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| 02:15.44 | nonasuomy | Trying to convert my chan_sip setup to pjsip using 18.2 with voip.ms I seem to have outbound and inbound working but kept getting this error when I turn logger on "<--- Transmitting SIP response (361 bytes) to UDP:192.168.1.42:5060 ---> SIP/2.0 404 Not Found" |
| 02:17.24 | nonasuomy | I was reading about adding NoOp options so I tried to first add this to my inbound rule but it stopped inbound from working. |
| 02:19.05 | nonasuomy | https://pastebin.com/esqwbJSS |
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| 02:27.17 | igcewieling | sounds like qualifys. are they options packets? |
| 02:38.24 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 02:42.34 | UncleKiwi | Hola - i have a setup where i have a couple of hardphones and some ATAs the PBX is registed to a SIP provider and I cam make calls between internal phones and the PSTN. I can use a phone that is connect to the ATA and make a call to the PSTN network and then transfer that call to another phone and if it is unanswered the call will return to me . When I try this same situation with the |
| 02:42.34 | UncleKiwi | hardphone the call is ended upon the target of the transfer reaching busy tone. |
| 02:43.00 | UncleKiwi | just wondering how i can have failed xfers return to this hardphone |
| 02:44.02 | Samot | UncleKiwi: talk to the provider |
| 02:44.32 | UncleKiwi | the issue is in my PBX i think Samot |
| 02:45.26 | Samot | Oh i missed that. Sorry. So what hard phones? |
| 02:45.48 | UncleKiwi | its a GXP2160 |
| 02:46.43 | UncleKiwi | this phone seems to behave different in a call xfer than the ATA |
| 02:46.52 | Samot | Where is the forwarding being handled? |
| 02:47.01 | Samot | From the phone? |
| 02:47.16 | UncleKiwi | I think so |
| 02:47.37 | Samot | How is the transfer initiated? |
| 02:47.56 | UncleKiwi | using the BLF keys or the softkeys |
| 02:49.14 | Samot | What do sip debugs show between the calls? |
| 02:50.32 | UncleKiwi | i'll have to get that info |
| 02:50.37 | UncleKiwi | sorry |
| 02:51.38 | UncleKiwi | i find it interesting that the ATA can deal with the situation nicely |
| 02:52.37 | UncleKiwi | flash - extn - ringing - hangup - if call is not answered -- the phone rings again with the call |
| 03:03.49 | drmessano | What is a hardphone? |
| 03:04.37 | UncleKiwi | https://getvoip.com/blog/2019/04/04/hardphones-vs-softphones/ |
| 03:04.50 | UncleKiwi | i thought it was a common term |
| 03:05.08 | UncleKiwi | to describe a physical sip phone |
| 03:06.11 | drmessano | Well |
| 03:06.16 | drmessano | I don't call them hardphones |
| 03:06.19 | drmessano | or sip phones |
| 03:06.22 | drmessano | They're phones |
| 03:06.28 | drmessano | or maybe a desktop phone |
| 03:07.06 | drmessano | But the term "softphone" doesn't require other phones to be qualified as "hardphones" |
| 03:14.55 | Samot | UncleKiwi: ATAs dont have blind xfer as a primary option |
| 03:15.14 | Samot | A lot of sip phones will |
| 03:15.31 | Samot | This could be a blind xfer |
| 03:16.02 | Samot | Without seeing how a call is handled hard to say |
| 03:33.27 | nonasuomy | @igcewieling full burp looks like this https://pastebin.com/raw/h0CTBMRk |
| 03:35.27 | UncleKiwi | Thanks Samot |
| 03:37.53 | Samot | nonasuomy: that looks like a MWI subscribe |
| 03:39.16 | nonasuomy | Is there a way for that to cleanly complete without a 404 not found message? |
| 03:40.02 | Samot | Well 404 means it didnt find what was being subscribed for |
| 03:40.41 | Samot | You have mailbox= set in the AOR? |
| 03:41.03 | Samot | or mailboxes= actaully |
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| 03:48.47 | nonasuomy | This is my full config of pjsip.conf/extensions.conf https://pastebin.com/UeWL6Evn |
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| 04:10.32 | Samot | OK you have an phone trying to register for MWI and nothing setup to accept MWI registrations. |
| 04:10.37 | Samot | So you're going to have that error. |
| 04:10.57 | Samot | Your options are 1) Set up Asterisk to allow it or 2) Turn it off in the phone. |
| 04:11.22 | nonasuomy | How do I fix that from the config above as I'm not sure how I process that. |
| 04:11.52 | nonasuomy | (I've only added those settings above to the server) |
| 04:12.27 | nonasuomy | Oh the phone itselfs config. |
| 04:12.54 | nonasuomy | hmm |
| 04:20.22 | nonasuomy | So I see that there https://pastebin.com/raw/y6Gr5Y7g when I was using sip I was using that because I was specifying a mailbox for it to dial. |
| 04:22.24 | nonasuomy | Oh for the message indicator |
| 04:22.46 | nonasuomy | So I need to add this for pjsip |
| 04:24.05 | nonasuomy | all of the phone sets point to the same mailbox |
| 04:25.03 | nonasuomy | so how I revise the above to enable this feature to work again with pjsip? |
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| 04:46.10 | nonasuomy | so mailboxes=201@default,203@default,204@default |
| 04:51.48 | nonasuomy | Or do I just do mailboxes=203@default in all the phone set [###] |
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| 04:56.37 | nonasuomy | 1st one didn't seem to work. |
