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HT-503 as an Asterisk gateway
Hello,I have a couple of HT-503's connected to an Asterisk (FreePBX) server that function as both an ATA with a SIP account and an FXO gateway for PSTN lines.Like with many others, my HT503's were less than reliable. To the point where I had to write watchdog code on a server to power cycle the HT503 after it had crashed. I used an X10 module for this. Other problems included the fact that if I received a call on the FXO port, a SIP call would be sent to Asterisk and continue to ring even if the FXO caller immediately hung up. That SIP call would continue until VM answered and there was a few seconds of silence message - along with a new message lamp trigger, very annoying.I made a small change to how FreePBX answers incoming calls from the HT-503 that has improved the gateway's reliability (so far).- The 503's FXO port is setup as a Trunk in Asterisk. Context=from-trunk and very importantly: port=5062. You must change the port number or Asterisk will send the outbound trunk call to the FXS port and cause issues. The FXS SIP account runs on default port 5060.- I send the incoming FXO calls to a Ring Group because I have several extensions that receive the FXO calls. This also allows me to insert a silence annoucement, as described below.- I then add an "announcement" of 1 second of silence (Silence-1) in that Ring Group. The FXO caller hears nothing out of ordinary because the "silence" announcement falls between rings.The net effect is that now the 503 has a "completed" call and it terminates the trunk call if FXO hangs up. The other plus is that, at least in my case, the HT503 has not crashed in over 45 days since I've made this change. This is significant since it had crashed over 75 times in the previous 90 days.FreePBX setup:SIP TRUNK:Maximum Channels: 1 (you can only receive/make 1 call at a time over a PSTN)outgoing settings:context=from-trunkusername=ENTER FXO SIP ACCOUNT HEREtype=peersecret=ENTER FXO PASSWORD HEREport=5062host=dynamicdtmfmode=rfc2833HT503 setup:Unconditional call forward to VOIP: userid = ring groupserver = your server's IPport =5060FXO setup:DTMF audio=noDTMF via RFC2833=yesEnable Call Features=noDial plan={ [x*]+ }Number of rings=2PSTN ring-thru=NoStage method=1Hope this helps.JR
Grandstream Support .... if you do read these posts, please fix this problem. I never saw this with FX100p that I was told to replace by experts with HT503.
JRoque,Thank you for sharing the solution on the problem. Your's is the only post that I found on the net with a solution. I just bought ht-503 and installed it in Freepbx/asterisk/Elastix environment. Today I had my first lockup. Fortunately the lockup was total and the phone connected to FXS also stopped working, otherwise all the incoming calls would have gone unanswered. And worst of all, although failover has been set to auto, FXO port did not fail-over. I do have old answering machine connected to the same line in parallel that picked up the messages but it is sad for asterisk installment. :-(BTW I do not understand how 1 second annoucement in the ring group completes the call. I was under impression that once the call gets in to asterisk ring group, it has been picked up from FXO stand-point. The rining tone heard by caller is generated by Asterisk. How does adding 1 second silence change it?
