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| 00:29.17 | gtjoseph | lvlinux: Restart=always ... RestartSec=4 |
| 00:29.46 | lvlinux | gtjoseph: really? that easy? Thanks! |
| 00:29.56 | gtjoseph | yep, that's it. |
| 00:30.18 | lvlinux | RestartSec is wait 4 seconds before restarting I assume? |
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| 00:33.48 | gtjoseph | yep |
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| 04:08.59 | snadge | is this incorrect syntax? |
| 04:09.16 | snadge | exten => _X.,n,ExecIF($[${CUT(SIP_HEADER(P-Asserted-Identity),@,1):5}=613xxxxxxxx]?Hangup(54)) |
| 04:09.39 | snadge | so the idea is.. it pulls the caller id out of the asserted identity field.. then if it matches the 613xxxx number, it hangs up |
| 04:10.34 | snadge | <PROTECTED> |
| 04:30.34 | snadge | never mind.. i fixed it myself.. i love you guys :p |
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| 08:42.57 | dan_j | Hi. Is there any way to have multiple email addresses for voicemail email notification? |
| 08:43.04 | dan_j | Without using aliases. Use case, two people want to receive voicemail emails for one mailbox. One is using gmail, other is using yahoo, so no features for them to set up a distribution address. |
| 08:50.29 | ChannelZ | not that I'm aware of, you'd have to do it on the local side (IE a local alias on the mail server that could mail to multiple addresses) |
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| 09:01.24 | dan_j | Ok. Time to look at the asterisk source code since the users arent able to set up aliases on their mail server. Unless I set up a mail server just to handle aliases. |
| 09:01.40 | dan_j | Thanks for the confirmation. |
| 09:04.25 | WIMPy | What's wrong with a local mail server to handle it? |
| 09:04.47 | WIMPy | Might be a good idea anyway. |
| 09:14.54 | dan_j | Local meaning local to asterisk? |
| 09:15.32 | WIMPy | sure |
| 09:16.57 | dan_j | Most of my asterisk config can be altered by clients using an online portal Ive built up over the years. I don't really want to expose the local mail system if I can help it. |
| 09:17.47 | WIMPy | Looks like you have to. |
| 09:17.50 | dan_j | Plus, it's easier if I can get asterisk to send to multiple addresses because then it will simply be the case of putting multiple addresses in the voicemail realtime db. I wouldnt need to go editting the mail config files each time. |
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| 11:04.27 | tparcina | I have installed Asterisk, on my CentOS 6 server, form asterisk-11 repo. |
| 11:04.58 | tparcina | How can I check is it installed with support for: res_http_websocket, res_crypto and chan_sip? |
| 11:05.36 | tparcina | (I'm trying to setup WebRTC following those instructions: https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 ) |
| 11:07.15 | jwpierce3 | does pjsip work on a polycom 550? |
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| 11:12.58 | dan_j | jwpierce3: why wouldnt it? It's still sip at the end of the day. |
| 11:13.57 | dan_j | tparcina: Are the options selected when you run menuselect? |
| 11:14.07 | wyoung | <3 snom |
| 11:14.54 | dan_j | <3 Yealink |
| 11:15.12 | jwpierce3 | first time working with it. I figured it should work like chan_sip as far as the transport mechanism is concerned. I'm using it only for my "endpoints" but I'm not seeing any traffic on asterisk console |
| 11:16.52 | dan_j | The config is slightly different to chansip. Chansip only has one transport. PJSIP runs multiple transports so that you can bind it to different IPs etc. |
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| 11:17.16 | dan_j | Are you familiar with chansip? |
| 11:17.54 | wyoung | dan_j: snom > Yealink |
| 11:18.09 | wyoung | snom docs are readable |
| 11:18.50 | DanQuinney | Are the snom docs as badly translated from Chinese as the Yealink ones are? |
| 11:19.05 | dan_j | Yealink is about 5 minutes drive from me so I can't really complain |
| 11:19.36 | dan_j | jwpierce3: if you are familiar with chansip, this might help you move to pjsip. https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip |
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| 11:22.22 | dan_j | Fyi, the conversation script didn't really work for me. |
| 11:24.13 | WIMPy | DanQuinney: Mixing up German and Chinese? |
| 11:24.47 | DanQuinney | I'm fluent in neither so it's possible |
| 11:27.01 | tparcina | dan_j: I didn't install Asterisk from source, but from package. Do I still have menuselect or not? |
| 11:29.29 | jwpierce3 | dan_j, it ain't workin for me neither |
| 11:29.55 | jwpierce3 | this damn thing isn't even doing udp |
| 11:29.58 | wyoung | DanQuinney: snom is german, and yes they can speak / writeenglish better than the chinese |
| 11:30.08 | wyoung | well, the chinese in Yealink any way |
| 11:30.37 | wyoung | Yealink sent us the bomb, all your base |
| 11:30.55 | wyoung | although from memory that was korean |
| 11:31.30 | WIMPy | Yealink is also german. It's the cheapo brand of Tiptel. |
| 11:31.48 | wyoung | really? The docs sounds like chinglish |
| 11:33.16 | DanQuinney | also didn't realise Yealink was German |
| 11:33.37 | WIMPy | Either they have been written in chinese to make sure they're cheap or someone used google translate. |
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| 11:43.11 | wyoung | WIMPy: someone as in the person responsible for translating from german to english? |
| 11:43.15 | wyoung | from yealink |
| 11:43.30 | WIMPy | Possibly. |
| 11:43.56 | WIMPy | I haven't had contact with their stuff so far. |
| 11:44.33 | wyoung | I have |
| 11:44.46 | wyoung | Could not get provisioning working with it by following the docs |
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| 11:44.51 | wyoung | snom was easy |
| 11:45.27 | WIMPy | Yes. Their wiki is quite good. |
| 11:46.24 | wyoung | Do you like their PBX software? umm vodia something or other? Or are there equiv asterisk based systems out there ? |
| 11:46.34 | wyoung | well I know there is one from digiym |
| 11:46.36 | wyoung | digium * |
| 11:46.46 | wyoung | and not sure if you can not freepbx in the same light |