| 05:07.49 | nonasuomy | so added it to all the sets like this https://pastebin.com/9HG7S9f0 |
| 05:08.43 | nonasuomy | and all their message indicators are blinking properly as there is a new message sitting in the inbox but the 404 error message still persists when you turn the pjsip set logger on. |
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| 05:20.18 | nonasuomy | SIP/2.0 200 OK |
| 05:21.31 | nonasuomy | After putting it properly in the AOR like you suggested. |
| 05:22.08 | nonasuomy | And this gave me more information to clarify it as well https://www.spinics.net/lists/asterisk/msg170464.html |
| 05:23.23 | nonasuomy | @Samot thank you for slapping me in the right direction! |
| 05:27.39 | nonasuomy | https://pastebin.com/raw/vJsQpn4j |
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| 06:20.31 | nonasuomy | Seems I have another issue when I dial outbound the server works and inbound works. When I let the server sit for awhile and then call into the server it no longer works until I call outbound again. |
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| 07:45.09 | igcewieling | classic nat translation closing |
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| 07:45.39 | igcewieling | try setting your qualify frequency to 20 seconds or switch to using TCP. |
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| 11:14.57 | DMJC | quick SIP question. DTMF negotiation. If the SIP INVITE doesn't include the formats for RFC2833 or RFC4733 is it still possible to negotiate those formats? |
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| 11:15.48 | file_ | no. |
| 11:15.58 | DMJC | thanks. |
| 11:16.26 | DMJC | that's exactly what I wanted to know. Carrier isn't offerint RTP EVENTs |
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| 14:16.49 | nonasuomy | I'm looking for information for qualify freqency for pjsip but seem to be all related to chan_sip, what is the way to do it for pjsip? |
| 14:19.05 | Samot | ... |
| 14:19.09 | Samot | It's in the docs. |
| 14:26.50 | nonasuomy | so it spelt different qualify_frequency = 15 found this example https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Wizard |
| 14:39.54 | Samot | As are most of the settings shared by both |
| 14:40.05 | Samot | fromdomain in chan_sip is from_domain in chan_pjsip. |
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| 17:03.59 | igcewieling | file: you around? |
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| 17:04.36 | file | barely |
| 17:05.06 | igcewieling | The AGISIGHUP channel variable determines if Asterisk sends a SIGHUP to an AGI if the channel hangs up while the AGI is running. My question is, does it need to be set before the AGI is run, or can it be changed/set from within the AGI. It looks to me, like it must be set before. |
| 17:06.40 | seanbright | looks at the code |
| 17:07.10 | file | before |
| 17:07.16 | igcewieling | seanbright: when I look at the code, all I get is a headache. C isn't my strong point. lol. |
| 17:07.49 | igcewieling | file: thanks. As always your answers are helpful/ |
| 17:08.20 | seanbright | before is confirmed. |
| 17:17.56 | *** join/#asterisk movozo (5e6446e7@94.100.70.231) |
| 17:18.29 | movozo | Hi guys. Hope you are doing well. Can you plese help on this one? |
| 17:18.36 | movozo | Call Recording is on. Call from Outside (03437123456) comes in. Ext 11 answers the call. 11 then does Attended transfer to 12. CDR will then think 11 and 12 talked all the time but in the End 03437123456 was talking to 12. |
| 17:18.37 | movozo | But when transferting the call from 11 to 12 the Phone (Fanivl X6U) can see the original CID in the Screen (0343712345). |
| 17:18.37 | movozo | Is it possible for Asterisk to see that CID (03437123456) as well so we can use it in the Dialplan? |
| 17:18.38 | movozo | Or would be a Custom Dilplan possible that records everything from the beginning to the end and can have all the involved Extensions in this call at the end? |
| 17:18.38 | movozo | Thank you |
| 17:19.47 | movozo | Attended Transfer is going from Fanivl X6U to Fanvil X6U. Those are initializing a new call. |
| 17:20.13 | movozo | Trunk is SIP. Phones use PJSIP. |
| 17:31.39 | ckb | Hi friends! What's the easiest way I can listen in on an active call? AMI? |
| 17:32.50 | ckb | for reference: I'm building a web app, that I will want to be able to originate/receive calls with. |
| 17:34.42 | Samot | You would have to spy on the call. |
| 17:36.02 | ckb | (I understand FCC compliance here)... but would asterisk allow me to stream it over a websocket? |
| 17:36.55 | Samot | You need to spy on the call. Whether or not you do that over websocket is up to you. |
| 17:37.17 | ckb | okay, so where do I need to start looking about "spying" on the call? |
| 17:38.10 | Samot | https://wiki.asterisk.org |
| 17:38.17 | Samot | https://wiki.freepbx.org |
| 17:38.28 | ckb | no other details? |
| 17:41.42 | Samot | Are you unable to read documents? |
| 17:42.34 | ckb | Samot: ChanSpy(), yes. but how does another device listen? |
| 17:42.43 | ckb | OH |
| 17:42.59 | ckb | 555 would ChanSpy on a channel? |
| 17:43.25 | Samot | Well in FreePBX, yes |
