i have a freepbx server running FreePBX 188.8.131.52. when sip isnt used and dahdi is used instead, the call stays connected untill i run from the cli “dahdi restart”. there is alot of articles/forums only on “disconnect supervision” but i have yet to see a simple solution for the bunch of people who are experiencing this issue, especially thoese using freepbx. could someone from the freepbx community give us a simple solution to resolve this problem within the freepbx gui or do i have to modify asterisk config files?
Unplug and replug the phone line quickly (~500ms) if the call drops call your telco. If it doesn’t drop then you need to do more troubleshooting on your side. That is the simple quick separation between your problem/not your problem
thanks for your response. so i made a test call from the dahdi line to my cell phone and went to my pbx server to disconnect it for half a second. the call did indeed disconnect so now i need to call my telco. since its a pots line, i call the provider of my pots line? also, what do i tell them or ask them?
Tell them there is no aparent line supervision and you are not seeing a cpc disconnect
i spoke with my telo and told them what you said and they have no idea what im talking about. is there anything i can do on my side?
You need to escalate. Maybe 2 or 3 times. You can fake supervision but it can end up dropping calls.
Sometimes when your provider doesnt know what they are doing it is best to move to another provider.
after doing research it is known that my tisp provider (bezeq) does not provide line supervision. my provider is the most reliable provider in my country and they are the only telco provider for POTS lines here in my country. im not sure how i can fake supervision. today the same thing happened to a sip line and after doing sip reload the line still didnt connect. its been connected for 70 hours. should i open a new ticket for this issue or is it posible that both issues are related.
you fake supervision by adding the following to your chan_dahdi config.
please ignore where i previously mentioned that this issue happened with sip. it may have been because of an update and i don’t have enough information to prove anything. lets stick with dahdi here. you provided me with a method on how i can fake supervision using asterisk config files but each freepbx save will rewrite these settings. i need to know how i can apply these settings to freepbx and not directly to the asterisk core files.
your answers are greatly appreciated. ~bump