network*CLI> debug level 4
Debugging level set to 4, file '<any>'
network*CLI> sip debug
SIP Debugging enabled

network*CLI>
<-- SIP read from 192.168.1.100:5061:
INVITE sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-9d05f9c8;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 330 <sip:330@192.168.1.100:5061>
Expires: 240
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 420
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 9296 9296 IN IP4 192.168.1.100
s=-
c=IN IP4 192.168.1.100
t=0 0
m=audio 28030 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (14 headers 19 lines)---
Using INVITE request as basis request - 108a6658-ce0a6457@192.168.1.100
Sending to 192.168.1.100 : 5061 (NAT)
Reliably Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-9d05f9c8;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aa50e50
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Proxy-Authenticate: Digest realm="voip.whisker.org.uk", nonce="3ec57029"
Content-Length: 0


---
Scheduling destruction of call '108a6658-ce0a6457@192.168.1.100' in 15000 ms
Found user '330'
network*CLI>
<-- SIP read from 192.168.1.100:5061:
ACK sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-9d05f9c8;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aa50e50
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 101 ACK
Max-Forwards: 70
Contact: 330 <sip:330@192.168.1.100:5061>
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 0


--- (10 headers 0 lines)---
network*CLI>
<-- SIP read from 192.168.1.100:5061:
INVITE sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-ed696b3c;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="330",realm="voip.whisker.org.uk",nonce="3ec57029",uri="sip:02075449494@192.168.1.250:5052",algorithm=MD5,response="0e1c4c82710fd2ed05bec3f5b250d2a7"
Contact: 330 <sip:330@192.168.1.100:5061>
Expires: 240
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 420
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 9296 9296 IN IP4 192.168.1.100
s=-
c=IN IP4 192.168.1.100
t=0 0
m=audio 28030 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (15 headers 19 lines)---
Using INVITE request as basis request - 108a6658-ce0a6457@192.168.1.100
Sending to 192.168.1.100 : 5061 (NAT)
Found user '330'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.100:28030
Found description format PCMA
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x71e (gsm|ulaw|alaw|g726|g729|speex|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x51c (ulaw|alaw|g726|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 02075449494 in fromspa2 (domain 192.168.1.250)
list_route: hop: <sip:330@192.168.1.100:5061>
Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-ed696b3c;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Length: 0


---
network*CLI> Jan 21 04:00:50 WARNING[18549]: pbx.c:5972 pbx_builtin_setvar_old: SetVar is deprecated, please use Set instead.
Jan 21 04:00:50 WARNING[18549]: ast_expr2.fl:176 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_RP, expecting $end; Input:
"4420" = "4480")
               ^
    -- Executing Goto("SIP/330-847a", "passworded|*#902075449494|1") in new stack
    -- Goto (passworded,*#902075449494,1)
    -- Executing SetVar("SIP/330-847a", "SH=*#") in new stack
Jan 21 04:00:50 WARNING[18549]: pbx.c:5972 pbx_builtin_setvar_old: SetVar is deprecated, please use Set instead.
    -- Executing SetVar("SIP/330-847a", "STARS=1") in new stack
    -- Executing SetVar("SIP/330-847a", "HASH=1") in new stack
    -- Executing Goto("SIP/330-847a", "902075449494|1") in new stack
    -- Goto (passworded,902075449494,1)
    -- Executing Macro("SIP/330-847a", "enum-call|442075449494|lcr|02075449494") in new stack
    -- Executing SetCallerID("SIP/330-847a", "442070436532") in new stack
Jan 21 04:00:50 WARNING[18549]: ast_expr2.fl:176 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_RP, expecting $end; Input:
"4420" = "4480")
               ^
Jan 21 04:00:50 WARNING[18549]: ast_expr2.fl:180 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source.
If you have questions, please refer to doc/README.variables in the asterisk source.
    -- Executing GotoIf("SIP/330-847a", "0?54") in new stack
    -- Executing EnumLookup("SIP/330-847a", "442075449494") in new stack
Jan 21 04:00:50 WARNING[18549]: app_enumlookup.c:99 enumlookup_exec: The application EnumLookup is deprecated.  Please use the ENUMLOOKUP() function instead.
The application EnumLookup is deprecated.  Please use the ENUMLOOKUP() function instead.
Destroying call '68a9aeca04b7d8f9563c41e530829831@18866.co.uk'
network*CLI>
<-- SIP read from 192.168.1.100:5061:
INFO sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-f1ea84d0;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 103 INFO
Max-Forwards: 70
Proxy-Authorization: Digest username="330",realm="voip.whisker.org.uk",nonce="3ec57029",uri="sip:02075449494@192.168.1.250:5052",algorithm=MD5,response="b482d0a1de6113b30d9dcbfac4545a74"
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 24
Content-Type: application/dtmf-relay

