0

In provisioning a Cisco SPA 942 hardphone, branded Linksys, how do I finish the setup for getonsip (or onsip)?

SIP Address: foo@getonsip.com
Username: foo
Domain: getonsip.com
SIP Password: GHdjlRBfjdklHWD
Auth Username: getonsip_foo
Outbound Proxy: sip.onsip.com

In the SIP tab:

SIP Parameters

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

In the EXT 1 tab there's:

SIP settings

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

Also in the Ext 1 tab there's:

Proxy and Registration

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

But I'm not quite sure where the Auth Username and password from onsip go. Notably, they use getonsip.com in the SIP address and sip.onsip.com in the proxy.

1

Best option - contact onsip.com support.

Auth username used ONLY if you want use for authentification other(hidden) name. Since you know nothing about that, leave it blank or put value of username in it(both option will work same).

Password have be put in secret field.

0

This is what I have so far:

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:

which I think is correct. I haven't quite got all the kinks worked out, but this seems correct. 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
0

I tried out telnyx and setup Asterisk for outbound calling, works like a charm. For whatever reason, the provisioning tool onsip uses just doesn't work.

All I've really done is to establish that the phone physically works and that there are no problems with the network. There are a multitude of settings on the phone, don't really look forward to mucking about with them.

Dialing from the CLI through 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> 

Dialing from the 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> 

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