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| 08:45.01 | wyoung | team team team |
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| 14:11.59 | lovetruth | hello :) |
| 14:12.16 | lovetruth | I am so new to asterisk... (and voip in general) |
| 14:12.32 | lovetruth | although I am an IT guy (System Engineer)... :) |
| 14:13.15 | Samot | OK. |
| 14:13.16 | lovetruth | so... small question: how can I get asterisk a.s.a.p. to a conference (~20 people) setup?... |
| 14:13.42 | Samot | Do you have it installed now? |
| 14:13.53 | lovetruth | I mean, sip.conf, all .conf files... eventually which modules... conference bridge module?... |
| 14:14.04 | lovetruth | yes, on a small linux server (arm device) |
| 14:14.08 | lovetruth | asterisk 15 |
| 14:14.17 | Samot | That's EOL. |
| 14:14.21 | *** join/#asterisk samwierema (~samwierem@195.240.143.134) |
| 14:14.25 | Samot | So first step is to be on a version that is supported. |
| 14:14.32 | lovetruth | like which version?... |
| 14:14.45 | Samot | LTS: 13.25.0 (2019/2/15) 16.2.1 (2019/2/28) |
| 14:14.52 | Samot | Those are the two LTS versions. |
| 14:14.58 | lovetruth | so 13 would be ok?... |
| 14:14.58 | Samot | I'd go with 16. |
| 14:15.53 | lovetruth | it's actually an openwrt device (a wifi router, actually). it has 13 in it's repository |
| 14:15.59 | lovetruth | for 16 I might have to compile... |
| 14:16.13 | Samot | OK. Stop. |
| 14:16.21 | Samot | Put it on a real system. |
| 14:17.24 | lovetruth | Samot: the thing is that my wife has to go to some foreign country and they need some software to help them get the translator's guide into their own language... :) |
| 14:17.41 | lovetruth | because their hardware broke (which they used to communicate)... |
| 14:17.54 | lovetruth | So I thought a conference call on some voip server might save the day... :) |
| 14:18.12 | lovetruth | and I thought that asterisk might be able to do it |
| 14:18.31 | Samot | Yes, Asterisk can do it. |
| 14:18.42 | Samot | But putting 20 actives calls on your OpenWRT router is a bad idea. |
| 14:18.46 | Samot | Just completely bad. |
| 14:19.23 | lovetruth | but they can't get some laptop or some PC with them all the time... they need something really portable. |
| 14:19.50 | lovetruth | the router supports 20 people connecting to it, but actually a part of them know the language. So it would be less than 15 probably who would connect |
| 14:20.09 | Samot | And how much CPU does this thing have? |
| 14:20.15 | Samot | RAM? |
| 14:20.19 | lovetruth | dual core 600 Mhz |
| 14:20.51 | Samot | That is really meant for like some home/hobby stuff for one or two calls. |
| 14:21.08 | Samot | You can try it but 15 calls on the router plus everything else the router is doing...could cause issues. |
| 14:22.19 | lovetruth | I would love to try it... but my conf (on asterisk 15, gonna switch to 13, if you would recommend it?...) not sure if it will work well |
| 14:23.15 | Samot | You're going to need to setup the confbridge module and other modules that the OpenWRT docs don't cover. |
| 14:23.32 | Samot | Because again, that covers having a system that just does calls. Nothing special. |
| 14:23.38 | lovetruth | I had in my mind to learn asterisk for some time now, already... but sadly I didn't make enough time for it... :) Definitely gonna do it, tough :) |
| 14:24.12 | lovetruth | I thought of this configuration: https://fucking-it.com/articles/asterisk/19-asterisk-conference-bridge |
| 14:24.56 | Samot | OK, did you install ALL the modules when you installed Asterisk? |
| 14:25.18 | lovetruth | yes, the full asterisk 15 and the conf bridge |
| 14:25.49 | Samot | OK so what is the issue? |
| 14:27.01 | lovetruth | well, for some reason asterisk (15) didn't open the port when I started the service... :) - and so mizudroid (or is something else you'd thing would be better) wasn't able to connect to it... |
| 14:27.19 | Samot | What port? |
| 14:27.38 | Samot | 5060? |
