| 00:47.16 | *** join/#asterisk jcpeters02 (~jcpeters0@104.193.30.44) |
| 00:48.17 | jcpeters02 | hello everyone, I have a freePBX install and I am trying to figure out how to get my domain to be reflected in the TO field when setting up an outbound call. |
| 00:49.15 | Samot | What do you mean? |
| 00:49.32 | Samot | When an endpoint makes a call to the PBX? |
| 00:49.39 | Samot | Or when the PBX sends a call to the provider? |
| 00:50.02 | jcpeters02 | PBX sends call to provider |
| 00:50.07 | Samot | No. |
| 00:50.13 | Samot | Because that's not how it works. |
| 00:50.20 | Samot | TO is for where you are sending the call TO |
| 00:50.24 | Samot | Hence the name, TO |
| 00:50.34 | Samot | So the TO field would have the provider's IP/domain. |
| 00:50.41 | Samot | Your domain would be in the FROM field... |
| 00:50.51 | Samot | Because that's where the call is, well, from. |
| 00:50.59 | jcpeters02 | Called SIP/BluIP/1502XXXXXXX@502YYYYYYYY (but the 502YYYYYYY needs to be mydomain.com) |
| 00:51.07 | Samot | No. |
| 00:51.08 | Oeaa | what |
| 00:51.13 | Samot | That's not how that works. |
| 00:51.16 | Oeaa | that doesnt make sense |
| 00:51.19 | Samot | You are the FROM domain. |
| 00:51.26 | Samot | Because the call is coming FROM you. |
| 00:51.33 | jcpeters02 | I have to specify the provider domain |
| 00:51.33 | Samot | The TO domain is where the call goes. |
| 00:51.37 | jcpeters02 | yes |
| 00:51.40 | Samot | OK |
| 00:51.48 | jcpeters02 | I need to send my calls to number@domain.com |
| 00:51.48 | Samot | So change the host name to be the domain and not an IP |
| 00:52.02 | Samot | In the trunk. |
| 00:52.52 | Oeaa | send yoru calls to number@domain.com? |
| 00:52.55 | Oeaa | i am so confused |
| 00:53.01 | Samot | Yes. |
| 00:53.10 | Samot | It's called the TO-URI |
| 00:53.21 | Samot | All TO are number@domain |
| 00:53.26 | Samot | That's how it works. |
| 00:53.46 | Samot | More exactly it's user@domain as the SIP user doesn't have to be a number. |
| 00:54.08 | Samot | If you have a trunk with Flowroute the Host is sip.flowroute.com |
| 00:54.27 | Samot | So when you send calls to them it's 1NXXNXXXXXX@sip.flowroute.com <-- That's the TO header. |
| 00:56.02 | jcpeters02 | ok, I have replaced the host name and it still wants to send the call to @number |
| 00:56.42 | jcpeters02 | i have domain, fromdomain and realm all in outbound trunk settings |
| 00:57.03 | jcpeters02 | I live in Broadsoft hosted, don't do much with SIP Trunking |
| 00:57.03 | Samot | This a PJSIP trunk? |
| 00:57.08 | jcpeters02 | just SIP |
| 00:57.28 | Samot | Broadsoft does a lot of SIP trunking. |
| 00:57.29 | Samot | OK |
| 00:57.36 | Samot | ~pb |
| 00:57.36 | infobot | somebody said pastebin was 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: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 00:57.48 | Samot | Pastebin what you have your your chan_sip trunk settings. |
| 01:00.52 | jcpeters02 | Samot, I think I understood what you are asking for. http://paste.lisp.org/display/359365 |
| 01:01.35 | Samot | host=techconnexgroup.com <-- This can't be right. |
| 01:01.50 | Samot | No one uses the general root domain for sip routing. |
| 01:01.57 | Samot | First, it's a no-non |
| 01:02.05 | Samot | First, it's a no-no |
| 01:02.06 | jcpeters02 | My SIP Domain on Broadsoft is that |
| 01:02.29 | Samot | So you have SIP records for that |
| 01:02.37 | jcpeters02 | on a different address |
| 01:02.41 | Samot | SRV/NAPTR? |
| 01:03.08 | jcpeters02 | yes, before you suggested changing the host it was something else |
| 01:03.13 | Samot | OK does that domain only resolve to the broadsoft server? |
| 01:03.20 | Samot | What was it before? |
