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Daniel Bareiro

2009-Aug-17 18:12 UTC

[asterisk-users] Accessing to ekiga.net through Asterisk

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi all!

I'm trying to connect to ekiga.net through a client connected to my
Asterisk server. For it I am being based on this [1] document. Next I
put the configurations that I am using.

/etc/asterisk/sip.conf:

; Outgoing to ekiga.net
[ekiga]
type=friend
username=MyUser
secret=MyPass
host=ekiga.net
canreinvite=no
qualify=300
nat = yes
stunaddr = stun.exiga.net
insecure=port,invite  ; required for incoming ekiga.net calls

/etc/asterisk/extensions.conf:

[from-internal]
...
exten => _8.,1,Dial(SIP/ekiga/${EXTEN:1},20,r))


I tried a echo test, dialing in my case to 8500, but in spite of seeing
traffic towards Internet, nothing is heard. Setting 'sip set debug', I
get
the following thing:


<--- SIP read from 10.1.0.65:5060 --->
INVITE sip:8500 at 10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: <sip:8500 at 10.1.0.10>
From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
CSeq: 183 INVITE
Contact: <sip:201 at 10.1.0.65>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.2
Content-Length: 247

v=0
o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
s=-
c=IN IP4 10.1.0.65
t=0 0
m=audio 8000 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
- --- (13 headers 12 lines) ---
Sending to 10.1.0.65 : 5060 (NAT)
Using INVITE request as basis request - mrsyiysrdkwmkeg at
defiant.freesoftware.org

<--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKjxcxrrks;received=10.1.0.65;rport=5060
From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
To: <sip:8500 at 10.1.0.10>;tag=as095989a3
Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
CSeq: 183 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="76b2dfe8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'mrsyiysrdkwmkeg at
defiant.freesoftware.org' in 32000 ms (Method: INVITE)
Found user '201'
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
ACK sip:8500 at 10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: <sip:8500 at 10.1.0.10>;tag=as095989a3
From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
CSeq: 183 ACK
User-Agent: Twinkle/1.2
Content-Length: 0


<------------->
- --- (9 headers 0 lines) ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
INVITE sip:8500 at 10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKoilauqhp
Max-Forwards: 70
Proxy-Authorization: Digest
username="201",realm="asterisk",nonce="76b2dfe8",uri="sip:8500
at
10.1.0.10",response="d49c0fdf11c9977fcd1fce6a50f445fe",algorithm=MD5
To: <sip:8500 at 10.1.0.10>
From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
CSeq: 184 INVITE
Contact: <sip:201 at 10.1.0.65>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.2
Content-Length: 247

v=0
o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
s=-
c=IN IP4 10.1.0.65
t=0 0
m=audio 8000 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
- --- (14 headers 12 lines) ---
Sending to 10.1.0.65 : 5060 (NAT)
Using INVITE request as basis request - mrsyiysrdkwmkeg at
defiant.freesoftware.org
Found user '201'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 10.1.0.65:8000
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.0.65:8000
Looking for 8500 in from-internal (domain 10.1.0.10)
list_route: hop: <sip:201 at 10.1.0.65>

<--- Transmitting (no NAT) to 10.1.0.65:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
To: <sip:8500 at 10.1.0.10>
Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
CSeq: 184 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:8500 at 10.1.0.10>
Content-Length: 0


<------------>
    -- Executing [8500 at from-internal:1] Dial("SIP/201-090ffff0",
"SIP/ekiga/500|20|r)") in new stack
Video is at 192.168.1.2 port 16080
Audio is at 192.168.1.2 port 14850
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x40000 (h261) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
INVITE sip:500 at ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4ea46842;rport
From: "Hector Bareiro" <sip:201 at 192.168.1.2>;tag=as2bb1b3cd
To: <sip:500 at ekiga.net>
Contact: <sip:201 at 192.168.1.2>
Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:36:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 331

v=0
o=root 4959 4959 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
b=CT:384
t=0 0
m=audio 14850 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 16080 RTP/AVP 31
a=rtpmap:31 H261/90000
a=sendrecv

- ---
    -- Called ekiga/500

<--- Transmitting (no NAT) to 10.1.0.65:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
To: <sip:8500 at 10.1.0.10>;tag=as1b0c8dab
Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
CSeq: 184 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:8500 at 10.1.0.10>
Content-Length: 0


<------------>
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK4ea46842;rport=28490;received=190.51.112.4
From: "Hector Bareiro" <sip:201 at
192.168.1.2:5060>;tag=as2bb1b3cd
To: <sip:500 at ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.9448
Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="ekiga.net",
nonce="4a899652000008b32e461e66fe61009ff6f4ffd9cb4e4ec2"
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (9 headers 0 lines) ---
Transmitting (no NAT) to 86.64.162.35:5060:
ACK sip:500 at ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4ea46842;rport
From: "Hector Bareiro" <sip:201 at 192.168.1.2>;tag=as2bb1b3cd
To: <sip:500 at ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.9448
Contact: <sip:201 at 192.168.1.2>
Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


- ---
Video is at 192.168.1.2 port 16080
Audio is at 192.168.1.2 port 14850
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x40000 (h261) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
INVITE sip:500 at ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5f88a0aa;rport
From: "Hector Bareiro" <sip:201 at 192.168.1.2>;tag=as2bb1b3cd
To: <sip:500 at ekiga.net>
Contact: <sip:201 at 192.168.1.2>
Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="danib",
realm="ekiga.net", algorithm=MD5, uri="sip:500 at
ekiga.net",
nonce="4a899652000008b32e461e66fe61009ff6f4ffd9cb4e4ec2",
response="950e5d853e07ad728da8ae8a02198034"
Date: Mon, 17 Aug 2009 17:36:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 331

v=0
o=root 4959 4960 IN IP4 192.168.1.2
s=session
c=IN IP4 192.168.1.2
b=CT:384
t=0 0
m=audio 14850 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=senon> for address/port to send to
set_destination: set destination to 86.64.162.35, port 5060
Transmitting (no NAT) to 86.64.162.35:5060:
ACK sip:500 at 86.64.162.35:5081 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK15031d34;rport
Route: <sip:86.64.162.35;lr=on>
From: "Hector Bareiro" <sip:201 at 192.168.1.2>;tag=as2bb1b3cd
To: <sip:500 at ekiga.net>;tag=as1603ca76
Contact: <sip:201 at 192.168.1.2>
Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


