[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 <sip:benjamoon@192.168.1.3:5061>;tag=2030974093
To: <sip:05566741999@192.168.1.3>;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 <sip:benjamoon@192.168.1.3:5061>;tag=2030974093
To: <sip:05566741999@192.168.1.3>;tag=as31ed7249
Contact: <sip:benjamoon@192.168.1.3:5061>
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