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

Wed Aug 19 06:04:17 CDT 2009

Daniel,

I'm a little confused as to what I'm seeing here. You're bounding 
through two RFC1918 address networks -- 10.1.0.X and 192.168.2.X.   Is 
this some sort of dual NAT scenario?

Perhaps if you can explain a little more about your network setup.

N.

Daniel Bareiro wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> SIP wrote:
>
>   
>> Daniel,
>>     
>
> Hi SIP.
>
>   
>> Check your stunaddr setting. Is it misspelled, or do they really use
>> stun.exiga.net instead of stun.ekiga.net ?
>>     
>
> Thanks to indicate that error to me. I doing the test again. I don't
> believe that this solves what I commented before about 192.168.1.2
> direction, but, just in case, I copy the output of debugging when trying
> to communicate to ekiga.net. The problem continues persisting after the
> correction.
>
> 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=z9hG4bKrqslryke
> Max-Forwards: 70
> To: <sip:8500 at 10.1.0.10>
> From: "Hector" <sip:201 at 10.1.0.10>;tag=typwm
> Call-ID: kafgeaflkmsdgij at defiant.freesoftware.org
> CSeq: 709 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 933572867 1938524932 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 -
> kafgeaflkmsdgij 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=z9hG4bKrqslryke;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=typwm
> To: <sip:8500 at 10.1.0.10>;tag=as0a3a462b
> Call-ID: kafgeaflkmsdgij at defiant.freesoftware.org
> CSeq: 709 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
> nonce="497d879d"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog
> 'kafgeaflkmsdgij 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=z9hG4bKrqslryke
> Max-Forwards: 70
> To: <sip:8500 at 10.1.0.10>;tag=as0a3a462b
> From: "Hector" <sip:201 at 10.1.0.10>;tag=typwm
> Call-ID: kafgeaflkmsdgij at defiant.freesoftware.org
> CSeq: 709 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=z9hG4bKxpraybjr
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="201",realm="asterisk",nonce="497d879d",uri="sip:8500 at 10.1.0.10",response="9cb53107d4d15b7a2e7df8599e851b80",algorithm=MD5
> To: <sip:8500 at 10.1.0.10>
> From: "Hector" <sip:201 at 10.1.0.10>;tag=typwm
> Call-ID: kafgeaflkmsdgij at defiant.freesoftware.org
> CSeq: 710 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 933572867 1938524932 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 -
> kafgeaflkmsdgij 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=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=typwm
> To: <sip:8500 at 10.1.0.10>
> Call-ID: kafgeaflkmsdgij at defiant.freesoftware.org
> CSeq: 710 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 10112
> Audio is at 192.168.1.2 port 12592
> 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 (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=z9hG4bK651d88ba;rport
> From: "Hector Bareiro" <sip:201 at 192.168.1.2>;tag=as7bab61b8
> To: <sip:500 at ekiga.net>
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 6149754405a8a52d5cae9ad92c8131eb at 192.168.1.2
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 21:30:25 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 12592 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 10112 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=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060
> From: "Hector" <sip:201 at 10.1.0.10>;tag=typwm
> To: <sip:8500 at 10.1.0.10>;tag=as37d19c71
> Call-ID: kafgeaflkmsdgij at defiant.freesoftware.org
> CSeq: 710 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=z9hG4bK651d88ba;rport=10003;received=190.51.105.123
> From: "Hector Bareiro" <sip:201 at 192.168.1.2:5060>;tag=as7bab61b8
> To: <sip:500 at ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1918
> Call-ID: 6149754405a8a52d5cae9ad92c8131eb at 192.168.1.2
> CSeq: 102 INVITE
> Proxy-Authenticate: Digest realm="ekiga.net",
> nonce="4a89cd1d00000360086ff884818a5d318b81c0d065d2743f"
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (9 headers 0 lines) ---
> Transmitting (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=z9hG4bK651d88ba;rport
> From: "Hector Bareiro" <sip:201 at 192.168.1.2>;tag=as7bab61b8
> To: <sip:500 at ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1918
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 6149754405a8a52d5cae9ad92c8131eb 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 10112
> Audio is at 192.168.1.2 port 12592
> 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 (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=z9hG4bK6e8ff8ba;rport
> From: "Hector Bareiro" <sip:201 at 192.168.1.2>;tag=as7bab61b8
