Re: [asterisk-users] Accessing to ekiga.net through Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 El miércoles 19 de agosto del 2009 a las 08:04:17 -0300, SIP escribió: Daniel, Hi SIP. 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. This it is a scheme of my network configuration: +--+ +-+ ___/\__ | | | | / \ | GNU/Linux eth1-+ ADSL Router +-| Internet | | Firewall/ | | | \__ __/ | Asterisx eth0++-+ \_/ | | | +--+ | | +--+--+ | LAN switch | +-+ The ADSL router is configured to connect itself to Internet for its own means (I don't use any software PPPoE in the GNU/Linux box). This router uses the private IP 192.168.1.1. In the GNU/Linux box the eth1 interface uses the private IP 192.168.1.2. The eth0 interface (10.1.0.10) is the point of connection to the rest of the LAN (10.1.0.0/24). Firewall makes NAT of all the originating traffic of eth0 through eth1. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkqZQ/8ACgkQZpa/GxTmHTcxAgCfQwO4PxNarZO7nAFwQSVK1EW/ /wYAnR3KQF6+6p2jkKo1spZxi1RjT4de =gCuK -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Accessing to ekiga.net through Asterisk
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:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 INVITE Contact: sip:2...@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 - kafgeaflkmsd...@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:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10;tag=as0a3a462b Call-ID: kafgeaflkmsd...@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 'kafgeaflkmsd...@defiant.freesoftware.org' in 32000 ms (Method: INVITE) Found user '201' alderamin*CLI --- SIP read from 10.1.0.65:5060 --- ACK sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10;tag=as0a3a462b From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@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:8...@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:8...@10.1.0.10,response=9cb53107d4d15b7a2e7df8599e851b80,algorithm=MD5 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE Contact: sip:2...@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 - kafgeaflkmsd...@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:2...@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:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE User-Agent: Asterisk PBX Allow:
Re: [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 Bareirodaniel-lis...@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:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: sip:8...@10.1.0.10
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 183 INVITE
Contact: sip:2...@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 -
mrsyiysrdkwm...@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:2...@10.1.0.10;tag=uucwz
To: sip:8...@10.1.0.10;tag=as095989a3
Call-ID: mrsyiysrdkwm...@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
'mrsyiysrdkwm...@defiant.freesoftware.org' in 32000 ms (Method: INVITE)
Found user '201'
alderamin*CLI
--- SIP read from 10.1.0.65:5060 ---
ACK sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: sip:8...@10.1.0.10;tag=as095989a3
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@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:8...@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:8...@10.1.0.10,response=d49c0fdf11c9977fcd1fce6a50f445fe,algorithm=MD5
To: sip:8...@10.1.0.10
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 184 INVITE
Contact: sip:2...@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 -
mrsyiysrdkwm...@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:2...@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
Re: [asterisk-users] Accessing to ekiga.net through Asterisk
-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:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 INVITE Contact: sip:2...@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 - kafgeaflkmsd...@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:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10;tag=as0a3a462b Call-ID: kafgeaflkmsd...@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 'kafgeaflkmsd...@defiant.freesoftware.org' in 32000 ms (Method: INVITE) Found user '201' alderamin*CLI --- SIP read from 10.1.0.65:5060 --- ACK sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10;tag=as0a3a462b From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@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:8...@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:8...@10.1.0.10,response=9cb53107d4d15b7a2e7df8599e851b80,algorithm=MD5 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE Contact: sip:2...@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 - kafgeaflkmsd...@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:2...@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:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:8...@10.1.0.10 Content-Length: 0 -- Executing [8...@from-internal:1] Dial(SIP/201-0900, 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 0x4 (h261) to SDP Adding non-codec 0x1 (telephone-event)
Re: [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:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: sip:8...@10.1.0.10
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 183 INVITE
Contact: sip:2...@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 -
mrsyiysrdkwm...@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:2...@10.1.0.10;tag=uucwz
To: sip:8...@10.1.0.10;tag=as095989a3
Call-ID: mrsyiysrdkwm...@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
'mrsyiysrdkwm...@defiant.freesoftware.org' in 32000 ms (Method: INVITE)
Found user '201'
alderamin*CLI
--- SIP read from 10.1.0.65:5060 ---
ACK sip:8...@10.1.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks
Max-Forwards: 70
To: sip:8...@10.1.0.10;tag=as095989a3
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@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:8...@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:8...@10.1.0.10,response=d49c0fdf11c9977fcd1fce6a50f445fe,algorithm=MD5
To: sip:8...@10.1.0.10
From: Hector sip:2...@10.1.0.10;tag=uucwz
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq: 184 INVITE
Contact: sip:2...@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 -
mrsyiysrdkwm...@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:2...@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:2...@10.1.0.10;tag=uucwz
To: sip:8...@10.1.0.10
Call-ID: mrsyiysrdkwm...@defiant.freesoftware.org
CSeq:
Re: [asterisk-users] Accessing to ekiga.net through Asterisk
-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. alderamin*CLI --- SIP read from 10.1.0.65:5060 --- INVITE sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 709 INVITE Contact: sip:2...@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 - kafgeaflkmsd...@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:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10;tag=as0a3a462b Call-ID: kafgeaflkmsd...@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 'kafgeaflkmsd...@defiant.freesoftware.org' in 32000 ms (Method: INVITE) Found user '201' alderamin*CLI --- SIP read from 10.1.0.65:5060 --- ACK sip:8...@10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: sip:8...@10.1.0.10;tag=as0a3a462b From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@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:8...@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:8...@10.1.0.10,response=9cb53107d4d15b7a2e7df8599e851b80,algorithm=MD5 To: sip:8...@10.1.0.10 From: Hector sip:2...@10.1.0.10;tag=typwm Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE Contact: sip:2...@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 - kafgeaflkmsd...@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:2...@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:2...@10.1.0.10;tag=typwm To: sip:8...@10.1.0.10 Call-ID: kafgeaflkmsd...@defiant.freesoftware.org CSeq: 710 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:8...@10.1.0.10 Content-Length: 0 -- Executing [8...@from-internal:1] Dial(SIP/201-0900, 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 0x4 (h261) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to
