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.
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.
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
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>
Tags networking phone voip sip asterisk