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| 01:40.27 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.1 (2018/11/14), Security Only: 15.6.2 (2018/11/14); 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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| 19:09.35 | Chotaire | Hi there, I seem to have a simple question today. Let's say I initiate an attended transfer with *2 and I dial the wrong number, how do I return to the caller? pressing *2 again would transfer the call. |
| 19:11.23 | Samot | Well it's an attended transfer. |
| 19:11.27 | Samot | That caller is on hold. |
| 19:12.19 | Chotaire | Return to caller on hold because I dialed the wrong number. |
| 19:12.24 | Chotaire | anyway, that's ** |
| 19:12.30 | Chotaire | just found out. have a good night! |
| 19:12.51 | Samot | Well that hangs up the current channel |
| 19:12.54 | Samot | https://wiki.asterisk.org/wiki/display/AST/Feature+Code+Call+Transfers |
| 19:13.00 | Chotaire | yes because it is the wrong number |
| 19:13.05 | Samot | Right |
| 19:13.12 | Samot | But ** isn't an attended transfer thing |
| 19:13.12 | Chotaire | why would i not want to hang up the wrong number? |
| 19:13.19 | Samot | It's a "hang up the call" thing |
| 19:13.26 | Chotaire | so you recommend i just disconnect both callers? |
| 19:13.31 | Samot | No. |
| 19:13.33 | Samot | Sigh |
| 19:13.36 | Samot | I just posted a link. |
| 19:13.47 | Samot | It has all the In-Call transfer settings you can use |
| 19:13.52 | Samot | Aborting, etc. |
| 19:14.15 | Samot | I'm just saying that ** isn't "transfer" specific feature. |
| 19:14.24 | Samot | It's the "Disconnect Active Call" feature. |
| 19:18.07 | Chotaire | ok then. I should have mentioned this is a freepbx installation. feature codes like *3 do not exist. anyway, on freepbx ** would disconnect the active call and return to the caller on hold. on asterisk, not sure if *3 is configured by default. |
| 19:18.32 | Chotaire | but on stock asterisk it is *3 (atxferabort). thanks. |
| 19:20.32 | Chotaire | funny, on the latest sangoma freepbx *3 would make that a three-way call, but it's not even listed under feature code admin. |
| 19:23.27 | Samot | All of this can be set in the _custom config files in FreePBX to expand this. |
| 19:24.06 | Samot | Literally the [general] section in the features config file calls on the _custom file so you can add more feature maps like this |
| 19:24.14 | Samot | I've added them to systems. Works fine. |
| 19:27.19 | Chotaire | ok i just experimented with the phone... when doing an attended transfer, *1 is hangup and return to caller, *2 is transfer call and *3 is bridge as three-way, *4 is swap calls. that is different to the asterisk standards. |
| 19:27.52 | Samot | You can make the whatever you want |
| 19:28.21 | Chotaire | just weird they do not show up in feature code admin. if that was the case I hadn't even asked. |
| 19:28.40 | Chotaire | *2 is the only transfer related option that is included |
| 19:30.09 | Samot | Because the majority of the users have IP phones/softphones that have actual "Transfer" buttons and functionality in them. |
| 19:30.39 | Samot | Thus removing the need for feature codes like that. |
| 19:31.37 | Samot | The only time I've used the transfer feature codes was the to test them. |
| 19:31.53 | Samot | Outside of that, my phone has all that handled. |
| 19:32.15 | Chotaire | well. transfer codes are working, but they are neither configured in /etc/asterisk/features* nor do they show up in feature admin. that is the issue i am talking about. |
| 19:32.22 | Chotaire | different story if they wouldn't be there at all |
| 19:32.36 | Samot | OK. |
| 19:32.57 | Samot | I'm trying to explain this to you. In world of FreePBX those feature codes aren't really needed. |
| 19:32.58 | Chotaire | plus, they are different from the asterisk defaults in the docs you just mentioned... sometimes freepbx is a little weird. |
| 19:33.10 | Samot | Again, _you can make them what you want_ |
| 19:33.28 | Samot | What is in the Asterisk wiki is what _Asterisk defaults them to in configs_ |
| 19:33.47 | Chotaire | I'm trying to explain this to you. There are hardcoded feature codes that are different from the asterisk defaults, they are not included in any templates, they are not included in feature admin, if I hadn't experiemted I wouldn't even know they exist. |
| 19:33.47 | Samot | FreePBX just controls Asterisk. |
| 19:34.01 | Samot | I know what you are saying |
| 19:34.19 | Samot | I'm saying they don't exists because they're are demands for them in the world of FreePBX users. |
| 19:34.31 | Samot | 2:30:08 PM <Samot> Because the majority of the users have IP phones/softphones that have actual "Transfer" buttons and functionality in them. |
| 19:35.48 | Chotaire | and yes, I need feature codes because I use a goddamn sophos xg firewall behind a goddamn branded fritzbox of which the combination of both completely destroys proper sip connectivity unless I am using a VPN (which obviously, my soft phone doesn't do). my hardware phone will do compoletely weird stuff when I am trying to use the built-in transfer features (and that has only just started a few weeks ago, with config untouched). so for now I am |
| 19:35.48 | Chotaire | left with feature codes :) |
| 19:36.12 | Chotaire | not "soft phone", "hardware phone" of course. |
| 19:36.18 | Samot | OK, well the beauty of FreePBX....customizable. |