| 11:47.12 | wyoung | I am just looking for a better GUI for end users to be able to do simple tasks like add extensions and transfer calls etc.. |
| 11:47.51 | WIMPy | FreePBX was the only one I could get installed at all and I was not impressed at all. |
| 11:48.06 | wyoung | yeah I was not impressed by FreePBX |
| 11:48.29 | wyoung | I have written a few GUIs myself but I would rather not maintain it :) |
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| 11:57.58 | BarthezZ | what would be the moast easy way to non-blocking originate a call from the cli? using FILE() and writing to spool/outgoing? |
| 11:58.34 | wyoung | BarthezZ: or monitor api |
| 11:58.38 | wyoung | AGI or whatever |
| 11:58.52 | wyoung | asterisk-java is one, there is one for python too and others |
| 11:59.02 | BarthezZ | sorry, cli -> dialplan |
| 11:59.26 | BarthezZ | I just noticed function originate is blocking, and I just need to call multiple users and dump them in a conference box |
| 11:59.37 | WIMPy | Call some external application. Or do it via AMI. |
| 12:00.57 | wyoung | AMI <3 |
| 12:01.05 | wyoung | WIMPy: where can I learn all about AMI with python? |
| 12:01.34 | wyoung | WIMPy: I would like to be able to show who is calling who as it happens. |
| 12:01.35 | WIMPy | That Question doesn't make sense :-) |
| 12:01.51 | WIMPy | You can learn about AMI and about Python. |
| 12:02.12 | wyoung | WIMPy: well, I would use java-asterisk which will sit in the system tray or background and show a notification when a call is coming in |
| 12:02.37 | wyoung | python has lousy GUI, although it's web framework stuff is the best |
| 12:02.42 | WIMPy | So where does Pyhon come ion to that picture then? |
| 12:02.43 | wyoung | (Flask and Django <3) |
| 12:02.59 | wyoung | WIMPy: I will make a web based vers |
| 12:03.03 | BarthezZ | WIMPy: it feels wrong... using the dialplan to call an external application to call asterisk... but I'm afraid there's not other decent option |
| 12:03.29 | wyoung | WIMPy: but to start off with I would like a java based app as I know java GUI more than python / wxwidget / gtk |
| 12:03.44 | WIMPy | BarthezZ: There's still the allmighty AMI. But that requires programming. |
| 12:04.05 | wyoung | WIMPy: how is your AMI? |
| 12:04.23 | WIMPy | Alive :-) |
| 12:04.53 | wyoung | Can you assist me in setting up a callback to act on incoming calls? |
| 12:04.59 | wyoung | I am guessing that is a channel creation? |
| 12:05.07 | wyoung | or would it be before that |
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| 12:06.20 | WIMPy | There are a number of events. 'newstate' is probably the one you want/ |
| 12:06.55 | wyoung | ah ok, then what? |
| 12:07.11 | WIMPy | That contains the caller ID. |
| 12:07.18 | wyoung | awesome. that is all I need |
| 12:07.23 | WIMPy | Or how much do you want to display? |
| 12:08.08 | wyoung | new question, I would like to dial a number and join it to a SIP phone, is it possible to set a SIP header so the phone auto picks up instead of having to answer it first before the outgoing number is called/ |
| 12:08.18 | wyoung | WIMPy: CallerID is enough for now |
| 12:08.47 | WIMPy | There are different headers available. |
| 12:09.35 | WIMPy | auto-answer and answer-after or something like that. Google or the phones manual should help there. |
| 12:10.56 | wyoung | WIMPy: I know what the header is, I just don't know how to set it via asterisk-java AMI library |
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| 12:12.18 | WIMPy | Might work with a setvar. Otherwise there's always the magic local channel. |
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| 12:31.46 | Greenlight | wyoung: I think you can add SIP headers using the Variable parameter of Originate |
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| 12:32.50 | dan_j | WIMPy: Yealink is not german? They are Bury, Manchester, UK |
| 12:33.35 | dan_j | tparcina: no idea if you've installed from a package. I've always used source but it's much harder to use source. |
| 12:33.47 | WIMPy | That's definitely even less cinese. |
| 12:34.02 | dan_j | jwpierce3: if it won't do UDP, then it could be a config issue or even a firewall issue. |
| 12:34.14 | dan_j | jwpierce3: Also make sure you've disabled chansip |
| 12:34.15 | jwpierce3 | I've got it working |
| 12:34.20 | jwpierce3 | udp working |
| 12:34.26 | tparcina | dan_j: It can be checked from console: module show like ... |
| 12:35.10 | jwpierce3 | I have both together. I've bound pjsip to 5061 |
| 12:35.46 | jwpierce3 | I'm having a problem with tls. It's crashing asterisk with no indication why in the log |
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| 12:39.13 | dan_j | jwpierce3: no idea. If you can catch gtjoseph, he's quite experienced with pjsip. |
| 12:39.25 | jwpierce3 | thanks dan_j |
| 12:40.11 | file | is it causing a segfault? |
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| 12:41.36 | jwpierce3 | file, this is all that is in messages: http://www.johnthecomputerguy.com/pastebin/messages |
| 12:42.20 | file | what about the full log or the console output when attached using asterisk -r? |
| 12:42.35 | file | and did Asterisk crash, or is TLS just not working? |
| 12:43.16 | jwpierce3 | If I add "bind = 173.199.126.116:5061" to pjsip.conf and restart asterisk, it won't start |
| 12:43.30 | jwpierce3 | with tls |
| 12:43.32 | file | what is the full configuration? |
| 12:43.42 | file | and if you manually start it using "asterisk -vvvvvvvvvgc" what is the output? |
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| 12:44.14 | jwpierce3 | hold on let me try |
| 12:44.30 | jwpierce3 | file, this is all that is in messages: http://www.johnthecomputerguy.com/pastebin/pjsip.conf |
| 12:46.13 | jwpierce3 | asterisk: symbol lookup error: /usr/lib64/asterisk/modules/res_pjsip.so: undefined symbol: pjsip_tls_transport_start2 |
| 12:46.36 | file | how was PJSIP built/installed? |
| 12:46.59 | file | and what version of it is present? |
| 12:47.18 | jwpierce3 | from source, version 2.5 |
| 12:47.31 | file | is there another version installed? |
| 12:47.35 | jwpierce3 | no |
| 12:47.47 | jwpierce3 | asterisk picked it up and built no problems |
| 12:48.39 | jwpierce3 | gonna run pjproject configure again and look to see what it finds |
