SKYPE to IPPI to Asterisk

Ok so i am trying to make a call from Skype to Asterisk via IPPI Messenger, i have ippi messanger running on the same machine as Asterisk, here are the outputs. Ports 5060 are open.

[code]<— SIP read from UDP:213.215.45.230:5060 —>
INVITE sip:s@192.168.1.81:5060 SIP/2.0
Record-Route: sip:213.215.45.230;lr;did=601.106257a2
Via: SIP/2.0/UDP 213.215.45.230:5060;branch=z9hG4bK6dfa.753c0c9.0
Via: SIP/2.0/UDP 213.215.45.239:5060;rport=5060;received=213.215.45.239;branch=z9hG4bK21c9ebd1
Max-Forwards: 12
From: "peter.minchington@skype.ippi.com" sip:peter.minchington@skype.ippi.com;tag=as49c79c90
To: sip:peter_minchington@ippi.fr
Contact: sip:peter.minchington@213.215.45.239:5060
Call-ID: 2a617cf83002cfa218c792e65a8ade06@skype.ippi.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.0
Date: Wed, 21 Aug 2013 07:49:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 337
DID-info: peter.minchington

v=0
o=root 391569625 391569625 IN IP4 213.215.45.239
s=Asterisk PBX 1.8.10.0
c=IN IP4 213.215.45.239
t=0 0
m=audio 19612 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
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
<------------->
— (17 headers 15 lines) —
Sending to 213.215.45.230:5060 (NAT)
Using INVITE request as basis request - 2a617cf83002cfa218c792e65a8ade06@skype.ippi.com
Found peer ‘ippi_incoming’ for ‘peter.minchington’ from 213.215.45.230:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format G729 for ID 18
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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=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 213.215.45.239:19612
Looking for s in from_ippi (domain 192.168.1.81)
list_route: hop: sip:213.215.45.230;lr;did=601.106257a2

<— Transmitting (NAT) to 213.215.45.230:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.215.45.230:5060;branch=z9hG4bK6dfa.753c0c9.0;received=213.215.45.230;rport=5060
Via: SIP/2.0/UDP 213.215.45.239:5060;rport=5060;received=213.215.45.239;branch=z9hG4bK21c9ebd1
Record-Route: sip:213.215.45.230;lr;did=601.106257a2
From: "peter.minchington@skype.ippi.com" sip:peter.minchington@skype.ippi.com;tag=as49c79c90
To: sip:peter_minchington@ippi.fr
Call-ID: 2a617cf83002cfa218c792e65a8ade06@skype.ippi.com
CSeq: 102 INVITE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:s@192.168.1.81:5060
Content-Length: 0

<------------>
– Executing [s@from_ippi:1] Dial(“SIP/ippi_incoming-00000001”, “SIP/my_phone”) in new stack
Really destroying SIP dialog ‘485bd9542064bd73450b8dc909108e68@[::1]:5060’ Method: INVITE
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/ippi_incoming-00000001’ status is ‘CHANUNAVAIL’

<— Reliably Transmitting (NAT) to 213.215.45.230:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 213.215.45.230:5060;branch=z9hG4bK6dfa.753c0c9.0;received=213.215.45.230;rport=5060
Via: SIP/2.0/UDP 213.215.45.239:5060;rport=5060;received=213.215.45.239;branch=z9hG4bK21c9ebd1
From: "peter.minchington@skype.ippi.com" sip:peter.minchington@skype.ippi.com;tag=as49c79c90
To: sip:peter_minchington@ippi.fr;tag=as408d296e
Call-ID: 2a617cf83002cfa218c792e65a8ade06@skype.ippi.com
CSeq: 102 INVITE
Server: Asterisk PBX SVN-branch-1.8-r396427
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0

<------------>

<— SIP read from UDP:213.215.45.230:5060 —>
ACK sip:s@192.168.1.81:5060 SIP/2.0
Via: SIP/2.0/UDP 213.215.45.230:5060;branch=z9hG4bK6dfa.753c0c9.0
From: "peter.minchington@skype.ippi.com" sip:peter.minchington@skype.ippi.com;tag=as49c79c90
Call-ID: 2a617cf83002cfa218c792e65a8ade06@skype.ippi.com
To: sip:peter_minchington@ippi.fr;tag=as408d296e
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: OpenSIPS (1.8.2-tls (i386/linux))
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘2a617cf83002cfa218c792e65a8ade06@skype.ippi.com’ Method: ACK
[/code]

sip.conf

[code][general]
context=unauthenticated ; default context for incoming calls
allowguest=no ; disable unauthenticated calls
srvlookup=yes ; enabled DNS SRV record lookup on outbound calls
udpbindaddr=0.0.0.0 ; listen for UDP requests on all interfaces
tcpenable=no ; disable TCP support
register=>peter_minchington:*******@ippi.fr

[ippi_incoming] ; incoming call setup for ippi
type=peer
host=ippi.fr
context=from_ippi
nat=yes
canreinvite=no

[ippi_outgoing] ; configuration for outgoing calls for ippi
type=peer
host=ippi.fr
username=peter_minchington
secret=*********
fromuser=peter_minchington
fromdomain=ippi.fr
nat=yes
canreinvite=no

[my_phone] ; configure a SIP account on Asterisk with username my_phone and password my_password
type=friend
secret=*******
host=dynamic
context=home
nat=yes
[/code]

extensions.conf

[code][from_ippi] ; incoming calls to ring the ippi SIP account my_phone
exten => s,1,Dial(SIP/my_phone)

[home] ; all outgoing calls from the account my_phone are sent over the ippi network:
exten => _X.,1,Dial(SIP/ippi_outgoing/${EXTEN})
[/code]

any help would be great, thanks

It is channel unavailable. my phone is probably not registered.

If they are both on the same machine, you will need explicit port numbers for one of them.

Please read the documentation on type, nat, canreinvite, externhost/externip/stunaddr, as you have deprecated, bad practice, questionable and possibly missing values/names for these (all common problems).

I’m not sure if there is any guarantee that the first of two peers with the same host address will be the one that maches, so you may need to use remotesecret (recent versions), insecure=invite, earlier ones.