Re: New Network Update in ViaTalk

Re: New Network Update

Tue, 12 Aug 2008 20:35:27 EDT

hootch : I checked my pap and it is pointing to optimusprime, even though the control panel shows East, so I guess that answers my question if I have been migrated. Thanks for the tip to check the pap.

Re: New Network Update

Tue, 12 Aug 2008 13:52:04 EDT

dcurrey : Maybe the east now points to the new servers or the CP doesn't reflect the changes correctly.

If you have admin passwd what server does the pap point to?

I think we have,
megatron.vtnoc.net
galvatron.vtnoc.net
optimusprime.vtnoc.net
are all on the new network.

Re: New Network Update

Tue, 12 Aug 2008 13:42:57 EDT

TJ_in_IL : There should be EAST, WEST, CENTRAL, and NEW.
You would want the NEW.

Re: New Network Update

Tue, 12 Aug 2008 12:55:28 EDT

Aveamantium :

said by hootch :

Also, I use a Buffalo router with tomato firmware...does anybody know where to turn off the features that Brendon mentions? (the SPI or SIP ALG)

Don't worry about it with Tomato, it will work fine the way it is (you can't disable SPI in Tomato anyway). If you experience one way audio go ahead and forward ports. Other than that you should be fine.

Not sure about the West/East thing?

Re: New Network Update

Tue, 12 Aug 2008 12:17:23 EDT

hootch : I have a quick question about the new network rollout. I just got the email that my account is rolled over...I now notice that in the control panel, the server config has been changed from WEST to EAST. Is "EAST" the new network that I should be on? I live on the west coast.

Also, I use a Buffalo router with tomato firmware...does anybody know where to turn off the features that Brendon mentions? (the SPI or SIP ALG)

Re: New Network Update

Tue, 12 Aug 2008 12:11:23 EDT

Re: New Network Update

Thu, 07 Aug 2008 14:16:52 EDT

im_chandave : Sorry chemie . The info I gave was only for the DI-series of D-Link based on the Global Sun Tech AP reference design.

Actually, this reference design was used for several other vendors like Surecom, Trendnet, and Esso.

This also includes some of the D-Link WBR-series routers.

See ya...

d.c.

Re: New Network Update

Thu, 07 Aug 2008 09:37:48 EDT

TJ_in_IL :

said by chemie :

PAP2T suggests Mapped RTP Port is in fact 5080 which I specified (does 5060 really matter vs 5060?)

I found that with my Linksys, the new network did not like anything but the 5060. I had it on 5090, tried 5080, but settled in on 5060.
I also changed from DMZ to port forward. Since then, I have been rock solid on the new network. YMMV

TJ

Re: New Network Update

Thu, 07 Aug 2008 08:10:21 EDT

chemie : IM_Chandave

I assume this is for the DI-624?? DIR-655 does not even give me anything for »192.168.0.1/natlist.txt

(also game mode is replaced with specific SPI on/off" (which I have off). PAP2T suggests Mapped RTP Port is in fact 5080 which I specified (does 5060 really matter vs 5060?)

Re: New Network Update

Wed, 06 Aug 2008 23:28:05 EDT

im_chandave : I'm glad I could help.

Maybe VTBrendan can use the above for a CSR script to help people behind the D-Link DI routers. At least then he'll have something different from the "SIP ALG or SPI enabled" question ;) .

See ya...

d.c.

Re: New Network Update

Wed, 06 Aug 2008 15:26:04 EDT

degauss1 : WOW! That's a lot of info. I'll give these a try and see if I can use the new network.

**edit

d.c. That worked great.
The devil was in the details. Looks like it's working well for me. Volume on the calls seems louder than they were on sanjose-1a. Got through right away to the toll free numbers. DTMF working well. VM working well.

Many thanks!

Re: New Network Update

Tue, 05 Aug 2008 23:36:01 EDT

im_chandave : "Turning off SPI" on the DLink DI series of routers entails enabling Game mode. "Game mode" on that router series just means that any Internet host can send data through the pinhole established by an outbound connection from behind the LAN side of the router.

Also, due to a bug in the firmware, you will notice that your DI-624 will be more stable with UPnP enabled...even if you don't use UPnP to control your Internet Gateway Device (IGD--i.e the DI-624 in UPnP terminology).