| 14:28.20 | lovetruth | If my understanding is correct, 5060?... |
| 14:28.21 | lovetruth | yes |
| 14:28.32 | Samot | Chan_SIP? Chan_PJSIP? |
| 14:31.40 | lovetruth | is it a problem if I used PJSIP?... |
| 14:33.56 | Samot | No. |
| 14:34.09 | Samot | But since there are two drivers, it is helpful to know which one you're using. |
| 14:34.17 | Samot | So the proper help can be given. |
| 14:35.20 | Samot | Now, show your transport section of the pjsip.conf that you set the address and port PJSIP is bound to. |
| 14:36.42 | lovetruth | you have here some preferred pastebin site?... :) |
| 14:38.20 | Samot | Just pastebin.com |
| 14:38.41 | lovetruth | and also - should I reinstall asterisk -> meaning, to install 13 (after uninstalling the 15)?... or we'l test this one?... |
| 14:38.56 | Samot | Not right now. |
| 14:39.01 | Samot | Misconfigured is misconfigured |
| 14:39.07 | lovetruth | thanks |
| 14:49.39 | Samot | Well? |
| 14:54.57 | lovetruth | I got I had a misconfiguration... but I was reading up how&what&why... :) -> here is the file (although I think it is definitely misconfigured, as you suggested me... :) ) ->https://pastebin.com/SNYdYJCk |
| 14:55.19 | lovetruth | I got I had a misconfiguration... but I was reading up how&what&why... :) -> here is the file (although I think it is definitely misconfigured, as you suggested me... :) ) ->https://pastebin.com/SNYdYJCk |
| 14:55.46 | lovetruth | I'd use UDP |
| 14:56.21 | Samot | Uhm, these are Chan_SIP peers. |
| 14:56.37 | Samot | Not even close to the transport section of the pjsip.conf |
| 14:57.22 | Samot | lovetruth: You just can't copy and paste things from two different how-to blogs and them not be related to this. |
| 14:57.40 | *** join/#asterisk samwierema (~samwierem@195.240.143.134) |
| 14:58.06 | Samot | You setup using PJSIP per the OpenWRT does but then blindly followed a confbridge doc that wasn't created for anything higher than Asterisk 11 which is Chan_SIP only. |
| 14:59.35 | lovetruth | Samot exactly... :) that is so right... :) I'm still reading up these... but I'm a little nervous that my wife won't have something and I keep trying these... (tomorrow she leaves country...) |
| 15:01.14 | Samot | How is she going to use this anyways? |
| 15:03.32 | Samot | Why doesn't she need a portable PBX? |
| 15:08.08 | *** join/#asterisk defsdoor_ (~Andrew@cpc120600-sutt6-2-0-cust232.19-1.cable.virginm.net) |
| 15:09.26 | *** join/#asterisk lovetruth (050cf6a2@gateway/web/freenode/ip.5.12.246.162) |
| 15:09.43 | Samot | lovetruth: ??? |
| 15:10.02 | Samot | 11:01:15 AM <Samot> How is she going to use this anyways? |
| 15:10.02 | Samot | 11:03:34 AM <Samot> Why does she need a portable PBX? |
| 15:10.14 | lovetruth | sorry, I connected to the router and changed IP -> it resulted in ping timeout... :) |
| 15:11.08 | lovetruth | She'll keep this in the bag, with some 11000mah battery (can give 10 hours of functioning...) |
| 15:11.28 | Samot | For? |
| 15:11.36 | Samot | Again, what is the use case for it? |
| 15:11.44 | Samot | Why does she need a portable PBX? |
| 15:12.23 | Samot | How will she or the other users call/connect to the PBX to use the conference bridge? |
| 15:12.43 | lovetruth | there is a translator, and the translator tells the people the translation in a conference (2 hours) |
| 15:13.09 | lovetruth | I saw this configuration, people have mizudroid or some SIP client, configured for some PBX server |
| 15:13.27 | Samot | So everyone is going to have to install a softphone app to use the PBX? |
| 15:13.27 | lovetruth | then they call 800 (or some specified number) and they are in the conference call |
| 15:13.51 | lovetruth | they have it already |
| 15:14.00 | lovetruth | I mean, part of the group have it already |
| 15:14.02 | Samot | And who is going to support these people? |
| 15:14.14 | Samot | Why does the PBX need to be portable? |
| 15:14.19 | Samot | If they all have softphones.. |
| 15:14.23 | lovetruth | you mean technical support? or the translation?... |
