Al aprovisionar unCisco SPA 942teléfono duro, marca Linksys, ¿cómo termino la configuración?sorber(osorbete)?
SIP Address: [email protected]
Username: foo
Domain: getonsip.com
SIP Password: GHdjlRBfjdklHWD
Auth Username: getonsip_foo
Outbound Proxy: sip.onsip.com
En la SIPpestaña:
SIP Parameters
SIP Server Name: getonsip.com
SIP User Agent Name: foo
SIP Reg User Agent Name: getonsip_foo
En la EXT 1pestaña hay:
SIP settings
SIP Port:
EXT SIP Port:
SIP Proxy-Require:
También en la Ext 1pestaña hay:
Proxy and Registration
Proxy:
Use Outbound Proxy:
Outbound Proxy:
Use OB Proxy In Dialog:
Pero no estoy muy seguro de dónde Auth Usernamevan la contraseña y la contraseña de onsip. Cabe destacar que se utilizan getonsip.comen la dirección SIP y sip.onsip.comen el proxy.
Respuesta1
La mejor opción: comuníquese con el soporte de onsip.com.
El nombre de usuario de autenticación se utiliza SÓLO si desea utilizar otro nombre (oculto) para la autenticación. Como no sabes nada sobre eso, déjalo en blanco o ponle el valor del nombre de usuario (ambas opciones funcionarán igual).
La contraseña se debe ingresar en un campo secreto.
Respuesta2
Esto es lo que tengo hasta ahora:
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:
lo cual creo que es correcto. No he resuelto todos los problemas, pero esto parece correcto. Variables:
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
Respuesta3
lo probételnyxy configurar Asterisk para llamadas salientes, funciona de maravilla. Por alguna razón, la herramienta de aprovisionamiento que utiliza onsip simplemente no funciona.
Todo lo que realmente hice fue establecer que el teléfono funciona físicamente y que no hay problemas con la red. Hay una multitud de configuraciones en el teléfono, no esperes jugar con ellas.
Marcando desde el CLI a través de 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>
Marcando desde el teléfono fijo:
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>