| 17:44.38 | Samot | Because that was the arbitrary extension they came up with |
| 17:44.41 | ckb | ChanSpy(...) is an asterisk function though, right? |
| 17:44.54 | Samot | Of course |
| 17:45.09 | file | dialplan application, specifically |
| 17:45.22 | ckb | exten=> 555,n,ChanSpy(...) would make 555 listen in on ...? |
| 17:45.59 | file | the dialing of extension 555 (if there was a priority of 1 as well) would listen in on whatever is given to ChanSpy |
| 17:46.08 | Samot | The wiki covers this |
| 17:46.10 | file | behavior documented on the wiki for the ChanSpy dialplan application. |
| 17:48.56 | ckb | thanks guys |
| 17:49.25 | Samot | You are welcome. |
| 17:49.51 | ckb | And if I'm originating/receiving calls, we're talking AMI, correct? |
| 17:50.09 | Samot | That is one option |
| 17:50.19 | file | AMI is for third party call control, initiating actions, it's not a VoIP protocol for calling itself |
| 17:50.30 | Samot | ^^^^ |
| 17:50.31 | file | you can't write a VoIP client that speaks only AMI to send and receive calls |
| 17:50.41 | ckb | Asterisk is the protocol to do so.... no? |
| 17:50.46 | Samot | No |
| 17:50.54 | Samot | Asterisk is a toolkit |
| 17:50.56 | file | Asterisk is a telecommunications toolkit |
| 17:51.09 | file | and AMI is one of those tools, for third party call control and events |
| 17:52.09 | ckb | https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients |
| 17:52.15 | ckb | is that what I'm looking for? |
| 17:52.26 | file | WebRTC is a mechanism by which web browsers can send and receive calls, yes. |
| 17:53.16 | ckb | Would it be easier to use webRTC for spying if I'm planning on going web client route? |
| 17:53.35 | Samot | Web client that does what? |
| 17:53.42 | ckb | sending and receiving calls |
| 17:53.47 | file | well since AMI is not a VoIP client... you couldn't use that to spy and deliver audio to a browser anyway |
| 17:53.48 | Samot | .... |
| 17:54.18 | file | WebRTC is an option for providing calling functionality in a browser. Whether it meets your needs, I do not know. |
| 17:54.20 | ckb | file: so webRTC... I was told AMI I could spy |
| 17:54.34 | Samot | Who said that? |
| 17:55.03 | file | you can originate calls from AMI, but your definition of originate and what AMI defines as originate differ in practice |
| 17:55.18 | file | in that AMI defines it as a mechanism by which you say "call Bob, upon answer send him here" |
| 17:55.48 | ckb | I can look back through the logs... but it's irrelevant |
| 17:56.01 | Samot | No one said that |
| 17:57.08 | file | WebRTC is complex. |
| 17:57.35 | file | (just throwing that out there) |
| 18:01.05 | ckb | file: I'm willing to take on the challenge. |
| 18:01.34 | Samot | For the love of Lemmy. |
| 18:01.38 | ckb | I can't find where someone told me that AMI could do the spying. |
| 18:01.46 | Samot | Because no one did. |
| 18:02.17 | igcewieling | I just use a plain sip client for spying. |
| 18:03.48 | Samot | ckb: Before you take on this challenge. Remember that asking questions on IRC every step of the way does not count. |
| 18:04.51 | ckb | Samot: This whole rabbit hole started with me asking how to list extensions and see what extensions had devices registered. |
| 18:05.15 | igcewieling | spying via a web page is a totally different rabbit. |
| 18:05.28 | ckb | I am still learning how VoIP works. In my case, whats my easiest way to spy? Using an extension? |
| 18:05.30 | Samot | Yeah. |
| 18:05.54 | Samot | An extension is a dialable location. |
| 18:05.59 | *** join/#asterisk hfb (~hfb@193.36.225.16) |
| 18:06.05 | Samot | I pick up my phone and dial 100, that is an extension. |
| 18:06.07 | ckb | Yes. But the dialplan to ChanSpy() |
| 18:06.16 | Samot | Now what happens with 100 is up the how the PBX is programmed |
| 18:06.23 | ckb | is that the EASIEST for me to do right now? |
| 18:06.24 | Samot | Does 100 cause a device to be called? |
| 18:06.29 | Samot | Does 100 send me to voicemail |
| 18:06.36 | Samot | Does 100 activate chanspy |
| 18:06.59 | ckb | Samot: I understand this. |
| 18:07.00 | Samot | We can't tell you what the easiest way to do is. |
| 18:07.09 | Samot | We have no idea what you are actually doing. |
| 18:07.14 | Samot | What is all involved. |
| 18:07.31 | ckb | How can I set the dialplan for #XNN(?) to spy? |
| 18:07.42 | Samot | Like any other dialplan. |
| 18:07.48 | ckb | is it literally just exten => #XNN? |
| 18:07.49 | Samot | Do you know how to make that happen? |
| 18:07.56 | Samot | No. |
| 18:08.27 | Samot | ckb: I've been saying this quite a bit. You need to read documentation. You need to learn how all this works. |
| 18:08.35 | Samot | And no, asking questions here doesn't count. |
| 18:09.14 | Samot | How to write dialplan is a basic thing of Asterisk. It is well documented. |
| 18:09.44 | ckb | I have looked over extensions.conf and I -KNOW- I need to know the syntax |
| 18:10.12 | file | you need to know a lot of things for what you're attempting to achieve, like, a lot |
| 18:10.23 | ckb | Fundamentally. If I have 100 extensions, and I want to spy on say 545. |