| 15:14.33 | Samot | Why does the PBX need to be portable and move from place to place? |
| 15:14.33 | lovetruth | you mean technical support? or the translation?... |
| 15:14.40 | Samot | The PBX support.. |
| 15:14.43 | lovetruth | because they have no internet |
| 15:14.43 | Samot | Come on. |
| 15:14.51 | lovetruth | it's in Africa... :) |
| 15:15.15 | Samot | So the PBX is also the AP for everyone to connect to? |
| 15:15.29 | lovetruth | well... the thing is that they broke their system and tomorrow they are leaving (they have the plane tickets already bought...) |
| 15:15.31 | lovetruth | Samot: yes |
| 15:15.50 | lovetruth | Samot: yes |
| 15:16.05 | Samot | So now you need to setup 15 accounts on a PBX plus a conference bridge and hope that it all works right? |
| 15:16.10 | lovetruth | they had some hard wired headsets (wireless)... but they don't work any more. |
| 15:16.16 | Samot | Have you even tested basic extension to extension calling? |
| 15:16.24 | Samot | WAit.. |
| 15:16.33 | lovetruth | So I thought of a portable PBX :)) -> I want to test it tonight |
| 15:16.35 | Samot | Where are the callers going to be located? |
| 15:16.43 | lovetruth | So I thought of a portable PBX :)) -> I want to test it tonight |
| 15:16.53 | Samot | If they don't have Internet and they are using the AP/PBX why do they need a conference bridge? |
| 15:16.56 | lovetruth | if the test passes, then that would be great... if not... dunno |
| 15:17.05 | lovetruth | well... around up to 15 meters from the router |
| 15:17.07 | lovetruth | if the test passes, then that would be great... if not... dunno |
| 15:17.14 | lovetruth | they'll stick together, as a group |
| 15:17.36 | lovetruth | the router has 2 x 5dbi antennas |
| 15:17.42 | Samot | You want a conference bridge for people that are going to be in a 50 Foot area? |
| 15:18.20 | Samot | Get a persona PA. |
| 15:18.25 | Samot | Get a personal PA. |
| 15:18.44 | Samot | You want people in a 50 foot area to connect to an AP, load a softphone... |
| 15:19.15 | Samot | To get into a conference bridge to have translation.. |
| 15:19.23 | Samot | A small PA system and two microphones. |
| 15:19.26 | Samot | Done. |
| 15:19.29 | Samot | That's your answer. |
| 15:19.49 | lovetruth | kinda something like that... if it works :) ... |
| 15:20.06 | Samot | No, the PBX for this is not the right solution. |
| 15:20.17 | Samot | This is just not the right option. |
| 15:20.24 | lovetruth | had this router lying around, I was thinking to test it... |
| 15:20.35 | lovetruth | ok... but how much can the pbx support?... |
| 15:20.49 | Samot | No.. |
| 15:20.51 | lovetruth | or... I can add a raspi or something in the setup, if I should?... |
| 15:20.59 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 15:21.01 | Samot | I'm saying using Asterisk for this as the solution is the wrong way. |
| 15:21.02 | lovetruth | dunno if I have time to get a system tomorrow before they leave country |
| 15:21.04 | Samot | No. |
| 15:21.21 | lovetruth | here is Sunday, so I can't easily find the system |
| 15:21.23 | Samot | You have people in an area 15 METER area. |
| 15:21.32 | Samot | They can literally all talk to each other IN PERSON |
| 15:21.52 | lovetruth | yes, but it's a conference |
| 15:21.56 | Samot | OK\ |
| 15:22.00 | lovetruth | they would be quite loud :) |
| 15:22.07 | Samot | 11:19:25 AM <Samot> A small PA system and two microphones. |
| 15:22.18 | *** join/#asterisk c0mrade (5d7ec5cf@gateway/web/freenode/ip.93.126.197.207) |
| 15:22.20 | Samot | Yes, a PERSON will be speaking for others to listen. |
| 15:22.21 | c0mrade | I've just created a Facebook Messenger Bot and it got approved by facebook so anyone should be able to see and use it :). I named it "Bot Inc." or @BotInc, when you try to message it you'll see a bunch of commands. I hope you guys can try it :D |
| 15:22.23 | Samot | With a translator. |
| 15:22.44 | Samot | c0mrade: What does this have to do with Asterisk? |
