sip.conf
Код: Выделить всё
register => 99051000152343:my_pass@sip.skype.com/99051000152343
[skype]
type=peer
host=63.209.144.201
fromuser=99051000152343
fromdomain=sip.skype.com
secret=my_pass
nat=no
canreinvite=no
insecure=port,invite
context=PRI
disallow=all
allow=gsm
allow=ulaw
allow=alaw
extensions.conf
Код: Выделить всё
[PRI]
exten => +79201022399,1,Set(CALLERID(all)=99051000152343)
exten => +79201022399,n,Dial(SIP/skype/${EXTEN})
sip*CLI> sip show registry
Код: Выделить всё
Host Username Refresh State Reg.Time
sip.skype.com:5060 990510001523 105 Registered Fri, 16 Dec 2011 14:53:39
Код: Выделить всё
-- Executing [+79201022399@PRI:1] Set("SIP/209-f4149ee0", "CALLERID(all)=99051000152343") in new stack
-- Executing [+79201022399@PRI:2] Dial("SIP/209-f4149ee0", "SIP/skype/+79201022399") in new stack
-- Called skype/+79201022399
-- SIP/skype-008080e0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/209-f4149ee0' status is 'CONGESTION'
Код: Выделить всё
Audio is at 146.247.0.44 port 11644
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 63.209.144.201:5060:
INVITE sip:+79201022399@63.209.144.201 SIP/2.0
Via: SIP/2.0/UDP 146.247.0.44:5060;branch=z9hG4bK47583317;rport
From: "99051000152343" <sip:99051000152343@sip.skype.com>;tag=as53258cdb
To: <sip:+79201022399@63.209.144.201>
Contact: <sip:99051000152343@146.247.0.44>
Call-ID: 0248e6f116f304e26097b4257915fa68@sip.skype.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 16 Dec 2011 12:11:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 19152 19152 IN IP4 146.247.0.44
s=session
c=IN IP4 146.247.0.44
t=0 0
m=audio 11644 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
sip*CLI>
<--- SIP read from 63.209.144.201:5060 --->
SIP/2.0 100 Trying
From: "99051000152343" <sip:99051000152343@sip.skype.com>;tag=as53258cdb
To: <sip:+79201022399@63.209.144.201>
Call-ID: 0248e6f116f304e26097b4257915fa68@sip.skype.com
CSeq: 102 INVITE
Via: SIP/2.0/UDP 146.247.0.44:5060;branch=z9hG4bK47583317;rport=5060
Contact: <sip:+79201022399@63.209.144.201:5060;maddr=63.209.144.201;transport=udp>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 63.209.144.201:5060 --->
SIP/2.0 404 Not Found
From: "99051000152343" <sip:99051000152343@sip.skype.com>;tag=as53258cdb
To: <sip:+79201022399@63.209.144.201>;tag=ca90d13f-13c4-4eeb277d-75d4e1cc-dc78ab7
Call-ID: 0248e6f116f304e26097b4257915fa68@sip.skype.com
CSeq: 102 INVITE
Via: SIP/2.0/UDP 146.247.0.44:5060;branch=z9hG4bK47583317;rport=5060
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Transmitting (no NAT) to 63.209.144.201:5060:
ACK sip:+79201022399@63.209.144.201 SIP/2.0
Via: SIP/2.0/UDP 146.247.0.44:5060;branch=z9hG4bK47583317;rport
From: "99051000152343" <sip:99051000152343@sip.skype.com>;tag=as53258cdb
To: <sip:+79201022399@63.209.144.201>;tag=ca90d13f-13c4-4eeb277d-75d4e1cc-dc78ab7
Contact: <sip:99051000152343@146.247.0.44>
Call-ID: 0248e6f116f304e26097b4257915fa68@sip.skype.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
