请问有人知如何将2b app set入freepbx吗？

lttliang

本人的2B PC版被停止服务了，现在买了28一个月的2b app ，想在freepbx里用，不知怎样set,有成功的朋友请分享一下啦 ，多谢

bigtoodog

add the followings into /etc/hosts:
;203.80.89.135        s2hkbntel.net     s21.hkbntel.net
203.80.89.139         s2hkbntel.net     s22.hkbntel.net
;203.80.89.146        s2hkbntel.net     s23.hkbntel.net
;203.80.89.150        s2hkbntel.net     s24.hkbntel.net

Trunk->Outgoing Setting->Peer Details:
type=peer
defaultuser=12345678hk
fromuser=12345678hk
secret=passwd
host=s2hkbntel.net   
fromdomain=s2hkbntel.net   
canreinvite=no
insecure=port,invite
qualify=yes
sendrpid=yes
;nat=force_rport,comedia
port=5060
context=from-trunk
dtmfmode=auto
disallow=all
allow=g729&ulaw&alaw

Trunk->Incoming Settings->User Details:
username=12345678hk
type=user
host=s22.hkbntel.net
fromdomain=s2hkbntel.net
context=from-trunk
outboundproxy=s22.hkbntel.net

Trunk->Registration->Registration String:
12345678hk:passwd@s2hkbntel.net/12345678

lttliang

回復 2# bigtoodog


    真是非常非常感谢，按照您提供的方法，已经加入freepbx中，并且可以正常接听拨打了，多晒！

richorus

我啱啱開始玩asterisk,elastix,freepbx. 我按ching設好咗2b啲setting, 但打唔出, hosts試咗上面4個到唔得, 唔知道有咩set錯呢?

    -- Executing [s@macro-dialout-trunk:22] Dial("SIP/383-00000006", "SIP/2bapp/61234567,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/2bapp/61234567
    -- No one is available to answer at this time (1:0/0/0)
    -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/383-00000006", "Dial failed for some reason with DIALSTATUS = NOANSWER and HANGUPCAUSE = 16") in new stack
    -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/383-00000006", "0?continue,1:s-NOANSWER,1") in new stack
    -- Goto (macro-dialout-trunk,s-NOANSWER,1)
    -- Executing [s-NOANSWER@macro-dialout-trunk:1] NoOp("SIP/383-00000006", "Dial failed due to trunk reporting NOANSWER - giving up") in new stack
    -- Executing [s-NOANSWER@macro-dialout-trunk:2] Progress("SIP/383-00000006", "") in new stack
    -- Executing [s-NOANSWER@macro-dialout-trunk:3] Playback("SIP/383-00000006", "number-not-answering,noanswer") in new stack
    -- <SIP/383-00000006> Playing 'number-not-answering.gsm' (language 'en')
       > 0x9143a00 -- Probation passed - setting RTP source address to 10.10.0.127:51402
    -- Executing [s-NOANSWER@macro-dialout-trunk:4] Congestion("SIP/383-00000006", "20") in new stack
[2015-07-02 14:56:51] WARNING[30360][C-00000003]: channel.c:4860 ast_prod: Prodding channel 'SIP/383-00000006' failed
  == Spawn extension (macro-dialout-trunk, s-NOANSWER, 4) exited non-zero on 'SIP/383-00000006' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 61234567, 5) exited non-zero on 'SIP/383-00000006'
    -- Executing [h@from-internal:1] Hangup("SIP/383-00000006", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/383-00000006'

ttmuskie

Incoming ok?

banana1012

大大們, 我個2B申請左幾日, 而家有$68蚊8個冧巴.
會唔會係有新SERVER? 因為個Sales以前問佢話冇, 上個月月中打卑我話upgrade左SYSTEM 話有新PLAN.

我跟住上面SET入FreePBX 都注冊唔到, 我FreePBX裝左入VM,
另一個問題, 我以前用ATCOM IP08用唔到第三條線打電話, 只係可以第一條用緊轉第二條.

banana1012

Asterisk System uptime: 8 hours, 53 minutes, 23 seconds Last reload: 29 minutes, 19 seconds
Active SIP Channel(s): 1	Active PJSIP Channel(s): 0	Active IAX2 Channel(s): 0
Sip Registry: 2	PJSip Registrations: 0	IAX2 Registry: 1
Sip Peers:     Online: 0     Online-Unmonitored: 0     Offline: 1     Offline-Unmonitored: 0	PJSip Endpoints:     Available: 0     Unavailable: 0     Unknown: 0	IAX2 Peers:     Online: 0     Offline: 0     Unmonitored: 0

角色

回復 7# banana1012

请你看PM。