The DI-series does not honor source-port preservation. As a result, if your SPA uses UDP port 5060 as a source port for SIP communication, there's a 100% chance that when your SIP packet exits your router, it will use a different (high 63,000 range) port number.

My suggestion:
- Goto your SPA's SIP > NAT Support Parameters and
  - Set "STUN Enable" to "yes"
  - Set "STUN Server" to "stun.vtnoc.net"
  - Set "STUN Test Enable" to "no"
  - Set "NAT Keep Alive Intvl" to 30 seconds
  - Set "Send Resp To Src Port" to "yes"
  - Ignore the "VIA received", "VIA rport", and "VIA Addr" stuff for now.
- Look at SPA' SIP > RTP Parameters and
  - Note the "RTP Port Min" and "RTP Port Max" numbers. This range represents the UDP port forwarding range you might need to setup on your DI-624...might if STUN is not resolving your call issues.
- Goto your Line 1 > NAT Settings and
  - Set "NAT Mapping Enable" to "yes"
  - Set "NAT Keep Alive Enable" to "yes"
- Goto your Line 1 > SIP Settings and
  - Set "SIP Port" to "5060"
  - Set "Restrict Source IP" to "no"
  - Make sure "EXT SIP Port" and "SIP Proxy-Require" are blank
- Take your SPA OUT OF the DMZ zone. Give it a static IP address within your LAN subnet.
- Remove any port forwarding for UDP port 5060 and the UDP ports range found in "RTP Port Min" to "RTP Port Max" inclusive.
- Turn off your SPA.
- Reboot your DI-624.
- After your DI-624 is fully up and running, get to a computer and retrieve the following file using the URL:
http://192.168.0.1/natlist.txt (assuming your DI-624's LAN IP address is 192.168.0.1). This is your NAT mappings on the router.
- Start up your SPA.
- Once the SPA is fully up, check your SPA's INFO > Line 1 Status:
  - Look at the "Mapped SIP Port" and take note of the number. It would probably be in the high 60K range.
  - Look at the "Registration State". Is it registered?
- Get the NATLIST again from your router via http://192.168.0.1/natlist.txt.
  - In the NATLIST.TXT, you should have a line listing your internal IP address of your SPA with port 5060 mapped to the port corresponding to "Mapped SIP Port" from your SPA. For example, if the "Mapped SIP Port" was 63456 and your SPA has the address of 192.168.0.2, you might see the following line in your NATLIST.TXT
If you don't have a line like that in your NATLIST.TXT, then make sure "Game mode" is still enabled on your DI-624.

If at this point everything has been going as I presented above, then try making an outbound call. While making the call, look at your SPA's INFO > Line 1 Status > Call 1 Mapped RTP Port. Grab a copy of the NATLIST.TXT from your DI-624 while the call is in progress. Is there a NATLIST UDP entry whose out_port matches the "Mapped RTP Port"? If not, then I can get back to you about how to enable Syslogging and then interpreting the SIP packet dumps that will appear in the Syslog server.

See ya...

d.c.

Re: New Network Update

Tue, 05 Aug 2008 17:03:58 EDT

degauss1 : Brendan,
Thanks for the reply.

I don't have the actiontec. I signed up with VZ before they offered that router and they gave me the DI-624.

The only settings that seem to sort of imply SPI/ALG is UPnP and Game mode. I have always run with those off. Again, I have REMOVED THE ROUTER completely from the mix. I do this by changing all the settings on the SPA 2002, release IP on the router, unplug SPA power, unplug WAN from the router, plug WAN into SPA 2002, power the SPA 2002 on and check results. I have same results with or without the router.

I do have an older belkin I can try...I know I could disable SPI on that one.

Re: New Network Update

Tue, 05 Aug 2008 16:47:19 EDT

VTBrendan : Hi,

Do you have the ability/access to check your router and maybe even the FiOS connection (likely actiontec?) for SPI or SIP ALG being enabled?

-Brendan

Re: New Network Update

Tue, 05 Aug 2008 16:06:18 EDT

degauss1 : Well, I have a SPA 2002 and it connects to a DI-624.
I have UPnP and Game mode disabled on the DI-624.
When I switch to megatron or galvatron I set port 5060 and disable outbound proxy as recommended in other posts. I have tried with and without STUN enabled.
I have tried connecting directly to my FiOS connection (no router in the mix at all).
I have tried using DMZ on the DI-624.

