Set up Obi212 as FXO for FreePBX, intermittent incoming audio issue
Nmennear:
Quote from: drgeoff on March 29, 2019, 12:39:16 pm
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?
1. Thanks, this makes sense
2. The dropped audio is always on my side but we're going to test based on your scenarios and i will report back. Our configuration is:
POTS LINE -- OBi212 LINE jack -- LAN -- PBX -- LAN -- Yealink t46s SIP endpoint
Calls between extensions (entirely internal calls) are working perfectly.
I will report back soon, thanks for your suggestions!
UPDATE: I've done the above tests and calls going to VM, pots calls made and pots calls received all have the audio drops. Any call made through FreePBX over the pstn has the same intermittent issue on the incoming audio (only).
As an aside, testing directly at the pots demarc shows it's not an issue at that point. Likewise, i made a couple calls from an analog phone attached to the OBi FXS port (directly to the pots line / bypassing the FreePBX entirely) which seem to not have the intermittent audio issue either.
My testing has shown the intermittent audio can start almost immediately after the start of the call or can take 5-10+ minutes to start.
buffaloco:
Your issue sounds almost exactly like the issue that we are experiencing and have been troubleshooting. Did you ever find any resolution to the issue?
drgeoff:
Quote from: Nmennear on March 29, 2019, 01:02:18 pm
Quote from: drgeoff on March 29, 2019, 12:39:16 pm
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?
1. Thanks, this makes sense
2. The dropped audio is always on my side but we're going to test based on your scenarios and i will report back. Our configuration is:
POTS LINE -- OBi212 LINE jack -- LAN -- PBX -- LAN -- Yealink t46s SIP endpoint
Calls between extensions (entirely internal calls) are working perfectly.
I will report back soon, thanks for your suggestions!
UPDATE: I've done the above tests and calls going to VM, pots calls made and pots calls received all have the audio drops. Any call made through FreePBX over the pstn has the same intermittent issue on the incoming audio (only).
As an aside, testing directly at the pots demarc shows it's not an issue at that point. Likewise, i made a couple calls from an analog phone attached to the OBi FXS port (directly to the pots line / bypassing the FreePBX entirely) which seem to not have the intermittent audio issue either.
My testing has shown the intermittent audio can start almost immediately after the start of the call or can take 5-10+ minutes to start.
I hadn't noticed your update until just now.
That you can make and receive calls OK using analogue phone - OBi212 - POTS suggests that all the analogue gubbins and some of the digital gubbins of the 212 is not responsible for the audio drops. But you do not seem to have tested digital/IP stuff from 212 to the PBX. Make a call from the analogue phone on the 212 to an extension on the PBX. And make a call from the analogue phone on the 212 using VoIP but not going through the PBX. That is using an SP on the 212 that is not the PBX and not POTS. If you don't have an ITSP that you can temporarily register one of the 212's SPs to, try the OBi test number **9 222 222 222.
If the first of those tests has the drops, that would point to the 212 to PBX section. If the second one never drops incoming audio that would point to the issue being more with the PBX, whereas if it does have drops it would point to a problem with the 212.
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