B
Premium,MVM
join:2000-10-28
edit:
August 20th, @06:32PM
| NOT? Working Okay Lately
Hi. Just checking in. This is a "no news is good news" post. I'm provisioned, and my firmware's still at 3.1.15(LS), so it looks as if I have NOT had my adapter upgraded.
I'm still on chicago-1i.vtnoc.net for both lines, so I guess (?) I'm still on the "old" network of SIP servers too.
And, while there have been some weird connection attempts from time to time, things have been pretty stable for the last few weeks. I sure hope it lasts. 
Edit 8/20/08: Seems I spoke too soon; title changed. 
-- B
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In a realm outside causality and function |
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digger16309
join:2007-06-26
00001
 ·VOIPo
·ViaTalk
| Re: Working Okay Lately
I'm in the exact same boat, just about. Firmware same as your version, and I'm still on chicago-1r.
I had a little hiccup last week but otherwise all has been well.
Since I plan to leave when my contract is up in November, I'd just as soon leave everything as it is and miss the upgrade altogether if possible. |
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unknvoip
join:2006-07-25
Rochester, NY
·ViaTalk
| reply to B
I too am in your boat. I am currently registered to chicago-1d on both lines and have the same firmware as you. No clue as to when I will be upgraded and switched to the new servers.
My recent hiccup was line 2 ringing, when line 1 did not. Very occasionally we get a fast busy when dialing out. |
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iLive4Apple
Hybrid power
Premium
join:2006-07-13
Helena, AL | reply to B
I am able to switch from the "new" server to "east" and all of that through the CP. Does that mean I have been migrated?
--
I get 29 MPG in my Toyota Highlander Hybrid! |
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jester121
join:2003-08-09
Lake Zurich, IL
·ViaTalk
| reply to B
Maybe Viatalk's problems are indeed capacity related. If they've moved half the people who used to connect to Chicago-1i off to the "beta" system, it's possible that your server now can handle the load easier and thus is more stable.
Just a pondering... |
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espaeth
Misanthrope
Premium
join:2001-04-21
Minneapolis, MN
·voip.ms
·Callcentric
·VoiceStick
·ViaTalk
 ·Comcast
 ·Embarq
| reply to B
The old servers (chicago-1 / richmond-1) were always pretty reliable for me. Were it not for the new OpenSER switches (and associated issues), I most likely would have done the $199 renewal instead of starting the process of moving my business elsewhere as I come to the end of my 2 year term. Given the current issues and stated direction to do a "forced march" migration onto the new switches, I didn't want to lock myself in for another 18 months.
I really hope the old servers will be around for a while. I got my folks signed up with Viatalk last year so they still have a little over a year left on their prepaid term. They're BYOD (using an adapter I gave them) into chicago-1 and things have been working without significant issue for them. |
|
B
Premium,MVM
join:2000-10-28
| said by espaeth  :The old servers (chicago-1 / richmond-1) were always pretty reliable for me. Were it not for the new OpenSER switches (and associated issues), I most likely would have done the $199 renewal instead of starting the process of moving my business elsewhere as I come to the end of my 2 year term. Given the current issues and stated direction to do a "forced march" migration onto the new switches, I didn't want to lock myself in for another 18 months. I really hope the old servers will be around for a while. I got my folks signed up with Viatalk last year so they still have a little over a year left on their prepaid term. They're BYOD (using an adapter I gave them) into chicago-1 and things have been working without significant issue for them. This is the first time I've seen mention of the specific software VT's using in the new architecture. Can you elaborate? (What else you know about the VT implementation, why you don't like it, etc.)
Strangely it seems that OpenSER, per se, does not even exist any more.
»www.voip-info.org/wiki/view/OpenSER
-- B
--
In a realm outside causality and function |
|
VTBrendan
Viatalk
Premium
join:2005-06-27
Clifton Park, NY
edit:
August 19th, @06:52PM
| openSIPS is the new fork that he is likely referring to. I'd imagine what he is referring to problem wise is the extra configuration necessary for some people depending upon how things are setup at home, once the migration is complete. We have successfully migrated more than half of our network at this point, and once people are moved over they have been having very few issues, especially at the 1st network level.
-Brendan |
|
espaeth
Misanthrope
Premium
join:2001-04-21
Minneapolis, MN
·voip.ms
·Callcentric
·VoiceStick
·ViaTalk
·Comcast
·Embarq
| 
Megatron incoming cap |
75% of the calls I made when configured to use megatron or galvatron didn't set outbound CID. Calls to 800 numbers often resulted in hearing this: »www.hearaduckquack.com/800call.au
When I started registering to megatron or galvatron all of my incoming SIP URI calls went straight to my network down number, and on occasion all calls would be routed to voicemail for no apparent reason.
In the course of troubleshooting over the last few weeks I've compiled and configured up my own home Asterisk instance so I could get better debug options than I could get out of just the PAP2T. I've seen the same behavior with my Asterisk instance running with a public IP where NAT and firewall issues are avoided. Based on the number of customers that have reportedly been moved to the new switches, I'm guessing that there is likely something more hosed up with my account than with the switch itself. I kinda wish it was something on my end, but so far I haven't seen the same issue with any of the other providers I've setup Asterisk to peer with:
Not to be completely negative, there's a lot of things that ViaTalk is doing right. The new control panel is a nicely coded UI, they have a very complete feature list, and with the exception of the last few weeks they provided me with nearly 2 years of good phone service. |
|
B
Premium,MVM
join:2000-10-28
| Thanks; very complete answer. You seem like exactly the kind of technically minded user who would be helpful in their beta -- but I gather they weren't able to pinpoint or resolve anything for you though. Too bad...
