AstT*CLI> sip set debug on
SIP Debugging enabled
[2015-09-05 12:08:45] Really destroying SIP dialog '141fa3eb2aae9fb17b0d1c6845c2f172@172.16.21.92:5060' Method: OPTIONS
[2015-09-05 12:08:47] 
<--- 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=1055492803
To: <sip:B-Number@route_name;user=phone>
Max-Forwards: 70
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000034458932927394
Call-ID: uReW7784205190501-AAAACMAG-@mci
CSeq: 7953 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=2C18A12000-0905-12084700;icid-generated-at=mci;orig-ioi=MSC2VIVA
Supported: 100rel
Content-Type: application/sdp
Contact: <sip:mci:5060;transport=UDP>
Content-Length: 363

v=0
o=- 7141455 7141455 IN IP4 mci
s=-
c=IN IP4 RTP_IP_MCI
t=0 0
a=sendrecv
m=audio 35864 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 12:08:47] --- (15 headers 19 lines) ---
[2015-09-05 12:08:47] Sending to mci:5060 (NAT)
[2015-09-05 12:08:47] Using INVITE request as basis request - uReW7784205190501-AAAACMAG-@mci
[2015-09-05 12:08:47] Found peer 'msco' for '+374A-Number' from mci:5060
[2015-09-05 12:08:47]   == Using SIP RTP TOS bits 184
[2015-09-05 12:08:47]   == Using SIP RTP CoS mark 5
[2015-09-05 12:08:47] Found RTP audio format 8
[2015-09-05 12:08:47] Found RTP audio format 0
[2015-09-05 12:08:47] Found RTP audio format 18
[2015-09-05 12:08:47] Found RTP audio format 96
[2015-09-05 12:08:47] Found RTP audio format 97
[2015-09-05 12:08:47] Found audio description format PCMA for ID 8
[2015-09-05 12:08:47] Found audio description format PCMU for ID 0
[2015-09-05 12:08:47] Found audio description format G729 for ID 18
[2015-09-05 12:08:47] Found audio description format G729 for ID 96
[2015-09-05 12:08:47] Found audio description format telephone-event for ID 97
[2015-09-05 12:08:47] Capabilities: us - (alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
[2015-09-05 12:08:47] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-09-05 12:08:47] Peer audio RTP is at port RTP_IP_MCI:35864
[2015-09-05 12:08:47] Looking for B-Number in from_msc (domain route_name)
[2015-09-05 12:08:47] list_route: hop: <sip:mci:5060;transport=UDP>
[2015-09-05 12:08:47] 
<--- Transmitting (NAT) to mci:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000034458932927394;received=mci;rport=5060
From: <sip:+374A-Number@mci;user=phone>;tag=1055492803
To: <sip:B-Number@route_name;user=phone>
Call-ID: uReW7784205190501-AAAACMAG-@mci
CSeq: 7953 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 12:08:47]     -- Executing [B-Number@from_msc:1] NoOp("SIP/msco-0000001d", "ER") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:2] Set("SIP/msco-0000001d", "Diversion=374") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:3] Set("SIP/msco-0000001d", "Diversion=374B-Number") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:4] NoOp("SIP/msco-0000001d", "div=374B-Number callerid=+374A-Number") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:5] GotoIf("SIP/msco-0000001d", "0?testb") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:6] GotoIf("SIP/msco-0000001d", "1?viva_numbers") in new stack
[2015-09-05 12:08:47]     -- Goto (from_msc,B-Number,22)
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:22] NoOp("SIP/msco-0000001d", "374B-Number") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:23] NoOp("SIP/msco-0000001d", "CALLERID(num)=+374A-Number") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:24] Set("SIP/msco-0000001d", "CALLERID(num)=374A-Number") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:25] Set("SIP/msco-0000001d", "privacy=") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:26] NoOp("SIP/msco-0000001d", "374A-Number") in new stack
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:27] GotoIf("SIP/msco-0000001d", "1?testa") in new stack
[2015-09-05 12:08:47]     -- Goto (from_msc,B-Number,38)
[2015-09-05 12:08:47]     -- Executing [B-Number@from_msc:38] Goto("SIP/msco-0000001d", "localoca,374B-Number,1") in new stack
[2015-09-05 12:08:47]     -- Goto (localoca,374B-Number,1)
[2015-09-05 12:08:47]     -- Executing [374B-Number@localoca:1] NoOp("SIP/msco-0000001d", "374B-Number") in new stack
[2015-09-05 12:08:47]     -- Executing [374B-Number@localoca:2] Read("SIP/msco-0000001d", "SWITCH,beep,1") in new stack
[2015-09-05 12:08:47]     -- Accepting a maximum of 1 digits.
[2015-09-05 12:08:47] Audio is at 15490
[2015-09-05 12:08:47] Adding codec 100004 (alaw) to SDP
[2015-09-05 12:08:47] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-05 12:08:47] 
<--- Reliably Transmitting (NAT) to mci:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000034458932927394;received=mci;rport=5060
From: <sip:+374A-Number@mci;user=phone>;tag=1055492803
To: <sip:B-Number@route_name;user=phone>;tag=as1dff6215
Call-ID: uReW7784205190501-AAAACMAG-@mci
CSeq: 7953 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 604620466 604620466 IN IP4 local_IP
s=Asterisk PBX 10.1.2
c=IN IP4 local_IP
t=0 0
m=audio 15490 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 12:08:47] 
<--- SIP read from UDP:mci:5060 --->
ACK sip:B-Number@local_IP:5060 SIP/2.0
From: <sip:+374A-Number@mci;user=phone>;tag=1055492803
To: <sip:B-Number@route_name;user=phone>;tag=as1dff6215
Max-Forwards: 70
Via: SIP/2.0/UDP mci:5060;branch=z9hG4bK00000016371807064923
Call-ID: uReW7784205190501-AAAACMAG-@mci
CSeq: 7953 ACK
Content-Length: 0