- ---
    -- SIP/ekiga-090cb900 answered SIP/201-090ffff0
Audio is at 10.1.0.10 port 14442
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
To: <sip:8500 at 10.1.0.10>;tag=as1b0c8dab
Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
CSeq: 184 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:8500 at 10.1.0.10>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 4959 4959 IN IP4 10.1.0.10
s=session
c=IN IP4 10.1.0.10
t=0 0
m=audio 14442 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
ACK sip:8500 at 10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKnovwlzvc
Max-Forwards: 70
Proxy-Authorization: Digest
username="201",realm="asterisk",nonce="76b2dfe8",uri="sip:8500
at
10.1.0.10",response="d49c0fdf11c9977fcd1fce6a50f445fe",algorithm=MD5
To: <sip:8500 at 10.1.0.10>;tag=as1b0c8dab
From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
CSeq: 184 ACK
User-Agent: Twinkle/1.2
Content-Length: 0


<------------->
- --- (10 headers 0 lines) ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK229d0a34;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as53f8b15a
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 6b6b26de041acc9b537b8d716cd18580 at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:37:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Sues
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK229d0a34
To: <sip:201 at 10.1.0.65>;tag=aacln
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as53f8b15a
Call-ID: 6b6b26de041acc9b537b8d716cd18580 at 10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.2
Supported: replaces,norefersub,100rel
Content-Length: 0


<------------->
- --- (13 headers 0 lines) ---
Really destroying SIP dialog '6b6b26de041acc9b537b8d716cd18580 at
10.1.0.10' Method: OPTIONS
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK10db4ef1;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as2ff24865
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 0a76cf0f0dd60b6855266e3c3105cd55 at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:37:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK10db4ef1;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as2ff24865
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.f0c5
Call-ID: 0a76cf0f0dd60b6855266e3c3105cd55 at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '0a76cf0f0dd60b6855266e3c3105cd55 at
192.168.1.2' Method: OPTIONS
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK77d011fa;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as1d024ca8
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 693813ae7b9c3e783112c4111b851071 at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:38:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supportnsmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7c056bcf;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as263b8e2b
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 544df8987fffac657cc726642845c34d at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:38:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK7c056bcf;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as263b8e2b
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d936
Call-ID: 544df8987fffac657cc726642845c34d at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '544df8987fffac657cc726642845c34d at
192.168.1.2' Method: OPTIONS
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK07c25ee9;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as587919f0
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 611ac37e0be64f463cdbe4d71ac4c74f at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:39:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK07c25ee9
To: <sip:201 at 10.1.0.65>;tag=doivz
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as587919f0
Call-ID: 611ac37e0be64f463cdbe4d71ac4c74f at 10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.2
Supporaces,norefersub,100rel
Content-Length: 0


<------------->
- --- (13 headers 0 lines) ---
Really destroying SIP dialog '611ac37e0be64f463cdbe4d71ac4c74f at
10.1.0.10' Method: OPTIONS
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK16aa33c2;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as361d1f0a
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 7e91ccff491d8a9d7856928c4c4f43e6 at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:39:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK16aa33c2;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as361d1f0a
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1b34
Call-ID: 7e91ccff491d8a9d7856928c4c4f43e6 at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '7e91ccff491d8a9d7856928c4c4f43e6 at
192.168.1.2' Method: OPTIONS
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK24a7bc95;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as5fa47440
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 49071c5656ab2a31252152a455139b09 at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:40:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK24a7bc95
To: <sip:201 at 10.1.0.65>;tag=sgply
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as5fa47440
Call-ID: 49071c5656ab2a31252152a455139b09 at 10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.2
Suppoorefersub,100rel
Content-Length: 0


<------------->
- --- (13 headers 0 lines) ---
Really destroying SIP dialog '49071c5656ab2a31252152a455139b09 at
10.1.0.10' Method: OPTIONS
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK13007a5c;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as5a0acdaf
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 3a193cd018c6a354017a0d501e7c59f8 at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK13007a5c;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as5a0acdaf
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.2e90
Call-ID: 3a193cd018c6a354017a0d501e7c59f8 at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '3a193cd018c6a354017a0d501e7c59f8 at
192.168.1.2' Method: OPTIONS
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK10cced95;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as076647df
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 7fc191e11f2509e7353b61d65b76f002 at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:41:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK10cced95
To: <sip:201 at 10.1.0.65>;tag=owawm
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as076647df
Call-ID: 7fc191e11f2509e7353b61d65b76f002 at 10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.2
Supported: replaces,norefersub,100rel
Content-Length: 0


<------------->
ers 0 lines) ---
Really destroying SIP dialog '7fc191e11f2509e7353b61d65b76f002 at
10.1.0.10' Method: OPTIONS
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3a587968;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as5139b49b
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 3f6b1291297e4eb20a2d65c662a67fb7 at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:41:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK3a587968;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as5139b49b
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.14a8
Call-ID: 3f6b1291297e4eb20a2d65c662a67fb7 at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '3f6b1291297e4eb20a2d65c662a67fb7 at
192.168.1.2' Method: OPTIONS
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK1c1f607a;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as416ac6cc
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 5ba8e26e43162c6b3c56b7787273d682 at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:42:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK1c1f607a
To: <sip:201 at 10.1.0.65>;tag=hplvm
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as416ac6cc
Call-ID: 5ba8e26e43162c6b3c56b7787273d682 at 10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,ESSAGE
Server: Twinkle/1.2
Supported: replaces,norefersub,100rel
Content-Length: 0


<------------->
- --- (13 headers 0 lines) ---
Really destroying SIP dialog '5ba8e26e43162c6b3c56b7787273d682 at
10.1.0.10' Method: OPTIONS
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK029714e0;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as686f2ada
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 6ca487dc5c8f40ab17428b3c76e6ff7c at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:42:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK029714e0;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as686f2ada
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.a004
Call-ID: 6ca487dc5c8f40ab17428b3c76e6ff7c at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '6ca487dc5c8f40ab17428b3c76e6ff7c at
192.168.1.2' Method: OPTIONS
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK4e32a4be;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as1d745b97
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 139126a231a61ca664de02153ee8cfc4 at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:43:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK4e32a4be
To: <sip:201 at 10.1.0.65>;tag=cvydb
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as1d745b97
Call-ID: 139126a231a61ca664de02153ee8cfc4 at 10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.2
Supported: replaces,norefe
<------------->
- --- (13 headers 0 lines) ---
Really destroying SIP dialog '139126a231a61ca664de02153ee8cfc4 at
10.1.0.10' Method: OPTIONS
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3513c9e8;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as348ceda1
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 3076a48850ae43bb5fd072736736ba52 at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:43:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK3513c9e8;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as348ceda1
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.dde9
Call-ID: 3076a48850ae43bb5fd072736736ba52 at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '3076a48850ae43bb5fd072736736ba52 at
192.168.1.2' Method: OPTIONS
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK0fd89b0f;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as204361ce
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 3418d0df540794014b7707011cb0bb9d at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:44:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK0fd89b0f
To: <sip:201 at 10.1.0.65>;tag=kwkmu
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as204361ce
Call-ID: 3418d0df540794014b7707011cb0bb9d at 10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.2
Supported: replaces,norefersub,100rel
Content-Length: 0


