AstT*CLI> sip set debug on
SIP Debugging enabled
[2015-09-05 12:07:22] Really destroying SIP dialog '5eb7df087723bb72219c74806375bce9@172.16.21.92:5060' Method: OPTIONS
[2015-09-05 12:07:25] 
<--- SIP read from UDP:mci:5060 --->
INVITE sip:B-Number@route_name;user=phone SIP/2.0
From: <sip:+374A-Number@mci;user=phone>;tag=1585592893
To: <sip:B-Number@route_name;user=phone>
Max-Forwards: 70
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000048099104676541
Call-ID: tFcP0572205190501-AAAABGFK-@mci
CSeq: 2945 INVITE
P-Asserted-Identity: <sip:+374A-Number@mci;user=phone>
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
P-Charging-Vector: icid-value=B5F4A12000-0905-12072307;icid-generated-at=mci;orig-ioi=MSC2VIVA
Supported: 100rel
Content-Type: application/sdp
Contact: <sip:mci:5060;transport=UDP>
Content-Length: 365

v=0
o=- 10696923 10696923 IN IP4 mci
s=-
c=IN IP4 RTP_IP_MCI
t=0 0
a=sendrecv
m=audio 35104 RTP/AVP 8 0 18 96 97
c=IN IP4 RTP_IP_MCI
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 G729/8000
a=fmtp:96 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40
<------------->
[2015-09-05 12:07:25] --- (15 headers 19 lines) ---
[2015-09-05 12:07:25] Sending to mci:5060 (NAT)
[2015-09-05 12:07:25] Using INVITE request as basis request - tFcP0572205190501-AAAABGFK-@mci
[2015-09-05 12:07:25] Found peer 'msco' for '+374A-Number' from mci:5060
[2015-09-05 12:07:25]   == Using SIP RTP TOS bits 184
[2015-09-05 12:07:25]   == Using SIP RTP CoS mark 5
[2015-09-05 12:07:25] Found RTP audio format 8
[2015-09-05 12:07:25] Found RTP audio format 0
[2015-09-05 12:07:25] Found RTP audio format 18
[2015-09-05 12:07:25] Found RTP audio format 96
[2015-09-05 12:07:25] Found RTP audio format 97
[2015-09-05 12:07:25] Found audio description format PCMA for ID 8
[2015-09-05 12:07:25] Found audio description format PCMU for ID 0
[2015-09-05 12:07:25] Found audio description format G729 for ID 18
[2015-09-05 12:07:25] Found audio description format G729 for ID 96
[2015-09-05 12:07:25] Found audio description format telephone-event for ID 97
[2015-09-05 12:07:25] Capabilities: us - (alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
[2015-09-05 12:07:25] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-09-05 12:07:25] Peer audio RTP is at port RTP_IP_MCI:35104
[2015-09-05 12:07:25] Looking for B-Number in from_msc (domain route_name)
[2015-09-05 12:07:25] list_route: hop: <sip:mci:5060;transport=UDP>
[2015-09-05 12:07:25] 
<--- Transmitting (NAT) to mci:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000048099104676541;received=mci;rport=5060
From: <sip:+374A-Number@mci;user=phone>;tag=1585592893
To: <sip:B-Number@route_name;user=phone>
Call-ID: tFcP0572205190501-AAAABGFK-@mci
CSeq: 2945 INVITE
Server: Asterisk PBX 10.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:B-Number@local_IP:5060>
Content-Length: 0


