dom0*CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [664040@STR-MAIN:1] Dial("SIP/101-00000014", "SIP/galacom/664040") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 10120
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 85.28.17.248:5060:
INVITE sip:664040@sip.galacom.ru SIP/2.0
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK2de8af73;rport
Max-Forwards: 70
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>
Contact: <sip:73812308877@92.124.129.58:5060>
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Tue, 03 Jul 2012 05:46:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296

v=0
o=root 19938248 19938248 IN IP4 92.124.129.58
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 92.124.129.58
t=0 0
m=audio 10120 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/galacom/664040

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK2de8af73;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=fe5a8b07ba70e85508daacf5e256d34d.eeb8
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.galacom.ru", nonce="T/KIVU/yhynQDlxNmRUOLaw7tAON4Wz1"
Server: kamailio (3.1.4 (x86_64/linux))
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:664040@sip.galacom.ru> for address/port to send to
set_destination: set destination to 85.28.17.248:5060
Transmitting (NAT) to 85.28.17.248:5060:
ACK sip:664040@sip.galacom.ru SIP/2.0
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK2de8af73;rport
Max-Forwards: 70
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=fe5a8b07ba70e85508daacf5e256d34d.eeb8
Contact: <sip:73812308877@92.124.129.58:5060>
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---
Audio is at 10120
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 85.28.17.248:5060:
INVITE sip:664040@sip.galacom.ru SIP/2.0
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK077c11d1;rport
Max-Forwards: 70
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>
Contact: <sip:73812308877@92.124.129.58:5060>
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Proxy-Authorization: Digest username="73812308877", realm="sip.galacom.ru", algorithm=MD5, uri="sip:664040@sip.galacom.ru", nonce="T/KIVU/yhynQDlxNmRUOLaw7tAON4Wz1", response="1a35f0975bee19aa6b79a5526511c273"
Date: Tue, 03 Jul 2012 05:46:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296

v=0
o=root 19938248 19938249 IN IP4 92.124.129.58
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 92.124.129.58
t=0 0
m=audio 10120 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK077c11d1;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 103 INVITE
Server: kamailio (3.1.4 (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK077c11d1;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Date: Tue, 03 Jul 2012 05:46:18 GMT
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
llow-Events: telephone-event
Contact: <sip:664040@85.28.17.244:5060>
Record-Route: <sip:85.28.17.248;lr=on>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 246

v=0
o=CiscoSystemsSIP-GW-UserAgent 7427 121 IN IP4 85.28.17.244
s=SIP Call
c=IN IP4 85.28.17.244
t=0 0
m=audio 17502 RTP/AVP 0 101
c=IN IP4 85.28.17.244
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (15 headers 11 lines) ---
list_route: hop: <sip:85.28.17.248;lr=on>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 85.28.17.244:17502
    -- SIP/galacom-00000015 is making progress passing it to SIP/101-00000014

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK077c11d1;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Date: Tue, 03 Jul 2012 05:46:18 GMT
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
llow-Events: telephone-event
Contact: <sip:664040@85.28.17.244:5060>
Record-Route: <sip:85.28.17.248;lr=on>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 246

v=0
o=CiscoSystemsSIP-GW-UserAgent 7427 121 IN IP4 85.28.17.244
s=SIP Call
c=IN IP4 85.28.17.244
t=0 0
m=audio 17502 RTP/AVP 0 101
c=IN IP4 85.28.17.244
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (15 headers 11 lines) ---
list_route: hop: <sip:85.28.17.248;lr=on>
    -- SIP/galacom-00000015 is making progress passing it to SIP/101-00000014

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK077c11d1;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Date: Tue, 03 Jul 2012 05:46:18 GMT
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:664040@85.28.17.244:5060>
Record-Route: <sip:85.28.17.248;lr=on>
Content-Type: application/sdp
Content-Length: 246

v=0
o=CiscoSystemsSIP-GW-UserAgent 7427 121 IN IP4 85.28.17.244
s=SIP Call
c=IN IP4 85.28.17.244
t=0 0
m=audio 17502 RTP/AVP 0 101
c=IN IP4 85.28.17.244
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (15 headers 11 lines) ---
list_route: hop: <sip:85.28.17.248;lr=on>
set_destination: Parsing <sip:85.28.17.248;lr=on> for address/port to send to
set_destination: set destination to 85.28.17.248:5060
Transmitting (NAT) to 85.28.17.248:5060:
ACK sip:664040@85.28.17.244:5060 SIP/2.0
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK21b5190c;rport
Route: <sip:85.28.17.248;lr=on>
Max-Forwards: 70
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Contact: <sip:73812308877@92.124.129.58:5060>
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---
    -- SIP/galacom-00000015 answered SIP/101-00000014
    -- Remotely bridging SIP/101-00000014 and SIP/galacom-00000015
set_destination: Parsing <sip:85.28.17.248;lr=on> for address/port to send to
set_destination: set destination to 85.28.17.248:5060
Audio is at 10120
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 85.28.17.248:5060:
INVITE sip:664040@85.28.17.244:5060 SIP/2.0
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK5b005f2d;rport
Route: <sip:85.28.17.248;lr=on>
Max-Forwards: 70
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Contact: <sip:73812308877@92.124.129.58:5060>
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 249

