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Unread 07-23-2007, 10:35 AM   #21
v164
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Default Pennytel various tests

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Originally Posted by v164 View Post
I also tried:

Atcom AT-530 -> PBXes -> VoXaLot -> Pennytel

and the results were the same (haven't tried the reverse order though).

I've now tried the reverse order:


Atcom AT-530 -> VoXaLot (us.voxalot.com) -> PBXes -> Pennytel

and the results are the same.


I've also tried:

Atcom AT-530 -> VoXaLot (us.voxalot.com) -> PBXes -> VoXaLot( 2nd account ) (au.voxalot.com) -> Pennytel

and the results are also the same (outbound audio automatically unproxied, inbound audio unproxied only after manually pushing the "Hold" button).

(This is with "Optimize Audio Path" / "Audio Bypass" set to "Yes" everywhere).



I suppose this means that sip.pennytel.com is not responding (or not correctly responding) to VoXaLot's re-INVITE.

Is this behaviour allowable according to RFC 3261 (if not, we could reasonably petition Pennytel to fix it), or do we just have to live with it?

(edited: even if it's not required for RFC 3261 conformance, we could probably ask Pennytel to consider it. They seem to be keen to cooperate.)

Last edited by v164; 07-23-2007 at 11:17 AM.
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Unread 07-23-2007, 10:53 AM   #22
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Quote:
Originally Posted by v164 View Post
Is this behaviour allowable according to RFC 3261 (if not, we could reasonably petition Pennytel to fix it), or do we just have to live with it?

(edited: even if it's not required for RFC 3261 conformance, we could probably ask Pennytel to consider it. They seem to be keen to cooperative.)
We are more than happy to assist them in achieving this goal.
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Unread 07-28-2007, 12:06 PM   #23
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anyone could help me with this?

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this is working perfectly with my present setup of spa3k but if im out and place a call via pstn+spa3k at home i get only one way audio.they can't hear me.
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Unread 07-28-2007, 12:13 PM   #24
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anyone could help me with this?
I'm not a Sipura expert however I will take a stab at a suggestion. Is there some thing like a pass thru setting in this device?

I would hazzard a guess that you need to allow the PSTN interface to pass thru to the VoIP interface.

Hopefully a Sipura expert can add to this.
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Unread 07-28-2007, 03:32 PM   #25
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yes.i already using it. i did every concievable thing and everything working beyond my expectations and this is the last remaining missing piece
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Unread 07-28-2007, 03:42 PM   #26
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yes.i already using it. i did every concievable thing and everything working beyond my expectations and this is the last remaining missing piece
Does *600 provide 2 way audio when you dial in via PSTN?
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Unread 07-28-2007, 04:22 PM   #27
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you mean outside/remote pstn? no.
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Unread 07-29-2007, 06:59 AM   #28
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here's what i noticed but remember this problem was caused only when i tried the new feature of bypassing voxalot...the setting to No of "Symmetric NAT Handling"

setting that not working
1.voxalot registered in line 1 and as @gw1
2. dial plan (xx.<:@gw1>) in "pstn line"

but if i use another account as @gw2 with the following setting
1. voxalot still registered in line 1
2. create a second gateway with differrent account and NAT mapping = yes
3. dial plan (xx.<:@gw2>) in "pstn line"
this one works

may questions now is...
the setup as i describe as the one working..did i really bypass voxalot proxying or what i did was i end up in the same scenario im avoiding and resulting to the exact same thing as if im not bypassing voxalot?
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Unread 07-29-2007, 07:12 AM   #29
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may questions now is...
the setup as i describe as the one working..did i really bypass voxalot proxying or what i did was i end up in the same scenario im avoiding and resulting to the exact same thing as if im not bypassing voxalot?
For the 2nd account setup on gw2 is:

1. The symmetric NAT set to "No"
2. The providers setup within that account have "Optimize Audio Path" set to "Yes"
3. If yes to both 1 and 2 and you dial *600 and get 2 way audio and you make a call via one of the providers setup in step 2 then, yes there is an excellent chance your voice stream is bypassing our servers.
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Unread 11-28-2007, 12:20 PM   #30
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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.
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