| 00:27.53 | *** part/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 00:33.39 | *** join/#asterisk mzbotr (~mzbotr@196-102-35-72.mtaonline.net) |
| 00:34.05 | mzbotr | Is it not abnormal for some installers to place an asterisk usergroup, gid 114? |
| 00:34.09 | mzbotr | i thought dialout was all |
| 00:44.40 | *** part/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 00:47.49 | [TK]D-Fender | ? |
| 01:09.50 | WIMPy | I think most will use an exclusive asterisk group. |
| 01:23.42 | *** join/#asterisk Voyage (~user1@unaffiliated/voyage) |
| 01:25.33 | Voyage | Can find a voip ITSP provider for outgoing calls for $0.005 / min or near. All I am getting from goolge is home based customer adapter based calling plans. I need a real voip provider. Any suggestions? DiDLogic was on offering $0.007 but its not offering service at my locale |
| 01:25.38 | Voyage | cant* |
| 01:28.02 | Voyage | to US of course |
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| 03:11.09 | igcewieling1 | Yoi |
| 03:11.28 | igcewieling1 | Yoo might have trouble finding much at that price. |
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| 03:28.27 | agckemito | hi to all |
| 03:36.02 | Voyage | igcewieling1, ya but I saw / heard some offering a similar |
| 03:36.28 | Voyage | igcewieling1, maybe I am not using the correct google keywords? (I am just doing "voip providers") |
| 03:37.05 | Voyage | igcewieling1, if I cant find, at lease I wanted to have per second charge |
| 03:54.29 | igcewieling1 | You could try not being so cheap. 8-|. |
| 03:55.33 | igcewieling1 | You'll need to commit to a contract and call volumes if you want the best prices. |
| 04:02.11 | igcewieling1 | I pay 1.2/cents/min + small monthly for my personal voip number. What sorts of volume can you commit to? |
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| 12:43.14 | Voyage | Wil asterisk work with LCR or I need some special kind of switches and configs? thinq.com |
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| 15:48.04 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 15:51.44 | [TK]D-Fender | Voyage, Asterisk doesn't "word" with LACR, LCR is a concept you can choose to implement in your dialplan. |
| 15:55.33 | Voyage | <PROTECTED> |
| 15:55.47 | Voyage | [TK]D-Fender, sorry, I didnt got you well? |
| 15:56.01 | [TK]D-Fender | wok* |
| 15:56.09 | [TK]D-Fender | work |
| 15:56.27 | [TK]D-Fender | can't type yet today... |
| 15:57.04 | Voyage | [TK]D-Fender, so i cannot use thinq.com with *? |
| 15:59.24 | [TK]D-Fender | What does that say it does? |
| 15:59.42 | [TK]D-Fender | And what is your syste's involvement wuth it imply? |
| 16:00.01 | Voyage | am? |
| 16:00.24 | joako | If I set a SIP phone to forward calls, is there any way to configure Asterisk to send those calls through a different context? |
| 16:00.56 | [TK]D-Fender | And what is your system's involvement with it imply? |
| 16:01.18 | [TK]D-Fender | joako, no |
| 16:15.48 | *** join/#asterisk raypulver (~raypulver@179.43.145.19) |
| 16:16.55 | raypulver | hello. I'm just curious: If I wanted to provide a service like Google Voice or Textfree with voice and SMS, what exactly would I have to rent besides the server? |
| 16:17.34 | [TK]D-Fender | Depends what you intent to use as your back-end to provide those services |
| 16:17.55 | raypulver | asterisk is what I've used before |
| 16:18.52 | raypulver | is there a more economical way to accomplish this? |
| 16:20.17 | [TK]D-Fender | * is th MIDDLE-MAN |
| 16:20.31 | [TK]D-Fender | Asterisk doesn't jsut make Voice and SMS get delivered |
| 16:20.51 | [TK]D-Fender | Lient communicates with you .. you communicate with your back-end provider |
| 16:20.54 | [TK]D-Fender | client* |
| 16:21.57 | Voyage | hm |
| 16:22.51 | Voyage | [TK]D-Fender, honestly I dont know. i guess I have to ask the support of thinq.com but I dont trust they would guide a newbie like you did. |
| 16:23.31 | [TK]D-Fender | So you don't understand what their service even is? |
