Hello All,
I’ve installed Asterisk PBX thanks to Asterisk@Home 1.5 (it’s so easy 
) Yesterday i succeeded in calling outbound via Freeipcall but inbound call still doesn’t work. When I try, I get a “error 503 :service unavailable”.
I’m wondering if the fact that i do my tests in the same network with a nat and it doesn’t work are in relation ?
I’d like to know if my ‘200’ extension might be in context ‘from-sip’ or 'from-internal’
Maybe someone could try to call me at 1312607876@sip.freeipcall.com in order to test if inbound call works when the call really comes from outbound.
Thanks in advance,
Vincent.
Here is my log when i try to simulate an inbound call:
Sip read:
INVITE sip:200@84.96.113.126 SIP/2.0
Record-Route: <sip:1312607876@82.97.10.20;ftag=008551cbfabc7933a9d6e209b91c2544;lr>
Via: SIP/2.0/UDP 82.97.10.20;branch=z9hG4bK50b9.087c4659c1caa52a8cfd54d940fb2e4f.0
Via: SIP/2.0/UDP 82.97.10.20:5061;branch=z9hG4bKd71523651228f006b7b776b01ddc35f9;rport=5061
Max-Forwards: 69
From: Vince <sip:1312609475@82.97.10.20>;tag=008551cbfabc7933a9d6e209b91c2544
To: <sip:1312607876@82.97.10.20>
Call-ID: 727BDB6D-F72B-4E8A-A7E0-941394C01DEE@192.168.1.61
CSeq: 200 INVITE
Contact: Anonymous <sip:82.97.10.20:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 4020051583-1383473812-3467240325-3077901472
h323-conf-id: 4020051583-1383473812-3467240325-3077901472
Content-Length: 301
Content-Type: application/sdp
v=0
o=1312609475 13807062 13807078 IN IP4 84.96.113.126
s=X-Lite
c=IN IP4 192.168.1.210
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
16 headers, 13 lines
Using latest request as basis request
Sending to 82.97.10.20 : 5060 (non-NAT)
Found peer 'fipc-outgoing'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.210:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 200 in from-sip
Reliably Transmitting (NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 82.97.10.20;branch=z9hG4bK50b9.087c4659c1caa52a8cfd54d940fb2e4f.0;received=82.97.10.20;rport=5060
Via: SIP/2.0/UDP 82.97.10.20:5061;branch=z9hG4bKd71523651228f006b7b776b01ddc35f9
From: Vince <sip:1312609475@82.97.10.20>;tag=008551cbfabc7933a9d6e209b91c2544
To: <sip:1312607876@82.97.10.20>;tag=as4e85cd91
Call-ID: 727BDB6D-F72B-4E8A-A7E0-941394C01DEE@192.168.1.61
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:200@84.96.113.126>
Content-Length: 0
to 82.97.10.20:5060
asterisk1*CLI>
Sip read:
ACK sip:200@84.96.113.126 SIP/2.0
Via: SIP/2.0/UDP 82.97.10.20;branch=z9hG4bK50b9.087c4659c1caa52a8cfd54d940fb2e4f.0
From: Vince <sip:1312609475@82.97.10.20>;tag=008551cbfabc7933a9d6e209b91c2544
Call-ID: 727BDB6D-F72B-4E8A-A7E0-941394C01DEE@192.168.1.61
To: <sip:1312607876@82.97.10.20>;tag=as4e85cd91
CSeq: 200 ACK
User-Agent: Free IP Call v2.291004
Content-Length: 0
8 headers, 0 lines
Destroying call '727BDB6D-F72B-4E8A-A7E0-941394C01DEE@192.168.1.61'Here are my conf file :
[sip.conf] :
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
externip = 84.96.113.126 ; Address that we're going to put in SIP messages if we're behind a NAT
localnet = 192.168.1.0/255.255.255.0
disallow=all
allow=gsm
allow=ulaw
allow=alaw
context = from-sip
maxexpirey=180
defaultexpirey=160
tos=reliability
register => 1312607876:password@sip.freeipcall.com/200
[fipc-outgoing]
type=friend
nat=yes
canreinvite=no
secret=password
fromuser=1312607876
username=1312607876
host=sip.freeipcall.com
fromdomain=sip.freeipcall.com
dtmfmode=inband
callerid="varod" <1312607876@sip.freeipcall.com>
insecure=very [extension.conf]
[globals]
FIPCUSERID=1312607876
FIPCUSERNAME=varod
PHONE1=201
PHONE1VM=201
FIPCEXTEN=200
[from-sip]
exten => ${FIPCEXTEN},1,Dial(${PHONE1},30)
exten => ${FIPCEXTEN},2,Voicemail(u${PHONE1VM})
exten => ${FIPCEXTEN},3,Hangup
exten => ${FIPCEXTEN},102,Voicemail(b${PHONE1VM})
exten => ${FIPCEXTEN},103,Hangup