Signal=#
Duration=100

--- (11 headers 2 lines)---
Receiving INFO!
* DTMF-relay event received: #
Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-f1ea84d0;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 103 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Length: 0


---
Destroying call '7279a18672ee9d6c6d0da9667404f293@iptel.org'
Destroying call '1fa170ea5f2f1537759aa3a55bdd32dd@192.168.1.250'
network*CLI> Jan 21 04:00:50 WARNING[18549]: app_setcidname.c:71 setcallerid_exec: SetCIDName is deprecated, please use Set(CALLERID(name)=value) instead.
Jan 21 04:00:50 WARNING[18549]: app_setcidnum.c:76 setcallerid_exec: SetCIDNum is deprecated, please use Set(CALLERID(number)=value) instead.
    -- Executing Macro("SIP/330-847a", "lcr|02075449494|") in new stack
    -- Executing Goto("SIP/330-847a", "lcr|02075449494|1") in new stack
    -- Goto (lcr,02075449494,1)
  == Channel 'SIP/330-847a' jumping out of macro 'lcr'
  == Channel 'SIP/330-847a' jumping out of macro 'enum-call'
    -- Executing GotoIfTime("SIP/330-847a", "8:00-17:59|mon-fri|*|*?lcr_day|02075449494|1") in new stack
    -- Executing GotoIfTime("SIP/330-847a", "*|mon-fri|*|*?lcr_eve|02075449494|1") in new stack
    -- Executing Goto("SIP/330-847a", "lcr_any|02075449494|1") in new stack
    -- Goto (lcr_any,02075449494,1)
    -- Executing Macro("SIP/330-847a", "Finarea|00442075449494") in new stack
    -- Executing SetVar("SIP/330-847a", "FINAREA=VS") in new stack
    -- Executing GotoIf("SIP/330-847a", "1?4") in new stack
    -- Goto (macro-Finarea,s,4)
    -- Executing GotoIf("SIP/330-847a", "1?6") in new stack
    -- Goto (macro-Finarea,s,6)
    -- Executing GotoIf("SIP/330-847a", "1?8") in new stack
    -- Goto (macro-Finarea,s,8)
    -- Executing GotoIf("SIP/330-847a", "1?10") in new stack
    -- Goto (macro-Finarea,s,10)
    -- Executing GotoIf("SIP/330-847a", "0?12") in new stack
    -- Executing Macro("SIP/330-847a", "VS|00442075449494") in new stack
    -- Executing GotoIf("SIP/330-847a", "1?4") in new stack
    -- Goto (macro-VS,s,4)
    -- Executing SetVar("SIP/330-847a", "Starhash=") in new stack
    -- Executing GotoIf("SIP/330-847a", "1?8") in new stack
    -- Goto (macro-VS,s,8)
    -- Executing SetVar("SIP/330-847a", "Starhash=") in new stack
    -- Executing SetAMAFlags("SIP/330-847a", "billing") in new stack
    -- Executing SetCIDName("SIP/330-847a", "Whisker") in new stack
Jan 21 04:00:50 WARNING[18549]: app_setcidname.c:71 setcallerid_exec: SetCIDName is deprecated, please use Set(CALLERID(name)=value) instead.
    -- Executing SetCIDNum("SIP/330-847a", """") in new stack
Jan 21 04:00:50 WARNING[18549]: app_setcidnum.c:76 setcallerid_exec: SetCIDNum is deprecated, please use Set(CALLERID(number)=value) instead.
    -- Executing Dial("SIP/330-847a", "SIP/00442075449494@VS|300|") in new stack
We're at 83.67.204.141 port 28050
Adding codec 0x8 (alaw) to SDP
13 headers, 8 lines
Reliably Transmitting (no NAT) to 80.239.235.201:5060:
INVITE sip:00442075449494@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 83.67.204.141:5052;branch=z9hG4bK08b99da6;rport
From: "Whisker" <sip:whiskerp1@voipstunt.com>;tag=as77af15eb
To: <sip:00442075449494@sip.voipstunt.com>
Contact: <sip:whiskerp1@83.67.204.141:5052>
Call-ID: 1859300d75cbf232285080065aadb0a8@voipstunt.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 21 Jan 2006 04:00:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 162