| 19:36.36 | Samot | So if there are things you want Asterisk to do that FreePBX didn't setup. You can do that |
| 19:36.53 | Chotaire | feature codes save my ass until I replace the sophos xg with a different firewall that does not have completely wrecked SIP ALG (known issue) |
| 19:37.07 | Samot | This is why FreePBX has so many _custom.conf files that are just there and blank. |
| 19:37.25 | Samot | And why the dialplan calls on -custom variations of contexts. |
| 19:39.46 | Chotaire | freepbx should not have hardcoded transfer feature codes and then not document them. that's a bad thing to do, especially if they are different from the defaults. I just grepped all of /etc/asterisk and it's nowhere. are there any other secret feature codes I should be aware of that are not listed in feature admin? |
| 19:41.30 | Samot | Nothing is "hard coded" |
| 19:42.14 | Samot | You are free to modify an of the Feature Codes in Feature Code Admin that are listed. |
| 19:42.58 | Chotaire | they are NOT listed. they are NOT in /etc/asterisk, they are NOT in advanced config. so how do I modify or even find something that is not listed in either of 3 places? |
| 19:43.15 | Samot | I don't know what you are talking about at this point. |
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| 19:43.33 | Samot | The additional xfer features are not including in FreePBX because of the lack of need for them. |
| 19:43.42 | Samot | included* |
| 19:44.07 | Chotaire | <PROTECTED> |
| 19:44.21 | Chotaire | <PROTECTED> |
| 19:44.27 | Samot | There are no "Asterisk Standards" |
| 19:44.43 | Chotaire | any idea where to find these non-standards? |
| 19:44.51 | Samot | 2:27:52 PM <Samot> You can make them whatever you want |
| 19:45.01 | Chotaire | i would like to know what they are and where they are. |
| 19:45.04 | Samot | What are you talking about? |
| 19:45.10 | Samot | What standards? |
| 19:45.15 | Chotaire | <PROTECTED> |
| 19:45.23 | Samot | I gave you a link. |
| 19:45.25 | Chotaire | <PROTECTED> |
| 19:45.25 | Chotaire | <PROTECTED> |
| 19:45.30 | Samot | It showed examples. |
| 19:45.35 | Chotaire | ok thanks. I give up :) |
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| 19:45.37 | Samot | For _Asterisk_ |
| 19:45.40 | Samot | Not for FreePBX |
| 19:45.48 | Chotaire | I'll grep the entire system and databases then |
| 19:45.51 | Samot | You're making a deal because a GUI doesn't have something. |
| 19:45.59 | Samot | There is no GUI in Asterisk. |
| 19:46.05 | Samot | Go to #freepbx and complain there. |
| 19:46.09 | Chotaire | there is also no gui in /etc/asterisk |
| 19:46.28 | Samot | GUI = Graphical User Interface. |
| 19:46.35 | Samot | IE. your web browser. |
| 19:46.46 | Chotaire | thanks. |
| 19:47.02 | Samot | Asterisk is configured and setup by config files you edit by hand. |
| 19:47.08 | Chotaire | and where are these config files? |
| 19:47.16 | Samot | The ones in /etc/asterisk |
| 19:47.20 | Samot | Those are ASTERISK CONFIGS |
| 19:47.21 | Chotaire | awesome. |
| 19:47.31 | Chotaire | now freepbx is modifying these. |
| 19:47.36 | Samot | Yes. |
| 19:47.51 | Chotaire | do you think there are any other additional feature codes located outside of /etc/asterisk? |
| 19:48.24 | trmg | makes a batch of pocorn |
| 19:49.02 | Chotaire | because if not, then it was jesus holy freaking christ who personally came down to the planet surface and injected feature codes into my asterisk. |
| 19:49.02 | trmg | FWIW that's how FreePBX works. It re-writes the Asterisk configs and structures them based on how you configure things via their web UI. |
| 19:49.22 | Chotaire | nn |
| 19:51.44 | Samot | There is nothing out of /etc/asterisk for Asterisk configs. |
| 19:51.47 | Samot | All the files are there. |
| 19:52.14 | Samot | 2:37:06 PM <Samot> This is why FreePBX has so many _custom.conf files that are just there and blank. |
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| 20:40.19 | phrearch | hello |
| 20:41.17 | phrearch | i'm trying to setup presence subscription with asterisk and sipjs, but somehow i don't get beyond a 481 response (SIP/2.0 481 Subscription does not exist) |
| 20:44.14 | [TK]D-Fender | Show us |
| 20:44.37 | phrearch | https://paste.ubuntu.com/p/xz9MnJzT7k/ |
| 20:45.16 | phrearch | i probably did something wrong with the hint and the extension dial definition? |
| 20:47.36 | phrearch | siptrace: https://paste.ubuntu.com/p/8MGG5NcDnn/ |
| 20:58.22 | phrearch | ah, i think i got it |
| 20:58.56 | phrearch | code i used before was subscribing on dialog, while i needed to subscribe to presence |
| 20:59.24 | phrearch | one step at a time :-) |
| 20:59.55 | trmg | I need to learn about the whole subscription thing so I can get BLF line appearances working. |
| 21:00.12 | phrearch | same here :) |
| 21:00.29 | trmg | Is that what you are doing? |
| 21:00.32 | trmg | lol |
| 21:01.13 | phrearch | i now got it working for asterisk 13 and chan_sip. need to figure it out for pjsip as well |
| 21:02.08 | trmg | Ahh |
| 21:02.49 | phrearch | i'll document the steps on the way. working on an opensource webrtc sip softphone for asterisk. new to asterisk, but did some work on webrtc conferencing tooling before. |
| 21:03.13 | phrearch | in case anyone is interested: https://github.com/garage11/ca11 (very early work) |
| 21:03.45 | trmg | Nice |
| 21:04.14 | phrearch | i tried the asterisk sfu demo two weeks before, and was really impressed by that. it's on top of the feature list |
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