| 18:10.45 | file | conceptually it's easy to say that - but Asterisk is about piecing things together to make a solution, so that's what you have to do |
| 18:10.46 | ckb | so I want to dial #545 and have that happen |
| 18:11.16 | Samot | OK so you do that and call on ChanSpy() with the right options to spy on a specific channel |
| 18:11.30 | ckb | so why can't I do exten => #XNN,... |
| 18:11.44 | Samot | You could but that's incomplete. |
| 18:11.51 | ckb | I can probably figure it out from there |
| 18:11.58 | file | if the dialplan were configured appropriately then it would. |
| 18:11.59 | ckb | (Gotta figure out the active channel, etc) |
| 18:12.12 | Samot | It's not that hard. |
| 18:12.35 | file | ChanSpy has options to control its behavior, including searching for channels |
| 18:12.43 | ckb | perfect |
| 18:12.46 | ckb | that's all I need |
| 18:12.56 | ckb | but #XNN will match 545, yes? |
| 18:13.01 | Samot | No. |
| 18:13.02 | ckb | wait NXX XD |
| 18:13.15 | Samot | Because that's not how pattern matching works in Asterisk. |
| 18:14.53 | ckb | _NXXNXXXXXX... ? |
| 18:15.04 | ckb | I'll google... thanks |
| 18:15.44 | ckb | why the _? |
| 18:16.42 | Samot | READ THE DOCUMENTS |
| 18:16.56 | Samot | Stop asking questions that can be answered by reading the documents. |
| 18:19.00 | ckb | https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 18:19.07 | ckb | Does NOT explain what _ is |
| 18:19.41 | Samot | It is how it knows to look for a pattern match. |
| 18:19.55 | ckb | Thank you. |
| 18:19.57 | Samot | Otherwise it will try to do an exact match. |
| 18:25.17 | ckb | one last question. # and * are valid digits even in pattern matching? |
| 18:37.03 | *** join/#asterisk hfb (~hfb@193.36.225.12) |
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| 18:55.15 | igcewieling | yes |
| 19:27.31 | igcewieling | @*@ <- a Princess Leia Tribble. |
| 19:38.13 | ckb | So does exten => 1234567890,... ; not match on outbound calls to 1234567890? (of course these are fake numbers) |
| 19:40.59 | ckb | I only have 3 directives. Answer, curl, and Hangup... I'm trying to dial out and get the curl to run. |
| 19:41.57 | ckb | (yes I have reload and restart) |
| 19:42.23 | *** join/#asterisk Zombie (~masterz@h221.23.191.173.dynamic.ip.windstream.net) |
| 19:42.34 | Zombie | Hey friends? |
| 19:42.58 | Zombie | I am trying yo add a sip peer in pjsip. |
| 19:43.26 | *** join/#asterisk hfb (~hfb@193.36.225.12) |
| 19:46.11 | Zombie | I'm trying to connect a system that has used chan_sip, and my system uses pjsip |
| 19:49.56 | Samot | That doesn't matter at all. |
| 19:56.24 | Zombie | I figured out a conversion path. |
| 20:00.15 | *** join/#asterisk movozo (5e6446e7@94.100.70.231) |
| 20:00.40 | Samot | A conversation path for what? |
| 20:00.53 | Samot | The other system using Chan_SIP doesn't matter. |
| 20:01.00 | Samot | SIP is SIP. |
| 20:02.27 | ckb | exten => 1339,n,CURL(https://test.tld/test.php) <<<< if I dial 1339 I get "your call cannot be completed as dialed" |
| 20:03.16 | igcewieling | Well, without a priority 1, it won't work. |
| 20:03.25 | igcewieling | Are you using FreePBX? |
| 20:03.49 | ckb | yes igcewieling |
| 20:03.57 | ckb | it's in extensions_custom.conf |
| 20:04.09 | ckb | and I changed it to 1, and still no dice |
| 20:04.15 | igcewieling | Try #FreePBX. |
| 20:04.32 | igcewieling | FreePBX is MUCH MUCH more complicated. |
| 20:04.38 | sibiria | try prio 's' |
| 20:05.07 | sibiria | (without apostrophes of course) |
| 20:05.15 | ckb | nope |
| 20:05.31 | sibiria | and you're reloading the dial plan between tries? |
| 20:05.33 | igcewieling | It is freepbx, I doubt the extension is even tried. |
| 20:05.50 | ckb | restart now everytime |
| 20:44.56 | *** join/#asterisk ryoshu (~kamil@netbsd/developer/kamil) |
| 20:59.26 | *** join/#asterisk allizom (~Thunderbi@unaffiliated/allizom) |
| 21:03.29 | allizom | Hi, I have analog phones connected to asterisk through an ATA, the latter has an option "Send Hook Flash Event (as DTMF)", is there a way for asterisk to react to pressing the R button on the phone? What I want is having multiple calls handled within asterisk. The ATA connects to asterisk via SIP |
| 21:07.12 | igcewieling | Generally the ATA would handle that without needing to send a hookflash to Asterisk. |
| 21:08.15 | allizom | igcewieling: yes, currently my ATA handles the multiple calls. That is what I'd like to avoid, using asterisk to bridge/handle the calls instead |
| 21:08.37 | igcewieling | allizom: I've never heard of it being done that way with SIP. |
| 21:08.47 | *** join/#asterisk eXistenZ (~pectic@bzq-109-67-190-6.red.bezeqint.net) |
| 21:08.52 | Samot | You mean 3 way calls? |
| 21:09.06 | allizom | Samot: including, but not limited to. |
| 21:09.20 | Samot | What else would there be? |
| 21:09.39 | allizom | more than 3 parties |
| 21:09.50 | Samot | On an ATA? |
| 21:10.02 | allizom | yes |
| 21:10.10 | allizom | on asterisk |
| 21:10.13 | Samot | Only if the ATA supports using a conference server |
| 21:10.56 | allizom | Samot: it should be a single call to asterisk from the ATA point to view |
| 21:11.18 | allizom | let asterisk take care of what to do if the user presses that button |
