SRVVOIP1*CLI>sip set debug ip freephonie.net
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
INVITE sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-7d5eff759e47110d-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@10.X.X.X:61399;rinstance=340261802ca46cef>
To: <sip:90950XXXXXX@WORKGROUP:5060>
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.19920.0
Content-Length: 407
 
v=0
o=3cxVCE 158336970 98167440 IN IP4 10.X.X.X
s=3cxVCE Audio Call
c=IN IP4 10.X.X.X
t=0 0
m=audio 40036 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40034 RTP/AVP 34
c=IN IP4 10.X.X.X
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv
 
<------------->
--- (13 headers 18 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 10.X.X.X : 61399 (no NAT)
Using INVITE request as basis request - Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
Found user '202' for '202'
 
<--- Reliably Transmitting (no NAT) to 10.X.X.X:61399 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-7d5eff759e47110d-1---d8754z-;received=10.X.X.X;rport=61399
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as2c365984
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="098e1d89"
Content-Length: 0
 
 
<------------>
Scheduling destruction of SIP dialog 'Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
ACK sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-7d5eff759e47110d-1---d8754z-;rport
Max-Forwards: 70
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as2c365984
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 1 ACK
Content-Length: 0
 
 
<------------->
--- (8 headers 0 lines) ---
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
INVITE sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:202@10.X.X.X:61399;rinstance=340261802ca46cef>
To: <sip:90950XXXXXX@WORKGROUP:5060>
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.19920.0
Authorization: Digest username="202",realm="asterisk",nonce="098e1d89",uri="sip:90950XXXXXX@WORKGROUP:5060",response="217b28fe239c388230418edca15ba4f9",algorithm=MD5
Content-Length: 407
 
v=0
o=3cxVCE 158336970 98167440 IN IP4 10.X.X.X
s=3cxVCE Audio Call
c=IN IP4 10.X.X.X
t=0 0
m=audio 40036 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40034 RTP/AVP 34
c=IN IP4 10.X.X.X
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1;
a=sendrecv
 
<------------->
--- (14 headers 18 lines) ---
Sending to 10.X.X.X : 61399 (no NAT)
Using INVITE request as basis request - Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
Found user '202' for '202'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - 0x4 (ulaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/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 10.X.X.X:40036
Looking for 90950XXXXXX in appel_interne_X (domain WORKGROUP)
list_route: hop: <sip:202@10.X.X.X:61399;rinstance=340261802ca46cef>
SRVVOIP1*CLI>
<--- Transmitting (no NAT) to 10.X.X.X:61399 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;received=10.X.X.X;rport=61399
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
To: <sip:90950XXXXXX@WORKGROUP:5060>
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:90950XXXXXX@10.X.X.X>
Content-Length: 0
 
 
<------------>
    -- Executing [90950XXXXXX@appel_interne_X:1] ChanIsAvail("SIP/202-0000000b", "SIP/90950XXXXXX@free-0950XXXXXX" ) in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Really destroying SIP dialog '3d93105b788f6f256b4ddcdb1e4defce@10.X.X.X' Method: INVITE
    -- Executing [90950XXXXXX@appel_interne_X:2] Dial("SIP/202-0000000b", "SIP/90950XXXXXX@free-0950XXXXXX),30,r" ) in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Really destroying SIP dialog '2870e9af28c7940d24f94dc775dcf1dd@10.X.X.X' Method: INVITE
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [90950XXXXXX@appel_interne_X:3] Congestion("SIP/202-0000000b", "" ) in new stack
 
<--- Reliably Transmitting (no NAT) to 10.X.X.X:61399 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;received=10.X.X.X;rport=61399
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as7c874b55
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 20
 
 
<------------>
  == Spawn extension (appel_interne_X, 90950XXXXXX, 3) exited non-zero on 'SIP/202-0000000b'
SRVVOIP1*CLI>
<--- SIP read from UDP://10.X.X.X:61399 --->
ACK sip:90950XXXXXX@WORKGROUP:5060 SIP/2.0
Via: SIP/2.0/UDP 10.X.X.X:61399;branch=z9hG4bK-d8754z-b65fcc098d7cc872-1---d8754z-;rport
Max-Forwards: 70
To: <sip:90950XXXXXX@WORKGROUP:5060>;tag=as7c874b55
From: "beber"<sip:202@WORKGROUP:5060>;tag=ff1a483e
Call-ID: Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.
CSeq: 2 ACK
Content-Length: 0
 
 
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'Mzg3YzZhYTYyMzIzMDEzOWQwNjBlZjhkMjA5NWYzYTM.' Method: ACK
SRVVOIP1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[SRVVOIP1.localdomain asterisk]#

