[maison] ; on déclare le contexte local qu'on a spécifié dans le sip.conf ; numéros "locaux" exten => 10,1,Dial(SIP/benjamoon) ; quand on compose le 10, le softphone branché sur le lien "moi" sonnera ; numéros externes exten => _9.,1,Dial(SIP/freephonie-out/${EXTEN:1}) ; quand on compose un numero qui commence par 9, on utilise le lien "freephonie" et on passe le numero au peer en ôtant le premier digit. [fromfree] ; on déclare le contexte de reception d'appels depuis freephonie ;exten => s,1,Dial(SIP/moi&SIP/toi) ; et on fait sonner les deux softphones, si on avait voulu faire sonner un seul client sip, on aurait écrit exten => s,1,Dial(SIP/moi) exten => s,1,Dial(SIP/benjamoon) ; et on fait sonner les deux softphones, si on avait voulu faire sonner un seul client sip, on aurait écrit exten => s,1,Dial(SIP/moi) Really destroying SIP dialog '5410b37778002ef25cfd561c17fa3a39@192.168.1.3' Method: OPTIONS kalida*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status benjamoon/benjamoon 192.168.1.3 D N 5061 OK (1 ms) freephonie-in 212.27.52.5 5060 OK (53 ms) freephonie-out/0950714910 212.27.52.5 N 5060 OK (44 ms) 3 sip peers [Monitored: 3 online, 0 offline Unmonitored: 0 online, 0 offline] --- (12 headers 14 lines) --- Sending to 192.168.1.3 : 5061 (NAT) Using INVITE request as basis request - 23FEA308-D4BA-4E71-511A-A92ED36EF264@192.168.1.3 Found user 'benjamoon' 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 192.168.1.3:8000 Found description format pcmu for ID 0 Found description format pcma for ID 8 Found description format gsm for ID 3 Found description format iLBC for ID 98 Found description format speex for ID 97 Found description format telephone-event for ID 101 Capabilities: us - 0x20c (ulaw|alaw|speex), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0x20c (ulaw|alaw|speex) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.3:8000 Looking for 05566741999 in maison (domain 192.168.1.3) <--- Reliably Transmitting (NAT) to 192.168.1.3:5061 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.3:5061;branch=z9hG4bK24161B1C90B8DFB84325B75D530B8D76;received=192.168.1.3;rport=5061 From: Benjamin Frank ;tag=2030974093 To: ;tag=as31ed7249 Call-ID: 23FEA308-D4BA-4E71-511A-A92ED36EF264@192.168.1.3 CSeq: 53482 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '23FEA308-D4BA-4E71-511A-A92ED36EF264@192.168.1.3' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.1.3:5061 ---> ACK sip:05566741999@192.168.1.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5061;rport;branch=z9hG4bK24161B1C90B8DFB84325B75D530B8D76 From: Benjamin Frank ;tag=2030974093 To: ;tag=as31ed7249 Contact: Call-ID: 23FEA308-D4BA-4E71-511A-A92ED36EF264@192.168.1.3 CSeq: 53482 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '68666283-8D40-6B6B-EC2D-661256BC60EC@192.168.1.3' Method: ACK