> To: <sip:500 at ekiga.net>
> Contact: <sip:201 at 192.168.1.2>
> Call-ID: 6149754405a8a52d5cae9ad92c8131eb 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="4a89cd1d00000360086ff884818a5d318b81c0d065d2743f",
> response="152416b836f298095455859a7c3f1696"
> Date: Mon, 17 Aug 2009ideo 10112 RTP/AVP 31
> a=rtpmap:31 H261/90000
> a=sendrecv
>
> - ---
> alderamin*CLI>
> <--- SIP read from 86.64.162.35:5060 --->
> SIP/2.0 100 Giving a try
> Via: SIP/2.0/UDP
> 192.168.1.2:5060;branch=z9hG4bK6e8ff8ba;rport=10003;received=190.51.105.123
> From: "Hector Bareiro" <sip:201 at 192.168.1.2:5060>;tag=as7bab61b8
> To: <sip:500 at ekiga.net>
> Call-ID: 6149754405a8a52d5cae9ad92c8131eb at 192.168.1.2
> CSeq: 103 INVITE
> Server: Kamailio (1.4.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> - --- (8 headers 0 lines) ---
> 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;received=190.51.105.123;branch=z9hG4bK6e8ff8ba;rport=10003
> Record-Route: <sip:86.64.162.35;lr=on>
> From: "Hector Bareiro" <sip:201 at 192.168.1.2:5060>;tag=as7bab61b8
> To: <sip:500 at ekiga.net>;tag=as38bf28ad
> Call-ID: 6149754405a8a52d5cae9ad92c8131eb at 192.168.1.2
> CSeq: 103 INVITE
> User-Agent: Ekiga.NET
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:500 at 86.64.162.35:5081>
> Content-Type: application/sdp
> Content-Length: 310
>
> v=0
> o=root 9963 9963 IN IP4 86.64.162.35
> s=session
> c=IN IP4 86.64.162.35
> b=CT:384
> t=0 0
> m=audio 10400 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 14488 RTP/AVP 31
> a=rtpmap:31 H261/90000
> a=sendrecv
>
> <------------->
> - --- (13 headers 16 lines) ---
> Found RTP audio format 8
> Found RTP audio format 101
> Found RTP video format 31
> Peer audio RTP is at port 86.64.162.35:10400
> Found audio description format PCMA for ID 8
> Found audio description format telephone-event for ID 101
> Found video description format H261 for ID 31
> Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0x40008
> (alaw|
> h261)/video=0x40000 (h261), combined - 0x40008 (alaw|h261)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 86.64.162.35:10400
> Peer video RTP is at port 86.64.162.35:14488
> list_route: hop: <sip:86.64.162.35;lwered SIP/201-090ffff0
> Audio is at 10.1.0.10 port 12994
> 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=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060
> rom: "Hector" <sip:201 at 10.1.0.10>;tag=typwm
> To: <sip:8500 at 10.1.0.10>;tag=as37d19c71
> Call-ID: kafgeaflkmsdgij at defiant.freesoftware.org
> CSeq: 710 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 12994 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=z9hG4bKpnwhfosw
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="201",realm="asterisk",nonce="497d879d",uri="sip:8500 at 10.1.0.10",response="9cb53107d4d15b7a2e7df8599e851b80",algorithm=MD5
> To: <sip:8500 at 10.1.0.10>;tag=as37d19c71
> From: "Hector" <sip:201 at 10.1.0.10>;tag=typwm
> Call-ID: kafgeaflkmsdgij at defiant.freesoftware.org
> CSeq: 710 ACK
> User-Agent: Twinkle/1.2
> Content-Length: 0
>
>
> <------------->
> - --- (10 headers 0 lines) ---
> 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=z9hG4bK3c1a71ce;rport
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as51f657b5
> To: <sip:201 at 10.1.0.65>
> Contact: <sip:asterisk at 10.1.0.10>
> Call-ID: 2096f1ac21f419aa029d7ccb5d8de044 at 10.1.0.10
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 21:30:29 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/UD=5060;branch=z9hG4bK3c1a71ce
> To: <sip:201 at 10.1.0.65>;tag=pecxh
> From: "asterisk" <sip:asterisk at 10.1.0.10>;tag=as51f657b5
> Call-ID: 2096f1ac21f419aa029d7ccb5d8de044 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
> '2096f1ac21f419aa029d7ccb5d8de044 at 10.1.0.10'
> Method: OPTIONS
> Reliably Transmitting (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=z9hG4bK2268d402;rport
> From: "asterisk" <sip:asterisk at 192.168.1.2>;tag=as2a2e4c13
> To: <sip:ekiga.net>
> Contact: <sip:asterisk at 192.168.1.2>
> Call-ID: 1a9ce3ad2f18ecf129c457d527603c8f at 192.168.1.2
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 17 Aug 2009 21:30:52 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=z9hG4bK2268d402;rport=10003;received=190.51.105.123
> From: "asterisk" <sip:asterisk at 192.168.1.2:5060>;tag=as2a2e4c13
> To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.c092
> Call-ID: 1a9ce3ad2f18ecf129c457d527603c8f 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
> '1a9ce3ad2f18ecf129c457d527603c8f at 192.168.1.2'
> Method: OPTIONS
>
>
>
>
> Thanks for your reply.
>
> Regards,
> Daniel
>
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>
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