first of all your info is so limited that almost noboby could help you...compatible obviously to a noob but resolving your question asterisk does not have default hold time...seems a timeout in your phones or lines tried to replicate but having no problems with hold times in my pbx´s
I think I may know what you are requesting as I have the same prob and just been living with it.
First we need to know, are you having this prob on "softphone" on one or more computers?,
or is this occurring on all phones (voip, hardline, etc...)? And, are you experiencing the same prob that I am that when you place a received call on hold (with your softphone) after a time of say 2 min to five mins it automatically hangs up the call or extension?
There is no setting in the Elastix or FreePBX for setting the hold time per extension that I am aware of. All the research I have done leads to beleive that it is a problem with the softphone software or phones being used.
As dicko suggested, it may be a problem within your router, I am more inclined to think it is a "keep-a-live" issue that my softphone is having, but like dicko suggested I will seek further into my router.
I can appreciate your frustration, as they say, been there - done that, I too have been placed in the position of having to seek info on what the boss can't find and he does that so you look bad and not him, so don't get too frustrated with him or us. Everyone on here is volunteer and some of us don't always related to the problems at hand and need a little more understanding of your problem.
So if you would just let us know what type of phones your using,
if the call that hangs up is one extension calling another, or a incoming call to an extension, or what. Kind of give us a better idea of what is occurring beside just a time limit on hold times.
Hope this helps, let us know if we can help any further and, Welcome to Elastix.
"Thanks for YOUR response Amphibian, you're a nice person.",
thanks for the complement, but before you go to far, I am probably one of the biggest A**holes known to mankind, I'm really not a nice person and really don't even like myself, and yes, I travel with the "D-H's" too. The difference is, I just put it in a little different perspective since I felt like you were experiencing the same prob I have.
But all this bickering aside, DH or not, the main question is, did what you post last solve your problem? If so then I will try it to. If not, then lets try to get it resolved if at all possible.
Hang in there, life gets better as you get older, after all we are living proof of that.
sorry willy....you dont even have a clue again
first of all
•rtpholdtimeout = Number : Max number of seconds of inactivity of RTP Stream before terminating a call on hold. Default 0 (no limit). (Since v1.2.x).
so inactivity...define inactivity...well inactivity is no rtp flow...so it only affects a call if no data is being moved from the server to the called as this menu from freepbx said also...
Terminate call if rtpholdtimeout seconds of no RTP or RTCP activity on the audio channel when we're on hold (must be > rtptimeout).
So as dicko said it is probably a network issue of keep alives affecting only some systems probably behind nat and thats why that timeout does not affect our systems because we do receive our rtp streams correctly
Hi my good friend lee your version was the old one from voip info wiki and is a little outdated (2 words jeje) we need to update that version to keed up to date the one from sip.conf 1.8 documentation !!!
Most if not all hard-phones send rtp keep-alive packets every few seconds (less than 30), many cheaper soft-phones and ATA's don't, these rtp settings are work-arounds for badly written software/hardware. Many routers that don't understand VOIP similarly arbitrarily drop udp connections after a while for no reason. A common misconfiguration is to allow "comfort-noise" on your end-points, this is a mistake in asterisk deployments and will fool it. These settings have no effect on IAX2 or DAHDI connections.