| 01:03.43 | jcpeters02 | ord-iad3.masteraccess.com |
| 01:03.58 | Samot | OK. |
| 01:04.06 | Samot | Who is the other domain? |
| 01:04.27 | jcpeters02 | the techconnexgroup.com? |
| 01:04.31 | Samot | Yes. |
| 01:04.34 | jcpeters02 | that is mine |
| 01:04.38 | Samot | OK. |
| 01:04.39 | Samot | So.. |
| 01:04.42 | Samot | Again.. |
| 01:04.48 | Samot | The host= is WHERE THE CALLS GO |
| 01:04.53 | Samot | You are not where the calls go |
| 01:04.59 | Samot | They do not go to your web server. |
| 01:05.02 | jcpeters02 | that is what I was thinking too. |
| 01:05.06 | Samot | OK. |
| 01:05.19 | Samot | I never said put that domain in that field. |
| 01:05.22 | Samot | At all. |
| 01:05.27 | jcpeters02 | I misunderstood you then, my apologies |
| 01:05.39 | Samot | Put it back the other way. |
| 01:05.56 | Samot | That other domain is the SIP server. |
| 01:06.15 | jcpeters02 | yes, ok, done |
| 01:06.31 | Samot | Now the provider wants what, exactly? |
| 01:06.45 | Samot | Is this documented somewhere? |
| 01:06.51 | jcpeters02 | not very well |
| 01:06.54 | Samot | That you can pastebin and show me the exact words. |
| 01:08.11 | jcpeters02 | Samot, refresh that pastebin |
| 01:09.07 | Samot | How did you get so different? |
| 01:10.44 | Samot | ord-iad3.masteraccess.com <-- This isn't even listed anywhere in that update |
| 01:10.51 | Samot | Like... |
| 01:10.58 | Samot | Your trunk is completely fubar'd. |
| 01:11.15 | Samot | Your config is grossly different. |
| 01:12.17 | jcpeters02 | yes, that is an example |
| 01:12.30 | jcpeters02 | that is not what I use exactly |
| 01:12.30 | Samot | OK, where is your data? |
| 01:12.56 | Samot | What did they give you as your data to put in those settings? |
| 01:13.11 | Samot | Even if those settings are generic.. |
| 01:13.25 | Samot | The values, your trunk has crap they don't have in their config example. |
| 01:13.53 | jcpeters02 | I provided you in the pastebin, the details I got |
| 01:14.06 | Samot | So the details you appended.. |
| 01:14.13 | Samot | Are *your* account details? |
| 01:14.22 | jcpeters02 | yes |
| 01:14.26 | Samot | OK |
| 01:14.27 | jcpeters02 | thats really all they provide |
| 01:14.37 | Samot | The second one? |
| 01:14.46 | Samot | The update you added to pastebin, right? |
| 01:15.05 | jcpeters02 | yes |
| 01:15.18 | Samot | OK, so if I fix it then I have the right settings. |
| 01:15.50 | Samot | https://www.irccloud.com/pastebin/imt2HRwA/ |
| 01:16.16 | Samot | OK, step 1) In the OUTGOING section of the trunk. Replace what you have with that. |
| 01:16.43 | Samot | Step 2) Go into the INCOMING tab.. |
| 01:17.05 | Samot | Where it says CONTEXT you put 7025550110 |
| 01:17.19 | Samot | You do not put anything in the peer/config box.. |
| 01:17.23 | Samot | That should be blank. |
| 01:17.37 | Samot | The last field, Register String you put: 7025550110@bluipdemo.com:1Q2W3E:7025550110@laxiad.masteraccess.com/7025550110 |
| 01:18.15 | jcpeters02 | Ok, one second |
| 01:18.15 | Samot | OH |
| 01:18.23 | Samot | If you're using FreePBX.. |
| 01:18.31 | Samot | Remove [BLUIP] from that |
| 01:18.38 | Samot | BLUIP is the "Trunk Name" |
| 01:18.52 | jcpeters02 | yes |
| 01:19.05 | Samot | Everything below [BLUIP] goes in the PEER Details. |
| 01:19.15 | Samot | User Context is your username |
| 01:19.27 | Samot | USER Details = nothing/blank |
| 01:19.34 | Samot | Register String is the string |
| 01:19.39 | jcpeters02 | ok, thats how I have it now |
| 01:19.49 | Samot | OK.. |
| 01:19.53 | jcpeters02 | I can receive inbound calls just fine |
| 01:19.58 | Samot | OK |
| 01:20.05 | Samot | Log via SSH |