<------------>
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:1] NoOp("SIP/msco-0000001b", "ER") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:2] Set("SIP/msco-0000001b", "Diversion=374") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:3] Set("SIP/msco-0000001b", "Diversion=374B-Number") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:4] NoOp("SIP/msco-0000001b", "div=374B-Number callerid=+374A-Number") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:5] GotoIf("SIP/msco-0000001b", "0?testb") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:6] GotoIf("SIP/msco-0000001b", "1?viva_numbers") in new stack
[2015-09-05 12:07:25]     -- Goto (from_msc,B-Number,22)
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:22] NoOp("SIP/msco-0000001b", "374B-Number") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:23] NoOp("SIP/msco-0000001b", "CALLERID(num)=+374A-Number") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:24] Set("SIP/msco-0000001b", "CALLERID(num)=374A-Number") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:25] Set("SIP/msco-0000001b", "privacy=") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:26] NoOp("SIP/msco-0000001b", "374A-Number") in new stack
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:27] GotoIf("SIP/msco-0000001b", "1?testa") in new stack
[2015-09-05 12:07:25]     -- Goto (from_msc,B-Number,38)
[2015-09-05 12:07:25]     -- Executing [B-Number@from_msc:38] Goto("SIP/msco-0000001b", "localoca,374B-Number,1") in new stack
[2015-09-05 12:07:25]     -- Goto (localoca,374B-Number,1)
[2015-09-05 12:07:25]     -- Executing [374B-Number@localoca:1] NoOp("SIP/msco-0000001b", "374B-Number") in new stack
[2015-09-05 12:07:25]     -- Executing [374B-Number@localoca:2] Set("SIP/msco-0000001b", "CALLERID(num)=from_user") in new stack
[2015-09-05 12:07:25]     -- Executing [374B-Number@localoca:3] Answer("SIP/msco-0000001b", "") in new stack
[2015-09-05 12:07:25] Audio is at 16036
[2015-09-05 12:07:25] Adding codec 100004 (alaw) to SDP
[2015-09-05 12:07:25] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-05 12:07:25] 
<--- Reliably Transmitting (NAT) to mci:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000048099104676541;received=mci;rport=5060
From: <sip:+374A-Number@mci;user=phone>;tag=1585592893
To: <sip:B-Number@route_name;user=phone>;tag=as149f6bca
Call-ID: tFcP0572205190501-AAAABGFK-@mci
CSeq: 2945 INVITE
Server: Asterisk PBX 10.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:B-Number@local_IP:5060>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 408188801 408188801 IN IP4 local_IP
s=Asterisk PBX 10.1.2
c=IN IP4 local_IP
t=0 0
m=audio 16036 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[2015-09-05 12:07:25] 
<--- SIP read from UDP:mci:5060 --->
ACK sip:B-Number@local_IP:5060 SIP/2.0
From: <sip:+374A-Number@mci;user=phone>;tag=1585592893
To: <sip:B-Number@route_name;user=phone>;tag=as149f6bca
Max-Forwards: 70
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000023538364262849
Call-ID: tFcP0572205190501-AAAABGFK-@mci
CSeq: 2945 ACK
Content-Length: 0

<------------->
[2015-09-05 12:07:25] --- (8 headers 0 lines) ---
[2015-09-05 12:07:25]     -- Executing [374B-Number@localoca:4] Dial("SIP/msco-0000001b", "SIP/multifon-out/C-Number,60") in new stack
[2015-09-05 12:07:25]   == Using SIP RTP TOS bits 184
[2015-09-05 12:07:25]   == Using SIP RTP CoS mark 5
[2015-09-05 12:07:25] Audio is at 10770
[2015-09-05 12:07:25] Adding codec 100004 (alaw) to SDP
[2015-09-05 12:07:25] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-05 12:07:25] Reliably Transmitting (NAT) to 193.201.229.35:5060:
INVITE sip:C-Number@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP real_ip:5060;branch=z9hG4bK2cd54ad6;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@193.201.229.35>
Contact: <sip:from_user@real_ip:5060>
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.1.2
Date: Sat, 05 Sep 2015 08:07:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1186090034 1186090034 IN IP4 real_ip
s=Asterisk PBX 10.1.2
c=IN IP4 real_ip
t=0 0
m=audio 10770 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[2015-09-05 12:07:25]     -- Called SIP/multifon-out/C-Number
[2015-09-05 12:07:25] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP real_ip:5060;received=real_ip;branch=z9hG4bK2cd54ad6;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@multifon.ru>
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 102 INVITE