Going on your "hosed up with my account" theory, I guess a good test would be for you try everything with someone else's credentials. Perhaps VT could provide you a "test user" account for debugging purposes?
-- B
--
In a realm outside causality and function |
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im_chandave
join:2005-07-28
Cleveland, OH
·ViaTalk
| reply to espaeth
said by espaeth :My issue is that the switch did screwy things on my account, like sending double SIP INVITEs for incoming calls and then issuing a CANCEL which caused the session to tear down. Were the two SIP INVITEs operationally identical (same CSeq, tag, and Call-ID)? Are "c=" and "m=" values in the SDP identical? Was the RTP stream originating from the correct source port from the Minneapolis L3 media gateway? Was the RTP stream being sent properly to your RTP UDP port as specified in your SIP 200 OK reply?
I've seen a situation like (SIP CANCEL from provider almost immediately after my SIP 200 OK) this one before. This happened when the media gateway started sending the RTP stream before my Asterisk box had a chance to poke the pinhole in my firewall. But, in your case, the RTP was started after the SIP CANCEL was sent.
The only thing I can think of is that the 2 SIP INVITEs you got from megatron.vtnoc.net were not operationally identical. In which case the received SIP CANCEL was to terminate one of the inbound calls. Hard to tell with the summary wireshark view.
Did you give this packet capture session to ViaTalk?
See ya...
d.c. |
|
espaeth
Misanthrope
Premium
join:2001-04-21
Minneapolis, MN
·voip.ms
·Callcentric
·VoiceStick
·ViaTalk
·Comcast
·Embarq
edit:
August 20th, @02:49PM
| said by im_chandave :Were the two SIP INVITEs operationally identical (same CSeq, tag, and Call-ID)? Are "c=" and "m=" values in the SDP identical? The packets are more than operationally identical, their entire payload _is_ identical.
said by im_chandave :Was the RTP stream originating from the correct source port from the Minneapolis L3 media gateway? Was the RTP stream being sent properly to your RTP UDP port as specified in your SIP 200 OK reply? The destination port matched exactly what was spec'd in the 200 session info, but the cancel had already been sent and ACK'd so by the time the gateway started sending RTP traffic the PAP2T had already stopped listening on that port.
said by im_chandave :Did you give this packet capture session to ViaTalk? Yep, I PM'd Brendan a link as requested but nobody ever grabbed a copy of the trace. |
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B
Premium,MVM
join:2000-10-28
| reply to B
Son of a bitch.
I jinxed my own damn self starting this thread.
Multiple outbound calls today are experiencing what seems to be one way audio. That is, they called me from home on the VT line, I answered on my cell, but they heard no ring or audio on my part as I repeated "hello", and they eventually hung up (I think I could hear background noise and breathing).
But eventually one call was successful.
Does this mean I've been whomped over to the beta system, or is it just a random glurp in the traditional network? I don't believe I've experienced this particular problem with Viatalk before now.
I'll check the adapter status when I get home... and then start going through the complaint threads if necessary. I'm provisioned; this is NOT supposed to happen.
-- B
--
In a realm outside causality and function |
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k2rj
Premium
join:2005-03-24
Solon, OH
·ViaTalk
·RoadRunner Cable
| That probably means you have been switched and need to forward ports. When this happened to me, I "thought" I had forwarded my ports, but found out that I had been caught by DHCP and my adapter was no longer on the IP address with the forwarded ports! (I fixed that with a static IP and a ticket to add it to my provisioning file.) |
|
VTBrendan
Viatalk
Premium
join:2005-06-27
Clifton Park, NY
| Hi,
Unless you moved yourself or were moved last week, you should not have been migrated as of yet. We have not moved any people this week in an effort to allow support to get completely caught up with the increase in volume that the migrations result in. You also would have received an email if you were migrated.
-Brendan |
|
iLive4Apple
Hybrid power
Premium
join:2006-07-13
Helena, AL | reply to B
Re: NOT? Working Okay Lately
How am I supposed to know if I am migrated or not?
--
I get 29 MPG in my Toyota Highlander Hybrid! |
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B
Premium,MVM
join:2000-10-28 | reply to B
Okay, I checked and I'm still at 3.1.15(LS) on chicago-1i.vtnoc.net.
So the problem was new.
-- B
--
In a realm outside causality and function |
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espaeth
Misanthrope
Premium
join:2001-04-21
Minneapolis, MN
edit:
August 21st, @02:12PM
| The strange thing is that when you answer on your cell phone, the voice traffic never hits your ATA so there's no way for that to be a factor. |
|
ptrowski
Got Helix?
Premium
join:2005-03-14
Putnam, CT
clubs:
·AT&T DSL Service
·ViaTalk
| reply to iLive4Apple
said by iLive4Apple :How am I supposed to know if I am migrated or not? Get the admin password and check. |
|
B
Premium,MVM
join:2000-10-28
| reply to espaeth
said by espaeth :The strange thing is that when you answer on your cell phone, the voice traffic never hits your ATA so there's no way for that to be a factor. I don't understand what you mean. The ATA is always a factor.
To be clear, the calls were made from my home VT line to me on my cell phone at a remote location.
Unless you were responding to someone else...
-- B
--
In a realm outside causality and function |
|