<------------->
[2015-09-05 12:08:47] --- (8 headers 0 lines) ---
[2015-09-05 12:08:48]     -- <SIP/msco-0000001d> Playing 'beep.gsm' (language 'en')
[2015-09-05 12:08:49]     -- User entered '1'
[2015-09-05 12:08:49]     -- Executing [374B-Number@localoca:3] GotoIf("SIP/msco-0000001d", "0?cnor") in new stack
[2015-09-05 12:08:49]     -- Executing [374B-Number@localoca:4] GotoIf("SIP/msco-0000001d", "1?rusas_mult") in new stack
[2015-09-05 12:08:49]     -- Goto (localoca,374B-Number,10)
[2015-09-05 12:08:49]     -- Executing [374B-Number@localoca:10] Set("SIP/msco-0000001d", "CALLERID(num)=from_user") in new stack
[2015-09-05 12:08:49]     -- Executing [374B-Number@localoca:11] Dial("SIP/msco-0000001d", "SIP/multifon-out/C-Number,60") in new stack
[2015-09-05 12:08:49]   == Using SIP RTP TOS bits 184
[2015-09-05 12:08:49]   == Using SIP RTP CoS mark 5
[2015-09-05 12:08:49] Audio is at 11652
[2015-09-05 12:08:49] Adding codec 100004 (alaw) to SDP
[2015-09-05 12:08:49] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-05 12:08:49] 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=z9hG4bK442f01d0;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as788340ba
To: <sip:C-Number@193.201.229.35>
Contact: <sip:from_user@real_ip:5060>
Call-ID: 1768775a565198e46456e6d416067361@multifon.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.1.2
Date: Sat, 05 Sep 2015 08:08:49 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 692923187 692923187 IN IP4 real_ip
s=Asterisk PBX 10.1.2
c=IN IP4 real_ip
t=0 0
m=audio 11652 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 12:08:49]     -- Called SIP/multifon-out/C-Number
[2015-09-05 12:08:49] 
<--- 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=z9hG4bK442f01d0;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as788340ba
To: <sip:C-Number@multifon.ru>
Call-ID: 1768775a565198e46456e6d416067361@multifon.ru
CSeq: 102 INVITE

