Wie kann ich ein festes Telefon für getonsip bereitstellen?

Beste Option - Wenden Sie sich an den Support von onsip.com.

Auth Benutzername wird NUR verwendet, wenn Sie zur Authentifizierung einen anderen (verborgenen) Namen verwenden möchten. Da Sie nichts davon wissen, lassen Sie das Feld leer oder geben Sie den Wert des Benutzernamens ein (beide Optionen funktionieren gleich).

Passwort muss in geheimes Feld gestellt werden.

Das habe ich bisher:

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: moh@linux.onsip.com 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: 

was ich für richtig halte. Ich habe nicht alle Knicke geklärt, aber das scheint richtig zu sein. Variablen:

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 

Ich habe Telnyx ausprobiert und Asterisk für ausgehende Anrufe eingerichtet, funktioniert wie ein Zauber. Aus irgendeinem Grund funktioniert das Provisioning-Tool, das onsip verwendet, einfach nicht.

Alles, was ich wirklich getan habe, ist festzustellen, dass das Telefon physisch funktioniert und es keine Probleme mit dem Netzwerk gibt. Es gibt eine Vielzahl von Einstellungen auf dem Telefon, freue mich nicht wirklich darauf, mit ihnen herumzuspielen.

Wählen von der CLI über 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>  

Wählen vom Hardphone aus:

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 $ to $ through $) [pbx_config] 3. Dial($/$,60) [pbx_config] 4. Playtones(congestion) [pbx_config] 5. Hangup() [pbx_config]  -= 1 extension (5 priorities) in 1 context. =- mordor*CLI>