVITE request as basis request - CALL_ID5_001346E0DEF0_T4301945@192.168.0.6
Found user '701'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Peer audio RTP is at port 192.168.0.6:10000
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.6:10000
Looking for 100 in from-internal (domain 192.168.0.1)
list_route: hop: <sip:701@192.168.0.6:5060>
trixbox1*CLI>
<--- Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK_001346E0DEF0_T0041A479;received=192.168.0.6
From: "701"<sip:701@192.168.0.1>;tag=001346E0DEF0_T4301945
To: <sip:100@192.168.0.1>
Call-ID: CALL_ID5_001346E0DEF0_T4301945@192.168.0.6
CSeq: 6787 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@192.168.0.1>
Content-Length: 0


<------------>
-- Executing [100@from-internal:1] Macro("SIP/701-087c46d0", "exten-vm|novm|100") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/701-087c46d0", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/701-087c46d0", "AMPUSER=701") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/701-087c46d0", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/701-087c46d0", "1|Set|REALCALLERIDNUM=701") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/701-087c46d0", "AMPUSER=701") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/701-087c46d0", "AMPUSERCIDNAME=DVG2004") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/701-087c46d0", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/701-087c46d0", "AMPUSERCID=701") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/701-087c46d0", "CALLERID(all)="DVG2004" <701>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/701-087c46d0", "REALCALLERIDNUM=701") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/701-087c46d0", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/701-087c46d0", "0?continue") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/701-087c46d0", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/701-087c46d0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/701-087c46d0", "Using CallerID "DVG2004" <701>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/701-087c46d0", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/701-087c46d0", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/701-087c46d0", "EXTTOCALL=100") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/701-087c46d0", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/701-087c46d0", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/701-087c46d0", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/701-087c46d0", "record-enable|100|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/701-087c46d0", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/701-087c46d0", "recordingcheck|20090512-153023|1242102623.26") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090512-153023|1242102623.26: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/701-087c46d0", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/701-087c46d0", "dial||tTr|100") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/701-087c46d0", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/701-087c46d0", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'DVG2004' number is '701'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 100 to extension map
dialparties.agi: Extension 100 has call forward set to 9746
> dialparties.agi: Primary ext is CF so disabling mastermode if it was set
> dialparties.agi: extnum 9746# has: cw: 0; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: Built External dialstring component for 9746: Local/9746@from-internal/n
-- dialparties.agi: Filtered ARG3: 9746
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/701-087c46d0", "Local/9746@from-internal/n||tTr") in new stack
-- Called 9746@from-internal/n
-- Executing [9746@from-internal:1] ResetCDR("Local/9746@from-internal-af76,2", "") in new stack

<--- Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK_001346E0DEF0_T0041A479;received=192.168.0.6
From: "701"<sip:701@192.168.0.1>;tag=001346E0DEF0_T4301945
To: <sip:100@192.168.0.1>;tag=as547069da
Call-ID: CALL_ID5_001346E0DEF0_T4301945@192.168.0.6
CSeq: 6787 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@192.168.0.1>
Content-Length: 0


<------------>
-- Executing [9746@from-internal:2] NoCDR("Local/9746@from-internal-af76,2", "") in new stack
-- Executing [9746@from-internal:3] Wait("Local/9746@from-internal-af76,2", "1") in new stack
Really destroying SIP dialog 'REGISTER_001346E0DEF0_T4286101@192.168.0.6' Method: REGISTER
-- Executing [9746@from-internal:4] Playback("Local/9746@from-internal-af76,2", "silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer") in new stack
-- <Local/9746@from-internal-af76,2> Playing 'silence/1' (language 'en')
-- <Local/9746@from-internal-af76,2> Playing 'cannot-complete-as-dialed' (language 'en')
-- <Local/9746@from-internal-af76,2> Playing 'check-number-dial-again' (language 'en')
Scheduling destruction of SIP dialog '6db75f0c02c97e9130958a713d823f25@192.168.0.1' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.0.6:5060:
NOTIFY sip:701@192.168.0.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0667ccb1;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as688e8861
To: <sip:701@192.168.0.6:5060>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 6db75f0c02c97e9130958a713d823f25@192.168.0.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 86

Messages-Waiting: no
Message-Account: sip:*97@192.168.0.1
Voice-Message: 0/0 (0/0)

