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Unread 08-28-2008, 03:12 AM   #1
paulies
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Hi all,

I regiserted my voip provider with a premium account so I can receive incoming calls my account is setup correctly in my ata and the control panel in voxalot says it is registered but my phone just does not ring.

Anybody got any idea what I could be doing wrong ?

Thanks Paul
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Unread 08-28-2008, 07:02 AM   #2
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What kind of ATA or SIP phone are you using? If it is behind a NAT router, are you using NAT_keep_alive or forwarding the SIP port?
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Unread 08-28-2008, 01:55 PM   #3
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Hi,

I'm using pap2 behind a router it works perfectly fine with mysipswitch but I get nothing through voxalot . For some reason when I dial the number from my mobile it tells me I dont have enough funds very strange .

Any other ideas, Nat keep alive is enabled.

Have I wasted $15

Thanks Paul
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Unread 08-28-2008, 07:22 PM   #4
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Are you able to receive any calls on your Voxalot number, such as calls from another Voxalot user or calls from Sipbroker PSTN numbers?

I don't think you have a NAT issue, but just in case, here are the settings to use NAT_Keep_Alive when ATA is behind one or more NAT routers.

(under SIP tab)
Handle_VIA_received: yes
Handle_VIA_rport: yes
Insert_VIA_received: yes
Insert_VIA_rport: yes
Substitute_VIA_Addr: yes
Send_Resp_To_Src_Port: yes
STUN_Enable: yes
STUN_Test_Enable: yes
STUN_Server: stun.voxalot.com:3478 (or any STUN server such as 'stun01.sipphone.com:3478' or 'stun.sipgate.net:10000')
NAT_Keep_Alive_Intvl: 179 (if your phone does not always ring on incoming calls try 119 or 59, use highest number that works)

(under Line_1 and Line_2 (or PSTN_Line) tabs)
NAT_Mapping_Enable: yes
NAT_Keep_Alive_Enable: yes
NAT_Keep_Alive_Msg: 0000
NAT_Keep_Alive_Dest: $PROXY
Register Expires: 3600

----------

Optional settings, steps 1 - 4:
Sometimes its preferable to operate the ATA independently of a STUN server. The benefit is that phone service can continue no matter if the STUN server is working. See steps 1 to 4 below.

1. Forward the SIP ports and the RTP port range from the router to the ATA.
2. Set "STUN Enable:" no
3. Set "NAT Keep Alive Enable:" no
4. The ATA must learn it's public IP address, either through normal SIP registration, or place your public IP address in the 'EXT IP:' field).

Last edited by boatman; 10-15-2009 at 09:47 PM.
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Unread 08-28-2008, 11:47 PM   #5
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Just to expand further on what BoatMan has above:

-A Screen by screen of PAP2 setup:
Setting up Linksys PAP2 - Voxalot FAQ

-How to test calling

Incoming calls

1. Call any SipBroker Access Number Local to you (see SIPBroker - PSTN Numbers ). When prompted enter *010123456 (replace 123456 with your actual 6 digit VoXalot number). Does that call come in, what's the audio like.

2. Go to SIPBroker - EziDial . Enter your VoXalot SIP URI in the field (eg. 123456@voxalot.com) and dial, do you receive the call from the Echo test Application? Is audio ok during the test?

Outgoing Calls

1. Will a call *500 or 500 reach the VoiceMail server?

2. Will a call to *600 or 600 reach the Echo Test server?


Also, is there any forwarding rules set in VoXalot that could be causing this "not enough funds" message to come from the forwarding provider?
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