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

Tue Aug 18 05:12:51 CDT 2009

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
>
>
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> =Ymjj
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>
>
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