-----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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: <sip:[email protected]>
From: "Hector" <sip:[email protected]>;tag=uucwz
Call-ID: [email protected]
CSeq: 183 INVITE
Contact: <sip:[email protected]>
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 - [email protected]
<--- 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:[email protected]>;tag=uucwz
To: <sip:[email protected]>;tag=as095989a3
Call-ID: [email protected]
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 '[email protected]'
in 32000 ms (Method: INVITE)
Found user '201'
alderamin*CLI>
<--- SIP read from 10.1.0.65:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: <sip:[email protected]>;tag=as095989a3
From: "Hector" <sip:[email protected]>;tag=uucwz
Call-ID: [email protected]
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:[email protected] 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:[email protected]",response="d49c0fdf11c9977fcd1fce6a50f445fe",algorithm=MD5
To: <sip:[email protected]>
From: "Hector" <sip:[email protected]>;tag=uucwz
Call-ID: [email protected]
CSeq: 184 INVITE
Contact: <sip:[email protected]>
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 - [email protected]
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:[email protected]>
<--- 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:[email protected]>;tag=uucwz
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 184 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
-- Executing [8...@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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4ea46842;rport
From: "Hector Bareiro" <sip:[email protected]>;tag=as2bb1b3cd
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=uucwz
To: <sip:[email protected]>;tag=as1b0c8dab
Call-ID: [email protected]
CSeq: 184 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
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:[email protected]:5060>;tag=as2bb1b3cd
To: <sip:[email protected]>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.9448
Call-ID: [email protected]
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4ea46842;rport
From: "Hector Bareiro" <sip:[email protected]>;tag=as2bb1b3cd
To: <sip:[email protected]>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.9448
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5f88a0aa;rport
From: "Hector Bareiro" <sip:[email protected]>;tag=as2bb1b3cd
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="danib", realm="ekiga.net", algorithm=MD5,
uri="sip:[email protected]",
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:[email protected]: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:[email protected]>;tag=as2bb1b3cd
To: <sip:[email protected]>;tag=as1603ca76
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=uucwz
To: <sip:[email protected]>;tag=as1b0c8dab
Call-ID: [email protected]
CSeq: 184 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
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:[email protected] 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:[email protected]",response="d49c0fdf11c9977fcd1fce6a50f445fe",algorithm=MD5
To: <sip:[email protected]>;tag=as1b0c8dab
From: "Hector" <sip:[email protected]>;tag=uucwz
Call-ID: [email protected]
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK229d0a34;rport
From: "asterisk" <sip:[email protected]>;tag=as53f8b15a
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=aacln
From: "asterisk" <sip:[email protected]>;tag=as53f8b15a
Call-ID: [email protected]
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 '[email protected]'
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:[email protected]>;tag=as2ff24865
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as2ff24865
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.f0c5
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK77d011fa;rport
From: "asterisk" <sip:[email protected]>;tag=as1d024ca8
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=as263b8e2b
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as263b8e2b
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d936
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK07c25ee9;rport
From: "asterisk" <sip:[email protected]>;tag=as587919f0
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=doivz
From: "asterisk" <sip:[email protected]>;tag=as587919f0
Call-ID: [email protected]
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 '[email protected]'
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:[email protected]>;tag=as361d1f0a
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as361d1f0a
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1b34
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK24a7bc95;rport
From: "asterisk" <sip:[email protected]>;tag=as5fa47440
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=sgply
From: "asterisk" <sip:[email protected]>;tag=as5fa47440
Call-ID: [email protected]
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 '[email protected]'
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:[email protected]>;tag=as5a0acdaf
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as5a0acdaf
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.2e90
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK10cced95;rport
From: "asterisk" <sip:[email protected]>;tag=as076647df
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=owawm
From: "asterisk" <sip:[email protected]>;tag=as076647df
Call-ID: [email protected]
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 '[email protected]'
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:[email protected]>;tag=as5139b49b
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as5139b49b
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.14a8
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK1c1f607a;rport
From: "asterisk" <sip:[email protected]>;tag=as416ac6cc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=hplvm
From: "asterisk" <sip:[email protected]>;tag=as416ac6cc
Call-ID: [email protected]
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 '[email protected]'
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:[email protected]>;tag=as686f2ada
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as686f2ada
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.a004
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK4e32a4be;rport
From: "asterisk" <sip:[email protected]>;tag=as1d745b97
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=cvydb
From: "asterisk" <sip:[email protected]>;tag=as1d745b97
Call-ID: [email protected]
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 '[email protected]'
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:[email protected]>;tag=as348ceda1
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as348ceda1
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.dde9
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK0fd89b0f;rport
From: "asterisk" <sip:[email protected]>;tag=as204361ce
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=kwkmu
From: "asterisk" <sip:[email protected]>;tag=as204361ce
Call-ID: [email protected]
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 '[email protected]' 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:[email protected]>;tag=as177de4d9
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as177de4d9
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.75a7
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK28825321;rport
From: "asterisk" <sip:[email protected]>;tag=as4bd66aee
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=ciqhf
From: "asterisk" <sip:[email protected]>;tag=as4bd66aee
Call-ID: [email protected]
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 '[email protected]'
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:[email protected]>;tag=as762c3fbe
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as762c3fbe
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.bd44
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK163239e7;rport
From: "asterisk" <sip:[email protected]>;tag=as05dfd44b
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=oqlta
From: "asterisk" <sip:[email protected]>;tag=aID:
[email protected]
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 '[email protected]'
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:[email protected]>;tag=as02eb79de
To: <sip:ekiga.net>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5060>;tag=as02eb79de
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.c991
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Kamailio (1.4.0-notls (i386/linux))
Content-Length: 0
<------------->
- --- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
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 [email protected] 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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