| 21:11.42 | Samot | How does asterisk know what to dial? |
| 21:12.29 | allizom | Samot: sorry, what do you mean? |
| 21:12.45 | Samot | You have call 1. |
| 21:12.51 | Samot | You want to add call 2 |
| 21:13.16 | Samot | You hit Flash, it puts call 1 on hold so you can use another line |
| 21:13.38 | Samot | You make call 2, you hit Flash again. 3 way call |
| 21:14.10 | allizom | Samot: my ATA should not put the call on hold, it should not react at all except for sending the event |
| 21:14.24 | Samot | And then what? |
| 21:14.38 | Samot | What does this event do? |
| 21:15.18 | allizom | Samot: it should trigger a dialplan action, whatever it may be |
| 21:15.48 | Samot | And how does that add more calls? |
| 21:16.26 | allizom | Samot: I will then probably wait for the phone to send an extension to call |
| 21:16.41 | Samot | So you need to dial digits |
| 21:16.58 | allizom | yes |
| 21:17.04 | Samot | That would require more DTMF |
| 21:17.42 | allizom | Samot: I'm not following your issue with that. That's what would happen, yes |
| 21:18.15 | Samot | Then how does Asterisk handle that? |
| 21:19.12 | allizom | WaitExten |
| 21:19.40 | Samot | No. |
| 21:19.51 | allizom | can you explain then? |
| 21:19.55 | Samot | If you are doing this over a live call, that's In-Call features. |
| 21:20.05 | Samot | You need to have Asterisk accept that DTMF to process it. |
| 21:20.18 | Samot | WaitExten does nothing in an active call. |
| 21:20.21 | allizom | ok |
| 21:20.38 | Samot | You not wanting to put a call on hold, which is how this works even in analog, is going to be problematic. |
| 21:22.34 | allizom | Samot: can I put the call on hold from within asterisk, while still being able to receive DTMF digits? |
| 21:23.05 | allizom | I suspect no at this point |
| 21:39.09 | ckb | can I get PAMI support here or is there another channel I can go to? |
| 21:45.20 | Samot | Don't know about in here but there's definitely not a #pami channel. |
| 21:46.28 | ckb | I'm just getting a read timeout on the AMI :( Is there an easy way to test it? |
| 21:51.12 | igcewieling | The question to ask is "How are other people who need to accomplish the same thing do this?" |
| 21:51.47 | igcewieling | that was for allizom |
| 21:55.31 | *** join/#asterisk allizom (~Thunderbi@unaffiliated/allizom) |
| 22:00.24 | ckb | PAMI is giving me a read timeout, but I can telnet just fine :( |
| 22:08.35 | seanbright | what is pami? |
| 22:08.53 | seanbright | ah, php |
| 22:09.29 | seanbright | lol. i love that you've already opened an issue on their github repo. |
| 22:12.03 | seanbright | ckb: can you pastebin the code you are using? |
| 22:12.30 | ckb | seanbright, one moment |
| 22:12.38 | ckb | I'm trying to just use telnet :| |
| 22:13.05 | seanbright | you just said telnet worked fine |
| 22:13.30 | ckb | with PHP lol |
| 22:13.40 | seanbright | annnnd i'm out |
| 22:13.43 | seanbright | good luck |
| 22:13.47 | ckb | no no no |
| 22:13.52 | ckb | I'm just trying these things |
| 22:14.14 | seanbright | ok, do me a favor first |
| 22:14.24 | seanbright | close the issue that you opened on the PAMI repo |
| 22:14.38 | seanbright | there is a 0% chance that it is a bug in that library |
| 22:15.26 | seanbright | https://github.com/marcelog/PAMI/issues/199 |
| 22:15.27 | seanbright | that one |
| 22:16.26 | Samot | Oh wow. |
| 22:16.31 | ckb | hastebin is not working |
| 22:17.21 | seanbright | ~pb |
| 22:17.22 | infobot | pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: https://pastebin.com, https://paste.ubuntu.com, http://paste.debian.net; or install pastebinit with yum or aptitude. |
| 22:18.07 | ckb | seanbright, https://privatebin.net/?a67ec90c3a80cc5d#8nejp9ihSjvCoqwM8sQVX1q3YYJvMUrNKu6gCPHQwrPy |
| 22:18.22 | seanbright | ckb: https://github.com/marcelog/PAMI/issues/199 |
| 22:18.24 | seanbright | close that issue |
| 22:18.39 | ckb | I am having an issue. |
| 22:18.59 | seanbright | we can argue over the definition of 'issue' at a later time |
| 22:20.00 | Samot | Is that 100 milliseconds? |
| 22:20.05 | Samot | For the timeout? |
| 22:20.17 | Samot | Is that measures in ms? |
| 22:20.26 | ckb | OOP |
| 22:20.28 | seanbright | actually, if you search for "read timeout" (with quotes) on their issue tracker you will see a lot of complaints |
| 22:20.30 | ckb | good call |
| 22:20.58 | Samot | Wair |
| 22:21.03 | Samot | I am asking |
| 22:21.16 | ckb | after changing to 1000000 it works fine |
| 22:21.24 | Samot | Is that measured in ms by the code? |
| 22:21.25 | ckb | so it HAS to be MS |
| 22:22.23 | ckb | The example had 10, and doesn't specify if it's in ms or not |
| 22:22.36 | Samot | Ok look |
| 22:22.42 | Samot | Its in the code |
| 22:24.30 | Samot | There is example code that detailed that fact |
| 22:24.39 | ckb | It very well might be, but I shouldn't have to dig through code for documentation, especially a quickstart guide? https://github.com/marcelog/PAMI/blob/master/doc/examples/quickstart/example.php |
| 22:24.40 | Samot | But you opened a ticket |
| 22:25.41 | Samot | http://marcelog.github.com/articles/pami_introduction_tutorial_how_to_install.html |
| 22:27.01 | ckb | I closed the ticket. And was going off the base .md |