[general]
 
Binaddr=0.0.0.0
Bindport=5060
Disallow=all
Allow=ulaw
Context=appel_interne_X
Dtmfmode=rfc2833
Allowoverlap=yes
Tos_sip=cs3
Tos_audio=ef
Tos=lowdelay
srvlookup=yes
language=fr
Context=free-0950XXXXXX
register => 0950XXXXXX:password@freephonie.net/0950XXXXXX
defaultexpirey=1800
 
 
 [0950XXXXXX]
; Venant du forfait telephonique (free)
 
type=peer
host=sip.freephonie.net
context=0950XXXXXX
language=fr
insecure=port
defaultuser=0950XXXXXX
secret=password
nat=yes
canreinvite=no
dtmfmode=auto
videosupport=no
restrictcid=no
amaflags=default
 
 
[200]
 
Defaultuser=200
Secret= password
callerid="bob" <200>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
Mailbox=200@voicemail
 
 
[201]
 
Defaultuser=201
Secret= password
callerid="val" <201>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
Mailbox=201@voicemail
 
 
[202]
 
Defaultuser=202
Secret= password
callerid="beber" <202>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
Mailbox=202@voicemail
 
[sets](!)
; Creation d'une section SIP (extension) pour un poste interne
; A repeter selon le nombre de postes
; Utilisation de template
 
Defaultuser=200
Secret=password
callerid="bob" <200>
Type=friend
Context=appel_interne_X
Host=dynamic
Qualify=yes
Nat=no
Directmedia=yes
Mailbox=200@voicemail
 
[203](sets)
[204](sets)
[205](sets)
 
 
; CONFIG IP PHONE CISCO 7960
 
;[pstn01]
;defaultuser=pstn01
;secret=pstn01
;type=friend
;host=dynamic
;allow=all
;context=appel_interne_X
 
;[pstn02]
;defaultuser=pstn02
;secret=pstn02
;type=friend
;host=dynamic
;allow=all
;context=appel_interne_X
 
;[test_cisco]
;defaultuser=test
;secret=test
;type=friend
;host=dynamic
;allow=all
;context=appel_interne_X
 

[appel_interne_X]
 
Include=>salle_de_conference
Include=>local_voicemail
Include=>horloge_parlante
Include=>auto_attendant
Include=>parkedcalls
Include=>client
Include=>globals
Include=>general
Include=>from-asterisk2
Include=>menu
Include=>menu1
Include=>menu2
Include=>menu3
Include=>free-0950XXXXXX
Include=>test
 
 
[general]
 
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
 priorityjumping=no
 
[globals]
 
GENERAL=SIP/200&SIP/201&SIP/202&SIP/0950XXXXXX
CONSOLE=Console/dsp
SIPINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
 
[client]
 
; APPELS INTERNES
 
exten => _2XX,1,Dial(SIP/${EXTEN},20,tr)
exten => _2XX,2,Voicemail(2${EXTEN:1}@default)
exten => _2XX,3,Hangup()
 
 
; MESSAGERIE
 
exten => _3XX,1,Answer()
exten => _3XX,2,Wait(1)
exten => _3XX,3,VoiceMailMain(${EXTEN}@default)
exten => _3XX,4,Hangup()
 
 
; APPELS SORTANTS
 
exten => _9.,1,ChanIsAvail(SIP/${EXTEN}@free-0950XXXXXX)
exten => _9.,2,Dial(SIP/${EXTEN}@free-0950XXXXXX),30,r)
exten => _9.,3,Congestion
 
exten => _9.,104,Dial(Zap/1/${EXTEN},30,r)
exten => _9.,105,Congestion
 
exten => _9.,206,SetLanguage(fr)
exten => _9.,207,Wait(1)
exten => _9.,208,Playback(all-circuits-busy-now)
exten => _9.,209,Hangup
 
; LOOPBACK DE TEST
 
exten => 99,1,Answer()
exten => 99,2,SetLanguage(fr)
exten => 99,3,Echo()
exten => 99,4,Playback(vm-goodbye)
exten => 99,5,Hangup()
 
 
[Menu]
; standard automatique
 
exten => s, 1, Background(/var/msg/Menu)
exten => s, 2, WaitExten(2)
exten => s, 3, Goto(Menu,s,1)
 
exten => 1, 1,SayNumber(1)
exten => 1, 2, goto(Menu1,s,1)
exten => 2, 1, SayNumber(2)
exten => 2, 2, Goto(Menu2,s,1)
 
exten => 3, 1, SayNumber(3)
exten => 3, 2, Goto(Menu3,s,1)
 
exten => 9, 1, SayNumber(9)
exten => 9, 2, Hangup()
 
