Unable to hear the person I am trying to talk to.
toddmd2:
I got OBi set up properly with my Google Voice number, I can place and receive calls no problem, but while the person I'm connected to can hear me I cannot hear them. I have been unsuccessful in troubleshooting this issue myself and wonder if anyone has any ideas?
QBZappy:
toddmd2,
Welcome.
Two things to try.
1) Port forward the RTP ports to the OBi:
Service Providers
ITSP Profile X
General
SIP
RTP <--
2) Setup a Stun server
toddmd2:
Thanks for the quick reply Zappy. Being a bit new to this whole thing I need to ask: When you talk about forwarding the ports, are you talking about the ports on my router? If so, I see no RTP options. I guess what I'm asking is if you can dumb it down a bit? Sorry.
QBZappy:
toddmd2,
Quote from: toddmd2 on November 09, 2012, 04:03:39 pm
I need to ask: When you talk about forwarding the ports, are you talking about the ports on my router? If so, I see no RTP options.
None of these suggestions may resolve your issue, however I suspect that you may not have covered some of the more basic trouble shooting steps.You did not give any specifics about your particular setup. You should mention the router brand/model. Is the call GV<->PSTN, GV<->sip, GV<->GV?
FAQ ( http://www.obihai.com/FAQ.html ) These should cover all the ports for an OBi110/100. The OBi202 has additional RTP ports to open. (SP3+SP4)
What ports should I keep open on my router/firewall?
In order for your OBi to be able to send packets w/o interruption, please configure your router as follows:
Allow Outgoing:
TCP Ports: 6800, 5222, 5223
UDP Ports: 5060, 5061, 10000 to 11000, 16600 to 16998, 19305
Allow Incoming on UDP Port: 10000
Additional troubleshooting steps in the event you decide to set up a sip account and encounter one way audio.
1) Try to disable the sip ALG setting in the router. In many routers this breaks sip. In your case you mentioned using GV. GV does not use a sip protocol, so I don't expect it to fix the issue. This is more a preventative measure for when you setup a sip account in the future.
2) Set up a Stun server. A stun server helps the router and the ATAs (OBi) in the communication to negogiate the WAN ip (internet) address and port numbers of the call automatically using the sip protocol. Note that once again this is developed for sip clients. In the case of GV I don't expect it to resolve one way audio.
Have a look here for port forwarding info. ( http://portforward.com/ )You should find info specificly related to your router brand. You will not find any reference to RTP in the router itself. The RTP ports are used to pass the voice part of the conversation. On the OBi you will find the specific port numbers to forward where I indicated with the arrow. They above ports from the FAQ is more complete.
This should keep you busy for awhile. :D
toddmd2:
I probably should have provided a bit more info. My router is a LinksysWRT54gTM (a t-mobile branded version of the standard 54g) and I am attempting to run a GV <-> PSTN.
On another post I found a solution that suggested forwarding port 19305, I did this and it fixed my problem, then it stopped working.
Additionally, do you find it better to run a network of static IPs so that the ports are always being forwarded to the same IP? Right now my OBi 100 is on .107, on a DHCP setup, that IP will change at some point so my port forwarding will need to change also correct?
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