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=yesCode:; 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
Merci d'avance à ceux qui pourront m'aider :)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>




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