<------------->
- --- (13 headers 0 lines) ---
Really P dialog '3418d0df540794014b7707011cb0bb9d at 10.1.0.10' Method:
OPTIONS
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK32b3f65d;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as177de4d9
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 04607ade51978546773a635538a52a21 at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:44:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK32b3f65d;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as177de4d9
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.75a7
Call-ID: 04607ade51978546773a635538a52a21 at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '04607ade51978546773a635538a52a21 at
192.168.1.2' Method: OPTIONS
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK28825321;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as4bd66aee
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 5a4422e21401e157268de2df0efd0db9 at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:45:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK28825321
To: <sip:201 at 10.1.0.65>;tag=ciqhf
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as4bd66aee
Call-ID: 5a4422e21401e157268de2df0efd0db9 at 10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.2
Supported: replaces,norefersub,100rel
Content-Leng- (13 headers 0 lines) ---
Really destroying SIP dialog '5a4422e21401e157268de2df0efd0db9 at
10.1.0.10' Method: OPTIONS
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK67ff02c7;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as762c3fbe
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 6dded6f57df9ec5d12b43e8f30289a26 at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:45:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK67ff02c7;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as762c3fbe
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.bd44
Call-ID: 6dded6f57df9ec5d12b43e8f30289a26 at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '6dded6f57df9ec5d12b43e8f30289a26 at
192.168.1.2' Method: OPTIONS
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->


<------------->
Reliably Transmitting (no NAT) to 10.1.0.65:5060:
OPTIONS sip:201 at 10.1.0.65 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK163239e7;rport
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as05dfd44b
To: <sip:201 at 10.1.0.65>
Contact: <sip:asterisk at 10.1.0.10>
Call-ID: 2d5b750607e46b92088457f43d114595 at 10.1.0.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK163239e7
To: <sip:201 at 10.1.0.65>;tag=oqlta
From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=aID:
2d5b750607e46b92088457f43d114595 at 10.1.0.10
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Server: Twinkle/1.2
Supported: replaces,norefersub,100rel
Content-Length: 0


<------------->
- --- (13 headers 0 lines) ---
Really destroying SIP dialog '2d5b750607e46b92088457f43d114595 at
10.1.0.10' Method: OPTIONS
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5611d4e0;rport
From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as02eb79de
To: <sip:ekiga.net>
Contact: <sip:asterisk at 192.168.1.2>
Call-ID: 5d11d978129dff027fb7a3721ffb0540 at 192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Aug 2009 17:46:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


- ---
alderamin*CLI>
<--- SIP read from 86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK5611d4e0;rport=28490;received=190.51.112.4
From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as02eb79de
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.c991
Call-ID: 5d11d978129dff027fb7a3721ffb0540 at 192.168.1.2
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0


<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '5d11d978129dff027fb7a3721ffb0540 at
192.168.1.2' Method: OPTIONS




Also I made sure to redirect the port 5060 of my router to the firewall. In
this scenery the softphone client is on a workstation with IP 10.1.0.65.
Firewall, that is where at the moment Asterisk is installed, has the LAN IP
10.1.0.10. The firewall interfaces in the network segment of router has IP
192.168.1.2, through which it doing NAT of everything what comes from the
internal network against router.

According to which I see, an answer is being sent to 201 at 192.168.1.2 and
and that would not be correct, since in any case it would have to become to
10.1.0.65. In this situation, how I could correct this?

Thanks in advance for your reply.

Regards,
Daniel

[1] http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net


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SIP

2009-Aug-17 19:02 UTC

head link

[asterisk-users] Accessing to ekiga.net through Asterisk

Daniel,

Check your stunaddr setting. Is it misspelled, or do they really use
stun.exiga.net instead of stun.ekiga.net ?

N.

