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Unread 02-24-2007, 05:08 PM   #1
rybshik
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Default in-bound calls either drop or do not stop ringing

I have Sipura 3000 behind the NAT (DSL modem +Linksys WRT54 router).
Sipura is registetered with Voxalot.

Out-bound calls are fine, but in-bound calls either drop on the first ring or do not stop ringing after Sipura's phone handset is picked up.

My Sipura's settings are exactly as in
Voxalot and VOIP Tutorial: A Comprehensive Walkthrough


I guess something needs to be changed in Sipura settings. Please advice

Last edited by rybshik; 02-24-2007 at 05:16 PM.
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Unread 02-24-2007, 07:18 PM   #2
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Unfortunately, connection problems due to firewalls are some of the most difficult to troubleshoot. Here's some advice (may have already been done):

0) Test calls from multiple sources, ideally both:
--a direct SIP URI call from a client like the Gizmo Project softphone
--a call from the PSTN to your DID that is either registered with Voxalot or mapped to Voxalot

1) Toggle from:

NAT Support Parameters heading:
Handle VIA received: No
Handle VIA rport: No
Insert VIA received: No
Insert VIA rport: No
Substitute VIA Addr: Yes
Send Resp To Src Port: Yes
STUN Enable: Yes
STUN Test Enable: No
STUN Server: stun.xten.com
EXT IP: (blank)
EXT RTP Port Min: (blank)
NAT Keep Alive Intvl: 30

to:

NAT Support Parameters heading:
Handle VIA received: Yes
Handle VIA rport: Yes
Insert VIA received: Yes
Insert VIA rport: Yes
Substitute VIA Addr: No
Send Resp To Src Port: No
STUN Enable: Yes
STUN Test Enable: No
STUN Server: stun.xten.com
EXT IP: (blank)
EXT RTP Port Min: (blank)
NAT Keep Alive Intvl: 30

2) Put your SPA-3000 in the WRT54's DMZ (or more exactly, forward the SIP and RTP port ranges used by the SPA to it).
see this thread to clarify why it might be good: http://www.dslreports.com/forum/remark,17884348

3) Toggle between Yes and No for the "Enable symmetric NAT handling" setting on your Voxalot portal.

4) Try temporarily to use a different router and start the process over.
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Unread 02-24-2007, 07:31 PM   #3
rybshik
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Quote:
Originally Posted by ctylor View Post
Test calls from multiple sources, ideally both:
--a call from the PSTN to your DID that is either registered with Voxalot or mapped to Voxalot
Ok, here are some test results. I recall that my Voxalot's Sipura is behind the NAT.

In-bound SIP URI calls from Callcentric and DIDWW drop after one ring or do not stop ringing.

In-bound SIP URI calls from my own remote (i.e. different external IP and different sub-net) asterisk are fine.
Martin, why might it be?

In-bound calls from Vitelity (which is registered to Voxalot) are fine.

Last edited by rybshik; 02-24-2007 at 11:58 PM.
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Unread 02-25-2007, 10:51 AM   #4
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Voxalot is very versatile in its handling of ATA's behind NAT so calls to DID's registered in Voxalot should work ok.

Are the In-bound SIP URI calls from Callcentric and DIDWW going via Voxalot?

If it works for some and not others, it might be due to the G729 Codec description in the SPA 3000. As a quick test try setting the preferred codec to G711u and set use preferred = yes and see if that makes a difference.
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