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Unread 01-25-2008, 11:59 AM   #1
philled
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Default Calling my Voxalot number with SIP URI

http://www.voxalot.com/action/tutorial?itemOID=68 says that I should be able to make a SIP URI call to my Voxalot number, but I can't. I've tried dialing xxxxxx@au.voxalot.com but it goes to voicemail (which I have disabled by the way - go figure!) rather than ringing my ATA.

My ATA is a SPA3000 which has my Voxalot account registered on the PSTN tab (I have a different VSP registered on the Line 1 tab so I can receive calls on that VSP's DID).

Does anyone know why the SIP URI dialing doesn't work? I'm calling it from a softphone on my PC which is registered to a different VSP.
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Unread 01-25-2008, 12:12 PM   #2
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Quote:
Originally Posted by philled View Post
Does anyone know why the SIP URI dialing doesn't work? I'm calling it from a softphone on my PC which is registered to a different VSP.
May be answering my own question here. Is is because I'm only on VoxBasic?
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Unread 01-25-2008, 01:39 PM   #3
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No that isn't. Must be possible with VoxBasic.
I think it have to do with use of same ports softphone and ata, but I am do not much about that.
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Unread 01-25-2008, 05:56 PM   #4
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Are you calling VoXID@au.voxalot.com from that same acct?

Try calling one of the SIPBroker access numbers (from a regular PSTN line), then enter *010VoXID, does it work fine then

Also, go to SIPBroker - EziDial ,enter your SIP address and press dial....you should receive a Call from the Echo test server...
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Unread 01-26-2008, 07:19 AM   #5
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I think PSTN line of SPA 3000|3102 never rings, only line 1.

Use line 1 or a softphone to try.
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Unread 01-26-2008, 07:32 AM   #6
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Have you tried xxxxxx@voxalot.com instead of au.voxalot.com?
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Unread 01-26-2008, 10:32 AM   #7
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Quote:
Originally Posted by Jorge View Post
I think PSTN line of SPA 3000|3102 never rings, only line 1.
Use line 1 or a softphone to try.
Unfortunately I can't use Line 1 as I have a VSP on there that has an incoming DID and it has to be registered
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Unread 01-26-2008, 03:33 PM   #8
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I think Jorge is correct. The PSTN line account is probably for bridging calls from VoIP to the local PSTN line. However the SPA310x should still pick up the call and wait for an access code to connect to the PSTN line. So it may also be a set up issue in the ATA which is part of the problem. I do not have first hand experience with this device, but I am assuming that it behaves similarly to a GS HT-488.

Also, to avoid port conflict, I find it better to offset the SIP (starting at 5060) and RTP (starting at 5004) ports by a count 2 for any SIP device behind the same NAT with the same public IP address.
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