Connecting Asterisk SIP PBX to Skype

Hi,
I'm a new Skype Business customer, I have a few user registered and I wish to connect my company Asterisk PBX to Skype. In details, I bought a Skype-In number, I would like that calls to such number will arrive to one of my PBX extension. Is there some way to do this?
Thank you in advance.

Could someone help me?
I'm having problems with my Elastix (Asterisk) / Skype Connect Configuration. I always get "All circuits are busy now..." message. My configuration is:
Trunk Name: skype_in
type=friend
username=xxxxxxx
fromdomain=sip.skype.com
fromuser=xxxxxxx
realm=sip.skype.com
host=sip.skype.com
dtmfmode=rfc2833
secret=password
nat=no
insecure=invite
qualify=yes
disallow=all
allow=alaw&ulaw
amaflags=default
trustrpid=no
sendrpid=yes
context=skype_in
Register String: [email protected]/xxxxxx From asterisk I get:
elastix*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
skype_in/99051000147265 204.9.161.164 5060 OK (177 ms)
and
sip show registry
sip.skype.com:5060 990510001472 105 Registered Thu, 06 Oct 2011 20:46:30
Every looks good but when I try to make a call... I get the busy message:
SIP Debugging Enabled for IP: 10.168.16.115:5060
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-b0d83df763d87d56-1---d8754z-
Max-Forwards: 70
Contact: 6001>
To: 20551141256555>
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 255
v=0
o=Zoiper_user 0 0 IN IP4 10.168.16.115
s=Zoiper_session
c=IN IP4 10.168.16.115
t=0 0
m=audio 8000 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 10.168.16.115 : 5060 (no NAT)
Using INVITE request as basis request - MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
<--- Reliably Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-b0d83df763d87d56-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>;tag=as2f3cba10
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60db2ce3"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.' in 32000 ms (Method: INVITE)
Found user '6001'
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-b0d83df763d87d56-1---d8754z-
Max-Forwards: 70
To: 20551141256555>;tag=as2f3cba10
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-
Max-Forwards: 70
Contact: 6001>
To: 20551141256555>
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Proxy-Authorization: Digest username="6001",realm="asterisk",nonce="60db2ce3",uri="sip:[email protected];transport=UDP",response="ed79aee161a6cc8a7e520cd011afe0bb",algorithm=MD5
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 255
v=0
o=Zoiper_user 0 0 IN IP4 10.168.16.115
s=Zoiper_session
c=IN IP4 10.168.16.115
t=0 0
m=audio 8000 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 10.168.16.115 : 5060 (NAT)
Using INVITE request as basis request - MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
Found user '6001'
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.168.16.115:8000
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.168.16.115:8000
Looking for 20551141256555 in from-internal (domain 10.168.16.3)
list_route: hop: 6001>
<--- Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: 20551141256555>
Content-Length: 0
<------------>
Audio is at 10.168.16.3 port 17408
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>;tag=as2650cb3c
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: 20551141256555>
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 14425 14425 IN IP4 10.168.16.3
s=session
c=IN IP4 10.168.16.3
t=0 0
m=audio 17408 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
<------------->
Scheduling destruction of SIP dialog 'MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 484 Address incomplete
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>;tag=as2650cb3c
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-
Max-Forwards: 70
To: 20551141256555>;tag=as2650cb3c
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
elastix*CLI> sip set debug off
SIP Debugging Disabled
Could someone help me?
Best regards
Eduardo6001>20551141256555>20551141256555>6001>20551141256555>20551141256555>6001>20551141256555>20551141256555>6001>6001>6001>20551141256555>6001>6001>20551141256555>20551141256555>6001>6001>20551141256555>6001>

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State of No Authentication in Asterisk SIP Registr...

I have been trying lots of combinations and permutations of registration entries in Asterisk's SIP.conf file.   I would be grateful if someone would let me know exactly what userid and password I need to include
The bulk of instructions I have been reading over the past half a day indicate that this form is correct.
register => [edited for privacy]
where bmdemo3 is the userid and REDACTED is the password for the Skype user I created under Skype Manager to represent Asterisk to Skype's SIP trunk.
In the output to the sip show registry command for Asterisk, there is one entry that suggests a registration for sip.skype.com:5060, however the column to the far right (entitled) State says "No Authentication".   That does not seem good.
The aforementioned userid and password log me in to that skype user's account (which I activated) and I can make and receive direct Skype calls using it.   Is there another userid and password I should be using instead?
I would greatly appreciate any guidance as to what I might be doing wrong.
Cheers,
Michael

Here is the output from the sip show registry command
ubuntu*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
sip.skype.com:5060                      N      bmdemo3            120 No Authentication
1 SIP registrations.
Here is the SIP REGISTER message being rejected:
<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 403 Forbidden
From: <sip:[email protected]>;tag=as333ae3e0
To: <sip:[email protected]>;tag=c990d13f-f1dd03ad-0-9c8aac96-0
Call-ID: [email protected]
CSeq: 102 REGISTER
Via: SIP/2.0/UDP 172.16.164.33:5060;branch=z9hG4bK0ae22ba3;rport=55060;received=96.241.233.63
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------->
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 0 [ 21]: SIP/2.0 403 Forbidden
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 1 [ 48]: From: <sip:[email protected]>;tag=as333ae3e0
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 2 [ 66]: To: <sip:[email protected]>;tag=c990d13f-f1dd03ad-0-9c8aac96-0
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 3 [ 51]: Call-ID: [email protected]
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 4 [ 18]: CSeq: 102 REGISTER
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 5 [ 93]: Via: SIP/2.0/UDP 172.16.164.33:5060;branch=z9hG4bK0ae22ba3;rport=55060;received=96.241.233.63
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 6 [ 12]: Expires: 120
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 7 [ 35]: Contact: <sip:[email protected]:5060>
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 8 [ 17]: Content-Length: 0
[Jul 1 16:21:41] VERBOSE[1081] chan_sip.c: --- (9 headers 0 lines) ---
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: = Looking for Call ID: [email protected] (Checking To) --From tag as333ae3e0 --To-tag c990d13f-f1dd03ad-0-9c8aac96-0
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #756
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
[Jul 1 16:21:41] WARNING[1081] chan_sip.c: Forbidden - wrong password on authentication for REGISTER for 'bmdemo3' to 'sip.skype.com'

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