Caller unable to Hear

[fch]

I purchased the Grandtream 286 ATA and config file about a month ago.
I have been unable to make my FPL service work completely.

When I make a call, I have no issues.. both caller can hear the voice properly and quality is good.

However, when someone calls me, then cannot hear me while I can hear them fine.
This has been going on for almost couple of weeks. I just call back the person again.

I have cable internet 10MBps/0.5MBps service from teksavvy.
The router is Asus WL500w with Asus firmware ver 3.0
I have assigned static IP address to ATA based on MAC address.
In port forwarding, I have enabled ports 5060 and 5061 to forward(UDP)
Virtual Server is enabled in the router setup, with ATA's static IP address and port 5060, 5061.
Router firewall is enabled, but no setup is defined.
Qos is set to assign highest pririoty to VOIP/Streaming, with highest priority to Static IP address of ATA

Questions:

1) Is port forwarding only helps with SIP registration and incoming calls?
2) What ports should be defined in port forwarding and virtual server section of router?
3) What is RTP? My ATA has RTP set to 5004, do I need to add 5004 to port forwarding and virtual server setup as well.
4) What codec should the ATA to be setup for? My SIP profile states G729..However the ATA from FPL (pre-config) and the online setting document states to use "PCMU" as the first choice codec

Any help is greatly appreciated.

Thanks,

[fch]

One more thing to add..

All the incoming calls on which caller to my number are unable to hear show up in my account call logs as multiple entry with identical time stamp.

Any idea what may be going on...

[FONGO_steve]

1) Is port forwarding only helps with SIP registration and incoming calls?

Port forwarding usually catches problems with inbound calls. A lot of routers still don't have stellar VOIP support, and thus don't know what to do when they see incoming VOIP traffic, so it just gets blocked (to put that in layman's terms). If you PM me your phone number, I could try forcing our server to always proxy your calls, which might help.

2) What ports should be defined in port forwarding and virtual server section of router?

You usually only need one or the other, that being port forwarding or virtual server. I'm not familiar with Asus router's enough to say which is the preferred choice for VOIP. I used to use Virtual Server on my D-Link router before moving over to a DD-WRT Linksys.

3) What is RTP? My ATA has RTP set to 5004, do I need to add 5004 to port forwarding and virtual server setup as well.

RTP is the audio triggering port, it's not nearly as important as the SIP ports when it comes to audio issues. If you're forwarding ports though, it doesn't hurt to add it in there.

4) What codec should the ATA to be setup for? My SIP profile states G729..However the ATA from FPL (pre-config) and the online setting document states to use "PCMU" as the first choice codec

Either will do to be honest, just PCMU will give you a little better quality (if your ear can even hear the difference)

All the incoming calls on which caller to my number are unable to hear show up in my account call logs as multiple entry with identical time stamp.

The second log is when the call jumps over to voicemail - we're still trying to find a way to make that more distinguishable in the call logs since they're pulled directly from another server.

[fch]

Steve,

I have pm'ed my number to you.

[fch]

How is this possible?

My seconds till voicemail is set to 30seconds. However, all the calls in which caller is unable to hear me and has multiple identical call log entires are less than 30 seconds.


fch wrote:
All the incoming calls on which caller to my number are unable to hear show up in my account call logs as multiple entry with identical time stamp.
Steve wrote:
The second log is when the call jumps over to voicemail - we're still trying to find a way to make that more distinguishable in the call logs since they're pulled directly from another server.