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Links: ·ALL ·Review Your VoIP Provider ·VoIP Providers ·VoIP FAQ ·Porting Rules ·What Codec?
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torwill

join:2011-01-28

Multiple ATA's/DID's and one way audio

Hello,

I am hoping to get some suggestions, Thanks!

Voip provider is voip.ms,

SPA3000 - one DID(line1), using proxy toronto.voip.ms:5060, SIP Port 5060, and STUN stun.sipgate.net

SPA112 - two DID
first DID(line1) using proxy toronto.voip.ms:5080, and SIP Port 5061, and no STUN
second DID(line2) using proxy toronto.voip.ms:5080, and SIP Port 5062, and no STUN

Both ATA's are behind my home router running DD-WRT.

I have no problem with only SPA3000 running, no one way audio, no problem registering.

But I am having problems when both ATA's are running. When I am on the phone using SPA3000, if someone wants to dial out using SPA112 and either line, one way audio occurs on SPA112's lines. With the following message in syslog,

02-08-2012 22:45:21 Kernel.Error 192.168.1.36 *** ICMP error on RTP ch 0. Err count 1.

I have tried using three different ports that voip.ms provides, and 3 different SIP port (5060,5061,5062) on each line, and it doesn't work. Also tried using a different STUN on SPA112 and no success there either.

Can anyone please give me suggestions on this set up? am I even doing this right? (I am pretty new to this).

Thank you.

gweidenh

join:2002-05-18
Houston, TX
kudos:1

Interesting, so you have multiple physical devices AND you are making multiple registrations to the same server on the same device. I can't say I've tried this combination before.

What you are doing with the SIP port (5060, 5061, 5062) is what I do and it works fine under pfsense router.

I would actually recommend getting rid of line 2 on the SPA112 temporarily. Keep the configuration the same.

I do wonder why you need the STUN server on the SPA3000. Does it not work properly without the STUN server?

And if the STUN server is needed, why not try it on the SPA112 as well?


gweidenh

join:2002-05-18
Houston, TX
kudos:1

reply to torwill
I would also take the SPA3000 offline temporarily and just try to get the SPA112 working on its own. First with 1 line, then with both lines.

If you can get 2-way audio working (try with and without STUN, leave the port settings the way they are) then add the SPA3000 back into the equation.


Stewart

join:2005-07-13
kudos:14

reply to torwill
Make sure that the three clients are registered to different sub-accounts, with NAT set to yes for all.

Set unique RTP port ranges for the two ATAs, e.g. one has RTP Port Min: 16384; RTP Port Max: 16432 and the other has RTP Port Min: 16434 and RTP Port Max: 16482.

Reboot router and both ATAs before retesting.

If no luck, post modem make/model and whether it's the ATA user or the remote party that receives no audio.


jason_m

join:2010-01-09
Peabody, MA

reply to torwill

about STUN

Yes, Stun isn't normally required these days unless your provider is one of the few that still requires it.

Stun is a way to obtain the external IP:port, but most providers send that information back in normal Sip conversations.

If something happened in the past that caused you to turn on Stun, then your Nat traversal parameters aren't quite right.

Of course, not using another provider's Stun server eliminates a possibility of your phones not working if Sipgate decides to have an outage, you would loose everything.

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