Re: [asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)
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 wrote: > > > On Tue, Nov 8, 2016 at 5:19 PM, Jonathan H 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(354:6)) >> 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
Re: [asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)
On Tue, Nov 8, 2016 at 5:19 PM, Jonathan H 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(354:6)) > 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
[asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)
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(354:6)) 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 * * -- _ -- 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
Re: [asterisk-users] Problem "re-parking" calls
> All; > I have a problem with regards to “re-parking” calls and I was > hoping someone could shed some light on the topic. Consider this scenario: > > (1) An inbound call comes in and the attendant answers it > (2) The attendant places the call on hold and the caller is sent to > extension 701 > (3) Blah, blah, blah. The attendant does something and tells John > Doe to pick up the call on extension 701 > (4) The attendant then picks up the call on 701 and tells the person > that John Doe will be right there to help them > (5) The attendant then re-parks the call but now the caller is sent to 702 > (6) John Doe can't find the call anymore > > > Is there something obvious that I am missing? Has anyone else found > this to be a problem? Any insight at all would be greatly appreciated. > Regards; > John V. Your problem occurs in step 4 & 5. I don't believe that you can pick up the call and then ever be guaranteed to get the same parking position when you put it back in park. What would happen if someone else parked a call in between steps 4 and 5 and they got 701 because it was free. Once parked, the call should remain so until it is picked up or times out back to the attendant. __ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com __ -- _ -- 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
[asterisk-users] Problem "re-parking" calls
All; I have a problem with regards to "re-parking" calls and I was hoping someone could shed some light on the topic. Consider this scenario: (1) An inbound call comes in and the attendant answers it (2) The attendant places the call on hold and the caller is sent to extension 701 (3) Blah, blah, blah. The attendant does something and tells John Doe to pick up the call on extension 701 (4) The attendant then picks up the call on 701 and tells the person that John Doe will be right there to help them (5) The attendant then re-parks the call but now the caller is sent to 702 (6) John Doe can't find the call anymore Is there something obvious that I am missing? Has anyone else found this to be a problem? Any insight at all would be greatly appreciated. Regards; John V. -- _ -- 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
[asterisk-users] Problem "re-parking" calls
All; I have a problem with regards to "re-parking" calls and I was hoping someone could shed some light on the topic. Consider this scenario: (1) An inbound call comes in and the attendant answers it (2) The attendant places the call on hold and the caller is sent to extension 701 (3) Blah, blah, blah. The attendant does something and tells John Doe to pick up the call on extension 701 (4) The attendant then picks up the call on 701 and tells the person that John Doe will be right there to help them (5) The attendant then re-parks the call but now the caller is sent to 702 (6) John Doe can't find the call anymore Is there something obvious that I am missing? Has anyone else found this to be a problem? Any insight at all would be greatly appreciated. Regards; John V. -- _ -- 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