| 16:23.44 | raypulver | right ok. about how much should I be paying per line? |
| 16:23.59 | raypulver | for voice and sms delivery |
| 16:24.01 | [TK]D-Fender | raypulver, Depends who you use as your back-end |
| 16:24.09 | [TK]D-Fender | The answer is "could be ANYTHING" |
| 16:24.24 | raypulver | who do the pros go through? I guess is my question |
| 16:24.24 | [TK]D-Fender | There are hundreds of potential providers |
| 16:24.36 | raypulver | ah, can you recommend one that is cheap? |
| 16:24.36 | [TK]D-Fender | Pros go though all sorts of different places |
| 16:25.11 | [TK]D-Fender | Clearly there is competition and geography involved, sometimes it's yours, sometimes it'd be your client's |
| 16:26.01 | raypulver | I'm looking for service in North America |
| 16:26.30 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 16:26.49 | raypulver | I've looked online but I am suspicious that the companies with the pagerank are not the ones with the best value |
| 16:26.51 | [TK]D-Fender | Time to go shop around |
| 16:26.56 | *** join/#asterisk kalz (~kalz@2602:fff6:f:1::d979:58e8) |
| 16:44.45 | raypulver | so do I need a SIP trunk for every phone number I plan to be able to provide? I see that DIDs are about $0.50/month each. is that all I need? |
| 16:45.50 | [TK]D-Fender | DID = phone number |
| 16:45.54 | [TK]D-Fender | trunk = nothing specific |
| 16:46.02 | [TK]D-Fender | that is a VAGUE term |
| 16:46.17 | raypulver | I see. |
| 16:46.28 | [TK]D-Fender | typically larger account have you pay for the number of simultaneous CHANNELS you intend to use |
| 16:46.44 | [TK]D-Fender | And then how they get used (inbound to DID, or outbound) is up to you |
| 16:46.52 | raypulver | so with a DID I can dial out from it and receive calls on it? |
| 16:47.03 | [TK]D-Fender | DID has nothing to do with dialing out |
| 16:47.15 | [TK]D-Fender | Direct INWARD Dial |
| 16:47.27 | raypulver | right that's what I was confused about |
| 16:47.34 | raypulver | but I suppose it was obvious |
| 16:47.50 | [TK]D-Fender | I don't need a specific DID to call out. |
| 16:48.13 | [TK]D-Fender | It'd be nice to HAVE a numebr to present in the CallerID so that the peron I'm calling knows who I am and can call me back... |
| 16:48.22 | [TK]D-Fender | (typically) |
| 16:48.28 | raypulver | alright that's how I figured it worked |
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| 16:53.03 | raypulver | so I can provide as many phone numbers as I want but in order for people to place calls simultaneously I need a trunk for every simultaneous call |
| 16:53.16 | *** join/#asterisk dms (~Adium@c-73-131-162-128.hsd1.sc.comcast.net) |
| 16:55.47 | raypulver | also .. I want to expose my service to the Internet. do I need an STUN server for this to work properly? |
| 16:56.40 | [TK]D-Fender | Don't call it a trunk |
| 16:56.43 | [TK]D-Fender | it isn't a "thing" |
| 16:57.09 | [TK]D-Fender | You need a service offering you as many channels as you intend to support |
| 16:57.25 | [TK]D-Fender | No, STUN is not required |
| 16:57.29 | [TK]D-Fender | it can be helpful. |
| 16:58.00 | [TK]D-Fender | You have a lot of reading to do to even understand the technologies and services |
| 16:59.03 | raypulver | I know some of the basics.. where should I start? |
| 17:00.14 | [TK]D-Fender | Learn how telcos even work. Look at providers to see who allows RESALE of thier services in the first place |
| 17:00.22 | raypulver | and is it sensible to use a public STUN server as opposed to renting a server with two publicly routable IP addresses? |
| 17:00.31 | [TK]D-Fender | Learn how * itself work as far as support for calls, SMS, etc |
| 17:00.47 | [TK]D-Fender | Go learn the LEGAL implication of actting as a CLEC |
| 17:00.51 | raypulver | I bring up STUN because I had problems using my asterisk server from behind NAT in the past |
| 17:01.13 | [TK]D-Fender | You think you're going to run a telco on a single server? |
| 17:01.28 | raypulver | for very few clients |
| 17:01.51 | [TK]D-Fender | NAT problems are alsmost always basic configuration failures or using actively hostile routers that do ALG, port re-writes, etc |