| 01:20.06 | jcpeters02 | I just cant get an outbound call to formulate correctly |
| 01:20.17 | Samot | asterisk -rvvvvvvvvv |
| 01:20.21 | Samot | sip set debug on |
| 01:20.27 | Samot | make an outbound call |
| 01:20.44 | Samot | pastebin the output from the moment you enter the Asterisk CLI to the hangup. |
| 01:28.35 | jcpeters02 | I updated the pastebin, but I cant make inbound calls anymore |
| 01:29.16 | jcpeters02 | let me try that again |
| 01:29.21 | jcpeters02 | I cannot receive calls anymore |
| 01:29.45 | Samot | Are you registered? |
| 01:30.03 | Samot | outbound calls don't require registration, just proper auth. |
| 01:30.07 | jcpeters02 | sip show registry says I am |
| 01:30.16 | Samot | OK |
| 01:30.20 | Samot | Run the debug again. |
| 01:30.22 | Samot | Make an inbound call |
| 01:30.26 | Samot | Show the results of that |
| 01:30.41 | Samot | Let's confirm if you're actually receiving the call. |
| 01:30.59 | Samot | But that outbound call went out |
| 01:32.03 | jcpeters02 | I got a switch intercept saying the number was not in service. It is, it is my cell phone |
| 01:32.49 | jcpeters02 | The paste has been annotated |
| 01:33.05 | Samot | bluipdemo.com <-- Is that what that should be? |
| 01:33.12 | Samot | Or should *that* be your domain? |
| 01:33.18 | jcpeters02 | that should be my domain |
| 01:33.24 | Samot | Then make it your domain. |
| 01:33.29 | jcpeters02 | let me paste what I actually have there |
| 01:33.32 | Samot | So your register string is messed up. |
| 01:33.49 | Samot | I asked if what was in that pasbebin was your details so I could fix your trunk. |
| 01:35.28 | jcpeters02 | the registration was not in that, but when you told me to update the trunk, I was taking the information you had and I populated my register key as 5022421853@techconnexgroup.com:XXXXXX:5022421853@ord-iad3.masteraccess.com/5022421853 |
| 01:36.23 | jcpeters02 | I am closer to the ORD pop, but their examples use the LAX POP |
| 01:36.28 | Samot | OK hang on a second. |
| 01:36.39 | Samot | this is why I hate single pastebin updates. |
| 01:36.44 | Samot | I missed the OB proxy settings. |
| 01:37.23 | Samot | OK, one second. |
| 01:39.58 | Samot | https://www.irccloud.com/pastebin/3N1Gw1UX/ |
| 01:40.56 | Samot | 5022421853@techconnexgroup.com:1Q2W3E:5022421853@ord-iad3.masteraccess.com/5022421853 |
| 01:41.15 | Samot | USER Conext is 5022421853 |
| 01:41.21 | Samot | USER Context is 5022421853 |
| 01:42.52 | jcpeters02 | I dont think that the example is complete that they provide. |
| 01:43.20 | jcpeters02 | I can pass traffic to the switch, as I get a switch intercept now, but it still is not happy |
| 01:43.45 | jcpeters02 | and I cant receive calls |
| 01:44.50 | Samot | OK, without having the full details of what y our settings are.. |
| 01:44.59 | Samot | And what they need in the trunk settings... |
| 01:45.15 | Samot | I've run broadsoft systems with Asterisk endpoints. |
| 01:45.33 | Samot | Like 1.8 Asterisk, I know it works. |
| 01:45.46 | jcpeters02 | I know, but when I make a call, I am passing 5021234567%405022422421853 |
| 01:46.09 | jcpeters02 | oops |
| 01:46.16 | Samot | Well I don't know if you're adding it |
| 01:46.20 | jcpeters02 | 5021234567%405022421853 |
| 01:46.23 | Samot | I asked for asterisk -rvvvvvvvvv as well |
| 01:46.36 | Samot | Which shows how the call is going through the dialplan. |
| 01:46.40 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 01:46.45 | jcpeters02 | One second |
| 01:46.50 | jcpeters02 | I did omit that |
| 01:46.52 | Samot | And can see if you're adding crap to the digits. |
| 01:48.25 | jcpeters02 | Ok, I updated the pastebin |
| 01:51.36 | jcpeters02 | and I fixed my inbound with insecure=port,invite |
