getonsip 用にハードフォンをプロビジョニングするにはどうすればよいですか?

getonsip 用にハードフォンをプロビジョニングするにはどうすればよいですか?

プロビジョニングではシスコ SPA 942ハードフォン、Linksysブランド、セットアップを完了するにはどうすればいいですか?ゲットンシップ(またはオンシップ)?

SIP Address: [email protected]
Username: foo
Domain: getonsip.com
SIP Password: GHdjlRBfjdklHWD
Auth Username: getonsip_foo
Outbound Proxy: sip.onsip.com

タブ内SIP:

SIP Parameters

SIP Server Name:  getonsip.com
SIP User Agent Name: foo
SIP Reg User Agent Name: getonsip_foo

タブにはEXT 1次のものがあります:

SIP settings

SIP Port:   
EXT SIP Port:
SIP Proxy-Require:

タブには次の項目もExt 1あります:

Proxy and Registration

Proxy:      
Use Outbound Proxy: 
Outbound Proxy:     
Use OB Proxy In Dialog:

しかし、onsip からの とパスワードがどこに行くのかよくわかりませんAuth Username。特に、それらはgetonsip.comSIP アドレスとsip.onsip.comプロキシで使用されます。

答え1

最善の選択肢 - onsip.com サポートにお問い合わせください。

認証ユーザー名は、認証に他の (非表示の) 名前を使用する場合にのみ使用されます。これについては何もわからないため、空白のままにするか、ユーザー名の値を入力してください (どちらのオプションも同じように機能します)。

パスワードは秘密フィールドに入力してください。

答え2

これまでのところ、次のものがあります:

General
Line Enable:  yes













Share Line Appearance 
Share Ext: private
Shared User ID:
Subscription Expires: 3600  





NAT Settings 
NAT Mapping Enable: no
NAT Keep Alive Enable: yes
NAT Keep Alive Msg: $NOTIFY
NAT Keep Alive Dest: $PROXY






 Network Settings 
SIP TOS/DiffServ Value: 0x68
SIP CoS Value: 3
RTP TOS/DiffServ Value: 0xb8
RTP CoS Value: 6
Network Jitter Level: high
Jitter Buffer Adjustment: up and down







SIP Settings 
SIP Transport:UDP
SIP Port: 5060
SIP 100REL Enable:no
EXT SIP Port: 
Auth Resync-Reboot:
SIP Proxy-Require: sip.linphone.org
SIP Remote-Party-ID:no
Referor Bye Delay: 0
Refer-To Target Contact:no
Referee Bye Delay: 0
SIP Debug Option:none
Refer Target Bye Delay:0 
Sticky 183:no
Auth INVITE:no
Ntfy Refer On 1xx-To-Inv:yes
Use Anonymous With RPID:yes
Set G729 annexb:none











Call Feature Settings 
Blind Attn-Xfer Enable: yes
MOH Server: [email protected]
Message Waiting:
Auth Page:no
Default Ring:1
Auth Page Realm: 
Conference Bridge URL:
Auth Page Password: 
Mailbox ID:
Voice Mail Server: 
State Agent:
CFWD Notify Serv:no
CFWD Notifier:  







Proxy and Registration 
Proxy: <custom_domain>.onsip.com
Use Outbound Proxy: yes
Outbound Proxy: sip.onsip.com
Use OB Proxy In Dialog: yes
Register: yes
Make Call Without Reg: no
Register Expires:3600
Ans Call Without Reg:yes
Use DNS SRV:no
DNS SRV Auto Prefix:no
Proxy Fallback Intvl:3600
Proxy Redundancy Method: normal




Subscriber Information 
Display Name: <first_name>
User ID: <sip_id>
Password: ***********************
Use Auth ID:yes
Auth ID: <custom_domain>
Mini Certificate: 
SRTP Private Key:  



Audio Configuration
 Preferred Codec:  G711u
Use Pref Codec Only: no
Second Preferred Codec: unspecified
Third Preferred Codec: unspecified
G729a Enable: yes
G723 Enable: yes
G726-16 Enable: yes
G726-24 Enable: yes
G726-32 Enable:G726-40 Enable: yes
Release Unused Codec:yes
DTMF Process AVT: yes

Silence Supp Enable:no

DTMF Tx Method: auto


Dial Plan 
Dial Plan:  (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
Caller ID Map: 
Enable IP Dialing: yes
Emergency Number:

これは正しいと思います。まだすべての問題点を解決できていませんが、正しいようです。変数:

custom_domain:   this is where you have sip_id@custom_domain.onsip.com
first_name:      I think this is just for display...?
sip_id:          for sip_id@custom_domain.onsip.com

答え3

試してみたテルニクス発信通話用に Asterisk を設定すると、うまく機能します。何らかの理由で、onsip が使用するプロビジョニング ツールが機能しません。

私が実際にやったことは、電話が物理的に動作し、ネットワークに問題がないことを確認することだけです。電話には多数の設定がありますが、それらをいじるのはあまり楽しみではありません。

Asterisk 経由で CLI からダイヤルする:

mordor*CLI> 
mordor*CLI> channel originate SIP/thufir extension 18889809750@outgoing
  == Using SIP RTP CoS mark 5
    -- Called thufir
    -- SIP/thufir-0000003a is ringing
    -- SIP/thufir-0000003a answered
    -- Executing [18889809750@outgoing:1] NoOp("SIP/thufir-0000003a", "") in new stack
    -- Executing [18889809750@outgoing:2] Log("SIP/thufir-0000003a", "NOTICE, Dialing out from "" <> to 8889809750 through SIP/TELNYX") in new stack
[Jul  3 01:11:07] NOTICE[5698][C-0000002a]: Ext. 18889809750:2 @ outgoing:  Dialing out from "" <> to 8889809750 through SIP/TELNYX
    -- Executing [18889809750@outgoing:3] Dial("SIP/thufir-0000003a", "SIP/TELNYX/8889809750,60") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/TELNYX/8889809750
       > 0x7f25a0053600 -- Probation passed - setting RTP source address to 192.168.1.5:16406
    -- SIP/TELNYX-0000003b is ringing
    -- SIP/TELNYX-0000003b answered SIP/thufir-0000003a
    -- Channel SIP/thufir-0000003a joined 'simple_bridge' basic-bridge <d5b17c07-f8df-4754-bc7f-447b26b71234>
    -- Channel SIP/TELNYX-0000003b joined 'simple_bridge' basic-bridge <d5b17c07-f8df-4754-bc7f-447b26b71234>
       > Bridge d5b17c07-f8df-4754-bc7f-447b26b71234: switching from simple_bridge technology to native_rtp
       > 0x7f2590009e80 -- Probation passed - setting RTP source address to 64.16.240.36:21662
    -- Channel SIP/thufir-0000003a left 'native_rtp' basic-bridge <d5b17c07-f8df-4754-bc7f-447b26b71234>
    -- Channel SIP/TELNYX-0000003b left 'native_rtp' basic-bridge <d5b17c07-f8df-4754-bc7f-447b26b71234>
  == Spawn extension (outgoing, 18889809750, 3) exited non-zero on 'SIP/thufir-0000003a'
mordor*CLI> 

ハードフォンからのダイヤル:

mordor*CLI> 
  == Using SIP RTP CoS mark 5
    -- Executing [18888980975@myphones:1] NoOp("SIP/thufir-0000003c", "") in new stack
    -- Executing [18888980975@myphones:2] Log("SIP/thufir-0000003c", "NOTICE, Dialing out from "thufir" <thufir> to 8888980975 through SIP/TELNYX") in new stack
[Jul  3 01:11:41] NOTICE[5702][C-0000002b]: Ext. 18888980975:2 @ myphones:  Dialing out from "thufir" <thufir> to 8888980975 through SIP/TELNYX
    -- Executing [18888980975@myphones:3] Dial("SIP/thufir-0000003c", "SIP/TELNYX/8888980975,60") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/TELNYX/8888980975
    -- SIP/TELNYX-0000003d is ringing
    -- SIP/TELNYX-0000003d is making progress passing it to SIP/thufir-0000003c
    -- SIP/TELNYX-0000003d answered SIP/thufir-0000003c
    -- Channel SIP/thufir-0000003c joined 'simple_bridge' basic-bridge <b113ae45-d191-4b8a-99fa-6f1aeba4a8dc>
    -- Channel SIP/TELNYX-0000003d joined 'simple_bridge' basic-bridge <b113ae45-d191-4b8a-99fa-6f1aeba4a8dc>
       > Bridge b113ae45-d191-4b8a-99fa-6f1aeba4a8dc: switching from simple_bridge technology to native_rtp
       > 0x7f25d000cbe0 -- Probation passed - setting RTP source address to 192.168.1.5:16408
       > 0x7f25fc0055f0 -- Probation passed - setting RTP source address to 64.16.240.36:24202
    -- Channel SIP/thufir-0000003c left 'native_rtp' basic-bridge <b113ae45-d191-4b8a-99fa-6f1aeba4a8dc>
    -- Channel SIP/TELNYX-0000003d left 'native_rtp' basic-bridge <b113ae45-d191-4b8a-99fa-6f1aeba4a8dc>
  == Spawn extension (myphones, 18888980975, 3) exited non-zero on 'SIP/thufir-0000003c'
mordor*CLI> 
mordor*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      
TELNYX/TELNYX             192.76.120.10                               Yes        Yes            5060     OK (105 ms)                                  
demo_alice                (Unspecified)                            D  Yes        Yes            0        UNKNOWN                                      
demo_bob                  (Unspecified)                            D  Yes        Yes            0        UNKNOWN                                      
hawat/hawat               (Unspecified)                            D  Yes        Yes            0        UNKNOWN                                      
thufir/thufir             192.168.1.5                              D  Yes        Yes            5062     OK (9 ms)                                    
5 sip peers [Monitored: 2 online, 3 offline Unmonitored: 0 online, 0 offline]
mordor*CLI> 
mordor*CLI> dialplan show 
__func_periodic_hook_context__  ael-builtin-h-bubble            ael-default                     ael-demo                        
ael-dundi-e164                  ael-dundi-e164-canonical        ael-dundi-e164-customers        ael-dundi-e164-local            
ael-dundi-e164-lookup           ael-dundi-e164-switch           ael-dundi-e164-via-pstn         ael-iaxprovider                 
ael-iaxtel700                   ael-international               ael-local                       ael-longdistance                
ael-std-exten-ael               ael-trunkint                    ael-trunkld                     ael-trunklocal                  
ael-trunktollfree               chanvar                         default                         demo                            
globals                         local                           outgoing                        parkedcalls                     
public                          myphones                        
mordor*CLI> 
mordor*CLI> dialplan show globals
   TOLL=SIP/TELNYX
   OUTBOUND-TRUNKMSD=1
   OUTBOUND-TRUNK="Zap/g2"
   IAXINFO-AEL=guest
   CONSOLE-AEL="Console/dsp"

    -- 5 variable(s)
mordor*CLI> 
mordor*CLI> dialplan show myphones
[ Context 'myphones' created by 'pbx_config' ]
  '1000' =>         1. Dial(SIP/1000)                             [pbx_config]
                    2. Hangup()                                   [pbx_config]
  '1001' =>         1. Dial(SIP/1001)                             [pbx_config]
                    2. Hangup()                                   [pbx_config]
  '201' =>          1. Answer()                                   [pbx_config]
                    2. Playback(tt-monty-knights)                 [pbx_config]
                    3. Hangup()                                   [pbx_config]
  '202' =>          1. Answer()                                   [pbx_config]
                    2. Playback(welcome)                          [pbx_config]
                    3. Playback(demo-echotest)                    [pbx_config]
                    4. Echo()                                     [pbx_config]
                    5. Playback(demo-echodone)                    [pbx_config]
                    6. Playback(vm-goodbye)                       [pbx_config]
                    7. Hangup()                                   [pbx_config]
  '4000' =>         1. Playback(tt-monkeys)                       [pbx_config]
  '5000' =>         1. Playback(tt-monkeysintro)                  [pbx_config]
  '555' =>          1. Playback(hello-world)                      [pbx_config]
                    2. Playback(echo-test)                        [pbx_config]
                    3. Echo()                                     [pbx_config]
                    4. Playback(demo-echodone)                    [pbx_config]
  '6001' =>         1. Dial(SIP/demo_alice,20)                    [pbx_config]
  '6002' =>         1. Dial(SIP/demo_bob,20)                      [pbx_config]
  '6003' =>         1. Dial(SIP/thufir,20)                        [pbx_config]
  '6004' =>         1. Dial(SIP/hawat,20)                         [pbx_config]
  Include =>        'outgoing'                                    [pbx_config]

-= 11 extensions (24 priorities) in 1 context. =-
mordor*CLI> 
mordor*CLI> dialplan show outgoing
[ Context 'outgoing' created by 'pbx_config' ]
  '_1NXXNXXXXXX' => 1. NoOp()                                     [pbx_config]
                    2. Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} through ${TOLL}) [pbx_config]
                    3. Dial(${TOLL}/${EXTEN:1},60)                [pbx_config]
                    4. Playtones(congestion)                      [pbx_config]
                    5. Hangup()                                   [pbx_config]

-= 1 extension (5 priorities) in 1 context. =-
mordor*CLI> 

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