| 22:27.35 | Samot | Where do you think I got that link? |
| 22:27.42 | Samot | In the .md |
| 22:27.49 | ckb | Then went to the source/doc/examples/quickstart and didn't see ANYTHING about timeouts |
| 22:28.26 | ckb | Yes, but when the "10" didn't work, I figured I'd try telnet to see. And here we are now. |
| 22:28.54 | Samot | With pami? |
| 22:29.08 | Samot | You used telnet with pami? |
| 22:29.14 | ckb | No. |
| 22:29.35 | ckb | I used telnet domain.tld 5038 |
| 22:30.07 | ckb | And logged in via telnet to see if it could be a library issue. Because "10" wasn't cutting it. |
| 22:31.06 | ckb | When telnet worked, and I got my results, I assumed it was the library because the quickstart in doc/examples has no details about parameters. |
| 22:32.33 | Samot | Ok but the install docs had that data |
| 22:32.43 | ckb | When it says "in-depth" I think "let me show you how to build a regex parser using php" type scenario. |
| 22:33.56 | ckb | Install docs had a link for "in-depth" not a "quickstart" |
| 22:34.29 | Samot | Read that link i posted |
| 22:34.34 | ckb | I did. |
| 22:34.42 | Samot | It has an example right in it |
| 22:34.59 | Samot | It tells you what each setting is required to be |
| 22:35.03 | ckb | but I wasn't going to try to go into an indepth tutorial when I just wanted a simple connection |
| 22:35.14 | Samot | No of course not |
| 22:35.20 | ckb | That should probably be like... in the README.md |
| 22:35.34 | Samot | Not really |
| 22:36.00 | ckb | k well thank you for your help |
| 22:36.26 | Samot | Well hold the thanks |
| 22:36.42 | Samot | This is looking to be a routine thing |
| 22:36.58 | ckb | mountain.... mole hill |
| 22:37.20 | Samot | Yes you are doing that |
| 22:38.59 | Samot | ckb: let me explain something. What you are doing, already have done it |
| 22:39.42 | Samot | Even more Ive made Asterisk run in a multi-tenant mode |
| 22:40.08 | Samot | Which basically made FreePBX obsolete for my needs |
| 22:40.48 | ckb | Right, but we're a VOIP reseller, essentially. so giving clients access to FPBX is useless |
| 22:41.02 | Samot | Im a telecom |
| 22:41.04 | ckb | It's not my business. It's just paying my bills |
| 22:41.24 | ckb | He asks me to build a system like X company, and I'm like "lets go" |
| 22:42.04 | Samot | Except that requires a certain skill and knowledge set |
| 22:42.34 | ckb | or the willpower to find a way |
| 22:42.49 | Samot | The way is to learn those things |
| 22:43.12 | *** join/#asterisk cation21 (cation21@gateway/vpn/protonvpn/cation21) |
| 22:43.21 | ckb | Uh huh |
| 22:43.38 | Samot | What was company x? |
| 22:44.19 | ckb | I can't remember the name. He pretty much wants me to make a drag and drop dialplan |
| 22:44.27 | Samot | Hahahahha |
| 22:44.30 | Samot | Hahahahahaha |
| 22:44.44 | ckb | Why is that funny? |
| 22:44.52 | Samot | Well |
| 22:45.14 | ckb | Just because I don't know my way around the CLI/AMI/ARI/AGI atm? |
| 22:45.14 | Samot | That wont work with FreePBX first off |
| 22:45.22 | Samot | No |
| 22:45.33 | ckb | Why won't it work with FPBX? |
| 22:45.35 | Samot | You dont know diaplan |
| 22:45.45 | ckb | Yes, and I'm learning. |
| 22:45.50 | Samot | Because it is not how freepbx is designed |
| 22:48.22 | ckb | of course, I'm pretty much just taking pieces of FPBX that are clearly possible, and then in the future, completely get rid of it. |
| 22:48.36 | ckb | for now it's serving it's purpose |
| 22:49.22 | Samot | Which is doing all the things no one knows how to |
| 22:49.42 | ckb | wdym? |
| 22:49.56 | Samot | Well |
| 22:50.13 | ckb | Someone has to know how to, because it's been done by a few companies so far |
| 22:50.28 | ckb | And it's been done by FPBX |
| 22:50.33 | Samot | Based on your conversations you boss has no idea what the current state of Asterisk and its features are |
| 22:51.02 | Samot | Other companies have people like me |
| 22:51.11 | ckb | Correct. Queue me, and I'm like a kid in a candy store learning about the Asterisk CLI/AMI/ARI/AGI |
| 22:51.16 | Samot | That actually know this stuff |
| 22:51.39 | Samot | But you dont know either |
| 22:51.59 | Samot | So how is cueing you a change? |
| 22:52.31 | ckb | But I definitely have an education and background of computer science vs him not knowing what an if else if else statement is |
| 22:52.53 | Samot | Ok |
| 22:53.13 | Samot | So you can write an if/then statement |
| 22:53.53 | Samot | There are people that know asterisk that cant program |
| 22:54.23 | ckb | If you want my credentials, I wrote a Jacobi algorithm in PHP, I wrote a regex parser in Java, I wrote a FTP client/server in C, I built an 8 bit computer (ALU, memory, and bus) using a breadboard |
| 22:54.31 | ckb | what do you want from me? |
| 22:54.33 | Samot | Ok |
| 22:54.46 | Samot | Reading docs |
| 22:55.09 | Samot | Not asking questions that are easily answer in documentation |
| 22:56.15 | ckb | here nor there, thank you for your continued help. I'd be a fool if I didn't atleast try to read the docs first. But asterisk and fpbx is a HEAVY load for someone to learn in a week. Some compassion would be wonderful. |
| 22:56.41 | Samot | Well you need time |