Daniel Bareiro wrote:> Hi all!
>
> I'm trying to connect to ekiga.net through a client connected to my
> Asterisk server. For it I am being based on this [1] document. Next I
> put the configurations that I am using.
>
> /etc/asterisk/sip.conf:
>
> ; Outgoing to ekiga.net
> [ekiga]
> type=friend
> username=MyUser
> secret=MyPass
> host=ekiga.net
> canreinvite=no
> qualify=300
> nat = yes
> stunaddr = stun.exiga.net
> insecure=port,invite  ; required for incoming ekiga.net calls
>
> /etc/asterisk/extensions.conf:
>
> [from-internal]
> ...
> exten => _8.,1,Dial(SIP/ekiga/${EXTEN:1},20,r))
>
>
> I tried a echo test, dialing in my case to 8500, but in spite of seeing
> traffic towards Internet, nothing is heard. Setting 'sip set
debug', I get
> the following thing:
>
>
> <--- SIP read from 10.1.0.65:5060 --->
> INVITE sip:8500 at 10.1.0.10 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
> Max-Forwards: 70
> To: <sip:8500 at 10.1.0.10>
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 183 INVITE
> Contact: <sip:201 at 10.1.0.65>
> Content-Type: application/sdp
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Supported: replaces,norefersub,100rel
> User-Agent: Twinkle/1.2
> Content-Length: 247
>
> v=0
> o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
> s=-
> c=IN IP4 10.1.0.65
> t=0 0
> m=audio 8000 RTP/AVP 8 0 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> <------------->
> --- (13 headers 12 lines) ---
> Sending to 10.1.0.65 : 5060 (NAT)
> Using INVITE request as basis request -
> mrsyiysrdkwmkeg at defiant.freesoftware.org
>
> <--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 10.1.0.65;branch=z9hG4bKjxcxrrks;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> To: <sip:8500 at 10.1.0.10>;tag=as095989a3
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 183 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
> nonce="76b2dfe8"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog
> 'mrsyiysrdkwmkeg at defiant.freesoftware.org' in 32000 ms (Method:
INVITE)
> Found user '201'
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> ACK sip:8500 at 10.1.0.10 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
> Max-Forwards: 70
> To: <sip:8500 at 10.1.0.10>;tag=as095989a3
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 183 ACK
> User-Agent: Twinkle/1.2
> Content-Length: 0
>
>
> <------------->
> --- (9 headers 0 lines) ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> INVITE sip:8500 at 10.1.0.10 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKoilauqhp
> Max-Forwards: 70
> Proxy-Authorization: Digest
>
username="201",realm="asterisk",nonce="76b2dfe8",uri="sip:8500
at
10.1.0.10",response="d49c0fdf11c9977fcd1fce6a50f445fe",algorithm=MD5
> To: <sip:8500 at 10.1.0.10>
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 INVITE
> Contact: <sip:201 at 10.1.0.65>
> Content-Type: application/sdp
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Supported: replaces,norefersub,100rel
> User-Agent: Twinkle/1.2
> Content-Length: 247
>
> v=0
> o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
> s=-
> c=IN IP4 10.1.0.65
> t=0 0
> m=audio 8000 RTP/AVP 8 0 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> <------------->
> --- (14 headers 12 lines) ---
> Sending to 10.1.0.65 : 5060 (NAT)
> Using INVITE request as basis request -
> mrsyiysrdkwmkeg at defiant.freesoftware.org
> Found user '201'
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 3
> Found RTP audio format 101
> Peer audio RTP is at port 10.1.0.65:8000
> Found audio description format PCMA for ID 8
> Found audio description format PCMU for ID 0
> Found audio description format GSM for ID 3
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe
> (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 10.1.0.65:8000
> Looking for 8500 in from-internal (domain 10.1.0.10)
> list_route: hop: <sip:201 at 10.1.0.65>
>
> <--- Transmitting (no NAT) to 10.1.0.65:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> To: <sip:8500 at 10.1.0.10>
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:8500 at 10.1.0.10>
> Content-Length: 0
>
>
> <------------>
>     -- Executing [8500 at from-internal:1]
Dial("SIP/201-090ffff0",
> "SIP/ekiga/500|20|r)") in new stack
> Video is at 192.168.1.2 port 16080
> Audio is at 192.168.1.2 port 14850
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x40000 (h261) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> INVITE sip:500 at ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4ea46842;rport
> From: "Hector Bareiro" <sip:201 at
192.168.1.2>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:36:38 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 331
>
> v=0
> o=root 4959 4959 IN IP4 192.168.1.2
> s=session
> c=IN IP4 192.168.1.2
> b=CT:384
> t=0 0
> m=audio 14850 RTP/AVP 8 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 16080 RTP/AVP 31
> a=rtpmap:31 H261/90000
> a=sendrecv
>
> ---
>     -- Called ekiga/500
>
> <--- Transmitting (no NAT) to 10.1.0.65:5060 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP
> 10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> To: <sip:8500 at 10.1.0.10>;tag=as1b0c8dab
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:8500 at 10.1.0.10>
> Content-Length: 0
>
>
> <------------>
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK4ea46842;rport=28490;received=190.51.112.4
> From: "Hector Bareiro" <sip:201 at
192.168.1.2:5060>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.9448
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 102 INVITE
> Proxy-Authenticate: Digest realm="ekiga.net",
> nonce="4a899652000008b32e461e66fe61009ff6f4ffd9cb4e4ec2"
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (9 headers 0 lines) ---
> Transmitting (no NAT) to 86.64.162.35:5060:
> ACK sip:500 at ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4ea46842;rport
> From: "Hector Bareiro" <sip:201 at
192.168.1.2>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.9448
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
> Video is at 192.168.1.2 port 16080
> Audio is at 192.168.1.2 port 14850
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x40000 (h261) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> INVITE sip:500 at ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5f88a0aa;rport
> From: "Hector Bareiro" <sip:201 at
192.168.1.2>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Proxy-Authorization: Digest username="danib",
realm="ekiga.net",
> algorithm=MD5, uri="sip:500 at ekiga.net",
> nonce="4a899652000008b32e461e66fe61009ff6f4ffd9cb4e4ec2",
> response="950e5d853e07ad728da8ae8a02198034"
> Date: Mon, 17 Aug 2009 17:36:38 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 331
>
> v=0
> o=root 4959 4960 IN IP4 192.168.1.2
> s=session
> c=IN IP4 192.168.1.2
> b=CT:384
> t=0 0
> m=audio 14850 RTP/AVP 8 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=senon> for address/port to send to
> set_destination: set destination to 86.64.162.35, port 5060
> Transmitting (no NAT) to 86.64.162.35:5060:
> ACK sip:500 at 86.64.162.35:5081 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK15031d34;rport
> Route: <sip:86.64.162.35;lr=on>
> From: "Hector Bareiro" <sip:201 at
192.168.1.2>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>;tag=as1603ca76
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 103 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
>     -- SIP/ekiga-090cb900 answered SIP/201-090ffff0
> Audio is at 10.1.0.10 port 14442
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
>
> <--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> To: <sip:8500 at 10.1.0.10>;tag=as1b0c8dab
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:8500 at 10.1.0.10>
> Content-Type: application/sdp
> Content-Length: 255
>
> v=0
> o=root 4959 4959 IN IP4 10.1.0.10
> s=session
> c=IN IP4 10.1.0.10
> t=0 0
> m=audio 14442 RTP/AVP 8 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> <------------>
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> ACK sip:8500 at 10.1.0.10 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKnovwlzvc
> Max-Forwards: 70
> Proxy-Authorization: Digest
>
username="201",realm="asterisk",nonce="76b2dfe8",uri="sip:8500
at
10.1.0.10",response="d49c0fdf11c9977fcd1fce6a50f445fe",algorithm=MD5
> To: <sip:8500 at 10.1.0.10>;tag=as1b0c8dab
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 ACK
> User-Agent: Twinkle/1.2
> Content-Length: 0
>
>
> <------------->
> --- (10 headers 0 lines) ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK229d0a34;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as53f8b15a
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 6b6b26de041acc9b537b8d716cd18580 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:37:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Sues
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK229d0a34
> To: <sip:201 at 10.1.0.65>;tag=aacln
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as53f8b15a
> Call-ID: 6b6b26de041acc9b537b8d716cd18580 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> --- (13 headers 0 lines) ---
> Really destroying SIP dialog
> '6b6b26de041acc9b537b8d716cd18580 at 10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK10db4ef1;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as2ff24865
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 0a76cf0f0dd60b6855266e3c3105cd55 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:37:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK10db4ef1;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as2ff24865
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.f0c5
> Call-ID: 0a76cf0f0dd60b6855266e3c3105cd55 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '0a76cf0f0dd60b6855266e3c3105cd55 at 192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK77d011fa;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as1d024ca8
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 693813ae7b9c3e783112c4111b851071 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:38:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supportnsmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7c056bcf;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as263b8e2b
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 544df8987fffac657cc726642845c34d at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:38:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK7c056bcf;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as263b8e2b
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d936
> Call-ID: 544df8987fffac657cc726642845c34d at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '544df8987fffac657cc726642845c34d at 192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK07c25ee9;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as587919f0
