При предоставленииCisco СПА 942аппаратный телефон, фирменный Linksys, как мне завершить настройкуgetonsip(илиглоток)?
SIP Address: [email protected]
Username: foo
Domain: getonsip.com
SIP Password: GHdjlRBfjdklHWD
Auth Username: getonsip_foo
Outbound Proxy: sip.onsip.com
На SIPвкладке:
SIP Parameters
SIP Server Name: getonsip.com
SIP User Agent Name: foo
SIP Reg User Agent Name: getonsip_foo
На EXT 1вкладке есть:
SIP settings
SIP Port:
EXT SIP Port:
SIP Proxy-Require:
Также во Ext 1вкладке есть:
Proxy and Registration
Proxy:
Use Outbound Proxy:
Outbound Proxy:
Use OB Proxy In Dialog:
Но я не совсем уверен, куда Auth Usernameидут и пароль от onsip. В частности, они используются getonsip.comв SIP-адресе и sip.onsip.comв прокси.
решение1
Лучший вариант — обратиться в службу поддержки onsip.com.
Имя пользователя аутентификации используется ТОЛЬКО если вы хотите использовать для аутентификации другое (скрытое) имя. Поскольку вы ничего об этом не знаете, оставьте его пустым или введите в него значение имени пользователя (оба варианта будут работать одинаково).
Пароль необходимо ввести в секретное поле.
решение2
Вот что у меня есть на данный момент:
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:
что я думаю правильно. Я не совсем разобрался со всеми загвоздками, но это кажется правильным. Переменные:
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
решение3
Я попробовалтельныкси настроить Asterisk для исходящих звонков, работает как часы. По какой-то причине инструмент подготовки, который использует onsip, просто не работает.
Все, что я действительно сделал, это установил, что телефон физически работает и что нет никаких проблем с сетью. На телефоне есть множество настроек, не надейтесь на то, что будете с ними возиться.
Набор номера из CLI через 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>
Набор номера с стационарного телефона:
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>