View Full Version : SIP to FXO tuning?

07-27-10, 12:45 AM
I have a couple of SPA942 linked up to the Warp as SIP phones needing to use the FXO ports to make outgoing calls over analog lines. Outbound rules seem fine as the dial plan routes the numbers correctly but I'm told by callers at the other end that they're having problems hearing due to drop outs. I can hear them just fine....

Could this be a codec issue going from SIP to analog? Or some other tuning parameter for the FXO that I've overlooked? Although the Warp is NAT'd behind a firewall, the SIP phones and the Warp are on the same subnet behind the firewall and the call is going out through analog so I can't see that the usual Asterisk configuration issues with NAT and SIP are at work here...

Are there any FXO audio tuning parameters I should look at first?


07-27-10, 04:13 PM
A few questions to start -
* Is this occurring on all calls?
* Are the dropouts occurring intermittently or is there a repeatable pattern?
* Do you know how long the dropouts last for?

I am not aware of any FXO tuning parameters. But I do have some thoughts on possible ways to proceed. First I would ensure that logging and any CPU intensive applications are disabled on Warp. Do you know what IP codec is being used?

Otherwise it sounds like this could related to network disturbances even though this is hard to believe considering it sounds like this is traversing a LAN. Have you tried other IP to IP conversations? have you tried FXS to FXO conversations to eliminate the IP leg? Is this reliable switch/hardware?

Just some thoughts.

07-27-10, 04:59 PM
Thanks for the follow up. To answer your questions:

It happens for any SIP handset call placed through an FXO trunk
The dropouts seem consistent and last apparently under a second. It was described to me as sounding like every fourth or fifth word went missing
Using a FXS handset does not exhibit a problem.

Which is the better choice for IP codec for the SIP phones based on the Warp CPU capacity - we're only talking about half dozen handsets maximum in a domestic geek setting with no more than two concurrent calls at a time, many times just a single call, so the overall network and CPU load on the Warp should be minimal, although I have seen the System Status screen in FreePBX show the CPU briefly flash to 100% or no readily apparent reason (sometimes when not on a call).

Are there any logs which would help me try to figure out where to look more closely?