| 17:02.08 | [TK]D-Fender | What's your idea of "a very few"? |
| 17:02.16 | raypulver | 3-5 |
| 17:03.25 | [TK]D-Fender | Ok, suitably perrty |
| 17:03.30 | [TK]D-Fender | petty* |
| 17:03.31 | [TK]D-Fender | gah |
| 17:03.37 | [TK]D-Fender | is just plain sloppy today... |
| 17:04.15 | raypulver | do I need anything besides a DID to setup * to send/receive SMS? |
| 17:04.21 | raypulver | I imagine I must |
| 17:04.32 | catphish | is there a standard way to do failover between asterisk servers by the way? |
| 17:05.27 | catphish | my usual mechanism of keepalived causes all kinds of problems with packets originating from the wrong IPs, so people tend to use an external loadbalancer? or is there a better way? |
| 17:07.41 | [TK]D-Fender | You need a provider that even supports it |
| 17:08.15 | [TK]D-Fender | Some use SIP Message as the means. Others use different API's |
| 17:08.22 | [TK]D-Fender | Time to go find a provider |
| 17:08.47 | catphish | between my providers, one uses HTTP for SMS, and one just refuse to bother to support it |
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| 17:12.01 | Voyage | [TK]D-Fender, correct. i dont know whats lcr |
| 17:12.57 | [TK]D-Fender | Well they describe what their service is.... |
| 17:13.16 | [TK]D-Fender | THEY act like the middle-man between you and multiple other providers and do the work FOR YOU |
| 17:13.30 | [TK]D-Fender | "LCR As A Service" just like they say |
| 17:13.47 | [TK]D-Fender | so they act as a SINGLE provider and do all the dirty work |
| 17:14.07 | [TK]D-Fender | Basically you hope they do their job properly and the people they pool through is actually good for you |
| 17:14.18 | [TK]D-Fender | And don't forget they get something off the top for what they offer |
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| 17:17.18 | [TK]D-Fender | heads out for a while |
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| 17:32.08 | Voyage | hm |
| 17:33.17 | Voyage | [TK]D-Fender, I see. GREAT help [TK]D-Fender . Thanks. you cleared me in few lines which I mean not have been clear off by reading a lot of goodle/wikis |
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| 19:48.50 | *** join/#asterisk gusto (~gusto@2001:4c50:62e:1413:8219:34ff:fecf:17f0) |
| 19:49.06 | gusto | hi |
| 19:49.16 | gusto | I have a smartphone and I installed csipsimple on it |
| 19:49.34 | gusto | somehow it has trouble authenticating with asterisk, it gives up on 401 unauthorized |
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| 19:53.26 | gusto | is 401 unauthorized the normal response, or is it somewhere blocked by ACLs? |
| 19:53.47 | gusto | because it does not even try to authenticate, it just says register and the answer is 401 unauthenticated |
| 19:53.53 | gusto | and then the jabber stops |
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| 20:05.21 | gusto | hm |
| 20:05.32 | gusto | after adjusting the realm it goes through but fails now on password |
| 20:05.56 | gusto | even though I am pretty sure that the password is correct, but I suspect another problem, because it looks like the formatting is different |
| 20:06.12 | gusto | he puts the response into the authorization |
| 20:08.08 | gusto | and now he succeeded because I adjusted the settings on csipsimple to plain password (which does not seem to be the plain password, but rather the plain password used to make the digest MD5 -> MD5-HMAC???) |
| 20:34.23 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 20:52.13 | gusto | that's cool |
| 20:52.30 | gusto | it looks like every LTE phone has IP calling integrated |
| 20:52.40 | gusto | so I did not even need the csipsimple |
| 20:53.11 | gusto | I could have gone over my regular phone app, but that one does not support TLS SIP transport, it only does UDP and TCP |
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| 21:22.31 | Voyage | [TK]D-Fender, I have installed * from ubuntu package manager and created a free flowroute account. Is there a tutorial you might want to recommend me from here ? (this is my first time and I know nothing else then some linux and my automated call requirement list) |