Bottom line, no matter what I do, I get registation, but no outbound calls. I get a dial tone, but no calls connect. No 800,866 or any regular exchanges. No outbound - period.

If I switch back to sanjose-1a and change the proxy settings everything works flawlessly as it has since I signed up.

It's quite frustrating. If the new network is the wave of the future for all users and I can't get it working it's gonna put a damper on using VT for phone service that's for sure.

***update***
Just tried xlite soft phone. Same results with toll free numbers. Get one-way outbound audio on standard toll exchanges. Shut down xlite, tried switching to megatron on the spa2002 and got same results. So, can dial and recieve one-way audio calls to standard exchanges but toll free never connect.

Again have tried with AND without router in the way, with and without stun etc as above.

Re: New Network Update

Tue, 05 Aug 2008 16:06:01 EDT

unknvoip : How is the firmware update, server roll-out going? Still waiting for the update here, but I may have been moved back in the queue since my adapter had to be replaced.

Re: New Network Update

Tue, 05 Aug 2008 15:56:16 EDT

espaeth :

said by VTBrendan :

Can you PM me the packet capture by chance?

Done, although nobody seems to have grabbed it yet.

Re: New Network Update

Tue, 05 Aug 2008 15:29:42 EDT

VTBrendan : Hi,

There is and will remain an option to turn on audio relaying through the new setup, such that it will route identically to how it used to.

-Brendan

Re: New Network Update

Tue, 05 Aug 2008 14:06:48 EDT

im_chandave :

said by VTBrendan :

It's currently 60 on the nose. There will very rarely be a case where this is worse, and often cases where it is better.

-Brendan

Well, if that's the case, then can you keep some of the old servers up for those of us that feel ViaTalk's routing to L3 is more consistent and predictable than our ISP's routing?

SBC/at&t likes to bounce it's NE Ohio customers between 2 redbacks--one in Ohio and one in Illinois. Routing to L3's LA POP differs greatly between the two redbacks. But, the routing to ViaTalk's Chicago-1a is very similar between the two redbacks.

See ya...

d.c.

Re: New Network Update

Tue, 05 Aug 2008 13:42:34 EDT

VTBrendan : It's currently 60 on the nose. There will very rarely be a case where this is worse, and often cases where it is better.

-Brendan

Re: New Network Update

Tue, 05 Aug 2008 13:22:24 EDT

im_chandave :

said by VTBrendan :

I believe it was setup TCP/IP in locations that were on-net with L3, thus the RTP path would generally stay on L3's network, but still route to the same endpoints you are seeing now. Overall you should be seeing consistently less latency now per the results of our testing.

-Brendan

Here's a traceroute from a DSL connection in CLV to chicago-1a.vtnoc.net:
You don't see my traceroute hit the destination but, 216.246.94.94 is in the same BGP peering region.

Here's a traceroute from the same machine to sv4.LosAngeles1.Level3.net (209.247.23.75), one of the L3 gateways handling 800 number termination:
Again, I can't hit the destination, but 4.68.102.39 is within L3's Los Angeles POP.

If your Chicago-1a's route to L3's LA media gateway took more than 60ms to the routing, then I would agree that our latency has gotten less with the new destination-based L3 media gateway selection.

See ya...

d.c.

Re: New Network Update

Tue, 05 Aug 2008 12:50:46 EDT

VTBrendan : I believe it was setup TCP/IP in locations that were on-net with L3, thus the RTP path would generally stay on L3's network, but still route to the same endpoints you are seeing now. Overall you should be seeing consistently less latency now per the results of our testing.

-Brendan

Re: New Network Update

Tue, 05 Aug 2008 12:45:12 EDT

im_chandave :

said by VTBrendan :

Thats not necessarily the case, as before it would go from you, to us, to whatever regional destination L3 routed the call to. Now it does the same thing, only taking us out of the loop.

-Brendan

So, does that mean that ViaTalk only had TCP/IP access to Level 3 media gateways? Or did ViaTalk connect to the Level3 media gateways through TDM? If it was via TDM, then there's a better chance of QOS between your systems and L3's. In that type of situation, my only concern would been the path between my SIP device and ViaTalk's media gateway.