v=0
o=root 19938248 19938250 IN IP4 192.168.1.104
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.1.104
t=0 0
m=audio 21000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK5b005f2d;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 104 INVITE
Server: kamailio (3.1.4 (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK5b005f2d;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Date: Tue, 03 Jul 2012 05:46:24 GMT
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 104 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:664040@85.28.17.244:5060>
Content-Type: application/sdp
Content-Length: 246

v=0
o=CiscoSystemsSIP-GW-UserAgent 7427 122 IN IP4 85.28.17.244
s=SIP Call
c=IN IP4 85.28.17.244
t=0 0
m=audio 17502 RTP/AVP 0 101
c=IN IP4 85.28.17.244
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 85.28.17.244:17502
set_destination: Parsing <sip:85.28.17.248;lr=on> for address/port to send to
set_destination: set destination to 85.28.17.248:5060
Transmitting (NAT) to 85.28.17.248:5060:
ACK sip:664040@85.28.17.244:5060 SIP/2.0
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK7dd6a59c;rport
Route: <sip:85.28.17.248;lr=on>
Max-Forwards: 70
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Contact: <sip:73812308877@92.124.129.58:5060>
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


--
set_destination: Parsing <sip:85.28.17.248;lr=on> for address/port to send to
set_destination: set destination to 85.28.17.248:5060
Audio is at 10120
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 85.28.17.248:5060:
INVITE sip:664040@85.28.17.244:5060 SIP/2.0
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK5d01e6d5;rport
Route: <sip:85.28.17.248;lr=on>
Max-Forwards: 70
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Contact: <sip:73812308877@92.124.129.58:5060>
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 105 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 249

v=0
o=root 19938248 19938251 IN IP4 92.124.129.58
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 92.124.129.58
t=0 0
m=audio 10120 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK5d01e6d5;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 105 INVITE
Server: kamailio (3.1.4 (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK5d01e6d5;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Date: Tue, 03 Jul 2012 05:46:28 GMT
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 105 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:664040@85.28.17.244:5060>
Content-Type: application/sdp
Content-Length: 246

v=0
o=CiscoSystemsSIP-GW-UserAgent 7427 123 IN IP4 85.28.17.244
s=SIP Call
c=IN IP4 85.28.17.244
t=0 0
m=audio 17502 RTP/AVP 0 101
c=IN IP4 85.28.17.244
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 85.28.17.244:17502
set_destination: Parsing <sip:85.28.17.248;lr=on> for address/port to send to
set_destination: set destination to 85.28.17.248:5060
Transmitting (NAT) to 85.28.17.248:5060:
ACK sip:664040@85.28.17.244:5060 SIP/2.0
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK2a7411af;rport
Route: <sip:85.28.17.248;lr=on>
Max-Forwards: 70
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Contact: <sip:73812308877@92.124.129.58:5060>
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 105 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0


---
Scheduling destruction of SIP dialog '12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:85.28.17.248;lr=on> for address/port to send to
set_destination: set destination to 85.28.17.248:5060
Reliably Transmitting (NAT) to 85.28.17.248:5060:
BYE sip:664040@85.28.17.244:5060 SIP/2.0
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK58f4cace;rport
Route: <sip:85.28.17.248;lr=on>
Max-Forwards: 70
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
CSeq: 106 BYE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Proxy-Authorization: Digest username="73812308877", realm="sip.galacom.ru", algorithm=MD5, uri="sip:664040@85.28.17.244:5060", nonce="T/KIVU/yhynQDlxNmRUOLaw7tAON4Wz1", response="829921ba5b9d2e8c4373f49a52297502"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (STR-MAIN, 664040, 1) exited non-zero on 'SIP/101-00000014'

<--- SIP read from UDP:85.28.17.248:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 92.124.129.58:5060;branch=z9hG4bK58f4cace;rport=5060
From: "New User" <sip:73812308877@sip.galacom.ru>;tag=as210389f1
To: <sip:664040@sip.galacom.ru>;tag=2D85FE8-262D
Date: Tue, 03 Jul 2012 05:46:28 GMT
Call-ID: 12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 106 BYE

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '12798ac5768bc3db33141d406ac4a2e5@sip.galacom.ru' Method: INVITE