| 21:23.02 | WIMPy | ~book |
| 21:23.07 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:23.33 | Voyage | cant read full book of course. Just need to get started |
| 21:23.47 | Voyage | my requirements are not large |
| 21:23.53 | Voyage | or extensive |
| 21:23.55 | WIMPy | Then tray and fail. |
| 21:24.33 | WIMPy | Asterisk is not a simple and easy thing. |
| 21:47.24 | [TK]D-Fender | Well.. it is from a limited perspective |
| 21:47.36 | [TK]D-Fender | Voyage, You need to learn how SIP works. |
| 21:47.42 | [TK]D-Fender | And you need to learn how the dialplan works |
| 21:47.50 | [TK]D-Fender | 2 VERY specific simple bits |
| 21:48.01 | [TK]D-Fender | And spaters for each. |
| 21:48.19 | [TK]D-Fender | And a substantial amount of guides on each |
| 21:48.22 | [TK]D-Fender | these are BASICS |
| 21:49.01 | Voyage | [TK]D-Fender, ok. SIP and dialplan are the first steps after installing *? |
| 21:49.17 | [TK]D-Fender | yes |
| 21:49.20 | Voyage | ok |
| 21:49.23 | Voyage | will do |
| 21:49.29 | [TK]D-Fender | SIP because that is what you'll be using to talk to other things |
| 21:49.30 | catphish | understanding basic security is the first step, ie making sure you don't allow anonymous people to make calls |
| 21:49.36 | [TK]D-Fender | And All call processing = dialplan |
| 21:49.45 | catphish | but otherwise yes, dialplan and sip config, and contexts |
| 21:49.48 | Voyage | catphish, ok |
| 21:49.54 | Voyage | [TK]D-Fender, ok |
| 21:49.56 | catphish | dialplan is at the heart of everything |
| 21:50.00 | [TK]D-Fender | Security = controlling what calls are permitted to do. |
| 21:50.15 | Voyage | first will understand SIP, then dialpan (provided with security concepts like catphish said) |
| 21:50.29 | [TK]D-Fender | And preventing brute forcing of account info, etc |
| 21:50.37 | Voyage | [TK]D-Fender, I will just block on ip based (with credentials of course |
| 21:50.55 | Voyage | [TK]D-Fender, like ssh bruteforce? |
| 21:50.58 | [TK]D-Fender | you need just a tiny amount of SIP to learn how to setup a basic phone and a peer for Flowroute (they even GIVE you that poart) |
| 21:51.13 | Voyage | poart? |
| 21:51.17 | [TK]D-Fender | the rest is making sure that calls from each land in the right place and that you do the appropraite steps |
| 21:51.23 | [TK]D-Fender | PART |
| 21:51.31 | [TK]D-Fender | Please try to figure out the obvious typos |
| 21:51.38 | Voyage | ok |
| 21:51.56 | catphish | i'm currently trying to learn SIP properly, it's quite a mess at first glance |
| 21:52.04 | Voyage | I will find a youtube video for sip for now. then dialplan |
| 21:52.15 | Voyage | catphish, what sources? |
| 21:53.19 | WIMPy | catphish: It will get worst when you understand it. |
| 21:54.32 | [TK]D-Fender | Voyage, This is all text files. There will be little of value on youtube for this |
| 21:54.58 | [TK]D-Fender | Voyage, This is not like some gui where you get a visual walkthrough of "click here", then fill in this blank. |
| 21:55.05 | catphish | WIMPy: so i've heard :) |
| 21:55.36 | Voyage | [TK]D-Fender, talking about concepts first. will use some wiki for text files.. |
| 21:55.47 | [TK]D-Fender | it's ALL text-files |
| 21:55.50 | Voyage | [TK]D-Fender, I am very used to config files |
| 21:55.52 | [TK]D-Fender | SIp config |
| 21:55.55 | [TK]D-Fender | Dialplan, everything |
| 21:56.05 | Voyage | yup. I am a linux user though not a geek |
| 21:56.08 | [TK]D-Fender | the BOOK has a few simple guides already |
| 21:56.12 | Voyage | is getting concepts from the video |
| 21:56.24 | Voyage | [TK]D-Fender, ok. will see the book too |
| 21:56.29 | [TK]D-Fender | What video? |
| 21:56.35 | [TK]D-Fender | link it |
| 21:56.39 | [TK]D-Fender | I'm curious |
| 21:57.30 | Voyage | https://www.youtube.com/watch?v=n9cT1Lq9Lg8 |
| 21:57.45 | Voyage | https://www.youtube.com/watch?v=PgKc8DGIEzM |