v=0
o=root 18531 18531 IN IP4 83.67.204.141
s=session
c=IN IP4 83.67.204.141
t=0 0
m=audio 28050 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

---
    -- Called 00442075449494@VS
network*CLI>
<-- SIP read from 80.239.235.201:5060:
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 83.67.204.141:5052;branch=z9hG4bK08b99da6;rport
From: "whisker"<sip:whiskerp1@voipstunt.com>;tag=as77af15eb
To: <sip:00442075449494@sip.voipstunt.com>;tag=c9ebef50c90078c2c93eddc243d0bdd23df6
Contact: sip:80.239.235.201:5060
Call-ID: 1859300d75cbf232285080065aadb0a8@voipstunt.com
CSeq: 102 INVITE
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Content-Type: application/sdp
Content-Length: 157

v=0
o=whiskerp1 123456 123456 IN IP4 80.239.235.161
s=SIP Call
c=IN IP4 80.239.235.161
t=0 0
m=audio 41550 RTP/AVP 8
a=rtpmap:8 pcma/8000
a=ptime:20

--- (11 headers 8 lines)---
Found RTP audio format 8
Peer audio RTP is at port 80.239.235.161:41550
Found description format pcma
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    -- SIP/VS-d787 is making progress passing it to SIP/330-847a
We're at 192.168.1.250 port 28058
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-ed696b3c;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Type: application/sdp
Content-Length: 394

v=0
o=root 18531 18531 IN IP4 192.168.1.250
s=session
c=IN IP4 192.168.1.250
t=0 0
m=audio 28058 RTP/AVP 8 0 3 2 18 110 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Destroying call '57e44a6760535fc738f22ab0354385d0@192.168.1.250'
Destroying call '58b79a2d3e37df32149161a955ef1f46@192.168.1.250'
Destroying call '0bef69064ee84d46093b84295c606ddd@sip.voipuser.org'
network*CLI>
<-- SIP read from 80.239.235.201:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 83.67.204.141:5052;branch=z9hG4bK08b99da6;rport
From: "whisker"<sip:whiskerp1@voipstunt.com>;tag=as77af15eb
To: <sip:00442075449494@sip.voipstunt.com>;tag=c9ebef50c90078c2c93eddc243d0bdd23df6
Contact: sip:80.239.235.201:5060
Call-ID: 1859300d75cbf232285080065aadb0a8@voipstunt.com
CSeq: 102 INVITE
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Content-Type: application/sdp
Content-Length: 157

v=0
o=whiskerp1 123456 123456 IN IP4 80.239.235.161
s=SIP Call
c=IN IP4 80.239.235.161
t=0 0
m=audio 41550 RTP/AVP 8
a=rtpmap:8 pcma/8000
a=ptime:20

