VTBrendan
Viatalk
Premium,VIP
join:2005-06-27
Clifton Park, NY
4 edits | New VT Beta Available For Testing
Hey All,
We have a new beta server going live today for testing if any of you are interested in playing around with it. The location is beta-1.vtnoc.net, and you can try it out by simply switching your device to connect to that location. This is a completely different system from the switches currently in production, and is something that has been in the works for quite a while.
This system does not relay audio through our servers by default, however if you have any one way audio issues, *51 turns audio relaying on, and *50 turns it off.
At the current time a few features such as the two line feature (incoming only), call forwarding, call hunt, and DND forward will not work when connected to this switch. Other features should work the same as they do everywhere else.
Please post feedback/comments/questions here. Thanks!
edit: The * codes need to be dialed independently, not as a pre-pendage to a number. Think of them as an on-off switch, not a per call thing. After this goes through, the setting will change and you will get a busy signal.
-Brendan |
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dcurrey
Premium
join:2004-06-29
1 edit | Might as well test that out also.
Dead air or fast busy after dialing.
Placed pap2 in dmz. I have had trouble with last beta server required port forwards to work. Didn't help |
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VTBrendan
Viatalk
Premium,VIP
join:2005-06-27
Clifton Park, NY
1 edit | Hi,
Did it register okay? Try doing the *51 thing. Also, were you dialing off network or on?
-Brendan |
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dcurrey
Premium
join:2004-06-29
 ·ViaTalk
4 edits | It took it a few seconds but it registered. Currently it now shows "Can't connect to login server"
Reboot put it back online.
Tried off net call and tried to call voicemail *123
I have tried entering the *51/*50 codes during the call and before dialing ie: *511513xxxyyyy.
Your post wasn't clear on how to do it.
Ok I just remember in addition to the port forwards from before with the beta server I had to set "NAT Mapping Enable:" to No. in the pap2. Outbound calls now work as long as ata is in dmz or the rtp ports 18384 to 18582 are forwarded. |
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VTBrendan
Viatalk
Premium,VIP
join:2005-06-27
Clifton Park, NY
1 edit | .
-Brendan |
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VTBrendan
Viatalk
Premium,VIP
join:2005-06-27
Clifton Park, NY
1 edit | Hi,
I should have clarified before, the * codes need to be dialed independently, not as a pre-pendage to a number. Think of them as an on-off switch, not a per call thing. After this goes through, the setting will change and you will get a busy signal.
-Brendan |
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dcurrey
Premium
join:2004-06-29 | reply to VTBrendan
Ok unless I am in the *51 mode I can't hear who I am talking to. |
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VTBrendan
Viatalk
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join:2005-06-27
Clifton Park, NY | Alrighty. Thats why that option is there, certain combinations of network setups and/or equip work better than others without the relaying.
-Brendan |
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VTJohn
Premium
join:2006-09-14
Clifton Park, NY
| reply to VTBrendan
We've put in a few fixes. Anyone having problems where one or both ends don't hear audio should dial *51. After you dial it you'll get a busy signal and you can hang up. In-network calls and normal incoming/outgoing calls should be functioning now. |
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dcurrey
Premium
join:2004-06-29
·ViaTalk
| reply to VTBrendan
Ok I guess I didn't need to set my router up like I did with the old beta servers (see above). Must have switched on the *51 mode before making the calls. All forwards and dmz are currently off and it still works. *50 is still dead however. 
Ummm was I the only one who tested or did everyone else use pm to communicate? |
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CyberSultan
Premium
join:2006-07-20
| said by dcurrey  :Ummm was I the only one who tested or did everyone else use pm to communicate? I have been PM'ing.  |
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ptrowski
Got Helix?
Premium
join:2005-03-14
Putnam, CT
clubs:
·VOIPo
 ·Metrocast Communic..
 ·AT&T DSL Service
·ViaTalk
| reply to dcurrey
said by dcurrey :Ok I guess I didn't need to set my router up like I did with the old beta servers (see above). Must have switched on the *51 mode before making the calls. All forwards and dmz are currently off and it still works. *50 is still dead however. Ummm was I the only one who tested or did everyone else use pm to communicate? Waiting to leave the office.  |
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GVG
join:2006-09-19
Charlotte, NC
| reply to VTBrendan
Made a couple of incoming and out going calls. They sounded fine.
Should I be concerned that my ping to this server is almost 100ms, its all the way over in San Francisco and I am in North Carolina? Will it affect echo? My ping to the richmond server is only 28ms. |
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VTJohn
Premium
join:2006-09-14
Clifton Park, NY
| reply to VTBrendan
One of the key points of this system is that even if your SIP packets go all the way to San Francisco your audio stream doesn't unless you force it to. So unless you've utilized the *51 code the latency between you and the server shouldn't affect the audio. |
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GVG
join:2006-09-19
Charlotte, NC | So for the terminally curious like me, can you explain how this works versus the standard way. Always have been curious as to how my calls get to and from POTs destinations. |
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GVG
join:2006-09-19
Charlotte, NC | Call Logs do NOT work with this server! |
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ptrowski
Got Helix?
Premium
join:2005-03-14
Putnam, CT
clubs: | reply to VTBrendan
I had it on line one of my two line uniden, sounded good. I did not tell the wife about it so the real test will be if she notices. |
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VTBrendan
Viatalk
Premium,VIP
join:2005-06-27
Clifton Park, NY | Hey,
We need some more testers. Please get in there and give it a go if you have some spare time. At this point you can start using most all of the features. Please post feedback here, thanks!
-Brendan |
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B
Premium,MVM
join:2000-10-28
| Would that be all features except recording? That is, 2 lines in/out, call forwarding, call hunt, DND forward, and call logs are now working on the beta?
Is there anything besides recording that's technically lost when using this variation?
-- B
--
In a realm outside causality and function |
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VTBrendan
Viatalk
Premium,VIP
join:2005-06-27
Clifton Park, NY
1 edit | said by B :Would that be all features except recording? That is, 2 lines in/out, call forwarding, call hunt, DND forward, and call logs are now working on the beta? Is there anything besides recording that's technically lost when using this variation? -- B The 2 line thing we are running into a bug with certain versions of the PAP2, that will be resolved soon. Call forwarding, call hunt, DND forward are working. Call logs are being written and stored, however not combined into the centralized system yet. The calls you make now on it will show there in the future.
edit: call recording is not in there yet, but it will be there soon.
-Brendan |
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