<------------->
[2015-09-05 12:07:25] --- (6 headers 0 lines) ---
[2015-09-05 12:07:25] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP real_ip:5060;received=real_ip;branch=z9hG4bK2cd54ad6;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@multifon.ru>;tag=SDr62t599-9AB6324631353641926D6500
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 102 INVITE
Proxy-Authenticate: Digest nonce="MTQ0MTQ0MDQ0NTqFI5+FFMgYkY8MHImIrh6N",opaque="MTQ0MTQ0MDQ0NTqFI5+FFMgYkY8MHImIrh6N",algorithm=md5,realm="BREDBAND",qop="auth"
Reason: SEM;cause=5;text="Need auth"
Content-Length: 0

<------------->
[2015-09-05 12:07:25] --- (9 headers 0 lines) ---
[2015-09-05 12:07:25] Transmitting (NAT) to 193.201.229.35:5060:
ACK sip:C-Number@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP real_ip:5060;branch=z9hG4bK2cd54ad6;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@193.201.229.35>;tag=SDr62t599-9AB6324631353641926D6500
Contact: <sip:from_user@real_ip:5060>
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


---
[2015-09-05 12:07:25] Audio is at 10770
[2015-09-05 12:07:25] Adding codec 100004 (alaw) to SDP
[2015-09-05 12:07:25] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-05 12:07:25] Reliably Transmitting (NAT) to 193.201.229.35:5060:
INVITE sip:C-Number@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP real_ip:5060;branch=z9hG4bK4f4b179f;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@193.201.229.35>
Contact: <sip:from_user@real_ip:5060>
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.1.2
Proxy-Authorization: Digest username="from_user", realm="BREDBAND", algorithm=MD5, uri="sip:C-Number@193.201.229.35", nonce="MTQ0MTQ0MDQ0NTqFI5+FFMgYkY8MHImIrh6N", response="cebccd8b5ec9bdaf4d499fbead69f20b", opaque="MTQ0MTQ0MDQ0NTqFI5+FFMgYkY8MHImIrh6N", qop=auth, cnonce="5bb7259b", nc=00000001
Date: Sat, 05 Sep 2015 08:07:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1186090034 1186090035 IN IP4 real_ip
s=Asterisk PBX 10.1.2
c=IN IP4 real_ip
t=0 0
m=audio 10770 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[2015-09-05 12:07:25] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP real_ip:5060;received=real_ip;branch=z9hG4bK4f4b179f;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@multifon.ru>
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 103 INVITE

<------------->
[2015-09-05 12:07:25] --- (6 headers 0 lines) ---
[2015-09-05 12:07:26] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP real_ip:5060;received=real_ip;branch=z9hG4bK4f4b179f;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@multifon.ru>;tag=SDr62t599-7DFD3246313536419A6D6500
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 103 INVITE
Content-Length: 156
Supported: 100rel,precondition,timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK
Contact: <sip:C-Number@193.201.229.35:5060;transport=udp>

v=0
o=- 0 0 IN IP4 193.201.229.19
s=-
c=IN IP4 193.201.229.19
t=0 0
m=audio 18978 RTP/AVP 8
b=AS:80
a=rtpmap:8 PCMA/8000
a=maxptime:20
a=ptime:20
<------------->
[2015-09-05 12:07:26] --- (12 headers 10 lines) ---
[2015-09-05 12:07:26] list_route: hop: <sip:C-Number@193.201.229.35:5060;transport=udp>
[2015-09-05 12:07:26] Found RTP audio format 8
[2015-09-05 12:07:26] Found audio description format PCMA for ID 8
[2015-09-05 12:07:26] Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[2015-09-05 12:07:26] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2015-09-05 12:07:26] Peer audio RTP is at port 193.201.229.19:18978
[2015-09-05 12:07:26]     -- SIP/multifon-out-0000001c is making progress passing it to SIP/msco-0000001b
[2015-09-05 12:07:30] Really destroying SIP dialog '28ab9f5b5b643d8442a40ae863fad8b9@172.16.21.103' Method: REGISTER
[2015-09-05 12:07:31] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP real_ip:5060;received=real_ip;branch=z9hG4bK4f4b179f;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@multifon.ru>;tag=SDr62t599-7DFD3246313536419A6D6500
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 103 INVITE
Content-Length: 0
Supported: 100rel,precondition,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,UPDATE
Contact: <sip:C-Number@193.201.229.35:5060;transport=udp>