---
trixbox1*CLI>
<--- SIP read from 192.168.0.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0667ccb1;rport
From: "Unknown"<sip:Unknown@192.168.0.1>;tag=as688e8861
To: <sip:701@192.168.0.6>
Call-ID: 6db75f0c02c97e9130958a713d823f25@192.168.0.1
CSeq: 102 NOTIFY
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '6db75f0c02c97e9130958a713d823f25@192.168.0.1' Method: NOTIFY
-- Executing [9746@from-internal:5] Wait("Local/9746@from-internal-af76,2", "1") in new stack
-- Executing [9746@from-internal:6] Congestion("Local/9746@from-internal-af76,2", "20") in new stack
-- Local/9746@from-internal-af76,1 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dial:8] Set("SIP/701-087c46d0", "DIALSTATUS=CONGESTION") in new stack
-- Executing [s@macro-dial:9] GosubIf("SIP/701-087c46d0", "0?CONGESTION|1") in new stack
== Spawn extension (from-internal, 9746, 6) exited non-zero on 'Local/9746@from-internal-af76,2'
-- Executing [h@from-internal:1] Macro("Local/9746@from-internal-af76,2", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("Local/9746@from-internal-af76,2", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("Local/9746@from-internal-af76,2", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("Local/9746@from-internal-af76,2", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("Local/9746@from-internal-af76,2", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("Local/9746@from-internal-af76,2", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("Local/9746@from-internal-af76,2", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/9746@from-internal-af76,2' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/9746@from-internal-af76,2'
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/701-087c46d0", "0?exit|return") in new stack
-- Executing [s@macro-exten-vm:11] Set("SIP/701-087c46d0", "SV_DIALSTATUS=CONGESTION") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/701-087c46d0", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/701-087c46d0", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/701-087c46d0", "DIALSTATUS=CONGESTION") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/701-087c46d0", "Voicemail is novm") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/701-087c46d0", "1?s-CONGESTION|1") in new stack
-- Goto (macro-exten-vm,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-exten-vm:1] NoOp("SIP/701-087c46d0", "IVR_RETVM: IVR_CONTEXT: ") in new stack
-- Executing [s-CONGESTION@macro-exten-vm:2] GotoIf("SIP/701-087c46d0", "0?exit|1") in new stack
-- Executing [s-CONGESTION@macro-exten-vm:3] PlayTones("SIP/701-087c46d0", "congestion") in new stack
Audio is at 192.168.0.1 port 13428
Adding codec 0x4 (ulaw) to SDP

<--- Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK_001346E0DEF0_T0041A479;received=192.168.0.6
From: "701"<sip:701@192.168.0.1>;tag=001346E0DEF0_T4301945
To: <sip:100@192.168.0.1>;tag=as547069da
Call-ID: CALL_ID5_001346E0DEF0_T4301945@192.168.0.6
CSeq: 6787 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@192.168.0.1>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 2757 2757 IN IP4 192.168.0.1
s=session
c=IN IP4 192.168.0.1
t=0 0
m=audio 13428 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
-- Executing [s-CONGESTION@macro-exten-vm:4] Congestion("SIP/701-087c46d0", "10") in new stack

<--- Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK_001346E0DEF0_T0041A479;received=192.168.0.6
From: "701"<sip:701@192.168.0.1>;tag=001346E0DEF0_T4301945
To: <sip:100@192.168.0.1>;tag=as547069da
Call-ID: CALL_ID5_001346E0DEF0_T4301945@192.168.0.6
CSeq: 6787 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@192.168.0.1>
Content-Length: 0


<------------>
== Spawn extension (macro-exten-vm, s-CONGESTION, 4) exited non-zero on 'SIP/701-087c46d0' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-CONGESTION, 4) exited non-zero on 'SIP/701-087c46d0'
trixbox1*CLI>
<--- SIP read from 192.168.0.6:5060 --->
ACK sip:100@192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK_001346E0DEF0_T0041A479
From: "701"<sip:701@192.168.0.1>;tag=001346E0DEF0_T4301945
To: <sip:100@192.168.0.1>;tag=as547069da
Call-ID: CALL_ID5_001346E0DEF0_T4301945@192.168.0.6
CSeq: 6787 ACK
User-Agent: G1104S 01.02
Contact: <sip:701@192.168.0.6:5060>
Max-Forwards: 70
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 'CALL_ID5_001346E0DEF0_T4301945@192.168.0.6' Method: ACK

CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 May 2009 04:33:38 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=z9hG4bK38e576de;rport
Allow: ACK,BYE,CANCEL,INMADDRFrom: "Unknown"<sip:Unknown@192.168.0.1>;tag=as3b5144d9
To: <sip:192.168.0.6>
Call-ID: 0192ebaa43c161884a92bafa354d0d19@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Retransmitting #2 (no 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=z9hG4bK38e576de;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as3b5144d9
To: <sip:192.168.0.6>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 0192ebaa43c161884a92bafa354d0d19@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 May 2009 04:33:38 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=z9hG4bK38e576de;rport
Allow: ACK,BYE,CANCEL,INMADDRFrom: "Unknown"<sip:Unknown@192.168.0.1>;tag=as3b5144d9
To: <sip:192.168.0.6>
Call-ID: 0192ebaa43c161884a92bafa354d0d19@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Retransmitting #3 (no 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=z9hG4bK38e576de;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as3b5144d9
To: <sip:192.168.0.6>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 0192ebaa43c161884a92bafa354d0d19@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 May 2009 04:33:38 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=z9hG4bK38e576de;rport
Allow: ACK,BYE,CANCEL,INMADDRFrom: "Unknown"<sip:Unknown@192.168.0.1>;tag=as3b5144d9
To: <sip:192.168.0.6>
Call-ID: 0192ebaa43c161884a92bafa354d0d19@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 192.168.0.6:5060 --->
INVITE sip:100@192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK_001346E0DEF0_T0043261F
Session-Expires: 180
From: "701"<sip:701@192.168.0.1>;tag=001346E0DEF0_T4400670
To: <sip:100@192.168.0.1>
Call-ID: CALL_ID7_001346E0DEF0_T4400670@192.168.0.6
CSeq: 6795 INVITE
Contact: <sip:701@192.168.0.6:5060>
Max-Forwards: 70
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Supported: 100rel,timer
User-Agent: ACT G1104S 01.02
Content-Type: application/sdp
Content-Length: 272

v=0
o=701 24304129 24304129 IN IP4 192.168.0.6
s=ACT G1104S 01.02
c=IN IP4 192.168.0.6
t=0 0
m=audio 10000 RTP/AVP 0 8 18 4
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:4 G723/8000/1
a=rtcp:10001
a=direction:both
a=sendrecv

<------------->
--- (14 headers 13 lines) ---
Sending to 192.168.0.6 : 5060 (no NAT)
Using INVITE request as basis request - CALL_ID7_001346E0DEF0_T4400670@192.168.0.6
Found user '701'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Peer audio RTP is at port 192.168.0.6:10000
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.6:10000
Looking for 100 in from-internal (domain 192.168.0.1)
list_route: hop: <sip:701@192.168.0.6:5060>
trixbox1*CLI>
<--- Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK_001346E0DEF0_T0043261F;received=192.168.0.6
From: "701"<sip:701@192.168.0.1>;tag=001346E0DEF0_T4400670
To: <sip:100@192.168.0.1>
Call-ID: CALL_ID7_001346E0DEF0_T4400670@192.168.0.6
CSeq: 6795 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@192.168.0.1>
Content-Length: 0


<------------>
-- Executing [100@from-internal:1] Macro("SIP/701-08786a68", "exten-vm|novm|100") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/701-08786a68", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/701-08786a68", "AMPUSER=701") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/701-08786a68", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/701-08786a68", "1|Set|REALCALLERIDNUM=701") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/701-08786a68", "AMPUSER=701") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/701-08786a68", "AMPUSERCIDNAME=DVG2004") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/701-08786a68", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/701-08786a68", "AMPUSERCID=701") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/701-08786a68", "CALLERID(all)="DVG2004" <701>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/701-08786a68", "REALCALLERIDNUM=701") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/701-08786a68", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/701-08786a68", "0?continue") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/701-08786a68", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/701-08786a68", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/701-08786a68", "Using CallerID "DVG2004" <701>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/701-08786a68", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/701-08786a68", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/701-08786a68", "EXTTOCALL=100") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/701-08786a68", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/701-08786a68", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/701-08786a68", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/701-08786a68", "record-enable|100|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/701-08786a68", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/701-08786a68", "recordingcheck|20090512-153341|1242102821.32") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090512-153341|1242102821.32: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/701-08786a68", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/701-08786a68", "dial||tTr|100") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/701-08786a68", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/701-08786a68", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'DVG2004' number is '701'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 100 to extension map
dialparties.agi: Extension 100 has call forward set to 9746
> dialparties.agi: Primary ext is CF so disabling mastermode if it was set
> dialparties.agi: extnum 9746# has: cw: 0; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: Built External dialstring component for 9746: Local/9746@from-internal/n
-- dialparties.agi: Filtered ARG3: 9746
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/701-08786a68", "Local/9746@from-internal/n||tTr") in new stack
-- Called 9746@from-internal/n
trixbox1*CLI>
<--- Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK_001346E0DEF0_T0043261F;received=192.168.0.6
From: "701"<sip:701@192.168.0.1>;tag=001346E0DEF0_T4400670
To: <sip:100@192.168.0.1>;tag=as1e729c62
Call-ID: CALL_ID7_001346E0DEF0_T4400670@192.168.0.6
CSeq: 6795 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@192.168.0.1>
Content-Length: 0