| 01:52.42 | Samot | <PROTECTED> |
| 01:52.54 | Samot | -- Called SIP/BluIP/15027166274@5022421853 |
| 01:53.15 | Samot | So you're using a PJSIP extension, correct? |
| 01:53.19 | jcpeters02 | Yes |
| 01:53.29 | Samot | And the trunk is Chan_SIP |
| 01:53.33 | jcpeters02 | Yes |
| 01:53.44 | Samot | So this looks like a bug in FreePBX |
| 01:53.48 | jcpeters02 | I have another trunk on the system and it works. |
| 01:54.04 | Samot | What type of trunk? |
| 01:54.08 | jcpeters02 | SIP |
| 01:54.14 | jcpeters02 | I can spin up a SIP endpoint |
| 01:54.33 | Samot | SIP/BluIP/15027166274@5022421853,300,T <-- That is wrong. |
| 01:54.55 | Samot | SIP/BluIP/15027166274,300,T <-- That is how it should be dialing the Chan_SIP Trunk. |
| 01:55.17 | Samot | PJSIP uses @ for Dialing contacts. |
| 01:57.26 | jcpeters02 | OK |
| 01:57.36 | Samot | Yeah for S&G spin up a Chan_SIP extension and try using that trunk. |
| 01:58.06 | Samot | And yeah, your inbound was on me. I forgot the insecure setting. |
| 02:02.35 | jcpeters02 | No biggie, I just appreciate you taking the time. |
| 02:06.20 | jcpeters02 | Same thing on a SIP Extension too |
| 02:06.47 | jcpeters02 | So, I wonder if this issue goes away if it is setup as a pjsip trunk |
| 02:07.43 | Samot | I'm curious as to why it's malforming the dial string for a Chan_SIP channel. |
| 02:08.25 | Samot | Doing SIP/peer/destination@something |
| 02:08.31 | Samot | Malforms the TO string. |
| 02:08.47 | Samot | Which I think is making the call go out weird. |
| 02:09.02 | jcpeters02 | This is a FreePBX Distribution, I have not done any system updates yet |
| 02:09.12 | Samot | Well, I do the updates. |
| 02:09.17 | Samot | And cross your fingers. |
| 02:09.24 | jcpeters02 | OK :) |
| 02:09.26 | Samot | Do the module updates |
| 02:09.33 | Samot | Those impact the dialplan. |
| 02:09.52 | Samot | So it could fix how it generates the Dial() string. |
| 02:15.30 | *** join/#asterisk iheartlinux (~jwpierce3@mail.trunkmasters.com) |
| 02:49.49 | jcpeters02 | Samot, if you are interested I setup that trunk with pjsip and it works |
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| 10:44.25 | slima | Hello, Please, hint me. I need call -> IVR for some press '1' (I can do it) and If user press '1' ring two lines (can do) but when they are busy Playback a message 'All lines are busy now, please hold' (can do); but what if are 3.. 4.. 5.. more 'queue' is on line, And I like to do something like: 'All lines are busy now, you are 2 on queue'. How to do that? |
| 10:47.42 | file | You configure a queue and use the Queue application generally |
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| 10:50.07 | slima | thanks file. |
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| 11:26.51 | slima | file: okey, but there is a way to play only one file with message 'you are sec on the line...' not 3 messages; queue-thereare.gsm,digits/2.gsm,queue-callswaiting.gsm ? |
| 11:27.39 | file | Um, no. You'd need a single message for every possible position. |
| 11:28.05 | file | The 3 sound files together do that. |
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| 11:32.32 | slima | I have 10 messages, for 10 positions, but I see, asterisk by default play 3 separated messages. |
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| 11:42.41 | file | yes, because it uses different sound files to construct the sentence |
| 11:48.47 | slima | So, my question is: is there a way to define single file for each queue position? |
| 11:48.52 | TandyUK | "you are at position" "three" "in the queue" for example |
| 11:49.22 | TandyUK | why would you want to do that? |
