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| 00:16.08 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 04:06.58 | ni291187 | hey all. if youâre using an ATA adapter, can you trigger functions/actions within asterisk by pushing touch tones on your analog phone DURING a phone call? |
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| 08:33.52 | legid | Hi everyone. I'm wondering if someone has an idea about the following: I have an application that uses AMI to originate a call that plays an audio file. What I'm trying to do now is to detect if the whole audio was played or if the remote party hung up before. Any ideas on how to do this? |
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| 08:41.33 | wraythe | use the h extension? |
| 08:48.46 | legid | I already know the h extension; it gets called as soon as I hang up. But how can I check if Playback() finished before the h extension gets called? |
| 08:56.34 | wraythe | not sure if the dialplan continues past the Playback, or if it jumps directly to h, |
| 08:56.53 | wraythe | if it jumps directly to h, you can set a variable after the playback to mark the playback is complete |
| 08:57.18 | wraythe | if by the time you reach h, and the variable is unset -> you know the playback did not complete |
| 08:57.30 | wraythe | not 100% if this will work, it would need to be tested |
| 09:03.00 | legid | that sounds good - I will give it a try. Do you know if I have to prefix the variable with a "_" to make it work? |
| 09:05.00 | wraythe | no you just use h |
| 09:05.08 | wraythe | ah wait sorry |
| 09:05.10 | wraythe | misread you |
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| 09:06.34 | wraythe | same = n, Set(CALLCOMPLETE=1) |
| 09:06.38 | wraythe | same => n, Set(CALLCOMPLETE=1) |
| 09:08.39 | legid | all right. I will try it later that day. Thank you so much! |
| 09:13.21 | wraythe | Hope it works :-) |
| 09:13.23 | wraythe | good luck! |
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| 11:41.08 | stan | could anyone recommend a source of example dialplans to follow current best practice? |
| 11:42.04 | wraythe | What are you wanting to do @stan? |
| 11:42.09 | wraythe | in your dialplans i mean |
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| 11:42.52 | wraythe | https://github.com/asterisk/asterisk/blob/master/README-SERIOUSLY.bestpractices.md |
| 11:44.00 | ni291187 | morning. I lost my connection last night when I asked this, apologies for asking again: if I have an asterisk server and analog phones connected via an ata adapter, is there a way to trigger functions/actions inside asterisk by pushing touch tones on the analog phone DURING a call? |
| 11:44.03 | stan | just simple small office setup, one ring group, few ddis |
| 11:45.25 | Samot | stan: There are numerous examples on the Internet. Dialplan is going to really be based on your needs. |
| 11:45.37 | Samot | stan: You might just want to look at FreePBX. |
| 11:45.54 | Samot | ni291187: Yes, ATA support incall feature codes. Just like a regular phone line. |
| 11:46.23 | ni291187 | awesome, thank you! |
| 11:47.07 | stan | ideally something like a git repo that's maintained with common macros |
| 11:48.26 | Samot | Stan: Then use FreePBX |
| 11:51.58 | stan | trying to migrate from freepbx to something more direct to the config files |
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| 11:54.50 | Samot | Uhm. Why? |
| 11:55.04 | Samot | You just said you wanted something maintained... |
| 11:56.31 | wraythe | performance or simplicity? |
| 11:57.16 | wraythe | i can see some benefit of using config-based, esp with GIT.. But eish, business continuity. If you have an issue, is there someone else who is skilled enough to cope? |
| 11:58.21 | Samot | First, there are no "common macros" |
| 11:58.32 | Samot | Second, Macro() is deprecrated. |
| 11:58.59 | Samot | Oddly enough, FreePBX has a collection of "common macros" that they use for their dialplan. |
| 12:00.16 | Samot | wraythe: Understand, it's all config based. FreePBX is just a GUI vs you editing it in w text editor. |
| 12:00.29 | Samot | It still generates the same dialplan commands as anyone else would use. |
| 12:02.31 | Samot | The one thing asked for was a Ring Group. That is not a function or an application or anything that exists in Asterisk pre-configured. It's a method. |
| 12:02.49 | Samot | How that method is a achieved is up to the person writing the ring group logic. |
| 12:12.54 | wraythe | yeah |
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| 12:20.17 | DannyA | is this doable with asterisk: let's say i have a small call center, and while one of my operators is on the line, i want them to be able to click a button on their computer, and a command (api?) is sent to asterisk which conferences in another caller onto the line (so now there are 3 parties on the line), and part 2 of the question: is there a way my operator can hang up, and parties 2 & 3 are still talking to each other? |
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| 13:35.18 | stan | freepbx seems complex and brittle on upgrade, would be more comfortable with a straight config files |
| 13:35.50 | stan | ideally without having to do it from scratch |
| 13:36.38 | Samot | Uhm. OK. |
| 13:36.50 | Samot | So if you install straight up Asterisk, the onus is on you to do everything. |
| 13:37.05 | Samot | All the stuff you didn't have to do in FreePBX, you have to do yourself. |
| 13:37.37 | Samot | Setting up how the modules load, which ones should/shouldn't. Getting all the other .conf files in place to make sure they are there and have what they need in them. |
| 13:38.19 | stan | the dialplan generated for the separate DDIs by freepbx looks very duplicated |
| 13:38.33 | Samot | Because each one can have it's own settings. |
| 13:38.44 | Samot | And by that logic, so does everything else. |
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| 13:39.01 | Samot | All the extensions, followme, ring groups, IVRs, etc. etc. etc. they all follow the same logic. |
| 13:39.23 | Samot | However, based on the settings in the GUI certain things will or will not be written out to the dialplan. |
| 13:39.31 | Samot | Or AstDB settings added, etc. |
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| 13:58.27 | [TK]D-Fender | stan, It is. Then again the GUI does all the work for you. |
| 14:09.11 | Samot | stan: Bottom line is this. If you understand and know Asterisk pretty well then what you want to do is fine. If you are dependent of pre-generated code/dialplan that you're just going to cut and paste into conf files, stick with FreePBX. |
| 14:10.24 | wraythe | Very well said Samot :-) |
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| 19:23.32 | DannyA | hey all. asked this question already this morning but didn't see any response so thought i'd try one more time: is this possible with asterisk: if i have an operator in a call center, and they're on the phone with a caller, can they push a button on their computer which will send an API command to asterisk to conference in another caller (so now it's a 3 way call), and then, next question, when they're ready, and they're not needed on the line |
| 19:23.32 | DannyA | <PROTECTED> |
| 19:25.19 | igcewieling | Yes, that is possible. I'm not aware of anything pre-built which is exactly like that, but things like iSymphony and Flash Operator Panel might be useful for that. |
| 19:27.25 | DannyA | @igcewieling doesn't even need to be something pre-built, im fine wiring things up myself, just wondering if the capabilities are there |
| 19:27.59 | DannyA | what im guessing is, you would hit asterisk and say "on this line, transfer party A and party B into a new conference line", and then make an outgoing call to Party 3, and when they're connected, put them into the same conf room? |
| 19:28.20 | DannyA | then if the operator (Party A) hangs up, it's fine, because Party B and Party C are in a conf room together... |
| 19:30.06 | Samot | No, you can use AMI for this. |
| 19:31.08 | Samot | You can send AMI action that will call Party C and redirect Party A and B into a confbridge. Then when Party C answers they can be added. |
| 19:35.55 | DannyA | @Samot as always, thank you!! :-) |
| 19:36.50 | DannyA | on a separate note, does anyone have a recommendation for an Asterisk implementation vendor in NYC? |
| 19:37.08 | DannyA | implementation + 24/7 critical support |
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