| 15:23.20 | c0mrade | Samot: You'd be able to control an Asterisk server through a simple Facebook Messenger |
| 15:23.29 | Samot | lovetruth: Using Asterisk to create a confbridge for people in the same room to call into on their softphones to listen to the speak and their translator is just the wrong option. |
| 15:23.47 | Samot | c0mrade: And how would one do that? Does you bot have commands for that? |
| 15:24.08 | Samot | c0mrade: How's your whole Mikrotik API/ Telnet thing going? |
| 15:24.32 | c0mrade | Samot: What? How did you know? |
| 15:24.41 | c0mrade | Samot: Yes it does of course. |
| 15:24.50 | Samot | How did I know what? |
| 15:24.56 | lovetruth | ah... the phones would pickup the sound, you mean?... Hm... |
| 15:25.14 | Samot | lovetruth: No. You're using a PBX when a PA system can be used. |
| 15:25.17 | lovetruth | ah... the phones would pickup the sound, you mean?... Hm... |
| 15:25.18 | lovetruth | We could mute the mics... :) |
| 15:25.28 | Samot | You just need to AMPLIFY the speaker and translator. |
| 15:25.31 | c0mrade | Samot: That am working with MikroTik API. |
| 15:25.51 | Samot | c0mrade: Because I was one of two people that said to use it. |
| 15:26.50 | c0mrade | But I was at #MikroTik not #asterisk |
| 15:26.58 | Samot | Yes. |
| 15:27.18 | Samot | I am in both channels. At once. |
| 15:27.23 | Samot | Crazy how IRC works. |
| 15:31.56 | *** join/#asterisk lovetruth (050cf6a2@gateway/web/freenode/ip.5.12.246.162) |
| 15:32.17 | lovetruth | anyway, Samot - I wanted to thank you for pointing me in the right direction(s)... :) |
| 15:32.37 | lovetruth | I think I'll test asterisk anyway tonight (as I wanted to get into asterisk anyway... :) ) |
| 15:33.23 | lovetruth | and might try to get some system (not an amplifier - we can't make that noise there) to get the translator to every member of the group who needs it |
| 15:34.00 | lovetruth | pjsip - there was the problem indeed :) |
| 15:34.55 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 15:54.32 | Samot | c0mrade: So no status on that yet? |
| 15:57.14 | c0mrade | Samot: Well there's a status, they don't need it because they have bought a fully automated system connected to cloud, it has analytics, dashboard with tons of features, and clients connect through their social network accounts |
| 15:57.36 | Samot | heh |
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| 16:12.40 | *** join/#asterisk samwierema (~samwierem@195.240.143.134) |
| 16:13.15 | *** join/#asterisk samwierema (~samwierem@195.240.143.134) |
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| 16:15.20 | *** join/#asterisk devdvd (17f248c1@gateway/web/freenode/ip.23.242.72.193) |
| 16:15.50 | devdvd | Morning all. With the directory application, is there any way to make it read the full name instead of letter by letter? |
| 16:16.05 | devdvd | and if not, can someone suggest another application that can? |
| 16:17.52 | file | Asterisk doesn't have a database of sounds of all the names in existence, so you'd need a text to speech system, and there's none built in for that or in the directory app |
| 16:17.59 | file | if the name has been recorded it will use that |
| 16:33.20 | *** join/#asterisk sysgrammer (~sysgramme@d50-117-140-107.yt.northwestel.net) |
| 16:33.56 | *** join/#asterisk sysgrammer_moe (~sysgramme@d50-117-158-201.yt.northwestel.net) |
| 16:35.32 | devdvd | @file There's a small number of names so that wouldn't be a problem. How do I tell the directory to use the sound file for that name? |
| 16:36.49 | file | it uses the one recorded in Voicemail if I recall |
| 16:38.24 | devdvd | ah, cool ill check that out, thanks! |
| 16:47.51 | *** join/#asterisk devdvd (17f248c1@gateway/web/freenode/ip.23.242.72.193) |
| 16:48.14 | devdvd | file, you were correct (not sure if you saw that a moment ago, i wasn't seeing anything appear in my chat) |
| 16:49.00 | Samot | Directory() works with voicemail |
