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| 03:24.24 | *** topic/#asterisk is AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.2 (2019/11/21) 16.6.2 (2019/11/21) Standard: 17.0.1 (2019/11/21); 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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| 05:14.11 | *** topic/#asterisk is AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.2 (2019/11/21) 16.6.2 (2019/11/21) Standard: 17.0.1 (2019/11/21); 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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| 12:56.19 | DonAlex | Heya peeps. Can someone help me debug WTF is going on here? I am perplexed by the error that is not showing with verbose 9 on but prevents me from dialling out. |
| 12:56.33 | DonAlex | https://pastebin.com/0ez2Yexy |
| 12:57.19 | DonAlex | It says syntax error: syntax error, unexpected '=', expecting $end; Input: but for the life of me I cannot see where in the stream it is seeing that ? |
| 12:57.36 | DonAlex | Am I blind ? |
| 12:57.42 | sibiria | there are several variables that you declare but without passing a value |
| 12:57.59 | sibiria | CDR(accountcode)=, CALLERPRES(*)= etc. |
| 12:58.13 | sibiria | at least set them to an empty string |
| 12:58.48 | sibiria | EMERGENCYCID= |
| 12:59.02 | sibiria | comparisons and declarations, i mean |
| 12:59.49 | Samot | It's a FreePBX problem. |
| 12:59.55 | Samot | I think someone opened a bug on it already. |
| 12:59.57 | DonAlex | O |
| 13:00.22 | DonAlex | is FALSE an empty string ? |
| 13:01.21 | DonAlex | I mean can I set EMERGENCYCID to FALSE if it is not used? |
| 13:06.56 | DonAlex | *sighs* So I have to wait for FReePBX to fix it then ? Bugger. . |
| 13:08.02 | Samot | It might already be fixed. Have you gotten the latest updates? |
| 13:09.56 | DonAlex | I am on FreePBX 14.0.13.12 |
| 13:10.24 | DonAlex | So yes I could upgrade to 15 I spose? |
| 13:11.06 | Samot | No. |
| 13:11.15 | DonAlex | Only that has just been released..so.. |
| 13:11.16 | Samot | I asked if you had the latest updates... |
| 13:11.23 | Samot | Not if you've upgraded. |
| 13:11.26 | DonAlex | To this.. yes.. |
| 13:11.33 | Samot | Are you sure? |
| 13:12.23 | DonAlex | fwconsole ma showupgrades |
| 13:12.23 | DonAlex | No repos specified, using: [standard,extended,unsupported,commercial] from last GUI settings |
| 13:12.23 | DonAlex | Up to date. |
| 13:14.03 | DonAlex | Unless there is something else I am missing ? |
| 13:15.25 | Samot | You could enable the Edge repo and get updates from there. There are a lot of updates in the edge repo. |
| 13:17.46 | Samot | Now whether or not that fixes the issue. I'm not sure. |
| 13:18.51 | DonAlex | I am looking through the extension profiles and filling any ones that seem to want to not be blank with a not null field. Maybe that is the simplest solution so far.. Assuming that all the variable declarations are defined in the GUI. |
| 13:20.27 | DonAlex | haha.. well I shifted the error.. Now I am getting all circuits are busy now ;) |
| 13:23.00 | DonAlex | Oh what a rabbit hole.. *sighs* |
| 13:24.06 | Samot | OK well now show the issue after the change |
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| 13:32.37 | DonAlex | Trying to debug now.. Still seeing some Set"($variable=)" |
| 13:40.55 | DonAlex | The problem is tying the variable to the gui parameter not always obvious |
| 13:45.04 | DonAlex | I mean I cannnot see anywhere in the extension set up where Set(CALLERPRES(name-pres)=)" might be ? |
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| 13:52.34 | DonAlex | Ahhh I thnk I found the bug. It shows the correct string but the values are blank in the GUI so you think it is already set.. but you need to set them in the GUI as well. |
| 13:53.49 | Samot | No, you shouldn't. |
| 13:54.06 | Samot | The outbound callerid and emegerency callerid fields are not required fields. |
| 13:54.12 | Samot | They can be blank. |
| 13:54.14 | Samot | It's a bug. |
| 13:54.24 | DonAlex | hmmm |
| 13:54.50 | DonAlex | Oh well.. Time for lunch I will look at it after perhaps.. |
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| 15:46.27 | martinvw | Is it possible to use HASH() to create global hashes? And if so, can I define global hashes in [globals] in my extensions.conf? |
| 15:48.30 | martinvw | "HASH(foo,bar)=bar" in [globals] leads to "Setting global variable '~HASH~foo~bar~' to 'bar'", while "TEST(bar,foo)=foo" leads to "ERROR[28242]: pbx_functions.c:699 ast_func_write: Function TEST not registered", so it does look like Asterisk interprets HASH in [globals] |
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| 15:52.31 | martinvw | I'm struggling to read from this hash though: neither ${HASHKEYS(foo)} nor ${HASHKEYS(GLOBAL(foo))} seems to work. |