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 611ac37e0be64f463cdbe4d71ac4c74f at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:39:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK07c25ee9
> To: <sip:201 at 10.1.0.65>;tag=doivz
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as587919f0
> Call-ID: 611ac37e0be64f463cdbe4d71ac4c74f at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supporaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> --- (13 headers 0 lines) ---
> Really destroying SIP dialog
> '611ac37e0be64f463cdbe4d71ac4c74f at 10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK16aa33c2;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as361d1f0a
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 7e91ccff491d8a9d7856928c4c4f43e6 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:39:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK16aa33c2;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as361d1f0a
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1b34
> Call-ID: 7e91ccff491d8a9d7856928c4c4f43e6 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '7e91ccff491d8a9d7856928c4c4f43e6 at 192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK24a7bc95;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as5fa47440
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 49071c5656ab2a31252152a455139b09 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:40:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK24a7bc95
> To: <sip:201 at 10.1.0.65>;tag=sgply
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as5fa47440
> Call-ID: 49071c5656ab2a31252152a455139b09 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Suppoorefersub,100rel
> Content-Length: 0
>
>
> <------------->
> --- (13 headers 0 lines) ---
> Really destroying SIP dialog
> '49071c5656ab2a31252152a455139b09 at 10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK13007a5c;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as5a0acdaf
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 3a193cd018c6a354017a0d501e7c59f8 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:40:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK13007a5c;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as5a0acdaf
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.2e90
> Call-ID: 3a193cd018c6a354017a0d501e7c59f8 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '3a193cd018c6a354017a0d501e7c59f8 at 192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK10cced95;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as076647df
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 7fc191e11f2509e7353b61d65b76f002 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:41:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK10cced95
> To: <sip:201 at 10.1.0.65>;tag=owawm
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as076647df
> Call-ID: 7fc191e11f2509e7353b61d65b76f002 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> ers 0 lines) ---
> Really destroying SIP dialog
> '7fc191e11f2509e7353b61d65b76f002 at 10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3a587968;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as5139b49b
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 3f6b1291297e4eb20a2d65c662a67fb7 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:41:29 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK3a587968;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as5139b49b
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.14a8
> Call-ID: 3f6b1291297e4eb20a2d65c662a67fb7 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '3f6b1291297e4eb20a2d65c662a67fb7 at 192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK1c1f607a;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as416ac6cc
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 5ba8e26e43162c6b3c56b7787273d682 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:42:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK1c1f607a
> To: <sip:201 at 10.1.0.65>;tag=hplvm
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as416ac6cc
> Call-ID: 5ba8e26e43162c6b3c56b7787273d682 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,ESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> --- (13 headers 0 lines) ---
> Really destroying SIP dialog
> '5ba8e26e43162c6b3c56b7787273d682 at 10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK029714e0;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as686f2ada
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 6ca487dc5c8f40ab17428b3c76e6ff7c at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:42:29 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK029714e0;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as686f2ada
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.a004
> Call-ID: 6ca487dc5c8f40ab17428b3c76e6ff7c at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '6ca487dc5c8f40ab17428b3c76e6ff7c at 192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK4e32a4be;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as1d745b97
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 139126a231a61ca664de02153ee8cfc4 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:43:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK4e32a4be
> To: <sip:201 at 10.1.0.65>;tag=cvydb
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as1d745b97
> Call-ID: 139126a231a61ca664de02153ee8cfc4 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefe
> <------------->
> --- (13 headers 0 lines) ---
> Really destroying SIP dialog
> '139126a231a61ca664de02153ee8cfc4 at 10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3513c9e8;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as348ceda1
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 3076a48850ae43bb5fd072736736ba52 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:43:29 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK3513c9e8;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as348ceda1
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.dde9
> Call-ID: 3076a48850ae43bb5fd072736736ba52 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '3076a48850ae43bb5fd072736736ba52 at 192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK0fd89b0f;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as204361ce
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 3418d0df540794014b7707011cb0bb9d at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:44:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK0fd89b0f
> To: <sip:201 at 10.1.0.65>;tag=kwkmu
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as204361ce
> Call-ID: 3418d0df540794014b7707011cb0bb9d at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> --- (13 headers 0 lines) ---
> Really P dialog '3418d0df540794014b7707011cb0bb9d at 10.1.0.10'
Method:
> OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK32b3f65d;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as177de4d9
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 04607ade51978546773a635538a52a21 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:44:29 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK32b3f65d;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as177de4d9
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.75a7
> Call-ID: 04607ade51978546773a635538a52a21 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '04607ade51978546773a635538a52a21 at 192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK28825321;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as4bd66aee
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 5a4422e21401e157268de2df0efd0db9 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:45:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK28825321
> To: <sip:201 at 10.1.0.65>;tag=ciqhf
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as4bd66aee
> Call-ID: 5a4422e21401e157268de2df0efd0db9 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Leng- (13 headers 0 lines) ---
> Really destroying SIP dialog
> '5a4422e21401e157268de2df0efd0db9 at 10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK67ff02c7;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as762c3fbe
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 6dded6f57df9ec5d12b43e8f30289a26 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:45:30 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK67ff02c7;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as762c3fbe
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.bd44
> Call-ID: 6dded6f57df9ec5d12b43e8f30289a26 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '6dded6f57df9ec5d12b43e8f30289a26 at 192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK163239e7;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as05dfd44b
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 2d5b750607e46b92088457f43d114595 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:46:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK163239e7
> To: <sip:201 at 10.1.0.65>;tag=oqlta
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=aID:
> 2d5b750607e46b92088457f43d114595 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> --- (13 headers 0 lines) ---
> Really destroying SIP dialog
> '2d5b750607e46b92088457f43d114595 at 10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5611d4e0;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as02eb79de
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 5d11d978129dff027fb7a3721ffb0540 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:46:30 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK5611d4e0;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as02eb79de
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.c991
> Call-ID: 5d11d978129dff027fb7a3721ffb0540 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '5d11d978129dff027fb7a3721ffb0540 at 192.168.1.2' Method: OPTIONS
>
>
>
>
> Also I made sure to redirect the port 5060 of my router to the
> firewall. In
> this scenery the softphone client is on a workstation with IP 10.1.0.65.
> Firewall, that is where at the moment Asterisk is installed, has the
> LAN IP
> 10.1.0.10. The firewall interfaces in the network segment of router has IP
> 192.168.1.2, through which it doing NAT of everything what comes from the
> internal network against router.
>
> According to which I see, an answer is being sent to 201 at 192.168.1.2 and
> and that would not be correct, since in any case it would have to
> become to
> 10.1.0.65. In this situation, how I could correct this?
>
> Thanks in advance for your reply.
>
> Regards,
> Daniel
>
> [1] http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net
>
>
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Patrick Plattes