--- (11 headers 8 lines)---
Found RTP audio format 8
Peer audio RTP is at port 80.239.235.161:41550
Found description format pcma
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:80.239.235.201:5060>
set_destination: Parsing <sip:80.239.235.201:5060> for address/port to send to
set_destination: set destination to 80.239.235.201, port 5060
Transmitting (no NAT) to 80.239.235.201:5060:
ACK sip:80.239.235.201:5060 SIP/2.0
Via: SIP/2.0/UDP 83.67.204.141:5052;branch=z9hG4bK7a241072;rport
From: "Whisker" <sip:whiskerp1@voipstunt.com>;tag=as77af15eb
To: <sip:00442075449494@sip.voipstunt.com>;tag=c9ebef50c90078c2c93eddc243d0bdd23df6
Contact: <sip:whiskerp1@83.67.204.141:5052>
Call-ID: 1859300d75cbf232285080065aadb0a8@voipstunt.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/VS-d787 answered SIP/330-847a
We're at 192.168.1.250 port 28058
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-ed696b3c;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Type: application/sdp
Content-Length: 394

v=0
o=root 18531 18532 IN IP4 192.168.1.250
s=session
c=IN IP4 192.168.1.250
t=0 0
m=audio 28058 RTP/AVP 8 0 3 2 18 110 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/330-847a and SIP/VS-d787
network*CLI>
<-- SIP read from 192.168.1.100:5061:
ACK sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="330",realm="voip.whisker.org.uk",nonce="3ec57029",uri="sip:02075449494@192.168.1.250:5052",algorithm=MD5,response="60688dd505f0f9919ddd3f39d1dfb303"
Contact: 330 <sip:330@192.168.1.100:5061>
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 0


--- (11 headers 0 lines)---
Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
network*CLI>
<-- SIP read from 192.168.1.100:5061:
ACK sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="330",realm="voip.whisker.org.uk",nonce="3ec57029",uri="sip:02075449494@192.168.1.250:5052",algorithm=MD5,response="60688dd505f0f9919ddd3f39d1dfb303"
Contact: 330 <sip:330@192.168.1.100:5061>
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 0


--- (11 headers 0 lines)---
Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
network*CLI>
<-- SIP read from 192.168.1.100:5061:
ACK sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="330",realm="voip.whisker.org.uk",nonce="3ec57029",uri="sip:02075449494@192.168.1.250:5052",algorithm=MD5,response="60688dd505f0f9919ddd3f39d1dfb303"
Contact: 330 <sip:330@192.168.1.100:5061>
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 0


--- (11 headers 0 lines)---
Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
Retransmitting #1 (no NAT) to 192.168.1.100:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-ed696b3c;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Type: application/sdp
Content-Length: 394

v=0
o=root 18531 18532 IN IP4 192.168.1.250
s=session
c=IN IP4 192.168.1.250
t=0 0
m=audio 28058 RTP/AVP 8 0 3 2 18 110 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
network*CLI>
<-- SIP read from 192.168.1.100:5061:
ACK sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="330",realm="voip.whisker.org.uk",nonce="3ec57029",uri="sip:02075449494@192.168.1.250:5052",algorithm=MD5,response="60688dd505f0f9919ddd3f39d1dfb303"
Contact: 330 <sip:330@192.168.1.100:5061>
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 0


--- (11 headers 0 lines)---
Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
network*CLI>
<-- SIP read from 192.168.1.100:5061:
ACK sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="330",realm="voip.whisker.org.uk",nonce="3ec57029",uri="sip:02075449494@192.168.1.250:5052",algorithm=MD5,response="60688dd505f0f9919ddd3f39d1dfb303"
Contact: 330 <sip:330@192.168.1.100:5061>
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 0


--- (11 headers 0 lines)---
Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
network*CLI>
<-- SIP read from 192.168.1.100:5061:
ACK sip:02075449494@192.168.1.250:5052 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="330",realm="voip.whisker.org.uk",nonce="3ec57029",uri="sip:02075449494@192.168.1.250:5052",algorithm=MD5,response="60688dd505f0f9919ddd3f39d1dfb303"
Contact: 330 <sip:330@192.168.1.100:5061>
User-Agent: Sipura/SPA2000-3.1.5
Content-Length: 0


--- (11 headers 0 lines)---
Transmitting (no NAT) to 192.168.1.100:5061:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.100:5061;branch=z9hG4bK-3068195b;rport;received=192.168.1.100
From: 330 <sip:330@192.168.1.250:5052>;tag=53aee664bd3b03bbo1
To: <sip:02075449494@192.168.1.250:5052>;tag=as4aecbc17
Call-ID: 108a6658-ce0a6457@192.168.1.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02075449494@192.168.1.250:5052>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing