We are having an issue with Misc Destinations on our FreePBX 22.214.171.124 system.
I have setup a Misc Destination to forward to my cell phone, if i setup VmX on my extension and press 1 when my voicemail picks up it forwards to my cell phone no problem.
If I setup the call to forward to the Misc Destination after 15 seconds with no answer the call rings through but there is no sound in or out.
Lack of rtp session survivability has nothing to do with FreePBX.
Look to your router’s capabilities and setup.
So why is this working fine with VmX and not with Timeout? same destination.
I would have thought the same thing but other then this our system works flawlessly for 100+ users.
Sorry I have no idea, they both work for me. Perhaps using tcpdump and/or setting rtp debug on will help you, again it is a network problem not FreePBX or Asterisk. VmX is a parallel process timeout is serial, look to your sip and rtp sessions on your router.
Thanks for the input, maybe I will try bypassing the router to confirm and then take it from there.
You were correct! Thanks again! I also found this that refers exactly to my problem
CONSIDER FORWARDING RTP MEDIA PORTS FROM YOUR FIREWALL TO YOUR FREEPBX: If you don’t forward the RTP Media Ports from your router to your FreePBX, you can encounter situations where callers who come in from VOIP SIP Trunks and who are transferred directly to another outside line without first receiving a voice prompt will have no audio. There are several ways to fix this. First, you can ensure that all calls are answered by something on your system (such as an auto attendant) before being transferred outside your system. Or, you can use only IAX Trunks. If do either of these, no other changes are needed.
However, if you plan to use SIP Trunks from external VOIP providers and you want to forward calls before they are answered by your system, then you’ll need to forward the RTP Media Ports from your Firewall to your FreePBX machine. Forward firewall UDP ports 10000 to 20000 to FreePBX server.
Please note that Port 10000 is also used for webmin (a tool that can be used to make substantial configuration changes on your machine using a web browser). If you have webmin on port 10000, either change webmin’s default port to something else (such as 9001), or change the default RTP Media Ports from 10000-20000 to 10001-20000.
A range of 10000 ports available for RTP Media is often unnecessarily large for most small systems, because one call requires only 4 active ports. Thus, you might consider narrowing the range of ports used for RTP Media. If you do narrow the range, keep the range somewhere within 10000 to 20000 (i.e. don’t select 43500 to 44500), as going outside this range can lead to call quality issues.
A fuller and way more verbose expanation than I could ever come up with, glad you are now working.