| 16:49.29 | Samot | You can set hidefromdir as an option for a voicemail user and it will not populate them when Directory is called. |
| 16:49.41 | devdvd | oh, good to know thanks |
| 16:50.39 | Samot | It's why the first option to set is the voicemail context. |
| 16:50.44 | Samot | Because that's where it pulls the users from. |
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| 18:08.27 | *** join/#asterisk Dovid (~dovid@ool-45738ae3.dyn.optonline.net) |
| 18:08.36 | Dovid | Does Asterisk have something like str_pad in php? |
| 18:08.49 | *** join/#asterisk life_of_e (~life_of_e@108-95-189-245.lightspeed.irvnca.sbcglobal.net) |
| 18:09.04 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 18:11.51 | Samot | Dovid: Not that I am aware of. |
| 18:11.58 | Dovid | Samot: Thanks |
| 18:12.06 | Samot | What are you looking to pad? |
| 18:13.11 | sibiria | one way is to call printf via System |
| 18:13.17 | sibiria | it's a bit "inelegant" but it gets the job done |
| 18:20.30 | *** join/#asterisk life_of_e (~life_of_e@108-95-189-245.lightspeed.irvnca.sbcglobal.net) |
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| 20:05.32 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:49.53 | life_of_e | how does one get the Read() application to say the numbers? When I use it I hear "Press" but no numbers |
| 20:51.22 | life_of_e | Nevermind, it was part of a recording |
| 21:12.10 | wyoung | yup |
| 21:24.23 | *** join/#asterisk Downlots (~Downlots@2a02:85f:1f01:b400:a5a2:cdd1:62a9:afe2) |
| 21:32.57 | life_of_e | I wish Read() could playback digits the way SayDigits can by passing in digits instead of having to build up a list of file names |
| 21:39.06 | life_of_e | I still can't find an example of using SIPSendCustomInfo() |
| 21:42.03 | file | it's a test application |
| 21:42.55 | life_of_e | Hmm, ok, I was hoping to use it to get around Asterisk's inability to send a flash to a SIP device |
| 21:44.10 | Samot | What are you trying to do exactly? |
| 21:44.26 | life_of_e | Flash my ATA adapter's FXO line |
| 21:44.49 | life_of_e | So the literal use of the flash event |
| 21:45.25 | Samot | OK, so you have an FXO ATA that is connected to Asterisk via SIP? |
| 21:45.29 | life_of_e | Yes |
| 21:45.39 | Samot | Not going to work the way you think anyways. |
| 21:45.45 | life_of_e | Why not? |
| 21:45.47 | Samot | The ATA is going to convert SIP to Anagalo. |
| 21:45.50 | Samot | The ATA is going to convert SIP to Analog. |
| 21:45.54 | life_of_e | I know |
| 21:45.56 | Samot | So it can go out a PSTN line. |
| 21:46.00 | Samot | OK so can your ATA do that? |
| 21:46.04 | life_of_e | I want to flash the FXO when a call waiting indicator comes in |
| 21:46.15 | life_of_e | Yes, it supports flashing the FXO on receipt of an INFO event 16 |
| 21:46.30 | Samot | What is the FXO connected to? |
| 21:46.33 | Samot | A POTS line? |
| 21:46.35 | life_of_e | Yep |
| 21:46.43 | Samot | So you want to flash your POTS line? |
| 21:46.47 | life_of_e | Right |
| 21:47.01 | life_of_e | When the POTS line beeps in with a call waiting tone |
| 21:47.20 | life_of_e | Manually of course, just using a feature code to send the flash |
| 21:48.17 | Samot | So an incoming call comes in over the POTS line... |
| 21:48.29 | life_of_e | An incoming call comes in *while I'm already on a call* |
| 21:48.32 | Samot | Then it hits the ATA and goes to the PBX... |
| 21:48.46 | Samot | And what happens? |
| 21:48.49 | Samot | Does it reach the ATA? |
| 21:48.52 | Samot | Does it reach the PBX? |
| 21:48.59 | life_of_e | Sure, analog incoming calls work fine |
| 21:49.08 | Samot | So the call waiting call hits the PBX? |
| 21:49.47 | life_of_e | The in-band tone for call waiting is heard in the background |
| 21:50.00 | Samot | In the background of what? |
| 21:50.03 | Samot | Your phone that you are on? |
| 21:50.10 | life_of_e | In the background of an existing phone call, yes the phone I'm on |
| 21:50.23 | Samot | Which is connected to the PBX via SIP" |