| 12:49.36 | Samot | Are you sure you're not having a listening port conflict? |
| 12:49.55 | file | that wouldn't cause the undefined symbol |
| 12:49.55 | jwpierce3 | yes, it works fine via udp |
| 12:50.14 | file | the undefined symbol would come from Asterisk being built against one version of PJSIP but at load time the system resolving against a different version |
| 12:51.27 | file | ldconfig -p | grep pj |
| 12:51.30 | file | will give you a list |
| 12:52.29 | jwpierce3 | file, this is all that is in messages: http://www.johnthecomputerguy.com/pastebin/ldconfig |
| 12:52.31 | wyoung | Greenlight: OK, can you give me an example? |
| 12:52.59 | file | what about if you do ldconfig before? |
| 12:53.20 | Greenlight | wyoung: It's something like __SIPHEADER=HeaderName:HeaderValue |
| 12:53.29 | jwpierce3 | file, what do you mean? |
| 12:53.35 | Greenlight | ALthough I seem to remember there also being a number needed |
| 12:53.38 | file | run ldconfig before running ldconfig -p | grep pj |
| 12:53.43 | wyoung | Greenlight: __SIPHEADER? That is a thing in Java? |
| 12:53.51 | file | and the output of ldd /usr/lib64/asterisk/modules/res_pjsip.so would be useful |
| 12:54.04 | jwpierce3 | same output |
| 12:54.15 | Greenlight | wyoung: You'd pass it as Varible in the Originate Action over the AMI |
| 12:54.48 | jwpierce3 | file, this is all that is in messages: http://www.johnthecomputerguy.com/pastebin/ldd |
| 12:55.03 | file | very very odd |
| 12:55.52 | wyoung | Greenlight: ok |
| 12:56.01 | Greenlight | wyoung: Although to be honest, it would be probably be cleaner to originate to a Local extension and then add the header in the dialplan |
| 12:56.27 | wyoung | Greenlight: I can dial a local extension with a random number to dial? |
| 12:56.39 | Greenlight | Sure |
| 12:57.18 | file | gtjoseph, thoughts? |
| 12:57.23 | Greenlight | wyoung: https://wiki.asterisk.org/wiki/display/AST/AMI+Examples |
| 12:57.43 | wyoung | thnx |
| 12:57.45 | wyoung | <3 |
| 12:58.42 | wyoung | Greenlight: hmmm, it has meetme in the example, is that deprecated? |
| 12:58.57 | wyoung | I find it hard to find up to date examples |
| 12:59.27 | Greenlight | That doesn't matter too much in regards what the example is showing, it could be a ConfBridge room |
| 12:59.30 | Samot | wyoung: What are you trying to do? |
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| 13:00.04 | Greenlight | You'd for example originate to Channel: Local/<PhoneExten>@auto-answer |
| 13:00.05 | jwpierce3 | file, http://www.johnthecomputerguy.com/pastebin/configure.log |
| 13:00.15 | Greenlight | And create a context called auto-answer, which set the header, and called your phone |
| 13:00.27 | Greenlight | Of course, this assumes your phone supports the auto-answer headers |
| 13:00.38 | Greenlight | I find hardphones generally do, softphones not so much |
| 13:00.40 | file | jwpierce3, what arguments are you passing to pjproject configure? |
| 13:00.47 | wyoung | Samot: a few things 1) set SIP headers before I tell a SIP phone to call a number, 2) create a GUI app to display the caller id of an incoming call to any extension |
| 13:01.01 | wyoung | Greenlight: ah that makes sense |
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| 13:01.26 | Samot | wyoung: Are the SIP headers going to be dynamic with each call? |
| 13:01.46 | wyoung | Greenlight: can I create a context that adds a sip header then includes my outgoing dial plan? |
| 13:01.52 | wyoung | Samot: yup |
| 13:02.04 | Samot | What's going to drive the change? |
| 13:02.08 | Samot | A database call? |
| 13:02.10 | Greenlight | wyoung: Yea, so just add the header, and then GoTo whereever you'd otherwise go |
| 13:02.30 | wyoung | Samot: a AMI call. Either from a GUI app or web based |
| 13:02.48 | Samot | Who's making the call? |
| 13:03.09 | Samot | You're going to originate the call via AMI? |
| 13:03.13 | wyoung | Samot: for 1) it is an AMI app |
| 13:03.28 | wyoung | Samot: for 2) any one calling from my ISTP |
| 13:03.31 | wyoung | ITSP |
| 13:03.50 | jwpierce3 | file, http://www.johnthecomputerguy.com/pastebin/pjprojectuseflags |
| 13:04.02 | Samot | AMI app doesn't tell me what it does. |
| 13:04.13 | file | jwpierce3, I don't know what that means, those aren't arguments passed to configure |
| 13:04.15 | Samot | Why do you need to change the SIP headers? |
| 13:04.26 | Greenlight | I'm assuming some sort of web form with a click-to-dial type function |
| 13:04.28 | Samot | What information do you need to put in the SIP headers? |
| 13:04.38 | Greenlight | He's wanting to add an auto-answer header |
| 13:05.14 | Samot | Greenlight: I'm looking for the overall scope. |
| 13:05.17 | jwpierce3 | each flag represents each configure argument, red is enabled |
| 13:05.27 | file | those aren't configure arguments for pjproject |
| 13:05.31 | jwpierce3 | nm you cant see it in color |
| 13:05.39 | file | well, some are |
| 13:05.44 | file | is prefix passed? |
| 13:06.01 | jwpierce3 | yes |
| 13:06.11 | *** join/#asterisk subvhome (~vgil@leparker127.l.subnet.rcn.com) |
| 13:06.14 | jwpierce3 | prefix=/usr |
| 13:07.12 | subvhome | Good morning all. Does asterisk require a SIP provider in order to communicate with the outside world? or is this something that can be done entirely by asterisk? |
| 13:07.19 | file | what's the output of ls /usr/lib/libpj* |
| 13:07.30 | subvhome | i presume that the answer is yes since someone needs to provide a number right?> |
| 13:08.01 | wyoung | Samot: to set auto answer for my snom phones |
| 13:08.28 | Greenlight | subvhome: Depends what you mean by outside world. Assuming you mean the PSTN, then you'd usually use SIP or ISDN. |
| 13:08.42 | jwpierce3 | file, http://www.johnthecomputerguy.com/pastebin/lslibpj |
| 13:08.51 | file | you have two PJSIP installs |
| 13:08.56 | Samot | wyoung: You keep giving pieces of a the puzzle with no connection. |
| 13:09.00 | file | you have one in /usr/lib64 and one in /usr/lib |
| 13:09.21 | Samot | wyoung: You want your ITSP customer to use an AMI app and your SNOMs to auto answer. |
| 13:09.35 | Samot | Who's SNOMs? Is this for some sort of support queue? |
| 13:09.36 | jwpierce3 | lib is a symlink |
| 13:09.53 | file | then I dunno - but it really is behaving as if you have two |