Now, with this new setup, I have to worry about the route between me and the Level3 LA gateway when calling 800 numbers. Then, I would have to worry about the route between me and the Level3 CLV gateway when calling NE Ohio. And, again, I would have to worry about the route to every single destination.

With the old way, during calls, I could generate traceroutes to the SIP registrar I using and get a reasonable estimation of the path of the RTP. Now, I've got to pry it out of the SIP 183 Replies before I can start doing the traceroutes.

See ya...

d.c.

Re: New Network Update

Tue, 05 Aug 2008 12:25:24 EDT

VTBrendan : Thats not necessarily the case, as before it would go from you, to us, to whatever regional destination L3 routed the call to. Now it does the same thing, only taking us out of the loop.

-Brendan

Re: New Network Update

Tue, 05 Aug 2008 12:20:30 EDT

im_chandave :

said by druber :

I'm guessing L3 would not want to do that. I'm sure they'd rather not use their bandwidth to haul your audio stream halfway across the country when they can make you go public :) For what its worth, I see your point (and I agree with you...)

Yes, druber , that's the exact point I was trying to highlight. With the new setup, it benefit's Level3 more than it benefits us the subscribers.

Sure, for local calls, it might produce better audio. But, for those of us that call to many parts of the US and Canada, our quality of audio could greatly vary due to the different routes to the Level3 media gateways closest to the call's termination.

I liked it better when I had a reasonable estimation of the routing my audio would transverse while in the Internet path.

See ya...

d.c.

Re: New Network Update

Tue, 05 Aug 2008 08:13:54 EDT

druber : I'm guessing L3 would not want to do that. I'm sure they'd rather not use their bandwidth to haul your audio stream halfway across the country when they can make you go public :) For what its worth, I see your point (and I agree with you...)

Re: New Network Update

Tue, 05 Aug 2008 04:29:24 EDT

im_chandave :

said by degauss1 :

Are ANY BYOD users having any luck with the new network?

I'm unable to get any outbound calls to go through no matter what I try.

I just switched to galvatron for my asterisk box. Inbound and outbound calls are ok. DTMF inbound and outbound are working fine as well.

As ViaTalk stated somewhere else, the new system does separate the media (RTP) system from the signaling (SIP) system. I call CLV weather and the RTP stream goes to Cleveland's Level 3. Call SFO weather and get directed to California. I did try several 800 numbers and they all seem to be serviced by the Los Angeles Level3 media gateway.

But I wonder about this destination-based selection of the media gateway.

I would think that using a media gateway closest to my SIP device would produce the best quality audio for the session. The audio would only be on the public Internet until it reached the media gateway. After that, the audio would be routed through Level3's private IP network or TDM circuit. While in Level3's private IP network, they can provide guarantees for quality of service. For the TDM circuit, there are regulations that guarantee quality of service.

However, now my audio no longer passes through Level3's private IP network for the majority of the journey. Instead, we are forced to use the public Internet to get to the media gateway that's closest to the destination. The quality of service is no longer the responsibility of Level3 for the majority of the journey.

Is there anyway ViaTalk can get Level3 to use source-based media gateway routing instead of destination-based?

See ya...

d.c.

Re: New Network Update

Mon, 04 Aug 2008 23:37:42 EDT

TJ_in_IL :

said by heavyt :

said by degauss1 :

Are ANY BYOD users having any luck with the new network?

I'm unable to get any outbound calls to go through no matter what I try.

Have you tried sip port 5060? What sort of BYOD you have, explain your problem so others can help.

I agree. Make sure you are on 5060, and that you are using G711u codec. These will cause problems.

TJ

Re: New Network Update

Mon, 04 Aug 2008 21:37:17 EDT

heavyt :

said by degauss1 :

Are ANY BYOD users having any luck with the new network?

I'm unable to get any outbound calls to go through no matter what I try.

Have you tried sip port 5060? What sort of BYOD you have, explain your problem so others can help.

Re: New Network Update

Mon, 04 Aug 2008 20:58:35 EDT

GVG : I have been on it from the beta days and it works perfect for me. Make sure you have turned off SPI in your routers firewall, and if you have any ALG's turn them off too.

Re: New Network Update

Mon, 04 Aug 2008 20:26:55 EDT

degauss1 : Are ANY BYOD users having any luck with the new network?

I'm unable to get any outbound calls to go through no matter what I try.