| 21:50.28 | life_of_e | It's not generated by Asterisk or the ATA, it's a tone from the central office |
| 21:50.39 | Samot | I get that. |
| 21:50.46 | *** join/#asterisk sa02irc (~sa02irc@155-079-043-212.ip-addr.inexio.net) |
| 21:50.58 | Samot | So it's PSTN -> FXO -> Asterisk -> SIP Phone? |
| 21:51.05 | life_of_e | Yes |
| 21:51.24 | Samot | OK you want to pick up the second call from your SIP Phone? |
| 21:51.28 | life_of_e | Right |
| 21:51.32 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 21:51.34 | Samot | That has nothing to do with your FXO |
| 21:51.40 | Samot | The call is already IN Asterisk. |
| 21:51.44 | life_of_e | It does because there's only one FXO port |
| 21:51.54 | Samot | You're not understanding.. |
| 21:51.59 | Samot | The call as already hit Asterisk.. |
| 21:52.04 | life_of_e | Right, I understand that |
| 21:52.12 | Samot | You are deciding that the call should Dial() your phone.. |
| 21:52.18 | life_of_e | No |
| 21:52.19 | Samot | That Channel has nothing to do with the FXO channel. |
| 21:52.28 | life_of_e | Let's start over |
| 21:52.56 | life_of_e | A call has come in over the PSTN, has been connected to the SIP phone and is ongoing |
| 21:53.02 | Samot | Right. |
| 21:53.17 | life_of_e | During that call the CO sends a tone saying another call is coming in over the PSTN (call waiting) |
| 21:53.19 | Samot | Via Asterisk. |
| 21:53.21 | Samot | I know. |
| 21:53.24 | Samot | I get what you are saying. |
| 21:53.31 | Samot | You're not listening or understanding me. |
| 21:53.39 | Samot | So let me start over. |
| 21:53.41 | life_of_e | I want to tell the FXO port on the ATA to quickly hook flash so I can listen to the second PSTN call |
| 21:53.52 | Samot | Call A comes in over the PSTN line, it hits the PBX.. |
| 21:54.02 | Samot | At that point, how the PBX is configured tells that call what to do. |
| 21:54.05 | Samot | It could go to a Queue |
| 21:54.10 | Samot | Or Vociemail... |
| 21:54.16 | life_of_e | It's already dialed the phone |
| 21:54.17 | Samot | Never leave the PBX. |
| 21:54.21 | Samot | Please wait. |
| 21:54.22 | life_of_e | I know that |
| 21:54.24 | Samot | OK |
| 21:54.25 | life_of_e | Ok |
| 21:54.33 | Samot | So when Call B comes in, same thing right? |
| 21:54.47 | Samot | YOU by programming the PBX have opted to DIAL those calls to your SIP device. |
| 21:54.50 | life_of_e | Not quite, the channel is already in use |
| 21:54.58 | Samot | There would be a NEW channel. |
| 21:54.59 | life_of_e | Call B comes in on the same wire |
| 21:55.03 | Samot | No it doesn't. |
| 21:55.08 | [TK]D-Fender | <life_of_e> I want to tell the FXO port on the ATA to quickly hook flash so I can listen to the second PSTN call <- ive never seen a SIP gateway device that offers this |
| 21:55.08 | Samot | Not to YOUR SIP PHONE |
| 21:55.10 | life_of_e | Yes it does, it's call waiting on PSTN |
| 21:55.14 | Samot | GD. |
| 21:55.18 | Samot | I know exactly what it is.\ |
| 21:55.20 | Samot | You have a PBX |
| 21:55.25 | Samot | That is a BACK TO BACK USER AGENT |
| 21:55.28 | life_of_e | No, I don't want to drop the channel |
| 21:55.36 | Samot | The PSTN to PBX side of the calls are not related to your SIP deivce. |
| 21:55.40 | Samot | JFC. |
| 21:55.57 | Samot | The call to your SIP phone exists between Asterisk and your SIP phone. |
| 21:55.58 | Samot | That's it. |
| 21:56.01 | life_of_e | I'm not talking about the SIP phone other than to say I want to tell Asterisk "Please send a hook flash to the ATA" |
| 21:56.17 | Samot | The PBX already HAS the call. |
| 21:56.22 | life_of_e | Yes |
| 21:56.33 | Samot | So you don't need to send anything to the teleco |
| 21:56.39 | life_of_e | No |
| 21:56.41 | Samot | You want to press a button on your SIP and answer the call... |
| 21:56.44 | [TK]D-Fender | life_of_e, go read your devices manual to see if it offers a way to signal it. Also don't bet that such a thing is offered at all |
| 21:56.45 | life_of_e | No |
| 21:56.58 | Samot | Then I have no clue what you want. |
