como provisiono um telefone para getonsip?

0

No provisionamento de Cisco SPA 942 hardphone, da marca Linksys, como faço para concluir a configuração para getonsip (ou onsip )?

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

Na guia SIP :

SIP Parameters

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

Na guia EXT 1 , há:

SIP settings

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

Além disso, na guia Ext 1 , há:

Proxy and Registration

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

Mas não tenho certeza de onde vão os Auth Username e a senha da onsip. Notavelmente, eles usam getonsip.com no endereço SIP e sip.onsip.com no proxy.

    
por Thufir 22.06.2016 / 13:43

3 respostas

1

Melhor opção - entre em contato com o suporte da onsip.com.

Nome de usuário de autenticação usado SOMENTE se você quiser usar para autenticação outro nome (oculto). Como você não sabe nada sobre isso, deixe em branco ou coloque o valor do nome de usuário nele (as duas opções funcionarão da mesma forma).

A senha foi colocada em um campo secreto.

    
por 22.06.2016 / 17:50
0

Isso é o que eu tenho até agora:

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:

que eu acho correto. Eu ainda não tenho todas as dobras funcionadas, mas isso parece correto. Variáveis:

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
    
por 28.06.2016 / 09:45
0

Eu testei o telnyx e configurei o Asterisk para chamadas de saída, funciona como um encanto. Por alguma razão, a ferramenta onsip da ferramenta de provisionamento simplesmente não funciona.

Tudo o que eu realmente fiz foi estabelecer que o telefone funciona fisicamente e que não há problemas com a rede. Há uma infinidade de configurações no telefone, não estou realmente ansioso para sair com elas.

Discando da CLI pelo Asterisk:

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> 

Discando do hardphone:

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> 
    
por 03.07.2016 / 10:37