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Unread 09-30-2009, 10:20 PM   #1
khabibul35
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Default One outbound provider works, other doesn't

Hi, I've just started up and have been receiving calls with Gizmo without a hitch. However, outgoing is a different story. At first, I thought outgoing was working fine as well. Then, I finally got past the 3 minute mark and realized I was calling through Gizmo. When I switched my outgoing to Betamax, suddenly when I called, the other party couldn't hear me, though I could hear them.

Now, usually, I would assume it's the VOIP connection or router, but under this circumstance it seems to me it's gotta be on voxalot's end. Are the any special settings I need to enter under the providers tab? I don't see much of difference in option except that gizmo is registered and that webcalldirect is not but that shouldn't matter for outgoing right?

Any ideas?
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Unread 10-01-2009, 04:08 AM   #2
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One-way audio problem are frequently (nearly always?) caused by routing/router/NAT issues. The first step is to try using a STUN server if you're not already. The next is to forward the RTP range of ports, and if still having problems, the SIP port(s).
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Unread 10-01-2009, 02:07 PM   #3
khabibul35
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STUN is on and port forwarding is also done. So... that's not the problem.

However, even if it was, I don't understand why Gizmo would work but Betamax wouldn't? Why would gizmo get through the ports but not betamax. Afterall, it's all PAP2 -> voxalot -> gizmo/betamax. Wouldnt the fact that gizmo connects indicate that it's a voxalot -> betamax problem and not a PAP2 ->voxalot problem by the virtue that the port works fine.
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Unread 10-01-2009, 07:00 PM   #4
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Quote:
Originally Posted by khabibul35 View Post
Why would gizmo get through the ports but not betamax?
I can't be sure without doing some research, but if you have "NAT Mapping Enable" off then it may be something to do with that.

First, set "Symmetric NAT Handling" to 'Yes' in your Voxalot account. Then install the following settings to your ATA and test again. Let us know if problem is resolved or not.

(under SIP tab)
Handle_VIA_received: yes
Handle_VIA_rport: yes
Insert_VIA_received: yes
Insert_VIA_rport: yes
Substitute_VIA_Addr: yes
Send_Resp_To_Src_Port: yes
STUN_Enable: yes
STUN_Test_Enable: yes
STUN_Server: stun.voxalot.com:3478 (or any STUN server such as 'stun01.sipphone.com:3478' or 'stun.sipgate.net:10000')
NAT_Keep_Alive_Intvl: 179 (if your phone does not ring try 119 or 59, use highest number that works)

(under Line_1 and Line_2 (or PSTN_Line) tabs)
NAT_Mapping_Enable: yes
NAT_Keep_Alive_Enable: yes
NAT_Keep_Alive_Msg: 0000
NAT_Keep_Alive_Dest: $PROXY
Register Expires: 3600

----------

Optional settings, steps 1 - 4:
Sometimes its preferable to operate the ATA independently of a STUN server. The benefit is that phone service can continue no matter if the STUN server is working. See steps 1 to 4 below.

1. Forward the SIP ports and the RTP port range from the router to the ATA.
2. Set "STUN Enable:" no
3. Set "NAT Keep Alive Enable:" no
4. The ATA must learn it's public IP address, either through normal SIP registration, or place your public IP address in the 'EXT IP:' field).
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