2009-Aug-18 10:12 UTC

head link

[asterisk-users] Accessing to ekiga.net through Asterisk

hi,

stunaddr = stun.exiga.net looks wrong ^^

in generally it looks like a nat problem.

bye,
 patrick

On Mon, Aug 17, 2009 at 8:12 PM, Daniel Bareiro<daniel-listas at gmx.net>
wrote:> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi all!
>
> I'm trying to connect to ekiga.net through a client connected to my
> Asterisk server. For it I am being based on this [1] document. Next I
> put the configurations that I am using.
>
> /etc/asterisk/sip.conf:
>
> ; Outgoing to ekiga.net
> [ekiga]
> type=friend
> username=MyUser
> secret=MyPass
> host=ekiga.net
> canreinvite=no
> qualify=300
> nat = yes
> stunaddr = stun.exiga.net
> insecure=port,invite ?; required for incoming ekiga.net calls
>
> /etc/asterisk/extensions.conf:
>
> [from-internal]
> ...
> exten => _8.,1,Dial(SIP/ekiga/${EXTEN:1},20,r))
>
>
> I tried a echo test, dialing in my case to 8500, but in spite of seeing
> traffic towards Internet, nothing is heard. Setting 'sip set
debug', I get
> the following thing:
>
>
> <--- SIP read from 10.1.0.65:5060 --->
> INVITE sip:8500 at 10.1.0.10 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
> Max-Forwards: 70
> To: <sip:8500 at 10.1.0.10>
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 183 INVITE
> Contact: <sip:201 at 10.1.0.65>
> Content-Type: application/sdp
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Supported: replaces,norefersub,100rel
> User-Agent: Twinkle/1.2
> Content-Length: 247
>
> v=0
> o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
> s=-
> c=IN IP4 10.1.0.65
> t=0 0
> m=audio 8000 RTP/AVP 8 0 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> <------------->
> - --- (13 headers 12 lines) ---
> Sending to 10.1.0.65 : 5060 (NAT)
> Using INVITE request as basis request - mrsyiysrdkwmkeg at
defiant.freesoftware.org
>
> <--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKjxcxrrks;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> To: <sip:8500 at 10.1.0.10>;tag=as095989a3
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 183 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="76b2dfe8"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog 'mrsyiysrdkwmkeg at
defiant.freesoftware.org' in 32000 ms (Method: INVITE)
> Found user '201'
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> ACK sip:8500 at 10.1.0.10 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
> Max-Forwards: 70
> To: <sip:8500 at 10.1.0.10>;tag=as095989a3
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 183 ACK
> User-Agent: Twinkle/1.2
> Content-Length: 0
>
>
> <------------->
> - --- (9 headers 0 lines) ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> INVITE sip:8500 at 10.1.0.10 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKoilauqhp
> Max-Forwards: 70
> Proxy-Authorization: Digest
username="201",realm="asterisk",nonce="76b2dfe8",uri="sip:8500
at
10.1.0.10",response="d49c0fdf11c9977fcd1fce6a50f445fe",algorithm=MD5
> To: <sip:8500 at 10.1.0.10>
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 INVITE
> Contact: <sip:201 at 10.1.0.65>
> Content-Type: application/sdp
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Supported: replaces,norefersub,100rel
> User-Agent: Twinkle/1.2
> Content-Length: 247
>
> v=0
> o=twinkle 2122879389 441437466 IN IP4 10.1.0.65
> s=-
> c=IN IP4 10.1.0.65
> t=0 0
> m=audio 8000 RTP/AVP 8 0 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> <------------->
> - --- (14 headers 12 lines) ---
> Sending to 10.1.0.65 : 5060 (NAT)
> Using INVITE request as basis request - mrsyiysrdkwmkeg at
defiant.freesoftware.org
> Found user '201'
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 3
> Found RTP audio format 101
> Peer audio RTP is at port 10.1.0.65:8000
> Found audio description format PCMA for ID 8
> Found audio description format PCMU for ID 0
> Found audio description format GSM for ID 3
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 10.1.0.65:8000
> Looking for 8500 in from-internal (domain 10.1.0.10)
> list_route: hop: <sip:201 at 10.1.0.65>
>
> <--- Transmitting (no NAT) to 10.1.0.65:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> To: <sip:8500 at 10.1.0.10>
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:8500 at 10.1.0.10>
> Content-Length: 0
>
>
> <------------>
> ? ?-- Executing [8500 at from-internal:1]
Dial("SIP/201-090ffff0", "SIP/ekiga/500|20|r)") in new stack
> Video is at 192.168.1.2 port 16080
> Audio is at 192.168.1.2 port 14850
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x40000 (h261) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> INVITE sip:500 at ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4ea46842;rport
> From: "Hector Bareiro" <sip:201 at
192.168.1.2>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:36:38 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 331
>
> v=0
> o=root 4959 4959 IN IP4 192.168.1.2
> s=session
> c=IN IP4 192.168.1.2
> b=CT:384
> t=0 0
> m=audio 14850 RTP/AVP 8 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 16080 RTP/AVP 31
> a=rtpmap:31 H261/90000
> a=sendrecv
>
> - ---
> ? ?-- Called ekiga/500
>
> <--- Transmitting (no NAT) to 10.1.0.65:5060 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> To: <sip:8500 at 10.1.0.10>;tag=as1b0c8dab
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:8500 at 10.1.0.10>
> Content-Length: 0
>
>
> <------------>
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK4ea46842;rport=28490;received=190.51.112.4
> From: "Hector Bareiro" <sip:201 at
192.168.1.2:5060>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.9448
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 102 INVITE
> Proxy-Authenticate: Digest realm="ekiga.net",
nonce="4a899652000008b32e461e66fe61009ff6f4ffd9cb4e4ec2"
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (9 headers 0 lines) ---
> Transmitting (no NAT) to 86.64.162.35:5060:
> ACK sip:500 at ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4ea46842;rport
> From: "Hector Bareiro" <sip:201 at
192.168.1.2>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.9448
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> - ---
> Video is at 192.168.1.2 port 16080
> Audio is at 192.168.1.2 port 14850
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x40000 (h261) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> INVITE sip:500 at ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5f88a0aa;rport
> From: "Hector Bareiro" <sip:201 at
192.168.1.2>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Proxy-Authorization: Digest username="danib",
realm="ekiga.net", algorithm=MD5, uri="sip:500 at
ekiga.net",
nonce="4a899652000008b32e461e66fe61009ff6f4ffd9cb4e4ec2",
> response="950e5d853e07ad728da8ae8a02198034"
> Date: Mon, 17 Aug 2009 17:36:38 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 331
>
> v=0
> o=root 4959 4960 IN IP4 192.168.1.2
> s=session
> c=IN IP4 192.168.1.2
> b=CT:384
> t=0 0
> m=audio 14850 RTP/AVP 8 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=senon> for address/port to send to
> set_destination: set destination to 86.64.162.35, port 5060
> Transmitting (no NAT) to 86.64.162.35:5060:
> ACK sip:500 at 86.64.162.35:5081 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK15031d34;rport
> Route: <sip:86.64.162.35;lr=on>
> From: "Hector Bareiro" <sip:201 at
192.168.1.2>;tag=as2bb1b3cd
> To: <sip:500 at ekiga.net>;tag=as1603ca76
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 2acb8bc830f595915de8e2774ca882ae at 192.168.1.2
> CSeq: 103 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> - ---
