Thank you - that makes sense. I've seen something about swapping and optimizing channels on the console, but I didn't realise "optimize" meant "not do what you wanted".
OK, so here's why I'm dialling anything at all: The first dial is because I MUST limit the incoming call to less than 60 minutes. The second dial, which carries the gH option, is because I want someone to be able to listen to a radio stream >From previous discussion here, it seems the only way to do that is the gH workaround above. If I'm not missing a trick here and there's no better way to do those to things, is there any way to force Asterisk to NOT "optimize" those channels? On 9 November 2016 at 00:09, Richard Mudgett <rmudg...@digium.com> wrote: > > > On Tue, Nov 8, 2016 at 5:19 PM, Jonathan H <lardconce...@gmail.com> wrote: >> >> Asterisk 14.1 >> >> Here's a bit of test dialplan, which works as expected and simulates >> exactly what I'm doing at the top of my large dialplan... >> >> [dial-pre-test] >> exten => s,1,NoOp() >> same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) >> same => n,Set(LIMIT_WARNING_FILE=time_limit_reached) >> same => n,Dial(Local/s@dial-test,3,L(3540000:60000)) >> same => n,Hangup() >> >> [dial-test] >> exten => s,1,NoOp() >> same => n,Dial(Local/s@dial-dest,,gH) >> same => n,Playback(goodbye) >> same => n,Hangup() >> >> [dial-dest] >> exten => s,1,Answer() >> same => n,MusicOnHold() >> same => n,Hangup() >> >> See what I'm doing here? I'm using a little fiddle to allow the caller >> to stop listening to music on hold. And it works..... the gH means >> that the caller can hang up the remote end. Great! >> >> BUT.... I have a large dialplan, and something, somehow, somewhere, is >> messing with "Disconnect Call". >> >> Because once through, nothing, not even star, does anything. It's like >> the receiving end (dial-dest in the example above) has become deaf! >> >> I've turned on debug and verbose to level 9, and there's nothing. It >> connects, starts music on hold, and then just ignores everything. >> >> Anything else I can add to the dialplan to see what might be causing >> this? (I've also tried dumpchan, too). >> >> It USED to work, and some point in the last week, it stopped working. >> (But the test dialplan above works). Mind boggled! >> >> Just to double check, yes, it's all set OK >> >> features show >> Builtin Feature Default Current >> --------------- ------- ------- >> Pickup *8 *8 >> Blind Transfer # # >> Attended Transfer >> One Touch Monitor >> Disconnect Call * * >> > > Beware of local channel optimization. You are putting state on local > channels > that can optimize out. When the local channels optimize out they take the > state with them. > > In the dialplan above you are creating the channel chain below. > > PJSIP/caller --> Local/s@dial-test;1 -- Local/s@dial-test;2 --> > Local/s@dial-dest;1 -- Local/s@dial-dest;2 > > PJSIP/caller gets the L() duration and sounds put on it. > The Local/s@dial-test;1 gets the L() duration put on it. > The Local/s@dial-test;2 gets the H dial option put on it. > > There is a bridge connecting PJSIP/caller and Local/s@dial-test;1 > There is a bridge connecting Local/s@dial-test;2 and Local/s@dial-dest;1 > > When Local/s@dial-dest;2 executes Answer it will allow Local/s@dial-test;1 > and ;2 to > optimize out because both ends are in a bridge. Thus the H dial option will > disappear from > the channel chain. > > Richard > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users