<------------->
[2015-09-05 12:08:49] --- (6 headers 0 lines) ---
[2015-09-05 12:08:49] 
<--- 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=z9hG4bK442f01d0;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as788340ba
To: <sip:C-Number@multifon.ru>;tag=SDctrm099-342C324631353641788E6500
Call-ID: 1768775a565198e46456e6d416067361@multifon.ru
CSeq: 102 INVITE
Proxy-Authenticate: Digest nonce="MTQ0MTQ0MDUyOTouNu9X0epjKiU3l9Fi3Bcd",opaque="MTQ0MTQ0MDUyOTouNu9X0epjKiU3l9Fi3Bcd",algorithm=md5,realm="BREDBAND",qop="auth"
Reason: SEM;cause=5;text="Need auth"
Content-Length: 0

<------------->
[2015-09-05 12:08:49] --- (9 headers 0 lines) ---
[2015-09-05 12:08:49] 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=z9hG4bK442f01d0;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as788340ba
To: <sip:C-Number@193.201.229.35>;tag=SDctrm099-342C324631353641788E6500
Contact: <sip:from_user@real_ip:5060>
Call-ID: 1768775a565198e46456e6d416067361@multifon.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


---
[2015-09-05 12:08:49] Audio is at 11652
[2015-09-05 12:08:49] Adding codec 100004 (alaw) to SDP
[2015-09-05 12:08:49] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-05 12:08:49] 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=z9hG4bK09d7a83f;rport
Max-Forwards: 70
From: "from_user" <sip:from_user@multifon.ru>;tag=as788340ba
To: <sip:C-Number@193.201.229.35>
Contact: <sip:from_user@real_ip:5060>
Call-ID: 1768775a565198e46456e6d416067361@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="MTQ0MTQ0MDUyOTouNu9X0epjKiU3l9Fi3Bcd", response="072eb12a5781d67afd7c6f50575dda63", opaque="MTQ0MTQ0MDUyOTouNu9X0epjKiU3l9Fi3Bcd", qop=auth, cnonce="7f8e671a", nc=00000001
Date: Sat, 05 Sep 2015 08:08:49 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 692923187 692923188 IN IP4 real_ip
s=Asterisk PBX 10.1.2
c=IN IP4 real_ip
t=0 0
m=audio 11652 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 12:08:50] 
<--- 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=z9hG4bK09d7a83f;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as788340ba
To: <sip:C-Number@multifon.ru>
Call-ID: 1768775a565198e46456e6d416067361@multifon.ru
CSeq: 103 INVITE

<------------->
[2015-09-05 12:08:50] --- (6 headers 0 lines) ---
[2015-09-05 12:08:50] 
<--- 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=z9hG4bK09d7a83f;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as788340ba
To: <sip:C-Number@multifon.ru>;tag=SDctrm099-231C324631353641808E6500
Call-ID: 1768775a565198e46456e6d416067361@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 11044 RTP/AVP 8
b=AS:80
a=rtpmap:8 PCMA/8000
a=maxptime:20
a=ptime:20
<------------->
[2015-09-05 12:08:50] --- (12 headers 10 lines) ---
[2015-09-05 12:08:50] list_route: hop: <sip:C-Number@193.201.229.35:5060;transport=udp>
[2015-09-05 12:08:50] Found RTP audio format 8
[2015-09-05 12:08:50] Found audio description format PCMA for ID 8
[2015-09-05 12:08:50] Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[2015-09-05 12:08:50] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2015-09-05 12:08:50] Peer audio RTP is at port 193.201.229.19:11044
[2015-09-05 12:08:50]     -- SIP/multifon-out-0000001e is making progress passing it to SIP/msco-0000001d
[2015-09-05 12:08:55] 
<--- 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=z9hG4bK09d7a83f;rport=5060
From: "from_user" <sip:from_user@multifon.ru>;tag=as788340ba
To: <sip:C-Number@multifon.ru>;tag=SDctrm099-231C324631353641808E6500
Call-ID: 1768775a565198e46456e6d416067361@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 12:08:55] --- (10 headers 0 lines) ---
[2015-09-05 12:08:55]     -- SIP/multifon-out-0000001e is ringing
AstT*CLI> sip set debug off
SIP Debugging Disabled
[2015-09-05 12:08:59]   == Spawn extension (localoca, 374B-Number, 11) exited non-zero on 'SIP/msco-0000001d'
AstT*CLI> 