> ? ?-- SIP/ekiga-090cb900 answered SIP/201-090ffff0
> Audio is at 10.1.0.10 port 14442
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
>
> <--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
10.1.0.65;branch=z9hG4bKoilauqhp;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> To: <sip:8500 at 10.1.0.10>;tag=as1b0c8dab
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:8500 at 10.1.0.10>
> Content-Type: application/sdp
> Content-Length: 255
>
> v=0
> o=root 4959 4959 IN IP4 10.1.0.10
> s=session
> c=IN IP4 10.1.0.10
> t=0 0
> m=audio 14442 RTP/AVP 8 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> <------------>
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> ACK sip:8500 at 10.1.0.10 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKnovwlzvc
> Max-Forwards: 70
> Proxy-Authorization: Digest
username="201",realm="asterisk",nonce="76b2dfe8",uri="sip:8500
at
10.1.0.10",response="d49c0fdf11c9977fcd1fce6a50f445fe",algorithm=MD5
> To: <sip:8500 at 10.1.0.10>;tag=as1b0c8dab
> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz
> Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org
> CSeq: 184 ACK
> User-Agent: Twinkle/1.2
> Content-Length: 0
>
>
> <------------->
> - --- (10 headers 0 lines) ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK229d0a34;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as53f8b15a
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 6b6b26de041acc9b537b8d716cd18580 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:37:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Sues
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK229d0a34
> To: <sip:201 at 10.1.0.65>;tag=aacln
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as53f8b15a
> Call-ID: 6b6b26de041acc9b537b8d716cd18580 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> - --- (13 headers 0 lines) ---
> Really destroying SIP dialog '6b6b26de041acc9b537b8d716cd18580 at
10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK10db4ef1;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as2ff24865
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 0a76cf0f0dd60b6855266e3c3105cd55 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:37:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK10db4ef1;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as2ff24865
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.f0c5
> Call-ID: 0a76cf0f0dd60b6855266e3c3105cd55 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '0a76cf0f0dd60b6855266e3c3105cd55 at
192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK77d011fa;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as1d024ca8
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 693813ae7b9c3e783112c4111b851071 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:38:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supportnsmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK7c056bcf;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as263b8e2b
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 544df8987fffac657cc726642845c34d at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:38:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK7c056bcf;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as263b8e2b
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d936
> Call-ID: 544df8987fffac657cc726642845c34d at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '544df8987fffac657cc726642845c34d at
192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK07c25ee9;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as587919f0
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 611ac37e0be64f463cdbe4d71ac4c74f at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:39:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK07c25ee9
> To: <sip:201 at 10.1.0.65>;tag=doivz
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as587919f0
> Call-ID: 611ac37e0be64f463cdbe4d71ac4c74f at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supporaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> - --- (13 headers 0 lines) ---
> Really destroying SIP dialog '611ac37e0be64f463cdbe4d71ac4c74f at
10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK16aa33c2;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as361d1f0a
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 7e91ccff491d8a9d7856928c4c4f43e6 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:39:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK16aa33c2;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as361d1f0a
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1b34
> Call-ID: 7e91ccff491d8a9d7856928c4c4f43e6 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '7e91ccff491d8a9d7856928c4c4f43e6 at
192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK24a7bc95;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as5fa47440
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 49071c5656ab2a31252152a455139b09 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:40:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK24a7bc95
> To: <sip:201 at 10.1.0.65>;tag=sgply
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as5fa47440
> Call-ID: 49071c5656ab2a31252152a455139b09 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Suppoorefersub,100rel
> Content-Length: 0
>
>
> <------------->
> - --- (13 headers 0 lines) ---
> Really destroying SIP dialog '49071c5656ab2a31252152a455139b09 at
10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK13007a5c;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as5a0acdaf
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 3a193cd018c6a354017a0d501e7c59f8 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:40:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK13007a5c;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as5a0acdaf
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.2e90
> Call-ID: 3a193cd018c6a354017a0d501e7c59f8 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '3a193cd018c6a354017a0d501e7c59f8 at
192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK10cced95;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as076647df
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 7fc191e11f2509e7353b61d65b76f002 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:41:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK10cced95
> To: <sip:201 at 10.1.0.65>;tag=owawm
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as076647df
> Call-ID: 7fc191e11f2509e7353b61d65b76f002 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> ers 0 lines) ---
> Really destroying SIP dialog '7fc191e11f2509e7353b61d65b76f002 at
10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3a587968;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as5139b49b
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 3f6b1291297e4eb20a2d65c662a67fb7 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:41:29 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK3a587968;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as5139b49b
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.14a8
> Call-ID: 3f6b1291297e4eb20a2d65c662a67fb7 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '3f6b1291297e4eb20a2d65c662a67fb7 at
192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK1c1f607a;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as416ac6cc
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 5ba8e26e43162c6b3c56b7787273d682 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:42:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK1c1f607a
> To: <sip:201 at 10.1.0.65>;tag=hplvm