| 13:09.59 | Samot | What is the goal of this when it's together. What is it SUPPOSED to do? |
| 13:10.14 | jwpierce3 | file, http://www.johnthecomputerguy.com/pastebin/symlink |
| 13:10.23 | subvhome | Greenlight: yes PSTN.. ok.. I'm going to do my research and dive into this. I have a pretty strong linux backrground and can RTFM... asterisk shouldn't be such a hair puâlling task .. right |
| 13:10.23 | subvhome | ? |
| 13:11.01 | Greenlight | subvhome: Yup, it's not a very steep learning curve, I imagine you'll have a phone ringing within the hour! :) |
| 13:11.13 | subvhome | gtfo :) |
| 13:11.22 | subvhome | if thats the case, I'll buy you a beer :) |
| 13:11.39 | Greenlight | :) |
| 13:12.11 | wyoung | Samot: ": |
| 13:12.21 | Greenlight | https://wiki.asterisk.org/wiki/display/AST/Getting+Started |
| 13:12.24 | wyoung | Samot: no, they are seperate concerns, which is why I marked them with different numbers |
| 13:12.57 | Samot | So let's cover the AMI app then. |
| 13:13.06 | Samot | What is it supposed to do? What's the goal of it? |
| 13:13.40 | Greenlight | ...to place a call from a Snom phone to a number |
| 13:13.50 | Greenlight | as I understood it anyway |
| 13:14.33 | Samot | Then why is AMI involved for a call originating from an IP Phone? |
| 13:15.34 | wyoung | Samot: Well both are AMI apps :) but the first one should be able to call a number for a SIP phone without the SIP phone first having to accept the call from the AMI call |
| 13:15.58 | Samot | How is the call ORIGINATED? |
| 13:16.02 | Samot | What triggers it?! |
| 13:16.17 | Samot | Do you want to connect that call back to the SNOM? |
| 13:16.18 | Greenlight | I'm assuming some sort of click-to-dial type webform |
| 13:16.19 | wyoung | Samot: the second app should display the caller id of an incoming call originating from my ITSP |
| 13:16.35 | Samot | Display it where? |
| 13:16.52 | wyoung | Samot: The app that has the AMI library in it triggers it, it will be a python based web app and a java based GUI app |
| 13:17.11 | WIMPy | wyoung: You can off course do both things directely with the phone as well. |
| 13:17.12 | wyoung | Samot: display it on the screen where the app is running |
| 13:17.24 | Samot | Alright.. |
| 13:17.26 | Samot | First part.. |
| 13:17.38 | Samot | Making a call on behalf of a SIP phone... |
| 13:17.39 | wyoung | WIMPy: I know but my computer screen is bigger than my phone's screen |
| 13:17.56 | Samot | So WHO initiates the call? |
| 13:18.01 | WIMPy | wyoung: No, I mean talking to the phone instead of AMI. |
| 13:18.19 | wyoung | Samot: the app |
| 13:18.25 | WIMPy | Or you can even have the phone call an URL. |
| 13:18.29 | Samot | HOW DOES THE APP DO IT? |
| 13:18.30 | wyoung | Samot: the app is web based of java based GUI |
| 13:18.35 | wyoung | Samot: using AMI |
| 13:18.41 | Samot | Automatically? |
| 13:18.50 | Samot | It just decides "Hey, I'm going to call this number" |
| 13:18.58 | wyoung | Samot: no, when the user types in the number or clicks on a number |
| 13:19.04 | Samot | So this is for the USERS |
| 13:19.08 | wyoung | yes |
| 13:19.10 | Samot | To call YOU |
| 13:19.15 | wyoung | no |
| 13:19.19 | wyoung | to call any number they want to |
| 13:19.20 | Samot | Who?! |
| 13:19.35 | Samot | So you need to initiate a call from a Local channel... |
| 13:19.39 | Samot | Call out your trunk... |
| 13:20.02 | wyoung | call out my trunk or a local channel, any number in the internal context |
| 13:20.07 | Greenlight | Indeed, that's what I said 20 minutes ago |
| 13:20.07 | Samot | Nope. |
| 13:20.11 | WIMPy | What are you actually talking about? |
| 13:20.15 | Samot | It's a Local channel. |
| 13:20.25 | Samot | You send the AMI command to Originate a call.. |
| 13:20.41 | Samot | It's going to pick up a LOCAL channel dial that call out your trunk. |
| 13:20.47 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-wixyvontxfobipdh) |
| 13:20.47 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:20.49 | Samot | If you have the dialplan correct. |
| 13:20.53 | wyoung | yes to a SIP phone, I want the SIP phone to auto pick up |
| 13:21.11 | Samot | How do you know then party they are calling is a SIP phone? |
| 13:21.12 | wyoung | as soon as I click dial on the GUI app I want the number to be called and the SIP phone to connect to it |
| 13:21.14 | Greenlight | So as I said, make a context that adds the auto-answer header, originate to that |
| 13:21.22 | Samot | Or do you want THEIR SIP phone to answer? |
| 13:21.27 | WIMPy | Or just dial from the phone. There's an URL to do it. |
| 13:21.41 | wyoung | WIMPy: yes there is, that is what I use atm |
| 13:21.42 | Greenlight | He wants *his* phone to answer automatically... |
| 13:21.53 | Samot | Stop Greenlight. |
| 13:21.54 | wyoung | WIMPy: I would like to do this via asterisk though |
| 13:22.01 | Samot | Let him answer. |
| 13:22.08 | wyoung | Samot: what Greenlight said |
| 13:22.10 | Greenlight | It's really pretty straightforward what he's looking to do |
| 13:22.11 | WIMPy | I know. - I remember from some time ago... |
| 13:22.14 | Samot | So far this is all over the place. |
| 13:22.18 | Greenlight | And I answered it 20 minutes ago |
| 13:22.30 | WIMPy | Ok, just set auto answer and there you go. |
| 13:22.33 | Samot | It's not HIS phone. |
| 13:22.40 | Greenlight | WHat? |
| 13:22.45 | wyoung | Samot: that is scenario 2 |
| 13:22.45 | Samot | He wants anyone to make the call via the APP |
| 13:22.51 | Samot | FFS. |
| 13:22.57 | Samot | Scenario ONE |
| 13:23.12 | Greenlight | No, the app will choose which phone the call should be originated from |
| 13:23.23 | wyoung | Samot: yes, I want any one to be able to tell the phone that is sitting next to them to call a number and the phone to auto answer |
| 13:23.27 | Greenlight | Presumably the user selecting the phone sitting next to them |
| 13:23.39 | Samot | wyoung: You can't. |
| 13:23.48 | Greenlight | Yes you can |
| 13:23.56 | Greenlight | Just add the auto-answer header like I said |
| 13:23.57 | wyoung | Samot: I have a list of computers (IP and MAcs) and a list of phones (IPs and Macs) |
| 13:24.14 | Samot | See now you are giving more details. |
| 13:24.25 | Greenlight | Where? |