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| 16:32.18 | igcewieling | When I'm using hash nothing technical works for me either. 8-| |
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| 16:54.22 | martinvw | seems to be a known bug that HASHKEYS won't work on global hashes :/ https://issues.asterisk.org/jira/browse/ASTERISK-24004 |
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| 17:31.54 | sysgrammer | Hello, I am looking for someone to hire to help me with analog rhino DAHDI card... Please let me know... |
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| 19:50.56 | sysgrammer | any DAHDI experts in the house??? |
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| 20:13.10 | nullr0ute | can you post your question? maybe someone can help |
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| 20:37.35 | brian1001 | Hello everyone, i'm looking for nice open source hotel billing managment solution for asterisk , does someone here has experience with it maybe? |
| 20:42.44 | Samot | I'm not really sure there is one. |
| 20:42.53 | Samot | What features does it need to have? |
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| 20:45.41 | nullr0ute | If there isn't one, then I see an opportunity for a developer |
| 20:45.52 | brian1001 | just a 'check in' , check'out out feature |
| 20:46.41 | Samot | That's the only feature? |
| 20:46.53 | Samot | How does that manage billing? |
| 20:48.30 | brian1001 | yes and indeed a billing feature based on how long someone called to a certain destination |
| 20:49.08 | Samot | So it doesn't need to tie into an existing PMS application? |
| 20:49.25 | brian1001 | no |
| 20:49.52 | Samot | How many rooms? |
| 20:50.01 | brian1001 | 15 |
| 20:51.50 | Samot | I'm not really sure there is something that would be current. |
| 20:52.08 | brian1001 | it amazed me but its indeed hard to find |
| 20:53.47 | Samot | The fact there is a hotel charging for calls still amazes me. |
| 20:55.36 | brian1001 | yes wish there was a good voip provider here with unlimited free calls |
| 20:55.49 | Samot | Where is here? |
| 20:57.12 | brian1001 | the netherlands |
| 20:57.38 | Samot | Well do they have a lot of usage already? |
| 20:58.03 | Samot | I really can't speak for that area of the world but with the amount of mobile phones in play... |
| 20:58.29 | Samot | The need for call from your hotel room is really not a thing anymore. |
| 20:58.31 | brian1001 | i think tehy just need 2 lines |
| 20:58.44 | brian1001 | yes exactly Samot |
| 21:02.21 | Samot | I think you're going to have to prepare them for the fact this is going to cost money. |
| 21:07.28 | electronic_eel | I'm from Germany and here it depends very much on the kind of number you are calling how much it is gonna cost you. If you call a regular landline, it is nearly for free and lot's of providers offer flatrates. If you are calling a cellphone, it is a bit more expensive and usually you won't be able to get a true flatrate for it. |
| 21:07.58 | electronic_eel | I think it is very similar in the Netherlands |
| 21:08.55 | electronic_eel | So if you wanted to offer free calls for your hotel guests, you'd have to block cellphone numbers, which wouldn't be a good service for your guests. |
| 21:09.25 | Samot | But that's my point. |
| 21:09.39 | Samot | This isn't 15 years ago. Pretty much everyone has a mobile phone. |
| 21:10.29 | electronic_eel | yes, and nobody sane is going to use a hotel phone for outgoing calls because they often charge ridiculous rates |
| 21:10.57 | electronic_eel | but you need a phone with outbound capability to get/keep your stars as a hotel |
| 21:12.24 | Samot | But at the end of the day if your month phone bill isn't crazy or even that high because guests aren't making calls... |
| 21:13.04 | Samot | Do you need a PMS just to manage rooms/checkin/checkout or do you need one that does that but can integrate to your PBX for billing of calls. |
| 21:13.46 | electronic_eel | you could maybe offer free calls, but implement some kind of limit, like 30 minutes per day and room or something. |
| 21:13.49 | Samot | If having all 15 rooms full for a weekend or whatever means you have to spend $50 in calls.... |
| 21:13.53 | Samot | It's worth it. |
| 21:14.05 | Samot | Or you do what they do here in the US |
| 21:14.12 | Samot | Limit the calling areas. |
| 21:14.29 | Samot | Some of my hotels don't let long distance calls happens. |
| 21:14.36 | Samot | Just local, toll free and 911 |
| 21:14.53 | Samot | And they don't even charge for calls. |
| 21:15.03 | Samot | And they have unlimited calling. |
| 21:16.46 | electronic_eel | yeah, but sometimes it is just that the owner of the hotel is thinking conventional and it has always been this way... |
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| 21:43.12 | brian1001 | ty both i indeed explain them this |