| 22:56.52 | Samot | You aint pulling this off in a few weeks |
| 22:57.01 | ckb | And if I do? |
| 22:57.17 | Samot | I will be impressed |
| 22:57.34 | ckb | (: |
| 22:58.24 | ckb | I already have call logs and graphs using the database. I'm sure there would have been a much easier way, BUT alas here we are. |
| 22:59.17 | ckb | Boss wanted me to check if the fpbx firewall was on using the database, I told him there's no f'ing way. Figured out there's a firewall daemon php script that runs and I can just check that. |
| 22:59.59 | Samot | So you answered without knowing |
| 23:00.15 | ckb | I've been in the dark, and now I have all the tools necessary. I just ask simple questions, and apparently, people like to go around their elbow to get to their butt. |
| 23:00.35 | Samot | Hahah |
| 23:01.50 | ckb | simple answer earlier "just use _override_pbx and -custom if you need a dialplan for 911 across your 100 servers |
| 23:02.10 | Samot | Dont know why you need override |
| 23:02.20 | ckb | well, if 911 is already defined |
| 23:02.32 | Samot | Then you should undefine it |
| 23:02.37 | ckb | -custom in _custom won't take? |
| 23:02.42 | Samot | Dont be sloppy |
| 23:02.43 | ckb | 100. servers. |
| 23:02.57 | ckb | (I didn't do this) |
| 23:03.01 | Samot | If it is in the GUI remove it |
| 23:03.12 | ckb | but I'm not going to sit around waiting for him to go remove it from 100 GUIs |
| 23:03.15 | Samot | So FreePBX doesnt create it |
| 23:03.27 | Samot | And then you use yours |
| 23:03.35 | ckb | because I don't have the GUI password for anything except 1 |
| 23:03.38 | Samot | So sloppiness it is |
| 23:03.47 | ckb | I can always move it?! |
| 23:03.57 | ckb | until he gets his ducks in a row |
| 23:04.23 | ckb | but tbh, 911 is definitely a use case where override would be warranted |
| 23:04.58 | ckb | can't really argue that, right? |
| 23:05.20 | Samot | Hey quick math |
| 23:05.35 | Samot | What is 10K x 100? |
| 23:05.55 | ckb | 100000? |
| 23:06.00 | ckb | wait lol |
| 23:06.18 | ckb | 10000000 |
| 23:06.21 | Samot | 1M |
| 23:06.47 | Samot | 10K is the base fine per PBX that fails 911 compliance |
| 23:06.50 | ckb | 4 zeros + 2 zeros = 6. 1 + 6 zeros |
| 23:07.18 | ckb | We are 911 compliant. |
| 23:07.33 | ckb | I'm literally just COPYING the 911 we have, and adding a CURL |
| 23:07.45 | Samot | Well as long as what you are doing works |
| 23:08.19 | ckb | I'm testing this out with my personal phone number first, and leaving things be until I know it works for XXX-XXX-XXXX |
| 23:09.27 | ckb | we just signed with a hotel with 50 rooms and did a random 3 room test with 911 and everything went PERFECT. (Not the server I'm testing on) |
| 23:10.02 | Samot | You used the same number and address for all three rooms? |
| 23:10.44 | ckb | I'm not sure how many numbers we had.... but I have e911 addresses listed on our webapp for every number on an account |
| 23:11.04 | Samot | OK, a hotel requires a location per room. |
| 23:11.08 | ckb | this whole app started with the need of SMS integration. |
| 23:11.19 | Samot | That has the exact location of the call. |
| 23:11.22 | ckb | INTERESTING |
| 23:11.35 | Samot | Welcome to 911 Compliance. |
| 23:11.36 | ckb | one moment. |
| 23:11.49 | Samot | Dispatchable Location is a requirement now. |
| 23:12.04 | Samot | It comes from room 202 it needs to say FL 2 RM 202 |
| 23:12.39 | ckb | Samot, this is critical information. |
| 23:12.44 | Samot | Of course it is. |
| 23:12.52 | ckb | One second. |
| 23:14.01 | ckb | If they all come from the same trunk/number.... how would I do that? |
| 23:14.21 | ckb | the carrier is the one with e911 info |
| 23:14.36 | ckb | are you saying I need a number for EVERY room? |
| 23:14.44 | Samot | You either need a provider that providers Dynamic Locations or you need to do it with DIDs. |
| 23:15.01 | ckb | so caller ID? |
| 23:15.07 | ckb | pretty much? |
| 23:15.30 | Samot | The number you present has to be associated with a location. |
| 23:15.53 | Samot | If your provider doesn't do Dynamic Locations, you need to use a DID per location. |
| 23:16.04 | ckb | which my carrier has an address for that # |
| 23:16.08 | Samot | So your 50 room hotel would need 50+ DIDs. |
| 23:16.38 | ckb | right it's 48 rooms and 3 admin phones |
| 23:16.42 | Samot | OK, Dispatchable Location means that the ADDRESS presented to 911 has the exact location of the CALL |
| 23:16.57 | Samot | So it must present FL 2 RM 202 when 202 calls 911 |
| 23:17.07 | Samot | And FL 1 RM 104 when 104 makes a call |
| 23:17.14 | ckb | I understand this |
| 23:17.21 | Samot | You'll need 49 registered locations. |
| 23:17.24 | ckb | but the DID would mean the caller ID for that room? |
| 23:17.38 | Samot | For 911 calls, yes. |
| 23:17.52 | ckb | like ######### <FL 2 Room 200> or whatever right? |
| 23:17.58 | Samot | No. |
| 23:18.00 | Samot | FFS. |
| 23:18.01 | *** join/#asterisk hvxgr (~wl2v_usrn@epjdn.zq3q.org) |
| 23:18.09 | Samot | You need to have a *REGISTERED LOCATION* |
| 23:18.19 | ckb | YES BUT FOR DID |
| 23:18.19 | Samot | With the PSAP/911 database. |
| 23:18.24 | Samot | NO PER ROOM |
| 23:18.41 | ckb | " you need to use a DID per location" <<<< per ROOM |