| 21:57.51 | Voyage | https://www.youtube.com/watch?v=RWggp5zHop8 |
| 21:59.50 | [TK]D-Fender | <Voyage> https://www.youtube.com/watch?v=n9cT1Lq9Lg8 <- not bad for the basics to understand going from VoIP to PSTN |
| 22:01.00 | [TK]D-Fender | Voyage> https://www.youtube.com/watch?v=PgKc8DGIEzM <- shows a bit of the role of a PBX |
| 22:01.10 | Voyage | hm |
| 22:01.24 | Voyage | RWggp5zHop8 looks comprehensive |
| 22:01.30 | [TK]D-Fender | using SIP as a protocol. Kinda vague, but bit more technical detail of the comms themselves |
| 22:02.44 | Voyage | hm |
| 22:02.50 | Voyage | this looks promising for * https://www.youtube.com/watch?v=bgAX3HS-YE4 |
| 22:03.23 | [TK]D-Fender | <Voyage> https://www.youtube.com/watch?v=PgKc8DGIEzM <- hard to get usable info as a beginner on this |
| 22:04.00 | [TK]D-Fender | first 15 min is meh |
| 22:04.02 | Voyage | hm |
| 22:04.25 | [TK]D-Fender | and that 30 imn video is a "table of contents" to OTHER training. |
| 22:04.29 | [TK]D-Fender | it isn't even any detail |
| 22:05.47 | Voyage | hm |
| 22:05.51 | Voyage | will skim through all |
| 22:06.01 | Voyage | I think this will take a weak |
| 22:06.04 | [TK]D-Fender | seriously.. the book covers this just fine |
| 22:06.12 | Voyage | ok. then. |
| 22:06.21 | Voyage | will see the book in few hours too |
| 22:06.33 | [TK]D-Fender | start up on diaplan basics and you'll get samles for setting up a simple softphone to test with |
| 22:06.45 | [TK]D-Fender | Sip config for this is under 10 lines of config |
| 22:06.50 | Voyage | I just hate reading (its not I am ignore the valueable advice) but I know I have to read a lot of text |
| 22:07.01 | Voyage | hm ok |
| 22:07.04 | [TK]D-Fender | then you'll be able to throw calls at your server and start learning how to process them |
| 22:07.10 | Voyage | I see |
| 22:07.16 | [TK]D-Fender | A call from a softphone is no different then a call from an ITSP |
| 22:07.37 | [TK]D-Fender | They may have a few more settings to properly ID them, but that's it |
| 22:07.58 | Voyage | [TK]D-Fender, so the sip part is short and I need to dig more into dialplan? |
| 22:08.22 | [TK]D-Fender | and SIP thing you want to talk to is probabl no more than a dozen lines of SIP config |
| 22:08.31 | [TK]D-Fender | How you PROCESS your calls is EVERYTHING |
| 22:08.51 | [TK]D-Fender | "Yes, this is FRED, now what do I *DO* with the number he dialed?" |
| 22:09.12 | [TK]D-Fender | Authing the call is TINY |
| 22:09.35 | [TK]D-Fender | properly defining the networking settings you need to tell your config about is a single small thing |
| 22:09.46 | [TK]D-Fender | Once that is done you need to PROCESS CALLS |
| 22:10.02 | [TK]D-Fender | And that = dialplan = extensions.conf |
| 22:10.40 | [TK]D-Fender | Think about an office PBX. You'd need menus. You'd need it so that "users" (people) can call other people. Maybe voicemail. Maybe calls from the outside need to have "menus" |
| 22:10.49 | [TK]D-Fender | all of that stuff = dialplan |
| 22:10.57 | [TK]D-Fender | 'therefor that's where all the real work is |
| 22:11.16 | [TK]D-Fender | https://www.youtube.com/watch?v=V1T9lHHuvM8 <- not a bad video really |
| 22:11.31 | [TK]D-Fender | I'd probably go through those as well |
| 22:12.41 | Voyage | hm |
| 22:13.17 | catphish | sip's "branch" is the part that's confusing me the most |
| 22:14.31 | Voyage | [TK]D-Fender, https://www.youtube.com/watch?v=6SBP43ecq3A |
| 22:14.40 | Voyage | <PROTECTED> |
| 22:14.47 | Voyage | Official Asterisk YouTube Channel\ |
| 22:16.38 | [TK]D-Fender | Good beginner video |
| 22:17.39 | Voyage | ok. I must be working on them and the book now |
| 22:17.48 | Voyage | [TK]D-Fender, thanks :) once again |
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| 22:22.11 | catphish | ah no, i get branches now |
| 22:22.39 | [TK]D-Fender | "branch" as in? |
| 22:25.38 | catphish | sip transactions are identified by a bunch of parameters, one is "branch" |
| 22:26.56 | catphish | i think i get it, but gonna read 3 more times |
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