| 21:43.35 | brian1001 | just a limit on the outgoing calls will probably do the trick and no long distance calls |
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| 21:50.49 | macaruchi | Hi! |
| 21:51.02 | macaruchi | I need to use Postgres for using with asterisk CDR |
| 21:51.29 | macaruchi | My question is i need to install ODBC for this or I can use directly libpq |
| 21:51.54 | macaruchi | When I use make menuselect I cant select CDR-PGSQL I get XXX |
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| 22:05.24 | macaruchi | ? |
| 22:09.01 | electronic_eel | usually all the database accesses go through odbc |
| 22:10.00 | electronic_eel | do you really want to use CDR? I prefer CEL |
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| 22:24.16 | dabukalam | OK. So I got hold of a Grandstream GXW 4108. And I have Asterisk on a raspberry pi. But I have to say the Grandstream web interface is... wow |
| 22:25.02 | dabukalam | (c) 2005 so it's 15 years old... I guess it makes sense |
| 22:30.26 | electronic_eel | you are lucky that it works with a current browser. I had one old ups that required an IE 4 or something like that to work, no other browser worked with their borked javascript or whatever |
| 22:32.40 | electronic_eel | was happy when it broke some years ago, good reason to throw it out |
| 22:33.21 | electronic_eel | needed a win xp in a vm, just so I could access it |
| 22:40.27 | dabukalam | hah, I was worried I would have to do the same with this to get onto it |
| 22:40.36 | dabukalam | they have some weird IP discovery script |
| 22:41.04 | dabukalam | but luckily it had DHCP enabled out of the box, so picked up an IP from my LAN |
| 22:42.43 | dabukalam | http://www.grandstream.com/sites/default/files/Faq/gxw410x_interop_asterisk.pdf |
| 22:42.48 | dabukalam | now I'm looking at this... |
| 22:43.06 | dabukalam | I have a freshly installed asterisk running on a raspi on the same LAN |
| 22:43.33 | dabukalam | I just want to make a phone call out to test it but even that looks very complicated so I might go to bed and try it another time |
| 22:49.51 | Samot | Yeah, the problem with it being 15 years old it is won't have anything about PJSIP. |
| 22:50.02 | Samot | Since Chan_SIP is deprecrated... |
| 22:57.46 | electronic_eel | doesn't look like it needs some chan_sip specific stuff to me, should be straightforward to use with pjsip |
| 22:59.52 | electronic_eel | just configure each channel as a separate sip account, just as they describe in the linked interop pdf |
| 22:59.56 | sysgrammer | hello all, I am stuck with a problem on an asterisk system I setup for a customer. I get no audio on the incoming analog lines. |
| 23:00.03 | sysgrammer | the only was I can hear the IVR over the analog lines is by going to line DAHDI/8-2 and switching back to 8-1. |
| 23:00.08 | sysgrammer | I can switch betweek 8-1 and 8-2 by pressing the discconnect button for a quick second |
| 23:01.18 | sysgrammer | later I foucn switching between channel 8-1 and 8-2 is called "three way audio" |
| 23:12.57 | Reinhilde | long distance barely costs me anything, except 1867, that's expensif |
| 23:17.22 | dabukalam | any idea if i need two-way dialling or one-way dialling? it tells me to refer to the FAQ on their site, but pretty sure the FAQ has changed quite a bit in the last 2 decades... |
| 23:19.41 | Reinhilde | what does one way dialling do, and what does two way dialling do |
| 23:19.48 | dabukalam | sorry s/way/stage/ |
| 23:20.30 | dabukalam | Maybe two-stage would support dialling out somewhere else once connected? I'm not sure |
| 23:21.23 | dabukalam | oooh maybe two-stage would support a dial tone once connected |
| 23:21.30 | dabukalam | and one-stage would just connect directly |
| 23:22.00 | electronic_eel | dabukalam: I'd go for the variant with a sip account for each line, that seems to be the simpler approach to me |
| 23:22.25 | dabukalam | electronic_eel: yup I followed your advice |
| 23:22.30 | electronic_eel | and use pjsip instead of chan_sip |
| 23:22.42 | dabukalam | the numbers in that pdf don't correspond to the methods |
| 23:22.59 | electronic_eel | the pjsip-wizard makes configuring such stuff easier, no manual aor setup and so on |
| 23:23.24 | dabukalam | is that still governed by sip.conf? |
| 23:23.40 | electronic_eel | no, pjsip.conf and pjsip_wizard |
| 23:23.49 | dabukalam | ahh, crap |
| 23:27.09 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 23:28.00 | Reinhilde | installs a very small telephone into clarjon1's skull |
| 23:28.04 | Reinhilde | ... that was bad |
| 23:28.28 | dabukalam | electronic_eel: so I just put the same things I put in sip.conf into pjsip.conf? |
| 23:28.45 | Reinhilde | no |
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