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Grandstream GX2000 PlusNet SIP accounr

adcombe
Newbie
Posts: 1
Registered: 17-10-2008

Grandstream GX2000 PlusNet SIP accounr

Hi, does anybody know I can register a Grandstream GX2000 on the Plusnet bband phone SIP account. I have tried to use the setting provided but am obviously missing something as it never registers. Successfully set up the device for SIPGATE and DRAYTEL account - any ideas?
Any help much appreiated.
Man ythanks  Smiley
2 REPLIES
Community Veteran
Posts: 6,256
Thanks: 432
Fixes: 39
Registered: 30-07-2007

Re: Grandstream GX2000 PlusNet SIP accounr

Gradwell have a setup guide here :- https://portal.gradwell.net/support.php/knowledgebase/view/category_id/34/article_id/224
For sip server use :- sip.plus.net:5060
For outbound proxy use :- nat.plus.net:5082 OR natproxy.plus.net:5082
And your PlusNet SIPid and SIP Password.
If this doesnt work and your SIP id starts with a 4 then try dropping the 4 and using the last 6 digits.
Hope that helps
puddy
Grafter
Posts: 1,571
Registered: 10-06-2007

Re: Grandstream GX2000 PlusNet SIP accounr

If droping the 4 does not work here are the working settings for my Grandstream GXP2000     
Regards
Puddy                                     
Settings For  Grandstream GXP2000
Account Name : Plusnet Voip
SIP Server: sip.plus.net
Outbound Proxy:natproxy.plus.net:5082
Sip User ID: 6Huh?Huh
Authenticate ID Huh?Huh
Authenticate Password Huh?Huh
Name: name (shows on display)
Use DNS SRV =no
User ID is phone number =no
Sip Reg = yes
Unregister On Reboot =no
Register expiration 60
Local SIP port: 5060
SIP T1 Timeout:1sec
SIP T2 Interval: 4sec
Sip Transport:=UDP
Use RFC3581 Symmetric Routing =no
Nat Traversal (STUN) =no
SUBSCRIBE for MWI:=no
PUBLISH for Presence:=no
Proxy-Require : blank
SendDTMF: Via RTP (RFC
Early Dial= no
Enable call feature = no
Session Exp =180
Min-SE =90
Caller req timer =no
Callee req Ref =no
Force timer =no
UAC Spec Ref =omit
UAS Spec Ref =UAC
Force INVITE  =no
Enable 100rel:=no
Account ring tone= you choose
Send Anonymous:= you choose/no
Anonymous Method: =use from header
Anonymous Call Rej=no
Auto Answer:=NO
Allow Auto Ans byCall:=no
Turn Off Speaker =no
Check SIP User ID: =no
Refer to Use Target:=no
Choice 1:=PCMU
2 TongueCMU
3: G.729A/B
4: G.723.1
5: G.726-32
6: iLBC
7:G.722(wide Band)
8:GSM
SRTP mode:=Disable
eventlist=blank
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