| 13:24.27 | WIMPy | thought that was done at least an hour ago or whenever it started. |
| 13:24.30 | Samot | This is different than "I want anyone to be able to enter a number in the app and then get the call" |
| 13:24.35 | wyoung | Samot: yes, bit you need to solve the first part first :) then we can move onto the second |
| 13:24.35 | Greenlight | thought so too |
| 13:24.51 | wyoung | Samot: and rthe third |
| 13:24.53 | Samot | anyone != a list of users. |
| 13:25.08 | wyoung | Samot: anyone within my organisation |
| 13:25.28 | Greenlight | Second part is going to be somewhat trickier |
| 13:25.29 | wyoung | Samot: any one within my organisation with a phone next to them |
| 13:25.36 | Samot | So you have it where it calls out and bridges the calls together? |
| 13:25.40 | wyoung | Greenlight: yes, which is why I broke it up into parts |
| 13:26.11 | wyoung | Samot: I can do that but the user needs to first accept the call from the AMI app first before the number is dialled |
| 13:26.28 | wyoung | Samot: I need to setup autoanswer for my snom |
| 13:26.40 | Samot | So every user is using an SNOM? |
| 13:26.47 | WIMPy | Do it! |
| 13:27.32 | Greenlight | FOr the second part, do you want a list of all current inbound calls with the callerid? |
| 13:27.53 | Greenlight | Or is it just for a screen-pop type thing for a user when their phone rings? |
| 13:29.01 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 13:29.04 | wyoung | Samot: yes |
| 13:29.11 | wyoung | WIMPy: thanx for the support! |
| 13:29.25 | wyoung | Greenlight: screen pop up when the call comes in |
| 13:29.46 | Samot | That part isn't AMI it's AGI |
| 13:29.53 | Greenlight | You *could* do it using the AMI, but to be honest, might be easier to add a CURL to your dialplan or an AGI call |
| 13:29.57 | Greenlight | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CURL |
| 13:30.10 | wyoung | Greenlight: thnx |
| 13:30.14 | Samot | You need to call on an AGI that will send the call information to your app. |
| 13:30.34 | Greenlight | Agreed |
| 13:30.46 | Greenlight | Or CURL |
| 13:30.48 | wyoung | Samot: ok |
| 13:30.53 | Samot | Honestly, this sounds like something for ARI. |
| 13:31.08 | WIMPy | Much lower resource use when using AMI. |
| 13:31.15 | WIMPy | ? |
| 13:31.39 | Greenlight | Sure, but I don't get the feeling we're talking call volumes here where that's an issue |
| 13:32.38 | wyoung | hmmm |
| 13:32.48 | wyoung | as long as I don't have to oikk |
| 13:32.51 | wyoung | poll* |
| 13:33.06 | WIMPy | It also depends on where you process the data, off course. |
| 13:33.10 | Greenlight | Naa, just get your app to listen for the events being pushed to it |
| 13:33.22 | Greenlight | Whatever method you choose, CURL, AGI or ARI |
| 13:34.44 | wyoung | pok |
| 13:34.46 | wyoung | ok |
| 13:36.15 | Samot | So how do I initiate a call from the app, have it ring my SNOM but have to accept the call before the AUTO ANSWER happens? |
| 13:36.45 | Greenlight | You don't have to accept the call? |
| 13:36.47 | Samot | I get that the App will send the AMI command to originate the call but when does it bridge the call to my SNOM? |
| 13:36.55 | Samot | He said he wants the call accepted. |
| 13:37.04 | Samot | Which goes against the auto-answer part. |
| 13:37.18 | Greenlight | Originate in the AMI originates *from* a channel *to* an extension |
| 13:37.35 | Samot | Then it makes the outbound call? |
| 13:37.45 | Samot | This is the part I'm trying to understand. |
| 13:37.49 | WIMPy | That's one possibility. |
| 13:38.05 | Greenlight | Well, whatever that extension does |
| 13:38.08 | Samot | Why am I using an App to make an outbound call when I'm sitting next to the phone the outbound call is going to happen on anyways? |
| 13:38.17 | Greenlight | It could be a conference, a external Dial, or whatever |
| 13:38.23 | Greenlight | Maybe some tt-monkeys... |
| 13:38.33 | Samot | Why am I using an App to make an outbound call when I'm sitting next to the phone the outbound call is going to happen on anyways? |
| 13:39.00 | Greenlight | I assume for ease of use reasons, eg, easy to copy paste a number to dial, or whatever |
| 13:39.29 | Samot | So why do I then need to accept it? |
| 13:39.36 | Samot | Or have auto answer? |
| 13:40.09 | Greenlight | If you didn;'t have auto-answer your snom phone would ring and wait to be answered *before* the call to extension was made |
| 13:40.16 | Samot | Punching in 10-digits is no slower than loading the app, copy and pasting or even putting in the numbers in the app. |
| 13:40.37 | Greenlight | I can copy-paste quicker than I can type 10 digits in a phone |
| 13:40.54 | Samot | Sure. |
| 13:40.55 | Samot | OK |
| 13:41.23 | Greenlight | Focus on the *how* not the *why* :) |
| 13:41.52 | Samot | Yeah, I focus on the why. |
| 13:42.14 | Samot | Because I've seen it too many times where focusing on the "how" was a waste. |
| 13:42.20 | Greenlight | I hear ya |
| 13:42.27 | Samot | Because in the end the "why" wasn't good enough. |
| 13:42.44 | Samot | Just because something can be done doesn't mean it should be done. |
| 13:43.14 | Samot | It's a waste of my time to load an app, put in a number, wait for it to call me, accept the call and then call the party I want. |
| 13:43.22 | Samot | I could have entered 10 digits and been connected . |
| 13:43.27 | WIMPy | Like trying to send realtime data over a packet switched network? |
| 13:43.47 | Greenlight | What if your job consists of making 100's of calls a day from numbers on emails or a CRM system? |
| 13:44.15 | Greenlight | I could imagine in that case being able to right click a number and press "call" would make your life a lot easier |
| 13:44.32 | Samot | He's an ITSP. |
| 13:44.37 | Greenlight | He is? |
| 13:44.41 | Samot | This is a solution for his workers. |
| 13:44.43 | Samot | Yes. |
| 13:45.09 | Greenlight | Well, still, I can see possible benefits for it |
| 13:46.15 | Samot | I'm not saying it doesn't have benefits... |
| 13:46.26 | Samot | But are they enough to make doing something like this worth it? |
| 13:47.27 | Greenlight | From a tinkering and learning perspective it's not a bad project |
| 13:47.49 | *** join/#asterisk rafaels (~rafaels@177.132.177.136) |
| 13:47.51 | *** join/#asterisk SoBlindWolf (~SoBlindWo@173-16-119-155.client.mchsi.com) |
| 13:48.25 | SoBlindWolf | Hey guys, is it possible to hook Google Voice into Asterisk and FreePBX? |
| 13:48.37 | wyoung | no idea |
| 13:48.39 | wyoung | ask WIMPy |
| 13:48.41 | wyoung | he is all knoeing |
| 13:48.46 | wyoung | knowing* |
| 13:48.51 | WIMPy | SoBlindWolf: #freepbx |
| 13:49.03 | SoBlindWolf | Okay thank you! |
| 13:49.03 | WIMPy | knows next to nothing. |
| 13:49.09 | Samot | Yes, you can but it violates the ToS |
| 13:49.15 | wyoung | SoBlindWolf: for your freepbx question , the asterisk one we can answer in here |
| 13:49.43 | SoBlindWolf | Okay |
| 13:49.46 | wyoung | SoBlindWolf: using g729 without a valid licence does too but everyone does it |
| 13:50.07 | SoBlindWolf | Wait so GV can be used just not really allowed? |
| 13:50.32 | Samot | Google Voice is personal use. |
| 13:50.40 | Samot | Putting a PBX on it violates it ToS. |
| 13:50.41 | SoBlindWolf | Also I am having issues understanding SIP stuff I am a noob at this and wanting to learn how to do this as a challenge but I am going up against a wall |
| 13:51.12 | jwpierce3 | file, fixed problem |
| 13:51.16 | file | jwpierce3, what was it? |
| 13:51.42 | jwpierce3 | ssl wan't building, even though I had told it to. had to do ./aconfigure |
| 13:51.43 | wyoung | SoBlindWolf: SIP is just telnet with headers and data |
| 13:51.53 | file | ah |
| 13:51.57 | SoBlindWolf | I would be using it for Home/Personal it would not be for business |
| 13:52.14 | jwpierce3 | ./aconfigure --prefix=/usr --with-ssl=gnutls blah blah blah |
| 13:52.53 | SoBlindWolf | wyoung: Would I be able to pm you so that I don't spam the chat with my mindless questions? |
| 13:53.41 | wyoung | SoBlindWolf: I love mind less quiestions! |
| 13:53.45 | wyoung | I will answer in kind though :P |
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| 14:28.06 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::3) |
| 14:29.32 | SoBlindWolf | hey creslin would you be able to help me get Google Voice working on a test home server? |
| 14:30.52 | cresl1n | haven't setup google voice with Asterisk before |
| 14:31.02 | cresl1n | I setup google talk a long time ago though |
| 14:31.38 | SoBlindWolf | Well I have installed the Google Voice motif |
| 14:32.17 | SoBlindWolf | It says Connected + Inbound Route - Send Unanswered Calls to Google Voicemail |
| 14:32.42 | file | that sounds FreePBX |
| 14:32.49 | SoBlindWolf | Yeah it is... |
| 14:32.58 | SoBlindWolf | How would I configure it all server side? |
| 14:33.06 | SoBlindWolf | Any good tuts? |
| 14:33.07 | cresl1n | I should try FreePBX sometime⦠|
| 14:33.55 | file | https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 14:35.15 | SoBlindWolf | Thanks |
| 14:35.37 | SoBlindWolf | So I need to do this just from terminal? |
| 14:36.09 | file | the tutorial is written without using a GUI, it requires basic knowledge of editing text files and using the Linux terminal - and if FreePBX is still present... then it's not a good idea to do it |
| 14:36.20 | file | because FreePBX can and will undo things |
| 14:37.14 | SoBlindWolf | Ah okay |
| 14:37.34 | SoBlindWolf | So just install asterisk headless and go from there? |
| 14:37.49 | SoBlindWolf | I love linux so it wouldn't be a problem |
| 14:38.00 | file | sure |
| 14:38.20 | SoBlindWolf | Is there an ISO for straight Asterisk? |
| 14:39.23 | tompaw | morning |
| 14:39.40 | *** join/#asterisk wil_syd (~wil_syd@c110-20-159-70.rivrw10.nsw.optusnet.com.au) |
| 14:39.43 | tompaw | SoBlindWolf: there is one for gay ;) |
| 14:39.59 | SoBlindWolf | Lol wut? |
| 14:40.37 | tompaw | "straight Asterisk" |
| 14:40.42 | file | no, there is no Asterisk distribution |
| 14:41.16 | tompaw | SoBlindWolf: what's wrong with asterisknow? |
| 14:42.11 | *** join/#asterisk aness (~aness@cm-84.215.174.224.getinternet.no) |
| 14:42.53 | SoBlindWolf | "<file> the tutorial is written without using a GUI, it requires basic knowledge of editing text files and using the Linux terminal - and if FreePBX is still present... then it's not a good idea to do it" |
| 14:43.43 | tompaw | ah, I see. |
| 14:43.54 | SoBlindWolf | See if this sounds right to anyone: http://jermsmit.com/adding-google-voice-to-freepbx/ |
| 14:44.18 | tompaw | building asterisk from scratch isn't that difficult really. follow the wiki and you'll be all set in 5 minutes |
| 14:45.22 | jwpierce3 | ok, tcp works, udp works, but tls for pjsip is giving me this error: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)' |
| 15:01.55 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 15:01.57 | Haris | hello all |
| 15:08.42 | Haris | I need to configure ws access for video calls through webrtc. I was going through this ( http://sipjs.com/guides/server-configuration/asterisk/ ) page. When I configure it as mentioned here, the web server from asterisk doesn't serve anything on port 8088. most responses comes out as not found or fobidden. what am I doing wrong ? |
| 15:14.05 | *** join/#asterisk puzzled (~puzzled@puzzled.xs4all.nl) |
| 15:24.14 | file | jwpierce3, what is the SIP URI? |
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| 15:27.33 | *** join/#asterisk mirela666 (~mirkob@89.184.168.160) |
| 15:28.37 | jwpierce3 | http://www.johnthecomputerguy.com/pastebin/pjsip.conf |
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| 15:29.14 | jwpierce3 | file, http://www.johnthecomputerguy.com/pastebin/messages |
| 15:29.42 | file | you need to configure the certificate information or else the transport probably won't even start |
| 15:29.56 | jwpierce3 | I got the same error |
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| 15:51.19 | Haris | guys ? |
| 15:51.45 | Haris | how can I check if asterisk has loaded the res_http_websocket.so module ? |
| 15:53.56 | hexanol | asterisk -rx "module show like res_http_websocket" ? |
| 15:56.14 | *** join/#asterisk elyob (~elyob@host86-160-116-229.range86-160.btcentralplus.com) |
| 16:04.16 | Haris | it says module is loaded |
| 16:04.42 | Haris | but when I connect with http://lan-IP:8088 (or 8089) Chrome says site cannot be reached ? |
| 16:05.46 | Haris | lan-ip:8088 brings 403 (not found) by asterisk web server |