| 21:57.04 | life_of_e | The manual says SIP INFO event 16 will send a hook flash to the FXO port |
| 21:57.09 | Samot | OK |
| 21:57.10 | life_of_e | It supports this |
| 21:57.14 | Samot | But does the call hit the PBX? |
| 21:57.21 | Samot | Do you see the call waiting call enter the PBX? |
| 21:57.25 | life_of_e | No |
| 21:57.32 | life_of_e | Because there's only one analog line |
| 21:57.36 | life_of_e | I hear the tone inband |
| 21:57.47 | *** join/#asterisk cryptic (~cryptic@142.196.139.17) |
| 21:57.52 | Samot | What FXO device are you using? |
| 21:57.58 | life_of_e | Grandstream HT813 |
| 21:58.29 | life_of_e | The PSTN is telling me it has another call wanting to come in. I want to signal the PSTN to change over to the new call |
| 21:58.52 | life_of_e | I do not want to drop the ongoing channel between the ATA and the SIP phone, I just want the PSTN to flip calls |
| 21:59.07 | [TK]D-Fender | I'm not aware of an option in * that will send out that kind of packet |
| 21:59.24 | life_of_e | It was sort of there as DTMF(f) but it doesn't do anything now |
| 21:59.37 | life_of_e | DTFM (f) |
| 21:59.47 | [TK]D-Fender | life_of_e, maybe you can use some other stack to send it. Then it'sa question of if the device will accept that comm from a different app... |
| 21:59.47 | life_of_e | argh, DTFM f |
| 22:01.13 | life_of_e | Perhaps, can I generally send a SIP INFO to a device from a program/device different than the one holding the channel? |
| 22:01.38 | Samot | What FXO device are you using? |
| 22:01.40 | [TK]D-Fender | life_of_e, if you're using dtmfmode=info then you could probably use a features.conf feature that will trigger a SendDTMF against that channel |
| 22:02.08 | [TK]D-Fender | life_of_e, if it's exp[ected in the same format as regular SIP INFO DTMF |
| 22:02.11 | life_of_e | TK: I tried that, to send the F code which used to be documented as flash but it's deadended |
| 22:02.27 | life_of_e | Samot: it's a Grandstream HT813 |
| 22:03.06 | life_of_e | TK: I dug through some of the SIP code and sending code F just gets ignored within the DTMF application |
| 22:03.40 | life_of_e | Hence the thought I could use the SIPSendCustomINFO application to get around that by building up my own |
| 22:04.16 | [TK]D-Fender | Ah, I don't recall there being an "F". Am a little tired. I do recall A-D being standard... not sure what "F" would even be |
| 22:04.59 | life_of_e | Yeah, F was to send event code 16 according to some very old documentation |
| 22:05.31 | life_of_e | Some others had changed it, for example I found that FreePBX had altered it to require an 'R' instead of an 'F' |
| 22:06.00 | life_of_e | Either way Asterisk doesn't bother sending anything now except 0-9A-D |
| 22:08.44 | [TK]D-Fender | Actually, I don't recall anything beyond that ever |
| 22:08.58 | [TK]D-Fender | SO not just "now" |
| 22:09.21 | [TK]D-Fender | <life_of_e> Some others had changed it, for example I found that FreePBX had altered it to require an 'R' instead of an 'F' <- How did FreePBX alter this? |
| 22:09.32 | [TK]D-Fender | No sure I understand what you're implying...\ |
| 22:09.36 | [TK]D-Fender | not* |
| 22:09.50 | life_of_e | Changing the source code to look for 'R' instead of 'F' |
| 22:10.38 | [TK]D-Fender | News to me... |
| 22:10.44 | life_of_e | I'm trying to find it again |
| 22:11.40 | life_of_e | Here was one thread |
| 22:11.40 | life_of_e | http://lists.digium.com/pipermail/asterisk-bugs/2012-September/106327.html |
| 22:14.23 | [TK]D-Fender | It refers to a patch... was it never merged? |
| 22:14.52 | [TK]D-Fender | Have you found the code and attempted to adapt it to your version? |
| 22:15.01 | life_of_e | I haven't found the code yet |
| 22:19.07 | Samot | Have you tried any of this with PJSIP? |
| 22:19.27 | life_of_e | I haven't finished moving over to PJSIP yet |
| 22:19.34 | life_of_e | So not yet |
| 22:19.43 | Samot | Well you should test it. |
| 22:19.56 | file | flash isn't something anyone has touched in PJSIP, so I doubt it would change things |
