AstT*CLI> sip set debug on
SIP Debugging enabled
[2015-09-08 16:05:11] Reliably Transmitting (NAT) to 193.201.229.35:5060:
OPTIONS sip:193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP self_real_IP:5060;branch=z9hG4bK0d6e454a;rport
Max-Forwards: 70
From: "Unknown" <sip:from_user@self_real_IP>;tag=as5306ba77
To: <sip:193.201.229.35>
Contact: <sip:from_user@self_real_IP:5060>
Call-ID: 08cb6598121e16a40214f1c04c3fa959@self_real_IP:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Tue, 08 Sep 2015 12:05:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
[2015-09-08 16:05:12] --- (7 headers 0 lines) ---
[2015-09-08 16:05:12] Really destroying SIP dialog '27d98b121c56a8b337d68dcf52109516@self_real_IP:5060' Method: OPTIONS
[2015-09-08 16:05:14] 
<--- 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=3610491221
To: <sip:B-Number@route_name;user=phone>
Max-Forwards: 70
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000038360398994279
Call-ID: u0bS3451608190501-AAAAAMLI-@mci
CSeq: 54529 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=2E184B2000-0908-16051307;icid-generated-at=mci;orig-ioi=MSC2VIVA
Supported: 100rel
Content-Type: application/sdp
Contact: <sip:mci:5060;transport=UDP>
Content-Length: 362

v=0
o=- 3625442 3625442 IN IP4 mci
s=-
c=IN IP4 inc_rtp_IP
t=0 0
a=sendrecv
m=audio 3026 RTP/AVP 8 0 18 96 97
c=IN IP4 inc_rtp_IP
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
<------------->
[2015-09-08 16:05:14] --- (15 headers 19 lines) ---
[2015-09-08 16:05:14] Sending to mci:5060 (NAT)
[2015-09-08 16:05:14] Using INVITE request as basis request - u0bS3451608190501-AAAAAMLI-@mci
[2015-09-08 16:05:14] Found peer 'mco' for '+374A-Number' from mci:5060
[2015-09-08 16:05:14]   == Using SIP RTP TOS bits 184
[2015-09-08 16:05:14]   == Using SIP RTP CoS mark 5
[2015-09-08 16:05:14] Found RTP audio format 8
[2015-09-08 16:05:14] Found RTP audio format 0
[2015-09-08 16:05:14] Found RTP audio format 18
[2015-09-08 16:05:14] Found RTP audio format 96
[2015-09-08 16:05:14] Found RTP audio format 97
[2015-09-08 16:05:14] Found audio description format PCMA for ID 8
[2015-09-08 16:05:14] Found audio description format PCMU for ID 0
[2015-09-08 16:05:14] Found audio description format G729 for ID 18
[2015-09-08 16:05:14] Found audio description format G729 for ID 96
[2015-09-08 16:05:14] Found audio description format telephone-event for ID 97
[2015-09-08 16:05:14] Capabilities: us - (alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
[2015-09-08 16:05:14] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-09-08 16:05:14] Peer audio RTP is at port inc_rtp_IP:3026
[2015-09-08 16:05:14] Looking for B-Number in from_msc (domain route_name)
[2015-09-08 16:05:14] list_route: hop: <sip:mci:5060;transport=UDP>
[2015-09-08 16:05:14] 
<--- Transmitting (NAT) to mci:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000038360398994279;received=mci;rport=5060
From: <sip:+374A-Number@mci;user=phone>;tag=3610491221
To: <sip:B-Number@route_name;user=phone>
Call-ID: u0bS3451608190501-AAAAAMLI-@mci
CSeq: 54529 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


<------------>
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:1] NoOp("SIP/mco-00000021", "ER") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:2] Set("SIP/mco-00000021", "Diversion=374") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:3] Set("SIP/mco-00000021", "Diversion=374B-Number") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:4] NoOp("SIP/mco-00000021", "div=374B-Number callerid=+374A-Number") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:5] GotoIf("SIP/mco-00000021", "0?testb") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:6] GotoIf("SIP/mco-00000021", "1?viva_numbers") in new stack
[2015-09-08 16:05:14]     -- Goto (from_msc,B-Number,22)
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:22] NoOp("SIP/mco-00000021", "374B-Number") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:23] NoOp("SIP/mco-00000021", "CALLERID(num)=+374A-Number") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:24] Set("SIP/mco-00000021", "CALLERID(num)=374A-Number") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:25] Set("SIP/mco-00000021", "privacy=") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:26] NoOp("SIP/mco-00000021", "374A-Number") in new stack
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:27] GotoIf("SIP/mco-00000021", "1?testa") in new stack
[2015-09-08 16:05:14]     -- Goto (from_msc,B-Number,38)
[2015-09-08 16:05:14]     -- Executing [B-Number@from_msc:38] Goto("SIP/mco-00000021", "localoca,374B-Number,1") in new stack
[2015-09-08 16:05:14]     -- Goto (localoca,374B-Number,1)
[2015-09-08 16:05:14]     -- Executing [374B-Number@localoca:1] NoOp("SIP/mco-00000021", "374B-Number") in new stack
[2015-09-08 16:05:14]     -- Executing [374B-Number@localoca:2] Read("SIP/mco-00000021", "SWITCH,beep,1") in new stack
[2015-09-08 16:05:14]     -- Accepting a maximum of 1 digits.
[2015-09-08 16:05:14] Audio is at 53020
[2015-09-08 16:05:14] Adding codec 100004 (alaw) to SDP
[2015-09-08 16:05:14] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-08 16:05:14] 
<--- Reliably Transmitting (NAT) to mci:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000038360398994279;received=mci;rport=5060
From: <sip:+374A-Number@mci;user=phone>;tag=3610491221
To: <sip:B-Number@route_name;user=phone>;tag=as4e073e1e
Call-ID: u0bS3451608190501-AAAAAMLI-@mci
CSeq: 54529 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 181369582 181369582 IN IP4 local_ip
s=Asterisk PBX 10.1.2
c=IN IP4 local_ip
t=0 0
m=audio 53020 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

<------------>
[2015-09-08 16:05:14] 
<--- SIP read from UDP:mci:5060 --->
ACK sip:B-Number@local_ip:5060 SIP/2.0
From: <sip:+374A-Number@mci;user=phone>;tag=3610491221
To: <sip:B-Number@route_name;user=phone>;tag=as4e073e1e
Max-Forwards: 70
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000073603226581562
Call-ID: u0bS3451608190501-AAAAAMLI-@mci
CSeq: 54529 ACK
Content-Length: 0

<------------->
[2015-09-08 16:05:14] --- (8 headers 0 lines) ---
[2015-09-08 16:05:15]     -- <SIP/mco-00000021> Playing 'beep.gsm' (language 'en')
[2015-09-08 16:05:15]     -- User entered '2'
[2015-09-08 16:05:15]     -- Executing [374B-Number@localoca:3] GotoIf("SIP/mco-00000021", "0?m") in new stack
[2015-09-08 16:05:15]     -- Executing [374B-Number@localoca:4] GotoIf("SIP/mco-00000021", "1?a") in new stack
[2015-09-08 16:05:15]     -- Goto (localoca,374B-Number,12)
[2015-09-08 16:05:15]     -- Executing [374B-Number@localoca:12] NoOp("SIP/mco-00000021", "374B-Number") in new stack
[2015-09-08 16:05:15]     -- Executing [374B-Number@localoca:13] Set("SIP/mco-00000021", "CALLERID(num)=from_user") in new stack
[2015-09-08 16:05:15]     -- Executing [374B-Number@localoca:14] MixMonitor("SIP/mco-00000021", "REC-201509081505_from_user.wav,b") in new stack
[2015-09-08 16:05:15]     -- Executing [374B-Number@localoca:15] Dial("SIP/mco-00000021", "SIP/multifon-out/+374C-NUmber,60") in new stack
[2015-09-08 16:05:15]   == Using SIP RTP TOS bits 184
[2015-09-08 16:05:15]   == Using SIP RTP CoS mark 5
[2015-09-08 16:05:15] Audio is at 30902
[2015-09-08 16:05:15] Adding codec 100004 (alaw) to SDP
[2015-09-08 16:05:15] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-08 16:05:15] Reliably Transmitting (NAT) to 193.201.229.35:5060:
INVITE sip:+374C-NUmber@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP self_real_IP:5060;branch=z9hG4bK6c354fd6;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@193.201.229.35>
Contact: <sip:from_user@self_real_IP:5060>
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.1.2
Date: Tue, 08 Sep 2015 12:05:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 781695874 781695874 IN IP4 self_real_IP
s=Asterisk PBX 10.1.2
c=IN IP4 self_real_IP
t=0 0
m=audio 30902 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

---
[2015-09-08 16:05:15]     -- Called SIP/multifon-out/+374C-NUmber
[2015-09-08 16:05:15]   == Begin MixMonitor Recording SIP/mco-00000021
[2015-09-08 16:05:15] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP self_real_IP:5060;received=self_real_IP;branch=z9hG4bK6c354fd6;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@multifon.ru>
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 102 INVITE

<------------->
[2015-09-08 16:05:15] --- (6 headers 0 lines) ---
[2015-09-08 16:05:15] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP self_real_IP:5060;received=self_real_IP;branch=z9hG4bK6c354fd6;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@multifon.ru>;tag=SDeocne99-9432324631353641DFB50602
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 102 INVITE
Proxy-Authenticate: Digest nonce="MTQ0MTcxMzkxNTqY/2ejA0HF7YiFw/CzQp6n",opaque="MTQ0MTcxMzkxNTqY/2ejA0HF7YiFw/CzQp6n",algorithm=md5,realm="BREDBAND",qop="auth"
Reason: SEM;cause=5;text="Need auth"
Content-Length: 0

<------------->
[2015-09-08 16:05:15] --- (9 headers 0 lines) ---
[2015-09-08 16:05:15] Transmitting (NAT) to 193.201.229.35:5060:
ACK sip:+374C-NUmber@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP self_real_IP:5060;branch=z9hG4bK6c354fd6;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@193.201.229.35>;tag=SDeocne99-9432324631353641DFB50602
Contact: <sip:from_user@self_real_IP:5060>
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


---
[2015-09-08 16:05:15] Audio is at 30902
[2015-09-08 16:05:15] Adding codec 100004 (alaw) to SDP
[2015-09-08 16:05:15] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-08 16:05:15] Reliably Transmitting (NAT) to 193.201.229.35:5060:
INVITE sip:+374C-NUmber@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP self_real_IP:5060;branch=z9hG4bK0cce1f00;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@193.201.229.35>
Contact: <sip:from_user@self_real_IP:5060>
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.1.2
Proxy-Authorization: Digest username="from_user", realm="BREDBAND", algorithm=MD5, uri="sip:+374C-NUmber@193.201.229.35", nonce="MTQ0MTcxMzkxNTqY/2ejA0HF7YiFw/CzQp6n", response="62368947dc352b40b25f40f7d69d6147", opaque="MTQ0MTcxMzkxNTqY/2ejA0HF7YiFw/CzQp6n", qop=auth, cnonce="7ee79df1", nc=00000001
Date: Tue, 08 Sep 2015 12:05:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 781695874 781695875 IN IP4 self_real_IP
s=Asterisk PBX 10.1.2
c=IN IP4 self_real_IP
t=0 0
m=audio 30902 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

---
[2015-09-08 16:05:16] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP self_real_IP:5060;received=self_real_IP;branch=z9hG4bK0cce1f00;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@multifon.ru>
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 103 INVITE

<------------->
[2015-09-08 16:05:16] --- (6 headers 0 lines) ---
[2015-09-08 16:05:23] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP self_real_IP:5060;received=self_real_IP;branch=z9hG4bK0cce1f00;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@multifon.ru>;tag=SDeocne99-8534324631353641E7B50602
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 103 INVITE
Allow: OPTIONS,CANCEL,UPDATE
Content-Length: 0
Contact: <sip:+374C-NUmber@193.201.229.35:5060;transport=udp>

