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| 00:18.28 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 02:36.31 | tehgooch | Using Asterisk 13.9.1 - is it possible to ring multiple phones, but return busy when any one of them is busy/in use? Did some googling, but the only things I find are people saying that queues are the solution, but they can't be bothered to describe how. AFAIK queues don't work like that. |
| 02:37.14 | tehgooch | Using PJSIP as well. |
| 02:39.30 | [TK]D-Fender | Nothing normally works like that |
| 02:39.55 | [TK]D-Fender | I can imagine one way to do it but it's a serious pain in the ass |
| 02:42.13 | [TK]D-Fender | dial multiple local channels. In each local channel do the outbound dial. Follwing the dial the first one to abort will hit the next line where you'll verify the cause code. Then you'll execute an external script that will poll for all of the other channels associated with the one that started it and subsequently kill them off one by one via AMI or CLI hangup command |
| 02:42.38 | [TK]D-Fender | there is nothing remotely normal as an alternative withing the bounds of * |
| 02:42.46 | [TK]D-Fender | So if you REALLY want this ... it' |
| 02:42.53 | [TK]D-Fender | s going to take some work |
| 02:43.01 | [TK]D-Fender | heads out for the evening |
| 02:43.24 | tehgooch | Well, thanks. I'll keep at it. |
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| 13:24.00 | skrusty | anyone know of an issue with PJSIP_HEADER? Doesn't seem to set the header on the leg b invite |
| 13:24.23 | skrusty | can see the dialplan function being called, no errors, but nothing in the invite |
| 13:24.25 | file | you have to execute it on that channel |
| 13:24.30 | file | using a pre-dial handler |
| 13:24.51 | skrusty | i've executed it on leg a, i assume... how do you execute it on leg b? |
| 13:24.56 | skrusty | sorry |
| 13:25.20 | skrusty | looks up pre-dial handlers :) |
| 13:25.52 | skrusty | reads wiki page :) |
| 13:26.48 | skrusty | ah. |
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| 13:28.32 | file | go go leveraging recentish stuff |
| 13:30.10 | skrusty | ;) |
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| 13:32.29 | skrusty | hmmmm :/ |
| 13:33.52 | skrusty | trying to set the header to "somevar" that's defined in the first channel |
| 13:34.29 | skrusty | tried using global (__) var (is that still right?) but that didn't work |
| 13:34.46 | file | I don't know what you mean... |
| 13:34.50 | file | PJSIP_HEADER is a dialplan function |
| 13:35.17 | file | https://wiki.asterisk.org/wiki/display/DIGIUM/Ringing+Types+and+Intercom+2.0 that's a small example of usage |
| 13:35.44 | skrusty | so in the predial handler, i set a sip header |
| 13:36.02 | skrusty | which is now there.. but the value of the header is a channel varaible from the endpoint that made the call |
| 13:36.13 | skrusty | my predial handler is of type 'b' (callee) |
| 13:36.26 | skrusty | my variable isn't in the predial handler... |
| 13:36.32 | skrusty | does that make sense? |
| 13:36.51 | file | pass it in as an argument to the pre-dial handler? |
| 13:37.02 | skrusty | oh, didn't see that in the spec, sorry... |
| 13:37.07 | skrusty | goes away again |
| 13:37.21 | skrusty | oh yeah, the big args bit... |
| 13:37.24 | skrusty | hangs head in shame |
| 13:37.32 | file | :D |
| 13:42.38 | skrusty | are the arguments accessed like those in a macro? ${ARG1} |
| 13:42.54 | file | if I recall correctly, yes |
| 13:44.39 | skrusty | hmm :/ |
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| 13:45.27 | skrusty | not sure it does... |
| 13:45.58 | skrusty | i can see arg passed to handler, but variable is empty. will look : code |
| 13:50.52 | skrusty | figure it out |
| 13:51.01 | skrusty | args need to be between () |
| 13:51.33 | skrusty | didn't spot them when looking at the syntax |
| 13:51.49 | skrusty | maybe i could add an example to the wiki including arguments, for clarity? |
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| 14:05.26 | file | Sure |
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| 14:30.04 | skrusty | need to think of some nice example mind you :) |
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| 18:13.32 | salviadud | Right now I'm attaching voicemails with ssmtp and I'm wondering if its valid to use a global variable such as ${CALLERID(NUM) inside of voicemail.conf |
| 18:27.00 | salviadud | Please excuse my stupidity, I just found out about ${VM_CIDNUM |
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