| 16:06.08 | Haris | if it has loaded res_http_websocket.so shouldn't it be giving something on this port ? |
| 16:06.16 | Haris | is there a way to test it ? |
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| 16:19.21 | qakhan | i have 2 groups of agents Group A (1001 - 1004) and Group B (1005 - 1008) in a queue, i want to send all calls first to Group A agents if no one answer the call for 30 seconds then call go to Group B agents |
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| 16:37.14 | *** join/#asterisk lgaetz (~lgaetz@66.185.28.100) |
| 16:38.39 | lgaetz | using a straightforward dialplan originate on both Asterisk 11 and 13 and seeing weird results. In 11 I can set the Account in the originate command and the CDR(accountcode) is populated but I am unable to set CDR(accountcode) at any point after the Originate |
| 16:39.00 | lgaetz | In 13 I can set CDR(accountcode) anywhere and the CDR populates fine |
| 16:40.24 | lgaetz | Correction, not using dialplan originate, I am using AMI originate, |
| 16:43.25 | lgaetz | So I guess I'm wondering if I have uncovered a bug, or if there are expected differences between Asterisk 13 and 11 for CDR(accountcode) |
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| 17:04.06 | Samot | lgaetz: You're setting the accountcode variable in the call file? |
| 17:05.02 | lgaetz | no call file, using AMI Originate with the Account parameter seems to be the only way I can get the CDR table to record the account code in 11 |
| 17:08.58 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 17:09.32 | Samot | So yeah, accountcode=12334 or whatever. |
| 17:09.33 | Samot | Hrm. |
| 17:09.39 | Samot | That should be passed as a global. |
| 17:09.45 | Samot | And hit all the channels. |
| 17:09.56 | Samot | Well it does in 13, so yeah it could be an 11 thing. |
| 17:10.49 | Samot | I know any custom vars I send over with AMI in 13 get passed through all the channels I use. |
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| 17:13.07 | Samot | So maybe you did find a bug. |
| 17:13.18 | Samot | A brand new shiny one. What will you name it? |
| 17:13.59 | Samot | If you send 5 bitcoins to @drmessano he'll make you a certificate of authentication and everything. |
| 17:14.49 | Greenlight | You can prefix variables with double underscore to make them passed along to all channels |
| 17:15.11 | Greenlight | You still reference then via the non-underscore name |
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| 17:15.20 | Samot | Yes but in 13 if you pass them through AMI they are treated as global |
| 17:15.37 | Samot | So you can call them on any channel you are using at any time. |
| 17:15.47 | Samot | It appears in 11 that's not happening. |
| 17:15.49 | lgaetz | I don't have bitcoins, will he take magic beans as payment? |
| 17:15.55 | Samot | I think he will. |
| 17:16.00 | Greenlight | Hmm... what if you needed channel specific variables |
| 17:16.08 | Greenlight | (Must admit I've not played with AMI in 13 yet) |
| 17:16.11 | Samot | You can assign them when you call the channel. |
| 17:16.30 | Samot | AMI just sends the call and calls a context to handle the call. |
| 17:16.49 | Greenlight | Oh, I thought you were meaning variables passed in the Originate command |
| 17:16.56 | Greenlight | *action |
| 17:17.01 | Samot | Yes, those should be treated as globals. |
| 17:17.11 | Greenlight | Hmm |
| 17:17.14 | Greenlight | That seems odd |
| 17:17.16 | Samot | I send custom vars through the AMI originate action all the time. |
| 17:17.27 | lgaetz | channel variables seem to work fine, it sure looks like a bug, after the call originates, i can set CDR(accountcode) a value, i can output that value back in the log with a noop(${CDR(accountcode)} but it will not get written to the CDR table, i only see the problem when going back thru cdr records |
| 17:17.29 | Greenlight | Me too, in 11, and they're all channel specific |
| 17:17.41 | Greenlight | I *need* them to be channel specific |
| 17:18.00 | Samot | You can call on them the channel you want. |
| 17:18.20 | Greenlight | It might be too late by then |
| 17:18.27 | Greenlight | I need them to be created with the channel on originate |
| 17:18.32 | Samot | OK. |
| 17:18.34 | Samot | It does. |
| 17:18.43 | Greenlight | Seems odd that they're global now in 13 |
| 17:18.51 | Greenlight | BUt as I say, I've not looked at AMI much in 13 |
| 17:19.07 | Greenlight | Sounds like I'll have lots of fun migrating code |
| 17:20.38 | Samot | Hrm..wait..I'm using AMI + callfiles so it could be because of the call file settings. |
| 17:20.41 | Greenlight | Like for example, I use Originate to "whisper" a audio file to a channel, and I use a variable to represent the audio file name |
| 17:21.05 | Greenlight | Ahh, that makes more sense |
| 17:21.12 | Samot | So I could have just been totally talking out my ass in regards to what lgaetz is doing. |
| 17:22.14 | Greenlight | You can certainly use Account with Originate in 11, if that's what he was asking |
| 17:22.29 | lgaetz | Ass talking, you need to be on America's Got Talent |
| 17:23.27 | Samot | I should be. |
| 17:23.53 | lgaetz | So to clarify, in Ast 11, the ONLY way I can get an account code to be recorded in the MySQL cdr table is to set the Account parameter in the AMI Originate, if i try to set it in dialplan after the originate, it will NOT be recorded in the CDR |
| 17:24.27 | lgaetz | In ast 13 I can set it in the Originate, or I can set it in the subsequent dialplan, both work |
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| 17:25.04 | Greenlight | So you're doing like Set(CDR(accountcode)=blah) in the dialplan, and it works in 13 and not in 11? |
| 17:25.13 | Greenlight | *before* the hangup in both cases? |
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| 17:36.52 | lgaetz | @greenlight, that is exactly correct, it appears to work, I can set the variable in Dialplan, I can read it back and write it to the full log with a noop, but the cdr table doesn't get populated in 11 |
| 17:38.02 | Greenlight | lgaetz: That's very strange, setting the accountcode that way has worked for like forever |
| 17:38.49 | lgaetz | yeah, tested on 2 systems running 11.21.2, don't have other 11 versions at hand to test with |
| 17:40.03 | Greenlight | Can you pastebin the output of your testing and the dialplan? |
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| 17:44.18 | Greenlight | Damn 30 degrees C in the office here today...It's tooo warm |
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| 17:49.10 | nixnothing | <--- used to live in Texas where is gets upto 38 C regularly in the summer |
| 17:49.40 | Greenlight | Bet you had air con though... |
| 17:55.01 | tompaw | what's the ETA for 13.9.2? |
| 17:56.37 | qakhan | i have 2 groups of agents Group A (1001 - 1004) and Group B (1005 - 1008) in a queue, i want to send all calls first to Group A agents if no one answer the call for 30 seconds then call go to Group B agents |
| 17:57.35 | Greenlight | qakhan: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Queue use the timeout |
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| 18:07.10 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 18:08.18 | _abc_ | Hello. I am building *-1.8-current from source on a slackware machine, which has *1-6.11 running on it. The config seems okay except chan_sip is X-ed in the config. I can't find anything relevant in the ./configure output. Where would one look for the relevant package or library message, without reading config... |
| 18:09.45 | WIMPy | Did you enable websockets? |
| 18:09.55 | WIMPy | (or read the last lies on the screen?) |
| 18:10.04 | _abc_ | lies? |
| 18:10.13 | Greenlight | Yes, don't beleive them! |
| 18:12.13 | _abc_ | I don't see anything relevant there eh. |
| 18:12.21 | _abc_ | config is okay, builds, but no chan_sip |
| 18:13.15 | Greenlight | Doesn't menuselect show the missing prerequisites for chan_sip down at the bottom? |
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| 18:16.51 | _abc_ | They are not missing... rtp and local are both on and in |
| 18:17.26 | _abc_ | I did a make distclean and now ./configure 2>&1 logging output |
| 18:17.38 | _abc_ | grepping -i for chan_sip or just sip brought no clues |
| 18:17.45 | _abc_ | Let's wait a few moments. |
| 18:18.42 | nixnothing | yeah I did have ac |
| 18:18.45 | Greenlight | DOesn't it depend on res_crypto |
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| 18:19.20 | Greenlight | If it's marked as XXX in menuselect it's because a dependency is missing, but it should show you which ones are required |
| 18:19.21 | nixnothing | central ac everywhere south, but would prob die if went outside |
| 18:19.38 | nixnothing | heatsroke was a thing they constantly warned kids about |
| 18:20.38 | Greenlight | No AC here (it's Scotland after all), but it's 30c *inside*. It's a stupid design of a building with metal roof and windows that don't open |
| 18:21.25 | Greenlight | _abc_: Do you have the SSL libraries installed? |
| 18:22.23 | _abc_ | I have I think. |
| 18:22.31 | Greenlight | openssl |
| 18:22.44 | _abc_ | I don't need crypto sip anyway. Openssl should be installed |
| 18:22.59 | Greenlight | I'm pretty sure chan_sip requires it regardless |
| 18:23.08 | _abc_ | libssl.so.x.x is present, 2 versions |
| 18:23.09 | Greenlight | But I'm confused why your menuselect isn't saying that |
| 18:23.28 | _abc_ | menuselect simply shows chan_sip X-ed and deps chan_local and *rtp |
| 18:23.31 | _abc_ | both of which are on |
| 18:23.45 | _abc_ | Where IS the generated config, or what symbol is related to SIP? |
| 18:24.07 | Greenlight | Is res_crypto showing any dependencies missing? |
| 18:26.19 | _abc_ | res_crypto is X-ed |
| 18:26.31 | Greenlight | Yea, I'm fairly sure you need that for chan_sip |
| 18:26.34 | _abc_ | Greenlight: where are the real config options placed? |
| 18:26.50 | Greenlight | SOrry, I'm not sure what you mean |
| 18:26.53 | WIMPy | Shouldn't be neccessary. |
| 18:26.57 | _abc_ | Well that depends on openssl which is installed. What does (E) mean in the depends? |
| 18:27.18 | _abc_ | Greenlight: I mean, WHAT symbol is defined/undefined and in what file to control chan_sip compilation |
| 18:27.46 | _abc_ | configure.log or the output of ./configure does not contain any reference to sip |
| 18:28.12 | WIMPy | No, why should they? |
| 18:32.01 | _abc_ | sigh |
| 18:32.57 | _abc_ | I think I asked a specific question: what symbol governs chan_sip building? |
| 18:33.12 | _abc_ | I would like to see its deps in the configure script |
| 18:33.29 | _abc_ | Do I have to reverse engineer the entire build script system to find out? |
| 18:34.06 | _abc_ | Also does not * maintain a .config file like the linux kernel does, with all settings in one place? |
| 18:34.24 | WIMPy | You should see them at the bottom of make menuconfig. |
| 18:34.55 | WIMPy | It's in menuselect.makeopts but fiddling around there is unlikely to help. |
| 18:34.58 | _abc_ | Okay so installing openssl (not just libs), then configure -> res_crypto is built, and chan_sip too. Thanks for the help. The questions remain? |
| 18:35.14 | _abc_ | WIMPy: I don't want to fiddle I want to know where to look next time when I have trouble |
| 18:35.26 | WIMPy | And you never answered if you have websockets enabled. |
| 18:35.54 | _abc_ | I don't even know what that is |
| 18:36.07 | WIMPy | You need them for chan_sip. |
| 18:36.30 | WIMPy | It's a resource module IIRC. |
| 18:36.56 | _abc_ | I found out what the protocol is |
| 18:37.46 | _abc_ | I do not see a res_websockets anywhere in the menu |
| 18:38.24 | _abc_ | Only place 'websockets' appears, with any case, is in ChangeLog |
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| 18:39.06 | _abc_ | And the record where it appears is from 2014. Are you asking whether the kernel has websockets enabled, or what, please? |
| 18:39.40 | _abc_ | It is not in kernel config fyi |
| 18:40.16 | _abc_ | Greenlight: okay, your info helped. Interesting, NOW the res_crypto appears as dep for chan_sip. Maybe I missed it before?! |
| 18:40.16 | WIMPy | No, Asterisk. |
| 18:40.36 | _abc_ | WIMPy: try to grep the sources yourself? grep -Rli websock * |
| 18:42.16 | WIMPy | Tons of stuff. |
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| 19:15.52 | _abc_ | Thanks for the help, will be back. Thanks. |
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