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Unread 07-21-2006, 07:48 PM   #1
nobre84
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Default Inbound calls problem

Hi guys, all my outbound calls are working great, the dial plans options and flexibility are awesome... But I still can't receive any calls...
I am currently registered with my Ata on voxalot and voxalot connected to voipdiscount.
When I log into a softphone from another voxalot account and dial my number, the call goes straight to my voice mail.
If I register to voxalot on my Ata, and set up Registering to voipdiscount in my profile, or even if I directly register to voipdiscount in my Ata, and try a direct call from either softphone or voipdiscount dialer, my Ata shows this message in debug mode:

7 <SIP 1> : Receive INVITE Request
38 <NetCon 1> : Found inbound voip peer by answer-address id(100)
39 <Call 1> : From Net - calledParty() callingParty(tanianobre)
40 <Call 1> : Terminated from(fffffff7) this(Local:InvalidNumber) b
efore(NULL) forced(0)
41 <NetEP 1> : Call TO <tanianobre> terminated reason Local:InvalidNumber)
42 <SIP 1> : Transaction Server (0 INVITE) Timeout (retry #1)
43 <SIP 1> : Send 404 Response

I'm trying to mail addpac support as they have helped me before on other issues, but shouldn't the Ata at least give the same error message when direct calling my 6 number ID from the softphone?
Thanks
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Unread 07-24-2006, 02:53 PM   #2
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Is there anything I should be looking to fix this? I am registered to voxalot in my ATA, but the calls always go straight to voice mail :/
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Unread 07-28-2006, 11:53 PM   #3
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Quote:
Originally Posted by nobre84
Is there anything I should be looking to fix this? I am registered to voxalot in my ATA, but the calls always go straight to voice mail :/
Are you behind a home/NAT router? If so, how did you setup your adapter and router? VoIP adapters require special settings to properly run behind NAT routers, and sometimes you also need to make security changes on the router to let in the traffic.

Remember, from the standpoint of your router, any inbound call you are receiving is inbound traffic coming to you, that you didn't "ask" for. So it's very common for incorrectly setup VoIP adapters and/or routers to let you call out (because you initiated the call), but block other people's attempts to call you (i.e. have your LAN see the incomming call as if it were an "internet attack")...
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Unread 07-30-2006, 02:11 AM   #4
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Quote:
Originally Posted by DracoFelis
Are you behind a home/NAT router? If so, how did you setup your adapter and router? VoIP adapters require special settings to properly run behind NAT routers, and sometimes you also need to make security changes on the router to let in the traffic.

Remember, from the standpoint of your router, any inbound call you are receiving is inbound traffic coming to you, that you didn't "ask" for. So it's very common for incorrectly setup VoIP adapters and/or routers to let you call out (because you initiated the call), but block other people's attempts to call you (i.e. have your LAN see the incomming call as if it were an "internet attack")...
hi, thanks for the input
im using speedstream 5200 in router mode, running at DMZ with 192.168.254.1 (my computer), the ATA is 192.168.254.2 , what should I do to proper receive inbound calls? Does forwarding 5060udp to ATA overrides the dmz running to the computer?
What other possible ports should I send to ata? there are currently no firewall settings running on any of them.
Cya
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Unread 07-30-2006, 05:25 AM   #5
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I think your first mistake is putting the computer in the DMZ , this effectively sends all inbounds to your computer.

If you MUST have your computer in the DMZ then forward both the SIP and RTP ports to the Telephone adapter

In a Sipura look for RTP Min and RTP max those are the RTP Ports, on the SIP Page usually 16384 to 16???. The SIP Ports are on the Line 1 and line 2 Pages usually 5060 and 5061. If you do this correctly toy will not need STUN or any other solution.

Last edited by MarkosJal; 07-30-2006 at 06:06 PM.
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Unread 07-30-2006, 07:45 AM   #6
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Quote:
Originally Posted by MarkosJal
I think your first mistake is putting the computer in the DMX , this rffectively sends all inbounds to your computer.

If you MUST have your computer in the DMZ then forward both the SIP and RTP ports to the Telephone adapter

In a Sipura look for RTP Min and RTP max those are the RTP Ports, on the SIP Page usually 16384 to 16???. The SIP Ports are on the Line 1 and line 2 Pages usually 5060 and 5061. If you do this correctly toy will not need STUN or any other solution.
I've done the following:
UDP 5060-5070 to ATA ip
UDP 8766-35000 to ATA ip

Calls still go straight to mail. How can I track the problem, is there a "debug mode" in the website?
PS: when directly registered to voipdiscount I receive the Sip Invite from softphone
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Unread 07-30-2006, 06:06 PM   #7
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Can you receive calls via voxalot?

If not, try deactivating the DMZ and see what happens
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Unread 07-31-2006, 02:53 AM   #8
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Those on the last post are already with DMZ off, I manually opened all the ports I need to my pc, and these to the ATA
I can call one voxalot account to another in 2 softphones, and also can dial from my ata to a voxalot account logged in a softphone, but my ata won't receive a sip INVITE from the softphone, it goes directly to mail
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Unread 09-04-2006, 06:51 PM   #9
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Same problem here, and I'm using a STUN server. My FWD calls are going into Voxalot voicemail.
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