| 11:50.05 | TandyUK | "you are at position" $queuepos "in the queue" works whether there are 1 or 1000 people in the queue |
| 11:50.34 | TandyUK | your way, you would need 1000 different files recorded |
| 11:51.40 | slima | nope, I have announce-position-limit = 10, But it is not my question. |
| 11:53.21 | file | unless there is a config file option, there isn't a way |
| 11:53.24 | file | you'd have to modify the source code |
| 11:55.06 | slima | okey, last question, How to use different sound path for 'digits/2.gsm' etc. in queue? |
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| 12:16.03 | [TK]D-Fender | slima, vi app_queue.c |
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| 13:30.42 | pawiecki | Very subjective matter, but do you use tabs or spaces in asterisk's conf files to separate code? I've found that my coworker is very sensitive on that matter, but i'm not :) |
| 13:30.56 | Samot | No. |
| 13:30.59 | Samot | You do not. |
| 13:31.09 | pawiecki | I do not, but I'm forced to, apparently |
| 13:31.27 | Samot | Asterisk shouldn't have spaces between things. |
| 13:31.36 | Samot | Just as a good measure. |
| 13:31.52 | Samot | It should not be host = 1.1.1.1 |
| 13:31.56 | Samot | It's host=1.1.1.1 |
| 13:32.37 | pawiecki | Samot: hmm, that's more restrictive. In sample files I've found both examples |
| 13:32.49 | Samot | Unwanted spaces and formatting in the dialplan can break things. |
| 13:32.56 | Samot | Yes, you can.. |
| 13:33.06 | Samot | But there are points where unwanted spaces can break things. |
| 13:33.35 | Samot | Or tabs or anything.. |
| 13:34.04 | pawiecki | For me, the "no unnecessary spacec and tabs" approach is better, and paradoxically it's more readable. |
| 13:34.14 | pawiecki | spaces* |
| 13:34.32 | [TK]D-Fender | This is code, not an artistic writing competition |
| 13:34.46 | [TK]D-Fender | Back away from the space bar Picasso |
| 13:34.51 | Samot | Well.. |
| 13:34.57 | Samot | There are standards for code. |
| 13:35.02 | Samot | Formatting, etc. |
| 13:35.11 | Samot | Depending on the language of the code. |
| 13:35.19 | pawiecki | Samot: we don't have one internally, so it's difficult to discuss it. |
| 13:35.38 | Samot | Well when I say standards I mean |
| 13:35.45 | Samot | PSR |
| 13:36.08 | Samot | if ($x = $y) is not the same as if($x=$y) |
| 13:36.10 | davidbowlby | tabs > * |
| 13:36.16 | Samot | First = good, second = bad |
| 13:36.23 | Samot | In the format standards world. |
| 13:36.39 | Samot | However, the first can break stuff in dialplan |
| 13:36.48 | Samot | because of spaces where it doesn't think spaces should be. |
| 13:43.23 | pawiecki | what about: if("$x" = "$y") |
| 13:43.47 | Samot | Well, it depends |
| 13:43.55 | Samot | Generally, quotes aren't need for variables. |
| 13:44.01 | Samot | But Asterisk is old school code. |
| 13:44.14 | fauxalliance | like regex with dimensia |
| 13:44.17 | Samot | Coding style that really doesn't exist anymore. |
| 13:45.07 | Samot | If you were to break if or while statements up into multi-line formatting like in most code in Asterisk dialplan.. |
| 13:45.15 | Samot | Your dialplan will be broken to all hell. |
| 13:45.30 | pawiecki | Samot: quote might help if you compare non-empty variable to an empty one. |
| 13:45.45 | Samot | In what |
| 13:45.49 | Samot | General? No. |
| 13:46.11 | Samot | There are functions for checking non-empty variables in code. |
| 13:46.46 | pawiecki | GotoIf($[ $a > $b ]?yes:no) - if one is empty, it does not have a type, so it breaks |
| 13:46.58 | pawiecki | forgot {} |
| 13:47.18 | Samot | OK. |
| 13:47.33 | Samot | No, one is not "empty" |
| 13:47.35 | pawiecki | I don't mean general programming, just the usual * config files |