 
[Menu1]
 
exten => s, 1, Background(/var/msg/Menu1)
exten => s, 2, WaitExten(2)
exten => s, 3, Goto(Menu1,s,1)
 
exten => 1, 1,SayNumber(1)
exten => 1, 2, goto(local,200, 1)
 
exten => 2, 1, SayNumber(2)
exten => 2, 2, Goto(local,201, 1)
 
exten => 3, 1, SayNumber(3)
exten => 3, 2, Goto(local,202, 1)
 
exten => 9, 1, SayNumber(9)
exten => 9, 2, Hangup()
 
 
[Menu2]
 
exten => s, 1, Background(/var/msg/Menu2)
exten => s, 2, WaitExten(2)
exten => s, 3, Goto(Menu2,s,1)
 
exten => 1, 1, SayNumber(1)
exten => 1, 2, Goto(local,200, 2)
 
exten => 2, 1, SayNumber(2)
exten => 2, 2, Goto(local,201, 2)
 
exten => 3, 1, SayNumber(3)
exten => 3, 2, Goto(local,202, 2)
 
exten => 9, 1, SayNumber(9)
exten => 9, 2, Hangup()
 
 
[Menu3]
 
exten => s, 1, Background(/var/msg/Menu3)
exten => s, 2, WaitExten(2)
exten => s, 3, Goto(Menu3,s,1)
 
exten => 1, 1, SayNumber(1)
exten => 1, 2, Goto(local,211, 1)
 
exten => 2, 1, SayNumber(2)
exten => 2, 2, Goto(local,298, 1)
 
exten => 3, 1, SayNumber(3)
exten => 3, 2, Goto(local,212, 1)
 
exten => 9, 1, SayNumber(9)
exten => 9, 2, Hangup()
 
 
 [free-0950XXXXXX]
 
exten => s,1,Answer
 
exten => s,2,JABBERSend(gtalk_account,0950XXXXXX:password@sip.freephonie.net/0950XXXXXX
exten => s,3,Playback(/home/xxxx/voip/voix_recherche_telephone)
exten => s,4,Ringing(4)
exten => s,5,Dial(SIP/200,10,tm)
exten => s,6,Ringing(4)
exten => s,7,Dial(SIP/201,10,tm)
exten => s,8,VoiceMail(1001)
exten => s,9,Hangup(16)
 
 
; test IP Phone cisco 7960
;[test]
;exten => cisco_line1,1,Goto(200,1)
;exten => 200,1,Dial(SIP/pstn01)
;exten => 200,2,Hangup()
 
;exten => cisco_line2,1,Goto(201,1)
;exten => 201,1,Dial(SIP/pstn01)
;exten => 201,2,Hangup()
 
;exten => 202,1,Dial(SIP/test_cisco)
;exten => 202,2,Hangup()
 
 
 [local_voicemail]
 
Exten=>888,1,Answer()
Exten=>888,2,VoiceMailMain()
Exten=>888,3,Hangup()
 
 
[horloge_parlante]
 
Exten=>3344,1,Answer()
Exten=>3344,2,SayUnixTime(,Europe/Paris,AdBY kM)
Exten=>3344,3,Hangup()
 
 
[auto_attendant]
 
Exten=>2233,1,Read(digit,/var/lib/asterisk/sounds/en/hello-word,1)
Exten=>2233,2,Gotoif($["${digit}" = "1"]?appel_interne_X,3344,1)
Exten=>2233,3,Gotoif($["${digit}" = "2"]?appel_interne_X,900,1)
Exten=>2233,4,Gotoif($["${digit}" = "3"]?appel_interne_X,888,1)
Exten=>i,1,Goto(2233,1)
 
[macro-appel_interne_X] :
 
Exten=>s,1,Answer()
Exten=>s,2,Dial(${ARG1},20,Ttr)
Exten=>s,3,VoiceMail(${ARG1}@voicemail)
Exten=>s,4,Playback(vm-goodbye)
Exten=>s,5,Hangup()

SRVVOIP1*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time
freephonie.net:5060             0950XXXXXX        1785 Registered           Wed, 14 Sep 2011 09:31:47
1 SIP registrations.
SRVVOIP1*CLI>

