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