| 13:47.37 | Samot | One is not set |
| 13:47.42 | Samot | Not setting $a |
| 13:47.45 | pawiecki | Samot: yes |
| 13:47.54 | Samot | IS not the same as Set(a=) |
| 13:48.05 | pawiecki | correct |
| 13:48.21 | Samot | So there is a different between empty and non-existent. |
| 13:48.28 | Samot | difference |
| 13:48.49 | pawiecki | that's what I meant, I just wasn't precise, sorry. |
| 13:48.57 | Samot | Well |
| 13:49.12 | Samot | Now you've entered a realm where being precise matters. |
| 13:49.23 | fauxalliance | words are minced, grammar is enforced and the sarcasm is always implied... |
| 13:50.54 | fauxalliance | heads back to writting 'witty' PERL oneliners |
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| 14:22.41 | pawiecki | Samot: by PSR did you mean this: https://en.wikipedia.org/wiki/PHP_Standard_Recommendation ? |
| 14:24.30 | Samot | Right. |
| 14:24.43 | Samot | I was just using that as an example |
| 14:25.24 | Samot | Projects/companies, etc will expressly say or want their code in PSR format. |
| 14:25.36 | Samot | Because that means if I leave.. |
| 14:25.46 | Samot | The next guy can come in and look at my code and understand it. |
| 14:25.58 | Samot | Follow the logic. |
| 14:26.15 | Samot | And part of the PSR is how and when to leave comments in the code. |
| 14:28.30 | Samot | Trying to follow guidelines like that in Asterisk dialplan isn't going to work. |
| 14:28.39 | Samot | I mean the little things like "80 character lines" |
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| 14:29.45 | Samot | Also things like Dial() or GoSub() that take arguments to pass do not like spaces between the commas and the argument values. |
| 14:30.07 | Samot | ARG1, ARG2, ARG3 <--fail |
| 14:30.18 | Samot | ARG1,ARG2,ARG3 <-- winning |
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| 16:48.49 | kadosh | Hello, I've been googling but not at actual answer |
| 16:49.04 | fauxalliance | ?ask @ kadosh |
| 16:49.08 | kadosh | my issue is that my installation of asterisk doesn't have chan_sip.so |
| 16:49.21 | kadosh | It's a new installation from source |
| 16:49.24 | fauxalliance | load it |
| 16:49.32 | fauxalliance | module load ... |
| 16:49.43 | kadosh | sorry im pretty new... gonna try that |
| 16:49.52 | fauxalliance | asterisk -r |
| 16:50.06 | fauxalliance | the module load TAB (completion) |
| 16:50.14 | kadosh | awesome :) |
| 16:50.18 | kadosh | yes.. it worked |
| 16:50.21 | fauxalliance | cool |
| 16:50.24 | kadosh | I thought was loaded by default |
| 16:50.26 | kadosh | thanks |
| 16:50.30 | fauxalliance | no sweat |
| 16:52.29 | [TK]D-Fender | it is loaded... if your config tells it to load modules |
| 16:52.37 | [TK]D-Fender | So far I'm suspecting you don't have proper configs |
| 16:53.41 | fauxalliance | i was just about to mention that hand compiled / coded asterisk is probably not the best spot to start..â[14:19] â<âkadoshâ>â sorry im pretty new... |
| 16:54.31 | fauxalliance | but there is a config file that controls modules at runtime... find it and modifiy it to suit your personal module needs |
| 16:54.33 | fauxalliance | ~book |
| 16:54.33 | 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 |
| 16:54.36 | fauxalliance | is a good read |
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| 17:47.32 | jamesaxl | hello |
| 17:48.48 | jamesaxl | is a vpn connection enough to connect 2 locals for using asterisk? |
| 17:50.49 | jamesaxl | for example I have local A in Paris and local B in Tokyo, I installed asterisk in Paris, and I want people in Tokya call from asterisk in Paris |
| 17:53.06 | [TK]D-Fender | VPN is jsut another path for packets to flow |
