[asterisk-users] incoming SIP call

[Jean Marc Le Fevre]

Hello all,


I'm having a quite simple configuration like:

SIP provider <=> asterisk SIP <=> lan

Everythings works fine but sometime I can't get incoming call.

here are some of the logs from set debug 25 set verbosity 25 sip show  
debug and sip.conf and a part of extension.conf
thanks in advance


Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK67c2df66;rport
From: "asterisk" <sip:asterisk at 82.XXX.XXX.XXX>;tag=as01265eaf
To: <sip:freephonie.net>
Contact: <sip:asterisk at 82.XXX.XXX.XXX>
Call-ID: 7263e88c20c9f38c34963cef6704cf07 at 82.XXX.XXX.XXX
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 18 Apr 2007 13:57:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch=z9hG4bK253c1a3d;rport
From: "asterisk" <sip:asterisk at 82.XXX.XXX.XXX>;tag=as372da2cb
To: <sip:freephonie.net>
Contact: <sip:asterisk at 82.XXX.XXX.XXX>
Call-ID: 793bf24e5290d562787c8d9451baedd7 at 82.XXX.XXX.XXX
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 18 Apr 2007 13:57:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Zpro*CLI>
<-- SIP read from 212.27.52.5:5060:
SIP/2.0 403 not registered
Call-ID: 7263e88c20c9f38c34963cef6704cf07 at 82.XXX.XXX.XXX
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk at 82.XXX.XXX.XXX>;tag=as01265eaf
To: <sip:freephonie.net>;tag=00-31057-001dc208-591e1ca81
Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 
5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66
Content-Length: 0


--- (7 headers 0 lines) ---
Destroying call '7263e88c20c9f38c34963cef6704cf07 at 82.XXX.XXX.XXX'
Zpro*CLI>
<-- SIP read from 212.27.52.5:5060:
SIP/2.0 403 not registered
Call-ID: 793bf24e5290d562787c8d9451baedd7 at 82.XXX.XXX.XXX
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk at 82.XXX.XXX.XXX>;tag=as372da2cb
To: <sip:freephonie.net>;tag=00-32700-001dc209-6fc2b3303
Via: SIP/2.0/UDP 82.XXX.XXX.XXX: 
5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d
Content-Length: 0

--- (7 headers 0 lines) ---
Destroying call '793bf24e5290d562787c8d9451baedd7 at 82.XXX.XXX.XXX'


sip.conf

[general]
context=incoming
realm=etatcritik.dyndns.org
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
maxexpiry=3600
defaultexpiry=1800
videosupport=yes
disallow=all
allow=ulaw
allow=ilbc
allow=alaw
allow=gsm
musicclass=default
language=fr
useragent=Asterisk PBX
dtmfmode = auto
register => 09XXXXXXXX:SECRET at freephonie.net
registertimeout=40
externip = 82.XXX.XXX.XXX
localnet=10.XXX.XXX.XXX/255.255.255.0
qualify=60000
nat = yes
[test]
type=friend
username=test
secret=test
host=dynamic
context=home
callerid =test <2222>
dmtfmode=rfc2833
authuser=test
fromuser=test
allow=all
[freephonie_outbound]
type=peer
allow=all
host=freephonie.net
secret=SECRET
fromuser=09XXXXXXX
username=09XXXXXXX
dtmfmode=inband
qualify=60000
fromdomain=freephonie.net
[freephonie_inbound]
type=peer
context=incoming
host=freephonie.net
qualify=60000
allow=all
deny=0.0.0.0/0.0.0.0
permit=212.27.52.5/255.255.255.255  ; ip de freephonie.net

etension.conf


...
[incoming]
exten => s,1,Ringing
exten => s,2,Noop(I receive a sip call);
exten => s,n,Goto(home,1000,1)
exten => s,n,Congestion
;
...








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