<------------>
-- Executing [9746@from-internal:1] ResetCDR("Local/9746@from-internal-cc96,2", "") in new stack
-- Executing [9746@from-internal:2] NoCDR("Local/9746@from-internal-cc96,2", "") in new stack
-- Executing [9746@from-internal:3] Wait("Local/9746@from-internal-cc96,2", "1") in new stack
Retransmitting #4 (no 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=z9hG4bK38e576de;rport
From: "Unknown" <sip:Unknown@192.168.0.1>;tag=as3b5144d9
To: <sip:192.168.0.6>
Contact: <sip:Unknown@192.168.0.1>
Call-ID: 0192ebaa43c161884a92bafa354d0d19@192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 May 2009 04:33:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '0192ebaa43c161884a92bafa354d0d19@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=z9hG4bK38e576de;rport
Allow: ACK,BYE,CANCEL,INMADDRFrom: "Unknown"<sip:Unknown@192.168.0.1>;tag=as3b5144d9
To: <sip:192.168.0.6>
Call-ID: 0192ebaa43c161884a92bafa354d0d19@192.168.0.1
CSeq: 102 OPTIONS
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
-- Executing [9746@from-internal:4] Playback("Local/9746@from-internal-cc96,2", "silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer") in new stack
-- <Local/9746@from-internal-cc96,2> Playing 'silence/1' (language 'en')
-- <Local/9746@from-internal-cc96,2> Playing 'cannot-complete-as-dialed' (language 'en')

на других экстеншенов вызов идет, но почему то именно для 100 перенаправляет на номер 9746:

-- dialparties.agi: Added extension 100 to extension map
dialparties.agi: Extension 100 has call forward set to 9746

sip show peer 100:

* Name : 100
Secret : <Set>
MD5Secret : <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 100@device
VM Extension : *97
LastMsgsSent : 0/0
Call limit : 50
Dynamic : Yes
Callerid : "device" <100>
MaxCallBR : 384 kbps
Expire : 533
Insecure : no
Nat : Always
ACL : Yes
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: Yes
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : IPсофтфона Port 5061
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 100
SIP Options : (none)
Codecs : 0x28000c (ulaw|alaw|h263|h264)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing: No
Status : OK (1 ms)
Useragent : Zoiper rev.3938
Reg. Contact : sip:100@85.15.70.174:5061;rinstance=689fccfc8b6050d2;transport=UDP

если поставить пароль, то DVG не регестрируеться на * и звонки вновь становяться не возможными..

данная проблема разрешилась СПАСИБО ВСЕМ ЗА ПОМОЩЬ!!!:)
но всё таки не понятно почему при звонке на 100 он переводит звонок на 9746, такого экстеншена вобще нет..

-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Port4' number is '704'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 100 to extension map
dialparties.agi: Extension 100 has call forward set to 9746
-- dialparties.agi: Filtered ARG3: 9746
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0

при звонке на другой номер, звонок идет нормаоьно.. вот лог:

-- Executing [s@macro-dial:3] AGI("SIP/704-b7806ae0", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Port4' number is '704'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 2005 to extension map
-- dialparties.agi: Extension 2005 cf is disabled
-- dialparties.agi: Extension 2005 do not disturb is disabled
-- dialparties.agi: dbset CALLTRACE/2005 to 704
-- dialparties.agi: Filtered ARG3: 2005
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/704-b7806ae0", "SIP/2005||tTr") in new stack
-- Called 2005
-- SIP/2005-0a076988 is ringing
-- SIP/2005-0a076988 answered SIP/704-b7806ae0

может кто знает, почему переброс идет, это вообще на любые входящие для 100 такая пропблема происходит..100 зарегистрирован на софтфоне.....