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|  | Onesuite VoIP and ArtDio IP phone | |  |
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Our entry level VoIP calculator allowing you to easily calculate your Voice over
IP bandwidth requirements.
Available for immediate download at only 99 US Dollars.
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Design and use your own VoIP calculators using drag and drop techniques.
We used this software to design our own VoIP calling
service. Now, we offer it for everyone to use.
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Simplifies the complicated design process involved in creating effective voice
networks.
Support both traditional circuit switching techniques and Voice over IP.
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| | This might be too late but try thess settings.
Onesuite VOIP configurations for Trixbox/Asterisk/ATA
Gateway: userid@voip.onesuite.com
GW1 AuthID: onesuite username
GW1 Password: onesuite SA password
GW! Nat Mapping Enable: YES
Proxy: voip.onesuite.com
UserID: myuserid
Password: onesuite_broadband_password
Display Name: MyName
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PROXY DOMAIN : voip.onesuite.com
PORT : 5060
DTMF : rfc2833/rtp
RTP PAYLOAD : 101
REGISTRATION TIMER : 180 seconds
AUDIO CODEC : GSM, G711U
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Display Name: myuserid
User Id: myuserid
Password: mypassword
Proxy: voip.onesuite.com
Use Auth Id: No
Register Expires: 3600
Use Outbound Proxy: No
NAT Mapping Enable: Yes
NAT KeepAlive: Yes
On the SipPage Tab under NAT Support Parameters
Substitute Via Address: Yes
Send Response to Src Port: Yes
STUN Enable: Yes
STUN Test Enable: No
STUN Server: stun.softjoys.com
Nat Keep Alive Intvl: 15
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This works for VM Trixbox (EasyPBX)
Trunk Name: OneSuite
Peer Details:
dtmfmode=rfc2833
fromdomain=voip.onesuite.com
fromuser=myusername
host=voip.onesuite.com
secret=mypassword
type=friend
username=myusername
Incoming Settings: [Blank]
Register Settings: [Blank]
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sip.conf
[general] ;reg context etc are set too.
tos=184 ; Set IP QoS to either a keyword or numeric val
tos=lowdelay ;lowdelay, throughput, reliability, mincost, none
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=alaw
allow=gsm
[onesuite]
type=friend
username=userID
secret=secret1
fromuser=userID
fromdomain=voip.onesuite.com
host=voip.onesuite.com
;callerid="abc Family"
;
extensions.conf
[onesuite-forced]
;exten => _69XXXXXXXXXXX,1,SetCallerID(2021234567) ; My phone number with this provider
;exten => _69XXXXXXXXXXX,n,SetCIDName("John Doe")
exten => _69XXXXXXXXXXX,1,Dial(SIP/${EXTEN:2}@onesuite,33,Ttr)
exten => _69XXXXXXXXXXX,n,Playback(tt-weasels)
exten => _69XXXXXXXXXXX,104,Playback(tt-monkeys)
---------------------------------------------------------------------------------------------------------------- | | Hi guys, I am a Onesuite VoIP user and so far it works like a charm using it on my computer (SJphone as a software).
But I was thinking of using it with a stand alone IP phone and my brother has a spare ArtDio IPF-2002L that I can use.
The problem is I can't seem to find the right settings. Maybe someone here is using Onesuite or ArtDio and and give a pointer or 2.
Thanks in advance. |
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