Re: New Network Update

Mon, 04 Aug 2008 18:44:41 EDT

VTBrendan :

said by espaeth :

said by VTBrendan :

Let me know how it turns out. I'm confident that once people are up and running on the new systems that they will be very pleased with the results.

Converting to megatron or galvatron still results in failed inbound calls for me. In doing a packet capture I actually see a SIP CANCEL message sent from the call manager immediately after the session establish is ACK'd. Outbound calls are unaffected.

How long will the legacy infrastructure be around? Things continue to work relatively flawlessly on chicago-1a / richmond-1, but I'm hesitant to re-up my contract if it locks me into the new infrastructure that still doesn't play nicely with my PAP2T.

Hi,

Can you PM me the packet capture by chance?

-Brendan

Re: New Network Update

Mon, 04 Aug 2008 17:57:59 EDT

RMKyote :

said by VTBrendan :

Let me know how it turns out. I'm confident that once people are up and running on the new systems that they will be very pleased with the results.

-Brendan

Well let's see, other than, as posted elsewhere in greater detail, the "Great Guy" helping me out and documenting all that he said needed to be sent to the engineers, and a detailed accounting of the problems, what he found and did to help me, which never made it into my records as per the last TSR's that did really help me. And his conversation with a TSR that I spoke to just before him that was dumb as a box of rocks that told the "Great Guy" that I was a waste of time, yes I heard it in the background and when let him know that I heard the conversation he said he didn't remember. Not real good on the short term memory there buddy. That's not good. This inturn created complications when a ticket that I had long since put in was finally acted on and since there was no record of his actions, I was put back on Galvatron. In the ticket reply that informed me that I was switched the new system, I replied that I had been there and my line became completely unusable. So I moved myself back as it was pointed out in another eMail that followed. This time we sync'd up and methodically moved me, at least that is what I was told was being done on VT's end. I was asked to monitor the situation, of which I was/am perfectly happy to do. But there is one problem with doing it through the ticket and that was tickets are taking a long time to be acted upon and when my line goes down or gets too funky, I can't wait a week or so. So I told the TSR upfront that I was willing to endure almost everything except when the line becomes completely unusable, when I would reply and if in a reasonable amount of time was not answered I would move myself back. OK, ground rules set.

All was really good until 6pm Friday night when 98% of my calls started to go to VoiceMail on their own, as are many others that have posted here, or when I picked up the phone. So for the next hour or so I played with this and saw that eMail Notification (the only accurate accounting), Phone Indicator, Website CP, the Auto-Attendant nor the Call Logs were showing the same activity. And when I, and this is still happening, deal with all my calls via *123#, not all the calls that are there are available, but when those that were are deleted the phone indicator goes out even though I have many more messages in the CP and in my eMail. I waited until about 8:30pm submitted a ticket saying that if I didn't hear anything or see a change by 10pm, I'd move myself back to a common server. Well I got distracted and when I realized that things were the same it was too late to call anyone. Bankers Hours. Oh and Brendan, why does Host Rocket have 24 hr phone support for web hosting that, even though I am a developer and have my own server farm and 24/7 phone support for the sake of proper customer service, I don't feel web hosting is a 24 hr service product, whereas, a phone line is and VT doesn't have 24 hour phone support? Submitting tickets after hours gets you nothing but on a waiting list.

Anyway, I finally went to bed at about 4am and things were still not working. Thank God for POTS. When I got up the next morning I called from another part of the house on the POTS line and still the same. So I called VT TS and got a very confident and capable young lady that confirmed that all the notes that I was told were being put into my account were not there and that is why when my original ticket was finally acted on, there was no current status on what had transpired. She asked a couple of questions, placed a couple of calls that failed then came back and said that she would have to escalate me. Fine. I told her I was going to be gone from 12 to 3 and they could have at it. When I got back no indicator lights nor eMail messages so even if things were fixed no one called. So I placed a test call and it worked. I checked to see if I was still on Galvatron and I was so that was a good sign, fingers crossed.

So over the past day or so I have had only one call go to VM when I picked up, I am able to call out with no problem as I have been able to all along, and as many others that have posted here are experiencing, and the noise has been at minimum. Today has been lean on calls, but the few that have come through have been good. I have yet to get any eMails telling me that they have tired to call and weren't able to get through. So . . . finger still crossed that things are looking up. Now with that said, I have spent more time and energy on having a VT line than I have on anything in my life. I am feeling like it's my new religion and I have to worship every waking minute. Plus all these new dance steps with VT are getting to be a little much. That's probably because I like to lead and that is not an option here.

