voip.ms discussion

[UberDave]

I'll know more as time goes by but so far only one number (almost local) that I just called this morning. Until now, even before the port, there weren't any issues calling out.

Of course after the number was ported I plugged the ATA into my home's wiring system and changed routers.

I decided to run with the tried and tested to be safe. Asus WL-520GU and Tomato 1.28.

I set up QoS and put the made the ATA's IP static. Otherwise nothing has been tweaked on the router.

[Mango]

With Tomato routers, as a general rule, you should enter Conntrack/Netfilter and set Unreplied UDP Timeout to 10 if it isn't already.

If possible, you might want to investigate QoS >> View Details while the problem is happening and make sure a VoIP connection is not improperly classified. If a VoIP connection is accidentally classified as low priority, that could certainly cause one-way audio.

Keep us posted on how things go!

m.

[UberDave]

Thanks for the help Mango. the Unreplied UDP is already set at 10. I realize now that for some reason my QoS settings for the PAP2T didn't save. Maybe I forgot to hit the 'save' button. Should I setup the QoS in Tomato based on the 'Src MAC' of the ATA or by the 'Src IP'. My thought was by the MAC address but I don't really know if it matters.

[FunkyTown]

Oh that's very clever Mango. I never would have thought of that trick with the INUM. Did you mean callback or DISA?

[Mango]

I was thinking of Callback. Theoretically, that would allow him to route an IVR to Callback, which you currently can't do natively. I'm interested to know if that works

[UberDave]

I lost a post to an administrator check earlier but I've tried a few more things. It seems like the problem is worse when I use premium routing than when I try calling out with the value routing. And when I call out using premium routing it doesn't sound like it is ringing on the other end even when the phone being called is ringing.

My issue sounds exactly like flyboy320's in post #1297. He wasn't sure whether it was coincidence but changing the RTP packet size to .02 fixed his issue. Mine is already set there so...

I tried changing servers to montreal - no effect. Tried changing RTP packet size down to 0.01 - no effect. I changed codecs to g729 from g711 - no effect.

I also added the QoS to tomato (It looks like I didn't save it when I set it up initially) and set it by device MAC address, to any port and set it at 'highest' priority. Does it matter if it is moved to the top of the list of QoS rules? I bumped it up and it looked to be working better when viewing the graphs during a call.

The other issue I'm having is that incoming audio breaks up and cuts out when I start speaking. This seems to be happing for incoming and outgoing calls.

Good thing we have a cell phone for a backup. My wife initially didn't care so much about teething pains but her patience is wearing out pretty quickly.

[Mango]

Last thing to try:

Remove your router and connect your VoIP device directly to the modem. You may need to powercycle the modem in order for this to work. Connect a phone directly to your VoIP device, bypassing your household wiring. If that doesn't help, we've narrowed your issue down to either a problem with your internet provider, or a routing problem at VoIP.ms. You might want to submit a support ticket to ask them for assistance.

They'll probably ask you if you can reproduce the issue with a softphone. If you want to do that beforehand to eliminate some of their troubleshooting, the free version of Zoiper should do the job.

m.

[gzink]

Can you get a toll-free number and just use DISA??
Perhaps I misunderstand!

[Mango]

Sure you can - that's useful for making calls from pay phones.

[gzink]

Then it would be good for long distance with only minutes applying on cell rather than LD charge. Good!

[Mikey0124]

Hello everyone! I have been a voip.ms customer for about 2 months. Everything works well but I am finding that people I talk to keep hearing like I am pushing a button on my phone once and a while. I am currently using the value route and I am guessing it is because of that but am wondering if it may be another issue. Does anyone have any thoughts about this?

Also I am setting up my QOS through a router with tomato firmware and I got everything working really well. I just have one question. How much bandwidth does my voip need to work well? I am trying to give my computer the max connection speed but without causing problems for my voip.

Thanks....

[Mango]

Try setting DTMF Tx Method to InBand. You may find this setting on the Line tab of your ATA. You also need to set DTMF Mode to INBAND on the VoIP.ms portal. You will find this either in Account Settings, or Edit Sub Account if you use a Sub Account. That should solve the problem.

With regards to bandwidth consumption, you will use approximately 87 kilobits per second, in both directions, if you use the G.711 (PSTN quality) codec. However...you typically need to reserve more than exactly 87 kilobits per second since your internet provider will not deliver 100% of rated speed at all times. The answer to your question is a bit of a guessing game. I like to leave more strict rules on my inbound QoS than I do on outbound. This is because it is harder for the router to do inbound QoS (though it is possible.)

m.

[Mikey0124]

Thanks for the quick reply Mango. I have adjusted my settings as you suggested. I will send an update in a few days and let you know if it helped. As for bandwidth right now I have about 400 kbit/s free for my voip.ms connection. I am hoping that is enough. I am going to try and leave it at about 128 Kbps free for my voip. I was reading on another forum that 128 kbit's is the optimal amount of bandwidth to leave for voip.

[UberDave]

Hey forum, I have noticed a rise in telemarketers calling since my number was ported over. I'm not sure why but to try and block them I've routed all calls to an IVR and added some filters but I'm having a problem.

Here's how it's set up:
DID setup to go to IVR by default.
647* --> Filtered to bypass the IVR and ring direct (a local area code)
6477260056 -->. Filtered to hang up. (telemarketer)

The number filtered out still bypasses the IVR and rings direct. Is there a way to setup these rules to work? I realize there is a conflict but I'm not sure how to handle this.

[recneps77]

Perhaps the order?
You have 647 wildcard before the specific, so it would take effect first.
If you had the specific one listed, it would read 6471234567, therefore, hang up.
If not 6471234567, move onto the next rule: 647 wildcard, therefore, ring through.

Just a thought. You might have already changed the order. If so, disregard