Set up Obi212 as FXO for FreePBX, intermittent incoming audio issue
Nmennear:
I'm in the process of rolling out a few FreePBX systems using OBi212s as FXO gateways. So far I've followed the instructions in the sticky and am able to successfully make & receive calls via the OBi212, but I am having persisting issues with incoming call audio dropping for 5-10 seconds intermittently. It seems to only be happening for the incoming call audio (from the PSTN) as the outside caller can hear me throughout the outage.
MY testing so far has all been done locally within the LAN and I've disabled SIP ALG in my router in the event it was causing issues. I've upgraded my OBi's to 3.2.2 firmware and have tried various settings minor settings in the Obi without any success.
I am having the same issue at every location so i suspect there is an important (and hopefully obvious) setting i am missing somwehere. My configuration at each location is identical - FreePBX installed on a physical Gigabyte BRIX, Ubiquiti Edgerouter, OBi212, netgear gigabit switches / WAP.
I've noticed that the packet counters on the OBi continue to show traffic during the dropped audio so it may be something on the FreePBX-side or network related.
Not sure what additional info I can add that would be helpful. Sorry for my ignorance, I've been trying to figure this out for the better part of the week and am unsure where to look at this point.
Thank you very much for any help you can provide!
Nmennear:
Just to add to my previous post..
A chan_sip trunk is configured to Asterisk using UDP.
Obi box is configued as bridged and a LAN cable from in place Netgear gigabit switch is connected to the WAN port on the Obi 212.
Settings we have tried:
- Using the LAN port instead
- T38Enable Off
- Using PJSIP and port instead (same issue)
- Force g729 codec instead
- Increase SilenceTimeThreshold
- DetectFarEndLongSilence Off
- DetectCPC Off
- SilenceDetectSensitivity to Low
- Disabled registration
- Adjusted MTU size
As mentioned watching the call status, packet counts continue to increase during the one-way audio dropout. Codecs remain as specified (711u) and there isn't anything abnormal in the displayed info (such as increased dropped packets, late packets, etc.).
It seems like the dropped audio problem happens more quickly when calling / receiving calls from cellular phones than from pots lines but this may be a coincidence.
azrobert:
This forum is more FreePBX knowledgeable:
https://www.dslreports.com/forum/voip
drgeoff:
1. A router actions SIP ALG only on WAN traffic. Your OBi to PBX packets are purely on the LAN.
2. Are you certain about where the incoming audio is getting dropped? If you are using an analogue phone on the 212 then you have
POTS LINE -- OBi212 LINE jack -- LAN -- PBX -- LAN -- OBi212 phone jack -- phone.
If an incoming POTS call goes to voicemail, are there drops in the recorded message?
Are the drops in incoming audio present in both POTS calls made and POTS calls received or in only one of those?
If you use the analogue phone on calls to and from a VoIP distant end do the drops occur?
Nmennear:
Thanks for the tip! I am posting there now as well.
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