Teise jaama IP aadress oli 192.168.252.42. Kasutajanimi ja parool olid samad, mis esimeses jaamas. Ühendamisel kasutasime SIP protokolli. Teises jaamas oli meil ainult üks kasutaja. Mõlemast jaamast sai otse valida teise jaama numbrit. Häälestasime jaamad, kasutasime veebilehte http://www.cadvision.com/blanchas/Asterisk/SIPtrunk.html Meie kõned algasid kell 10.25.28 ja lõppesid kell 11.50.39 helistasime üksteisele: 2011-09-14 10:25:28,IAX2/1004-478,1004,"""User4"" <1004>",1001,ANSWERED,18 2011-09-14 10:25:53,IAX2/1004-1012,1004,"""User4"" <1004>",1002,NO ANSWER,18 2011-09-14 10:26:23,IAX2/1004-6289,1004,"""User4"" <1004>",1002,ANSWERED,43 2011-09-14 10:28:28,IAX2/1003-3074,1003,"""User3"" <1003>",1004,ANSWERED,21 2011-09-14 10:29:43,IAX2/1004-241,1004,"""User4"" <1004>",1002,ANSWERED,10 2011-09-14 10:30:09,IAX2/1003-2775,1003,"""User3"" <1003>",1002,ANSWERED,18
html http://www.cadvisio n.com/blanchas/Asterisk/SIPtrunk.html PBX1 <=> PBX2 PBX2 <= PBX3. , SIP PBX1 PBX3 PBX2 => PBX3, . . , . IAX 2 PBX 1 PBX 2: (http://192.168.252.41 http://192.168.252.42). . IAX2 . FreePBX => Add Trunk => Add IAX2 Trunk , Outgoing Settings ( ) : [Trunk name] host username secret qualify=yes trunk=yes type=peer IAX2 ( PBX c PBX) : PBX 1: PBX 2: [200peer] [100peer] host=192.168.252.42 host=192.168.252.41 username=100user username=200user secret=1234 secret=1234
Mina tegin kaks reeglit: · AfterHours (peale tööd) Name: AfterHours Timeout: 10 Enable Directory: linnuke peal Directory Content: Default Enable Direct Dial: linnuke peal Announcement: None · BusinessHours (tööajal) Name: BusinessHours Timeout: 10 Enable Directory: linnuke Directory Content: Default Enable Direct Dial: linnuke peal Announcement: None Ideeks on suunata kõned teisele numbrile määratud ajal. Et suunamine töötaks, peaks reaalselt toimima IAX2 protokoli konto, mis ühendab telefonijaama välisvõrguga. · Viimane asi, mida ma tegin, on ,outbound route'. Nimelt see on tee, mis ühendab minu telefonijaama teise maailmaga. Selleks ma tegin kontot veebileheküljel www.voipjet.com Kuna see on testikonto, siis seda helistamiseks kasutada ei saa (algselt konto peal on 0.25$ kuid helistamiseks on vaja vähemalt 20$). Samas telefonijaam võtab voipjet serveriga ühendust. Kontot tegin vastavalt juhistele:
Routes, where it is needed to write the following: -7- Route Name must be a descriptive name of route Dial Patterns example of extension number Trunk Sequence this is the trunk that we configured As a result: For PBX 1: For PBX 2: [200-dial2xx] [100-dial1xx] [2xx] [1xx] [IAX2/200-peer] [IAX2/100-peer] The last step is to set some users or Extensions. All in all we need 25 users for PBX1 and 25 for PBX2. We will describe the creation of 2 users as the configuration of rest is absolutely the same. From FreePBX menu => Extensions where from Add an Extension menu we select «Generic SIP Device». On page which comes after that the following parameters were required:
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