<------------->
[2015-09-08 16:05:23] --- (9 headers 0 lines) ---
[2015-09-08 16:05:23] list_route: hop: <sip:+374C-NUmber@193.201.229.35:5060;transport=udp>
[2015-09-08 16:05:23]     -- SIP/multifon-out-00000022 is ringing
[2015-09-08 16:05:30] 
<--- SIP read from UDP:mci:5060 --->
BYE sip:B-Number@local_ip:5060 SIP/2.0
From: <sip:+374A-Number@mci;user=phone>;tag=3610491221
To: <sip:B-Number@route_name;user=phone>;tag=as4e073e1e
Max-Forwards: 70
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000029272745817835
Call-ID: u0bS3451608190501-AAAAAMLI-@mci
CSeq: 54530 BYE
Content-Length: 0

<------------->
[2015-09-08 16:05:30] --- (8 headers 0 lines) ---
[2015-09-08 16:05:30] Sending to mci:5060 (NAT)
[2015-09-08 16:05:30] Scheduling destruction of SIP dialog 'u0bS3451608190501-AAAAAMLI-@mci' in 32000 ms (Method: BYE)
[2015-09-08 16:05:30] 
<--- Transmitting (NAT) to mci:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000029272745817835;received=mci;rport=5060
From: <sip:+374A-Number@mci;user=phone>;tag=3610491221
To: <sip:B-Number@route_name;user=phone>;tag=as4e073e1e
Call-ID: u0bS3451608190501-AAAAAMLI-@mci
CSeq: 54530 BYE
Server: Asterisk PBX 10.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2015-09-08 16:05:30] Scheduling destruction of SIP dialog '42c6bdbd15c6d04c6752eab443240fc6@multifon.ru' in 6400 ms (Method: INVITE)
[2015-09-08 16:05:30] set_destination: Parsing <sip:+374C-NUmber@193.201.229.35:5060;transport=udp> for address/port to send to
[2015-09-08 16:05:30] set_destination: set destination to 193.201.229.35:5060
[2015-09-08 16:05:30] Reliably Transmitting (NAT) to 193.201.229.35:5060:
CANCEL sip:+374C-NUmber@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP self_real_IP:5060;branch=z9hG4bK0cce1f00;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@193.201.229.35>
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


---
[2015-09-08 16:05:30] Scheduling destruction of SIP dialog '42c6bdbd15c6d04c6752eab443240fc6@multifon.ru' in 6400 ms (Method: INVITE)
[2015-09-08 16:05:30]   == Spawn extension (localoca, 374B-Number, 15) exited non-zero on 'SIP/mco-00000021'
[2015-09-08 16:05:30]   == MixMonitor close filestream (mixed)
[2015-09-08 16:05:30]   == End MixMonitor Recording SIP/mco-00000021
[2015-09-08 16:05:30] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP self_real_IP:5060;received=self_real_IP;branch=z9hG4bK0cce1f00;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@multifon.ru>;tag=SDeocne99-8534324631353641E7B50602
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 103 CANCEL

<------------->
[2015-09-08 16:05:30] --- (6 headers 0 lines) ---
[2015-09-08 16:05:30] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP self_real_IP:5060;received=self_real_IP;branch=z9hG4bK0cce1f00;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@multifon.ru>;tag=SDeocne99-8534324631353641E7B50602
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 103 INVITE
Content-Length: 0

<------------->
[2015-09-08 16:05:30] --- (7 headers 0 lines) ---
[2015-09-08 16:05:30] set_destination: Parsing <sip:+374C-NUmber@193.201.229.35:5060;transport=udp> for address/port to send to
[2015-09-08 16:05:30] set_destination: set destination to 193.201.229.35:5060
[2015-09-08 16:05:30] Transmitting (NAT) to 193.201.229.35:5060:
ACK sip:+374C-NUmber@193.201.229.35:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP self_real_IP:5060;branch=z9hG4bK0cce1f00;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as13ed01cb
To: <sip:+374C-NUmber@193.201.229.35>;tag=SDeocne99-8534324631353641E7B50602
Contact: <sip:from_user@self_real_IP:5060>
Call-ID: 42c6bdbd15c6d04c6752eab443240fc6@multifon.ru
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


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[2015-09-08 16:05:30] Really destroying SIP dialog '42c6bdbd15c6d04c6752eab443240fc6@multifon.ru' Method: INVITE
AstT*CLI>