| 22:21.20 | Samot | I was more referring to the SendDTMF |
| 22:22.07 | *** join/#asterisk gugaua (~gugaua@unaffiliated/gugaua) |
| 22:24.12 | life_of_e | In the app_senddtmf.c it says "and f or F for a flash-hook if the channel supports flash-hook" |
| 22:24.18 | life_of_e | So that's where I got the 'F' from |
| 22:25.22 | life_of_e | But chan_pjsip.c doesn't show anything about supporting it at all while chan_sip.c looks like it considered it (ot advertises that it accepts it) |
| 22:27.36 | Samot | Is the Grandstream setup to just Forward PSTN to VoIP? |
| 22:28.55 | life_of_e | Yes |
| 22:36.10 | life_of_e | Followed the source code around. An ast_indicate is requested when an 'f' or 'F' is supplied in the SendDTMF and ast_indicate returns 0 for AST_CONTROL_FLASH |
| 22:39.41 | Samot | Flash is not really a SIP thing. |
| 22:40.31 | life_of_e | I know it's not generally, a SIP phone certainly doesn't need to make use of flash to answer other incoming calls. It just happens to be useful for SIP ATAs. |
| 22:41.06 | Samot | Doing FXO to SIP is not something that generally has CW as a feature. |
| 22:41.12 | Samot | As each FXO line is a single channel. |
| 22:41.26 | Samot | And CW might not even be enabled on the line to begin with. |
| 22:42.05 | life_of_e | Yeah, I get it, most gateways would be 4, 8, 16 FXO ports attached to all the PSTN lines where incoming calls roll over to the next free wire pair. |
| 22:42.16 | *** join/#asterisk gugaua (~gugaua@unaffiliated/gugaua) |
| 22:42.30 | life_of_e | But in the case of a residential or small business with a single line, CW is likely going to be present |
| 22:42.49 | Samot | I understand that but the market for that has shrunk. |
| 22:43.09 | life_of_e | if they keep the PSTN line. Obviously if they get a VoIP provider then it doesn't matter, you get an arbitrary number of "lines" |
| 22:44.01 | life_of_e | It would have been nice to use the built-in flash feature of the ATA in my case. There would be a lot of code to fix to get that working. |
| 22:44.15 | Samot | You're working off the assumption that in a home they will have a PBX |
| 22:45.22 | life_of_e | Yeah, but I did know two people years ago that had a PBX in their home. ISDN line coming in and I think the system was Nortel |
| 22:45.33 | life_of_e | So it's not a never :) |
| 22:46.07 | life_of_e | Oh, nope, just had to do an image search. They had an AT&T Merlin system |
| 22:46.42 | Samot | I'm not saying it doesn't happen but it isn't enough to base a market on it. |
| 22:48.00 | life_of_e | No, I get it, I just don't have the resources to fix the code either because I'd first have to dig in and understand what the existing code is doing to know exactly where to add the additional support |
| 22:48.58 | life_of_e | I can sort of get around the problem with some external hardware, I just tried to find soft solutions first (again if SIPSendCustomINFO could have helped for example) |
| 22:50.46 | life_of_e | I can hang a device inline on the PSTN line that listens for DTMF A-D and flashes or I could put GPIO hardware in the server to flash the line. It was just cleaner if the ATA did it itself. |
| 22:51.24 | life_of_e | I'll have to experiment with TK's suggestion of possibly sending SIP INFO in parallel to Asterisk, just not sure the ATA will accept that while it's already communicating with Asterisk. |
| 22:53.24 | life_of_e | I still lose out in the configuration anyway because CW CID isn't captured |
| 22:53.38 | life_of_e | So I will know a call is coming in but I won't know from whom |
| 22:55.00 | life_of_e | The workaround I've seen for that is to get a cheap VoIP service and set up the PSTN to call forward when busy to that number. |
| 22:57.57 | life_of_e | My ISP is not reliable enough to just move my PSTN service over to VoIP. |
| 22:59.33 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 23:05.37 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |