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Unread 05-17-2007, 08:15 PM   #1
pironato
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Default Incoming call problem

Hello,
I have a problem with incoming calls.
I have 3 VOIP accounts: 2 for incoming calls (one in Italy and one in Finland) and one for outgoing call.
Then the problem happen pretty often with incoming call: when i get the line i don't hear nothing; same in the other side they don't hear nothing as well.
This problem happen with both providers.
This doesn't happen with outgoing calls.
My Voxalot account is registered in a Grandstream budge tone 100.

Do you have some idea, suggestion to resolve this problem?

Thanks!
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Unread 05-18-2007, 08:54 AM   #2
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Could be a firewall problem. Do you have NAT enabled on your ATA?
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Unread 05-18-2007, 12:22 PM   #3
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Can you have good 2 way conversation to the Voxalot account?
Try calling your account from a landline using a SIP-Broker access number.
Also, have you tried calling test numbers (Eg. *600)?
If you DMZ the phone, there should be no firewall issues.

Do you have more than 1 SIP device or softphone running behind the same firewall?

Last edited by kurun; 05-18-2007 at 12:23 PM. Reason: Typo's
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Unread 05-18-2007, 08:20 PM   #4
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If it is a firewall problem can it be that some time works and some time doesn't work?
I checked out my phone setting(grandstream budge tone 100) and i can see only 2 parameters related to nat (as far as I can understand):
NAT Traversal: No
Use NAT IP: empty

I tried 2 way conversation to the voxalot account and it worked ok:
with *600 i was able to hear my voice repeated back with a very good latency.
i tried a SIP-broken access number as well and it worked just fine.

thanks guys!
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Unread 05-19-2007, 02:38 PM   #5
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If you have more than one SIP device active at the same time, I found it works far more reliably if the SIP Port is incremented for every SIP channel (5060, 5061, 5062, 5063, etc).

I currently have a GXP2000, HT496, and a BT102 running reliably behind the same router. I was losing the registration on the ATA (HT496) before I did this change, and incoming calls would ring on the GXP2000, but not on the ATA registered to the same Voxalot account.

Calling from the ATA to the GXP phone, or vice-versa, would result in one way or no conversation most times.

Specifying a STUN server also seemed to help make the set-up work reliably.

For the GS BT-10x, I suggest the following:

NAT Traversal - Yes
STUN Server is - stun.voxalot.com.au:3478
Keep Alive Interval - 20 sec.
Use NAT IP: empty (Not sure what this does)
Register Expiration: Set to between 2 - 5 minutes (Seems to work better, and I hope it does not cause any problems for the Voxalot servers)

[Added - By the way, does your BT100 have the latest firmware from the GS website?]

Once you establish that the Voxalot account is working reliably with your IP-Phone, then you can focus on the problems with the incoming calls from the registered accounts (Italy and Finland numbers).

If you can do a simple forward from these accounts to xxxxxx@voxalot.com, your incoming will work far more reliably, as there will be no registration issues between Voxalot and the other VSPs. Unfortunately, many VSPs do not allow this, in which case a registration in the Voxalot account is required.

_.

Last edited by kurun; 05-19-2007 at 03:03 PM.
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Unread 05-19-2007, 09:08 PM   #6
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Thanks Kurun for the support.
I forgot to tell you that i have only one sip device, sorry.

I setup my bt100 as you suggested and i'll check next days the situation.

I don't have the clue to understand what you said:
"If you can do a simple forward from these accounts to xxxxxx@voxalot.com, your incoming will work far more reliably, as there will be no registration issues between Voxalot and the other VSPs"

my accout is 637400
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