<------------->
[2015-09-05 12:07:31] --- (10 headers 0 lines) ---
[2015-09-05 12:07:31]     -- SIP/multifon-out-0000001c is ringing
[2015-09-05 12:07:35] 
<--- SIP read from UDP:mci:5060 --->
BYE sip:B-Number@local_IP:5060 SIP/2.0
From: <sip:+374A-Number@mci;user=phone>;tag=1585592893
To: <sip:B-Number@route_name;user=phone>;tag=as149f6bca
Max-Forwards: 70
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000007979397377796
Call-ID: tFcP0572205190501-AAAABGFK-@mci
CSeq: 2946 BYE
Content-Length: 0

<------------->
[2015-09-05 12:07:35] --- (8 headers 0 lines) ---
[2015-09-05 12:07:35] Sending to mci:5060 (NAT)
[2015-09-05 12:07:35] Scheduling destruction of SIP dialog 'tFcP0572205190501-AAAABGFK-@mci' in 32000 ms (Method: BYE)
[2015-09-05 12:07:35] 
<--- Transmitting (NAT) to mci:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000007979397377796;received=mci;rport=5060
From: <sip:+374A-Number@mci;user=phone>;tag=1585592893
To: <sip:B-Number@route_name;user=phone>;tag=as149f6bca
Call-ID: tFcP0572205190501-AAAABGFK-@mci
CSeq: 2946 BYE
Server: Asterisk PBX 10.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2015-09-05 12:07:35] Scheduling destruction of SIP dialog '41f13bb9170384e8164311e12f76b7a1@multifon.ru' in 6400 ms (Method: INVITE)
[2015-09-05 12:07:35] set_destination: Parsing <sip:C-Number@193.201.229.35:5060;transport=udp> for address/port to send to
[2015-09-05 12:07:35] set_destination: set destination to 193.201.229.35:5060
[2015-09-05 12:07:35] Reliably Transmitting (NAT) to 193.201.229.35:5060:
CANCEL sip:C-Number@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP real_ip:5060;branch=z9hG4bK4f4b179f;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@193.201.229.35>
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


---
[2015-09-05 12:07:35] Scheduling destruction of SIP dialog '41f13bb9170384e8164311e12f76b7a1@multifon.ru' in 6400 ms (Method: INVITE)
[2015-09-05 12:07:35]   == Spawn extension (localoca, 374B-Number, 4) exited non-zero on 'SIP/msco-0000001b'
[2015-09-05 12:07:35] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP real_ip:5060;received=real_ip;branch=z9hG4bK4f4b179f;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@multifon.ru>;tag=SDr62t599-7DFD3246313536419A6D6500
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 103 CANCEL

<------------->
[2015-09-05 12:07:35] --- (6 headers 0 lines) ---
[2015-09-05 12:07:35] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP real_ip:5060;received=real_ip;branch=z9hG4bK4f4b179f;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@multifon.ru>;tag=SDr62t599-7DFD3246313536419A6D6500
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 103 INVITE
Content-Length: 0

<------------->
[2015-09-05 12:07:35] --- (7 headers 0 lines) ---
[2015-09-05 12:07:35] set_destination: Parsing <sip:C-Number@193.201.229.35:5060;transport=udp> for address/port to send to
[2015-09-05 12:07:35] set_destination: set destination to 193.201.229.35:5060
[2015-09-05 12:07:35] Transmitting (NAT) to 193.201.229.35:5060:
ACK sip:C-Number@193.201.229.35:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP real_ip:5060;branch=z9hG4bK4f4b179f;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as73acd6a6
To: <sip:C-Number@193.201.229.35>;tag=SDr62t599-7DFD3246313536419A6D6500
Contact: <sip:from_user@real_ip:5060>
Call-ID: 41f13bb9170384e8164311e12f76b7a1@multifon.ru
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


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[2015-09-05 12:07:35] Really destroying SIP dialog '41f13bb9170384e8164311e12f76b7a1@multifon.ru' Method: INVITE
AstT*CLI>