Problème d'appels entrants
Bonjour, j'essaye de configurer mon asterisk avec ippi mais impossible de recevoir des appels, ça sonne occupé a chaque fois.
J'ai un asterisk 1.6.2.14 et le port 5060 de routé correctement (testé via netcat depuis un serveur externe, et l'asterisk reçoit bien ce que je lui dit).
Par contre en sortie, les appels passent correctement.
Voici mes confs (sip.conf et extensions.conf) ainsi qu'un "sip set debug peer ippi_incoming" lors de l'appel entrant :
Code:
; sip.conf
[general]
defaultexpirey=1800
dtmfmode=auto
qualify=yes
register => username:motdepasse@ippi.fr
externip=193.xxx.xxx.xx
localnet=192.168.2.0/255.255.255.0
[ippi_outgoing] ; appels sortants
type=peer
host=ippi.fr
username=username
secret=motdepasse
fromuser=username
fromdomain=ippi.fr
nat=yes
canreinvite=no
[ippi_incoming] ; appels entrants
type=peer
host=ippi.fr
context=from_ippi
nat=yes
canreinvite=no
qualify=yes
allow=all
insecure=port,invite
[rhaamo_n900] ; mon n900
; 10
type=friend
secret=unmotdepasse
host=dynamic
context=home
nat=yes
[test] ; test
; 11
type=friend
secret=unautremotdepasse
host=dynamic
context=home
nat=yes
Code:
; extensions.conf
[default]
include => from_ippi
[from_ippi]
exten => s,1,Dial(SIP/rhaamo_n900)
[home]
exten => 10,1,Dial(SIP/rhaamo_n900)
exten => 11,1,Dial(SIP/test)
exten => _X.,1,Dial(SIP/ippi_outgoing/${EXTEN})
Code:
SIP Debugging Enabled for IP: 213.xxx.xx.xxx:5060
<--- SIP read from UDP:213.xxx.xx.xxx:5060 --->
OPTIONS sip:193.xxx.xxx.xx:59517 SIP/2.0
Via: SIP/2.0/UDP 213.xxx.xx.xxx:5060;branch=0
From: sip:pinger@ippi.fr;tag=b5e188b1
To: sip:193.xxx.xxx.xx:59517
Call-ID: 2e9df1e4-f27f37f4-971e8@213.xxx.xx.xxx
CSeq: 1 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Looking for s in default (domain 193.xxx.xxx.xx)
<--- Transmitting (no NAT) to 213.xxx.xx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.xxx.xx.xxx:5060;branch=0;received=213.xxx.xx.xxx
From: sip:pinger@ippi.fr;tag=b5e188b1
To: sip:193.xxx.xxx.xx:59517;tag=as2462ef3a
Call-ID: 2e9df1e4-f27f37f4-971e8@213.xxx.xx.xxx
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:193.xxx.xxx.xx>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '2e9df1e4-f27f37f4-971e8@213.xxx.xx.xxx' in 32000 ms (Method: OPTIONS)
<--- SIP read from UDP:213.xxx.xx.xxx:5060 --->
OPTIONS sip:193.xxx.xxx.xx:59517 SIP/2.0
Via: SIP/2.0/UDP 213.xxx.xx.xxx:5060;branch=0
From: sip:pinger@ippi.fr;tag=c5e188b1
To: sip:193.xxx.xxx.xx:59517
Call-ID: 2e9df1e4-037f37f4-971e8@213.xxx.xx.xxx
CSeq: 1 OPTIONS
Content-Length: 0
<------------->
-- (7 headers 0 lines) ---
Looking for s in default (domain 193.xxx.xxx.xx)
<--- Transmitting (no NAT) to 213.xxx.xx.xxx:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.xxx.xx.xxx:5060;branch=0;received=213.xxx.xx.xxx
From: sip:pinger@ippi.fr;tag=c5e188b1
To: sip:193.xxx.xxx.xx:59517;tag=as3b930df8
Call-ID: 2e9df1e4-037f37f4-971e8@213.xxx.xx.xxx
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:193.xxx.xxx.xx>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '2e9df1e4-037f37f4-971e8@213.xxx.xx.xxx' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '2e9df1e4-330f37f4-b51e8@213.xxx.xx.xxx' Method: OPTIONS
Really destroying SIP dialog '2e9df1e4-430f37f4-b51e8@213.xxx.xx.xxx' Method: OPTIONS
Code:
natasha*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
ippi_incoming 213.xxx.xx.xxx N 5060 OK (21 ms)
ippi_outgoing/username 213.xxx.xx.xxx N 5060 OK (24 ms)
rhaamo_n900/rhaamo_n900 192.168.2.150 D N 63511 OK (104 ms)
test/test 192.168.2.2 D N 61102 OK (1 ms)
4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0 offline]
natasha*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
ippi.fr:5060 N username 1785 Registered Tue, 30 Nov 2010 11:15:59
1 SIP registrations.
natasha*CLI>
Merci d'avance à ceux qui pourront m'aider :)