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as416ac6cc
> Call-ID: 5ba8e26e43162c6b3c56b7787273d682 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,ESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> - --- (13 headers 0 lines) ---
> Really destroying SIP dialog '5ba8e26e43162c6b3c56b7787273d682 at
10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK029714e0;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as686f2ada
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 6ca487dc5c8f40ab17428b3c76e6ff7c at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:42:29 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK029714e0;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as686f2ada
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.a004
> Call-ID: 6ca487dc5c8f40ab17428b3c76e6ff7c at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '6ca487dc5c8f40ab17428b3c76e6ff7c at
192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK4e32a4be;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as1d745b97
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 139126a231a61ca664de02153ee8cfc4 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:43:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK4e32a4be
> To: <sip:201 at 10.1.0.65>;tag=cvydb
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as1d745b97
> Call-ID: 139126a231a61ca664de02153ee8cfc4 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefe
> <------------->
> - --- (13 headers 0 lines) ---
> Really destroying SIP dialog '139126a231a61ca664de02153ee8cfc4 at
10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3513c9e8;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as348ceda1
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 3076a48850ae43bb5fd072736736ba52 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:43:29 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK3513c9e8;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as348ceda1
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.dde9
> Call-ID: 3076a48850ae43bb5fd072736736ba52 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '3076a48850ae43bb5fd072736736ba52 at
192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK0fd89b0f;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as204361ce
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 3418d0df540794014b7707011cb0bb9d at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:44:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK0fd89b0f
> To: <sip:201 at 10.1.0.65>;tag=kwkmu
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as204361ce
> Call-ID: 3418d0df540794014b7707011cb0bb9d at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> - --- (13 headers 0 lines) ---
> Really P dialog '3418d0df540794014b7707011cb0bb9d at 10.1.0.10'
Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK32b3f65d;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as177de4d9
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 04607ade51978546773a635538a52a21 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:44:29 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK32b3f65d;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as177de4d9
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.75a7
> Call-ID: 04607ade51978546773a635538a52a21 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '04607ade51978546773a635538a52a21 at
192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK28825321;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as4bd66aee
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 5a4422e21401e157268de2df0efd0db9 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:45:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK28825321
> To: <sip:201 at 10.1.0.65>;tag=ciqhf
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as4bd66aee
> Call-ID: 5a4422e21401e157268de2df0efd0db9 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Leng- (13 headers 0 lines) ---
> Really destroying SIP dialog '5a4422e21401e157268de2df0efd0db9 at
10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK67ff02c7;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as762c3fbe
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 6dded6f57df9ec5d12b43e8f30289a26 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:45:30 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK67ff02c7;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as762c3fbe
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.bd44
> Call-ID: 6dded6f57df9ec5d12b43e8f30289a26 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '6dded6f57df9ec5d12b43e8f30289a26 at
192.168.1.2' Method: OPTIONS
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
>
>
> <------------->
> Reliably Transmitting (no NAT) to 10.1.0.65:5060:
> OPTIONS sip:201 at 10.1.0.65 SIP/2.0
> Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK163239e7;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as05dfd44b
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 2d5b750607e46b92088457f43d114595 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:46:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 10.1.0.65:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK163239e7
> To: <sip:201 at 10.1.0.65>;tag=oqlta
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=aID:
2d5b750607e46b92088457f43d114595 at 10.1.0.10
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Encoding: identity
> Accept-Language: en
> Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> Server: Twinkle/1.2
> Supported: replaces,norefersub,100rel
> Content-Length: 0
>
>
> <------------->
> - --- (13 headers 0 lines) ---
> Really destroying SIP dialog '2d5b750607e46b92088457f43d114595 at
10.1.0.10' Method: OPTIONS
> Reliably Transmitting (no NAT) to 86.64.162.35:5060:
> OPTIONS sip:ekiga.net SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5611d4e0;rport
> From: "asterisk" <sip:asterisk at
192.168.1.2>;tag=as02eb79de
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 5d11d978129dff027fb7a3721ffb0540 at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 17:46:30 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
192.168.1.2:5060;branch=z9hG4bK5611d4e0;rport=28490;received=190.51.112.4
> From: "asterisk" <sip:asterisk at
192.168.1.2:5060>;tag=as02eb79de
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.c991
> Call-ID: 5d11d978129dff027fb7a3721ffb0540 at 192.168.1.2
> CSeq: 102 OPTIONS
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> Really destroying SIP dialog '5d11d978129dff027fb7a3721ffb0540 at
192.168.1.2' Method: OPTIONS
>
>
>
>
> Also I made sure to redirect the port 5060 of my router to the firewall. In
> this scenery the softphone client is on a workstation with IP 10.1.0.65.
> Firewall, that is where at the moment Asterisk is installed, has the LAN IP
> 10.1.0.10. The firewall interfaces in the network segment of router has IP
> 192.168.1.2, through which it doing NAT of everything what comes from the
> internal network against router.
>
> According to which I see, an answer is being sent to 201 at 192.168.1.2 and
> and that would not be correct, since in any case it would have to become to
> 10.1.0.65. In this situation, how I could correct this?
>
> Thanks in advance for your reply.
>
> Regards,
> Daniel
>
> [1] http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.9 (GNU/Linux)
>
> iEYEARECAAYFAkqJnbAACgkQZpa/GxTmHTfnJgCeOKEq67+SlYwfN8DrPaTEkEyz
> kHsAoI31aNLNfNRjH7bKJdJypB0VVrO7
> =Ymjj
> -----END PGP SIGNATURE-----
>
>
> _______________________________________________
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