So basically things are working in an acceptable manner, for the moment, I got a laugh by being called a waste of time for looking out for my best interests, I have had several curious and few productive encounters with TSR's, been lied to, misled, and couldn't receive calls on a phone that I have been guaranteed would work. Fhew, I'm tired. Time for a cocktail by the pool and a nap.

Re: New Network Update

Mon, 04 Aug 2008 16:46:22 EDT

espaeth :

said by VTBrendan :

Let me know how it turns out. I'm confident that once people are up and running on the new systems that they will be very pleased with the results.

Re: New Network Update

Mon, 04 Aug 2008 15:48:24 EDT

VTBrendan : Let me know how it turns out. I'm confident that once people are up and running on the new systems that they will be very pleased with the results.

-Brendan

Re: New Network Update

Fri, 01 Aug 2008 16:02:26 EDT

RMKyote : UPDATE

I have been once again, but this time methodically, moved back to Galvatron (anyone else seeing Transformers in your mind when you say this?) and so far things are quite nice. Now that being said I have just jinxed myself. So we'll have to see what happens from here.

Will be in touch.

Re: New Network Update

Wed, 30 Jul 2008 23:06:25 EDT

RMKyote : I've been on Galvatron (new system) twice and nothing but problems. And they were big new ones that had to be documented and sent to the engineers as I was moved back to Chicago.

Doesn't look the like new system is anywhere near ready.

Re: New Network Update

Wed, 30 Jul 2008 09:34:09 EDT

heavyt :

said by GVG :

you should not have anything in the outbound proxy setting and the use outbound proxy should be turned off.

After I changed the sip port to 5060 from 5090 (Viatalk tech told me to use 5090 because they feared Comcast was causing my service to go down) I don't need to have anything in the Outbound Proxy for my service to work! Thanks everyone for all the help and suggestions.

Re: New Network Update

Wed, 30 Jul 2008 08:21:42 EDT

ptrowski :

said by heavyt :

said by TJ_in_IL :

New network with BYOD does not work for me.
Tried suggested tweeks, but still no go.

TJ

I am also a BYOD user who has no luck with the new network. I have a SPA-2102. I can receive calls but I can not hear the calling party, they can hear me. If I make a call I do not hear the ring nor due I hear the party when they answers, but they can hear me.

It seems the problem is in the Outbound Proxy setting. I can put the new server in the Proxy setting but I must have an old server address in the Outbound Proxy setting for my phone to work.

That sounds like an issue forwarding ports on your router will fix.
--
"A religious war is like children fighting over who has the strongest imaginary friend."

Have you been touched by his noodly appendage? »www.venganza.org

Re: New Network Update

Wed, 30 Jul 2008 01:03:44 EDT

KC9FOI : I can say that the new network has been working very well for me. No CID issues, no crappy calls. I spent several hours on the phone tonight with no delayed audio, no distortions and no dropped or failed calls. My up time has been very good lately. Keep it up VT! I'm beginning to seriously consider renewing in March when my free year is finished. It would be the first time I don't switch with my VoIP provider when my time is up.

Re: New Network Update

Tue, 29 Jul 2008 23:02:54 EDT

TJ_in_IL : I am refering to the VT Fax2Email service, not a fax machine. I know that does not work on VT (at least for me).

When I was having issues, they suggested the following in a ticket response:
In addition to change the codec to G711a, I would suggest that you change the following variables:

SIP Port -> 5090
RTP Port Min -> 19384
RTP Port Max -> 19482
RTP Packet Size -> 0.020

Please test the service with this setup, and let us know the results.
Now when I try the fax, it rings thru.
Not sure if they have tackled that issue yet.

TJ

Re: New Network Update

Tue, 29 Jul 2008 22:49:54 EDT

GVG : G711a is used in Europe, not in the USA. The telco system here is based on G711u and you will get unreliable connections if you try to use 711a. Faxing over VOIP is a crap shoot at best, there is no magic setting and what works one day will fail the next. The best you can do is set your PAP up for best voice service and if FAX works then you are being lucky that day.