| 23:18.44 | Samot | They don't care about CallerID Name |
| 23:19.00 | ckb | oh I'm thinking CLID |
| 23:19.03 | ckb | nevermind |
| 23:19.04 | Samot | There needs to be a registered location for each room. |
| 23:19.08 | ckb | I get it |
| 23:19.26 | Samot | If the carrier doesn't do Dynamic Locations, which allows a single DID to many locations... |
| 23:19.33 | Samot | You need to use a DID PER LOCATION. |
| 23:19.35 | ckb | I GET IT. Thank you. I'm going to hop on the phone right now and express this concern now. Thank you thank you thank you |
| 23:19.37 | Samot | The normal way. |
| 23:19.48 | Samot | Like you need to do this *per PBX* |
| 23:20.19 | ckb | Most of our PBXs only use a couple numbers at the same location (e911 address) |
| 23:20.33 | Samot | OK. |
| 23:20.35 | ckb | This is the ONLY case, but it's very useful info |
| 23:20.45 | ckb | So I commend you for it. You're a saint. |
| 23:21.36 | ckb | so if we violate this it's 10k (just for this hotel) |
| 23:22.03 | Samot | Initial. |
| 23:22.12 | Samot | Then $500/per day until you get it in place. |
| 23:22.29 | ckb | My carrier has Dynamic Location Routing |
| 23:22.39 | Samot | So then you need to sign up for it. |
| 23:22.42 | Samot | Get it setup. |
| 23:22.53 | Samot | But you still need to register the devices and locations. |
| 23:22.58 | Samot | You just get to use a single DID. |
| 23:23.10 | Samot | You're saved the cost of the DIDs. |
| 23:23.10 | ckb | Right I understand. |
| 23:23.17 | Samot | Not the 911 records. |
| 23:24.06 | Samot | See it's not Bandwidth's job to tell you all this. |
| 23:24.15 | Samot | They aren't on the hook for any of it. |
| 23:24.24 | ckb | That's kind of shady tbh :\ |
| 23:24.28 | Samot | No. |
| 23:24.30 | Samot | It's not. |
| 23:24.34 | Samot | You're their customer. |
| 23:24.39 | ckb | We just went through FCC compliance |
| 23:24.42 | Samot | They provide you with PSTN transit. |
| 23:25.00 | Samot | They don't have to be your only upstream. |
| 23:25.17 | ckb | I mean are they paying FCC fees? why do we have to as well? Just because we're on the "traceroute"? |
| 23:25.24 | Samot | They don't. |
| 23:25.33 | ckb | Our bill has FCC fees |
| 23:25.42 | ckb | (I think) |
| 23:25.47 | Samot | Are you a USF contributor? |
| 23:26.11 | ckb | I don't have access to the main BW account. So I have no idea what USF even means. |
| 23:26.17 | Samot | If you're not, then they charge you the taxes and fees. |
| 23:26.39 | Samot | If you are, you give them your FCC ID and they don't charge those anymore. |
| 23:26.44 | Samot | Since it's on you to do it. |
| 23:27.00 | ckb | USF means you're FCC compliant and pay your fees, right? |
| 23:27.21 | Samot | You can be registered with the FCC and not a USF contributor. |
| 23:27.22 | ckb | (FCC compliant lol... e911 convo) |
| 23:27.50 | Samot | And the new 911 laws apply to everyone. |
| 23:28.01 | Samot | Including regular ole PBX installers and admin's. |
| 23:28.06 | Samot | Not just carriers/providers. |
| 23:28.27 | Samot | Every PBX vendor had to update their systems to comply |
| 23:28.37 | Samot | If they needed to. |
| 23:29.18 | ckb | Okay and just for the record, would LEO say something if they get a distress call from said hotel and didn't have a room #? |
| 23:29.45 | Samot | That would be handled by the PSAPs. |
| 23:29.52 | Samot | How they report it, etc. |
| 23:29.55 | igcewieling | All it takes is for someone to file a complaint with the FCC. LEOs won't be involved in the FCC regulations. |
| 23:30.38 | Samot | Or when reporting is done. |
| 23:30.49 | Samot | Because there is reporting that is associated with this. |
| 23:30.53 | ckb | oh so the operator would then be the one to file the complaint? |
| 23:30.59 | Samot | Not just waiting for a complaint to be filed. |
| 23:31.08 | ckb | got it |
| 23:31.09 | Samot | The PSAP will be giving reports. |
| 23:31.20 | Samot | FCC will probably require LECs to report. |
| 23:31.36 | Samot | Interconnect VOIP Providers, etc. |
| 23:31.53 | ckb | LECs? |
| 23:31.58 | Samot | Wow. |
| 23:32.02 | Samot | Local Exchange Carriers. |
| 23:32.07 | ckb | OH doh |
| 23:32.07 | Samot | Bandwidth is a LEC |
| 23:32.25 | ckb | yeah don't mind me, I'm in 2 worlds right now. |
| 23:33.12 | ckb | very valuable info, Samot. I can't thank you enough. |
| 23:33.17 | *** join/#asterisk segnior (segnior@gateway/shell/xshellz/x-mzpvvqssqdmczaju) |
| 23:37.03 | *** join/#asterisk gregs (sid160074@gateway/web/irccloud.com/x-ocsyvqqhjjmvwedr) |
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| 23:46.01 | ckb | <PROTECTED> |
| 23:46.25 | Samot | Ok |
| 23:46.35 | ckb | alternatively, I'm reading over WebRTC atm, and I imagine it's only available for pjsips? |
| 23:48.17 | ckb | might be confusing myself here, but I am going to start showing that an extension is in an active call, so I'm wondering if it's best I start going down the RTC route. I already have an understanding of websockets, as I've built a real-time chat application. |
| 23:49.59 | igcewieling | I think I shall take a break for a few days. best of luck ckb. |
| 23:50.03 | *** part/#asterisk igcewieling (~ewieling@199.27.202.86) |