All calls go to Voicemail with ATA

Criatura83 · Post by **Criatura83** » Wed Mar 25, 2009 3:03 pm

Hello Guys,

Ok, i got my MJ last week and i set it up with my ATA (Sipura 3000), after dealing with the dial plans, i got it to work as i wanted but my problem is that i cant receive any calls at all, ALL calls go straight to voicemail, if i disconnect the ATA and plug the MJ to the computer i am able to receive calls just fine. But i want to use my ATA.

Incoming calls from my PSTN line are coming in just fine.

Can anybody help me out, what settings do you guys need to know to check if they are right?

Thank You.

szebrin · Post by **szebrin** » Wed Mar 25, 2009 4:06 pm

I had a similar experience with a PAP2 and made a few changes to the "line 1" configuration. I *think* this was the one that worked:

NAT Mapping Enable: YES
NAT Keep Alive Enable: YES
NAT Keep Alive Msg: $NOTIFY
NAT Keep Alive Dest: $PROXY

Hopefully this will work for you.

Criatura83 · Post by **Criatura83** » Wed Mar 25, 2009 4:20 pm

Thank you for the reply szebrin, that is exactly what i have in line 1.

i have even disable the nap mapping but nothing.

i also set the register expires to 840 and yesterday to 180 but no luck.

i dont know what to do, i'll keep on searching on the forum.

if anybody knows what might be causing this problem go ahead please.

mberlant · Post by **mberlant** » Wed Mar 25, 2009 7:31 pm

Have you set up STUN? Does your ATA say that it is Registered?

Criatura83 · Post by **Criatura83** » Wed Mar 25, 2009 8:30 pm

mberlant wrote:Have you set up STUN? Does your ATA say that it is Registered?

Yes, i have set up STUN but not sure if it is correct, YES the ATA says it is Registered.

This is what i have on the SIP Tab under NAT Support Parameters.

Handle VIA received: no
Insert VIA received: no
Substitute VIA Addr: yes
STUN Enable: yes
STUN Server: stun.ekiga.net:3478
EXT RTP Port Min: "Nothing Here"

Handle VIA rport: yes
Insert VIA rport: yes
Send Resp To Src Port:yes
STUN Test Enable:no
EXT IP: "Nothing Here"
NAT Keep Alive Intvl:20

Let me know if there is anything wrong.

Thanks.

mberlant · Post by **mberlant** » Wed Mar 25, 2009 10:23 pm

Try changing all of those parameters (except STUN Enable, of course) to no. The fact that you are Registered indicates that STUN is working for you and that the problem is in setting up an actual phone call.

Criatura83 · Post by **Criatura83** » Thu Mar 26, 2009 2:27 am

Thank you so much for your help mberlant, i change all the parameters to no and its working flawless now.

Hey mberlant, i would like to know if i can do a VOIP to PSTN and PSTN to VOIP gateway with my sipura and MJ, i am asking because i was searching the forum but i couldnt find anything about it.

Thanks.

mberlant · Post by **mberlant** » Thu Mar 26, 2009 4:49 am

You need to use an ATA with an FXO interface (to physically connect to a phone line) rather than an FXS interface (to physically connect to a telephone instrument). Devices like the Linksys/Sipura SPA-3000 or -3102, which have one FXS line and one FXO line, are perfect for this application.

Criatura83 · Post by **Criatura83** » Thu Mar 26, 2009 7:26 pm

Thank you, i will try to do it during the weekend, i have to read how to do it but if i have any problems i'll be posting lol.

Have a nice day.