Voxalot / SIP Broker Support Forums - View Single Post - Can't get voice packets to bypass VoXaLot

jerm · 11-28-2007, 12:20 PM

I clearly do not know enough about SIP, so excuse the possibly naive question.

I am trying to get an SPA3000 (ATA) on my PSTN line to call an SPA941 (IP phone) on the same subnet, with the ability to fallback to voicemail, on the voxalot account the SPA941 is registered to. I have tried many things.

One way I know I can get this to work (it worked on another VSP) is by having the SPA3000 'hotline' the call via a 2nd voxalot account to the account the SPA941 is registered to.

The problem here is the inefficiency of having all of this voice traffic (between two devices on the same subnet) traversing the net (and it is doubled up, right?).

My question is this : given that I have full control over the router, dns, phone settings etc. is there any setup that would allow the voice traffic to take the short route (directly across my subnet) between the two devices after the call setup has completed?

Thanks for any suggestions.

11-28-2007, 12:20 PM	#30
jerm Junior Member Join Date: Nov 2007 Posts: 11 Thanks: 0 Thanked 0 Times in 0 Posts	I clearly do not know enough about SIP, so excuse the possibly naive question. I am trying to get an SPA3000 (ATA) on my PSTN line to call an SPA941 (IP phone) on the same subnet, with the ability to fallback to voicemail, on the voxalot account the SPA941 is registered to. I have tried many things. One way I know I can get this to work (it worked on another VSP) is by having the SPA3000 'hotline' the call via a 2nd voxalot account to the account the SPA941 is registered to. The problem here is the inefficiency of having all of this voice traffic (between two devices on the same subnet) traversing the net (and it is doubled up, right?). My question is this : given that I have full control over the router, dns, phone settings etc. is there any setup that would allow the voice traffic to take the short route (directly across my subnet) between the two devices after the call setup has completed? Thanks for any suggestions.