Dead air on a misc destination

I have a misc destination that I can straight dial from my phone that works perfectly, however, whenever I assign it to a route time condition or anything I get dead air. Here is the relevant section from the logs:

app_dial.c: SIP/BW-SIP-A-0000008b answered SIP/BW-SIP-A-0000008a
[2019-05-06 17:37:24] VERBOSE[9110][C-00000039] bridge_channel.c: Channel SIP/BW-SIP-A-0000008b joined ‘simple_bridge’ basic-bridge <5127f1b1-f21b-4e8f-9d27-f8e7749298a1>
[2019-05-06 17:37:24] VERBOSE[9074][C-00000039] bridge_channel.c: Channel SIP/BW-SIP-A-0000008a joined ‘simple_bridge’ basic-bridge <5127f1b1-f21b-4e8f-9d27-f8e7749298a1>
[2019-05-06 17:37:54] NOTICE[5666] chan_sip.c: Disconnecting call ‘SIP/BW-SIP-A-0000008a’ for lack of RTP activity in 31 seconds

This is a recurring complaint, and it it probably due the fact that your PBX is behind a firewall without the RTP port range forwarded. You might be able to band-aid by enabling progressinband, or enable force answer on the inbound route.

The RTP port range is fine on the firewall. Already opened up. Force answer is enabled on the inbound route. Here’s some more info.

The misc destination that doesn’t work points to another FreePBX that we have running. If I straight up call the number it works fine. If I point anything on PBX-A to that misc destination it goes to dead air. If I put my cell phone in as a misc dest and point the same entries to it, it rings and works just fine. In fact it’s only this one misc destination that’s not working. I even deleted and recreated the misc destination with the same results. Weird.

By the way, Hey Lorne, it’s me.

Kafluke = Kris at Supplemental Health Care

Hi Kris!

It would not be very sporting of me to say “I told you so”, but it did turn out the that PBX SIP RTP range and the port range being forwarded at the router did not line up neatly.

That was a different PBX. This is a whole separate issue on a different server. I actually have another ticket into Sangoma for this one too. This is dead air on only one misc destination. All others work fine. RTP on this PBX matches the firewall perfectly. I can even call the number in the misc destination from my phone and it works. It’s only when the PBX tries to use this specific misc destination that we get dead air.

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mea culpa. This is your week for one-way audio.

Try this. At the Asterisk console run the command:

channel originate local/[email protected] application echo

with the xx’s being a DID that go out the trunk in question. Answer the call and you should hear your own voice echoed back. If you don’t investigate the RTP port forward.

yeah that worked fine. Weird thing, everything else on the PBX works. DID calling, internal and external extension calling. It’s only when it tries to dial that one misc destination that we get dead air. I set up a test inbound route that points to that misc destination if you feel like looking further into it. All the info is in the Sangoma ticket. (I even requested you by name lol)

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