; APPELS SORTANTS
 
exten => _9.,1,ChanIsAvail(SIP/${EXTEN}@free-0950XXXXXX)
exten => _9.,2,Dial(SIP/${EXTEN}@free-0950XXXXXX),30,r)
exten => _9.,3,Congestion
 
exten => _9.,104,Dial(Zap/1/${EXTEN},30,r)
exten => _9.,105,Congestion
 
exten => _9.,206,SetLanguage(fr)
exten => _9.,207,Wait(1)
exten => _9.,208,Playback(all-circuits-busy-now)
exten => _9.,209,Hangup

 == Everyone is busy/congested at this time (1:0/0/1)

SRVVOIP1*CLI> sip set debug peer 0950XXXXXX
SIP Debugging Enabled for IP: 212.27.52.5:5060
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 82.X.X.X:5060;branch=z9hG4bK1f55abcb;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@DOMAIN>;tag=as78adad8c
To: <sip:freephonie.net>
Contact: <sip:Unknown@82.X.X.X>
Call-ID: 1bf6409a4e27c9cd078a9d5f5722552a@DOMAIN
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Tue, 20 Sep 2011 13:46:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
 
 
---
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
SIP/2.0 501 Not Implemented Yet
Call-ID: 1bf6409a4e27c9cd078a9d5f5722552a@DOMAIN
CSeq: 102 OPTIONS
From: "Unknown" <sip:Unknown@DOMAIN>;tag=as78adad8c
To: <sip:freephonie.net>;tag=00-31101-1ed40e68-735604160
Via: SIP/2.0/UDP 82.X.X.X:5060;received=82.X.X.X;rport=1024;branch=z9hG4bK1f55abcb
Content-Length: 0
 
 
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1bf6409a4e27c9cd078a9d5f5722552a@DOMAIN' Method: OPTIONS
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [90950XXXXXX@appel_interne_X:1] Ringing("SIP/202-00000059", "" ) in new stack
    -- Executing [90950XXXXXX@appel_interne_X:2] Wait("SIP/202-00000059", "" ) in new stack
    -- Executing [90950XXXXXX@appel_interne_X:3] Answer("SIP/202-00000059", "" ) in new stack
    -- Executing [90950XXXXXX@appel_interne_X:4] Dial("SIP/202-00000059", "SIP/0950XXXXXX@freephonie.net),30,r" ) in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Everyone is busy/congested at this time (1:0/0/1)
SRVVOIP1*CLI>

SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
INVITE sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 23528-QA-102f2397-369c648f4@freephonie.net
Contact: <sip:172.X.X.X:5062>
Content-Type: application/sdp
CSeq: 264484018 INVITE
From: "0674420004" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=23528-XD-102f2398-54dd3b156
Max-Forwards: 27
Record-Route: <sip:C=on;t=UBFVF@212.27.52.5:5060;lr>
To: <sip:0950XXXXXX@172.X.X.X;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-UBFV-0e04cd9b-1b10a8d8
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 184
 
v=0
o=cp10 131659623199 131659623199 IN IP4 172.X.X.X
s=SIP Call
c=IN IP4 212.27.52.129
t=0 0
m=audio 31122 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv
 
<------------->
--- (13 headers 10 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (NAT)
Using INVITE request as basis request - 23528-QA-102f2397-369c648f4@freephonie.net
No user '06XXXXXXXX' in SIP users list
Found peer '0950XXXXXX-in' for '06XXXXXXXX' from 212.27.52.5:5060
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - 0x4 (ulaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
SRVVOIP1*CLI>
<--- Reliably Transmitting (NAT) to 212.27.52.5:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-UBFV-0e04cd9b-1b10a8d8;received=212.27.52.5
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=23528-XD-102f2398-54dd3b156
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as73716753
Call-ID: 23528-QA-102f2397-369c648f4@freephonie.net
CSeq: 264484018 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
 
 
<------------>
Scheduling destruction of SIP dialog '23528-QA-102f2397-369c648f4@freephonie.net' in 32000 ms (Method: INVITE)
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
ACK sip:0950XXXXXX@82.X.X.X:5060;transport=udp SIP/2.0
Call-ID: 23528-QA-102f2397-369c648f4@freephonie.net
CSeq: 264484018 ACK
From: "06XXXXXXXX" <sip:06XXXXXXXX@freephonie.net;user=phone>;tag=23528-XD-102f2398-54dd3b156
Max-Forwards: 27
To: <sip:0950XXXXXX@172.X.X.X;user=phone>;tag=as73716753
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-UBFV-0e04cd9b-1b10a8d8
Content-Length: 0
 
 
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '23528-QA-102f2397-369c648f4@freephonie.net' Method: ACK
SRVVOIP1*CLI>
<--- SIP read from UDP://212.27.52.5:5060 --->
Cirpack KeepAlive Packet
<------------->
SRVVOIP1*CLI>