Re: New Network Update

Tue, 29 Jul 2008 22:46:24 EDT

TJ_in_IL :

said by GVG :

G711a is not viatalk supported so whoever told you to us that was wrong or misspoke.

Once I change to g711a, my fax issues went away.
This was per ticket reponse from VT.

Now I am just faced with "INVALID" for CID, which I see is a whole other issue.

Re: New Network Update

Tue, 29 Jul 2008 22:44:12 EDT

GVG : G711a is not viatalk supported so whoever told you to use that was wrong or misspoke.

You may also find with the new stun settings you dont even need to port-forward. I do no forwarding at all and it works perfect.

Re: New Network Update

Tue, 29 Jul 2008 22:42:35 EDT

TJ_in_IL : check my edit

Re: New Network Update

Tue, 29 Jul 2008 22:42:04 EDT

GVG : well your problem all along was using 711a, Viatalk does not support that, only 711u.

Re: New Network Update

Tue, 29 Jul 2008 22:41:09 EDT

TJ_in_IL : OK, Finally got the new network to work.

Old settings for old network:
SIP Port 5090, codec G711a
Working settings new network:
SIP Port 5060, codec G711u

Ports forwarded:
5060 both TCP and UDP
10000-20000 both TCP and UDP

Now the real test is will Fax2Email work with the new network/configuration? Off to test that.... (hopefully)

TJ

edit- port/codec changed for fax2email to work, per vt tech suggestions.

Re: New Network Update

Tue, 29 Jul 2008 22:39:39 EDT

GVG : So change your sip ports to another range. The default is 5060, but you can change that to another value.

Re: New Network Update

Tue, 29 Jul 2008 22:31:27 EDT

TJ_in_IL :

said by GVG :

Try changing your Stun settings to these.

Tried, no luck.
Thanks anyway.
Like I said, think it is with the SIP ports.

TJ

Re: New Network Update

Tue, 29 Jul 2008 22:16:29 EDT

GVG : Try changing your Stun settings to these.

Re: New Network Update

Tue, 29 Jul 2008 22:13:47 EDT

GVG : you should not have anything in the outbound proxy setting and the use outbound proxy should be turned off.

Re: New Network Update

Tue, 29 Jul 2008 22:03:30 EDT

TJ_in_IL :

said by heavyt :

I can receive calls but I can not hear the calling party, they can hear me. If I make a call I do not hear the ring nor due I hear the party when they answers, but they can hear me.

Sounds like you need to try DMZ or port forward. Seems to be a common issue.
My problem is that I get fast busy outbound, and direct to voicemail inbound.
Try the DMZ or forward.

TJ

Re: New Network Update

Tue, 29 Jul 2008 21:45:34 EDT

heavyt :

said by TJ_in_IL :

New network with BYOD does not work for me.
Tried suggested tweeks, but still no go.

TJ

Re: New Network Update

Tue, 29 Jul 2008 21:03:35 EDT

TJ_in_IL : You know, while changing from DMZ to port forwarding, I noticed that my sip port is different than the norm. I wonder if that is the reason for not working. My SIP port is different because of the Fax2Email. Hmmmmm......

TJ

Re: New Network Update

Tue, 29 Jul 2008 20:33:32 EDT

TJ_in_IL : New network with BYOD does not work for me.
Tried suggested tweeks, but still no go.

TJ

Re: New Network Update

Tue, 29 Jul 2008 19:15:03 EDT

VTJosh : I don't have the specific numbers, but people are still being migrated to our new platform. Rather than flip the switch on the entire network, we're taking it slow to make sure there aren't any problems down the line.

There have been a few minor issues, such as the caller ID problem dcurrey mentioned, but the migration is going smoothly for most people.

- Josh

Re: New Network Update

Tue, 29 Jul 2008 18:43:33 EDT

dcurrey : Other than the 1 "Null" callerid problem I haven't noticed any issues with the new network.

New Network Update

Tue, 29 Jul 2008 18:30:13 EDT

jono_x : How about an update on the new network roll-out? I know ViaTalk will supposedly send out an email when they migrate a provisioned customer to the new network, but this sure seems to be taking a long time. It looks like a fair number of people are having problems with the new network? I don't see any rave reviews of the new network, so one could be led to believe that the new network isn't all that it is cracked up to be, and they are delaying the roll-out?