| 17:53.55 | [TK]D-Fender | It can add security for those calls, but otherwise the word "enough" doesn't mean anything |
| 17:54.09 | [TK]D-Fender | what is "enough" is any means of routing the packets from A to B intact |
| 17:54.47 | fauxalliance | the VPN revfers to the TCP/IP flow... you could use satellites, or radio, or a modem... it's irrelevant. |
| 17:55.22 | fauxalliance | you need a low latency and failry reliable connection for VoIP... |
| 17:55.30 | fauxalliance | latency counts... |
| 17:55.33 | jamesaxl | [TK]D-Fender: I meant, without using other devices |
| 17:55.47 | [TK]D-Fender | jamesaxl, Your framing still doesn't mean anything |
| 17:56.13 | fauxalliance | jamesaxl: the public internet is 'enough' |
| 17:56.19 | [TK]D-Fender | <jamesaxl> is a vpn connection enough to connect 2 locals for using asterisk? <- I don't need VPN to connect multiple * |
| 17:56.36 | [TK]D-Fender | I just need SOME networking between them to pass calls via VoIP |
| 17:56.57 | fauxalliance | eyes Codec2 and ax.25 |
| 17:57.02 | [TK]D-Fender | using VPN to make a secure tunnel between the sites is a BONUS but doesn't do anything else magical. |
| 17:57.07 | [TK]D-Fender | It is not required. |
| 17:57.13 | [TK]D-Fender | But it could be DESIRABLE |
| 17:57.18 | jamesaxl | [TK]D-Fender: Maybe I did not explain very well |
| 17:57.22 | [TK]D-Fender | Understand the difference |
| 17:57.36 | fauxalliance | jamesaxl: you clearly did not explain anything. |
| 17:58.31 | [TK]D-Fender | <jamesaxl> for example I have local A in Paris and local B in Tokyo, I installed asterisk in Paris, and I want people in Tokya call from asterisk in Paris <- * doesn't care what the path between the 2 looks like |
| 17:58.38 | fauxalliance | <haiku> TCP/IP, Learn how it fits together, There is no escape"</haiku> |
| 17:58.45 | [TK]D-Fender | Could you be using VPN? Yes. Do you need it it for * to pass calls? No. |
| 17:58.48 | [TK]D-Fender | do you WANT it? |
| 17:58.51 | fauxalliance | YES |
| 17:59.05 | fauxalliance | MSRP only $14,995 |
| 17:59.16 | fauxalliance | Today ONLY, FREE |
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| 18:00.48 | jamesaxl | I made a tunnel with openvpn between Paris and Tokyo (for a secure connection), I would like to know, should install a device, if yes in which side? should I install 2nd asterisk , and register the 1st asterisk like a provider |
| 18:01.04 | fauxalliance | meh, use IAX/2 |
| 18:01.14 | fauxalliance | then SIP/ PJSIP on the endpoints |
| 18:01.16 | [TK]D-Fender | jamesaxl, I still don't get your question at all |
| 18:01.23 | fauxalliance | me neither. |
| 18:01.26 | [TK]D-Fender | Install WHT device? WHY? |
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| 18:01.49 | jamesaxl | [TK]D-Fender: Mcro-litic |
| 18:01.55 | [TK]D-Fender | That's a ROUTER |
| 18:02.07 | [TK]D-Fender | Learn networking. |
| 18:02.14 | fauxalliance | comme jait adit on bon francis, shoot me in the face. |
| 18:02.37 | [TK]D-Fender | Should you have a ROUTER? |
| 18:02.39 | [TK]D-Fender | MAYBE |
| 18:02.44 | [TK]D-Fender | Do you see a reason to get one? |
| 18:02.53 | jamesaxl | [TK]D-Fender: I understand about networking. But I do not know how to explain :) |
| 18:03.01 | [TK]D-Fender | Do you understand what you are expecting it to DO? |
| 18:03.10 | [TK]D-Fender | Have you consdiered OTHER methods? |
| 18:03.28 | jamesaxl | [TK]D-Fender: reason if I have more the 5 places that I have to add |
| 18:03.36 | jamesaxl | then* |
| 18:03.57 | jamesaxl | [TK]D-Fender: USA, Canada, etc |
| 18:04.09 | [TK]D-Fender | What does quantity have to do with anything? |
| 18:04.14 | [TK]D-Fender | Why does QUANTITY matter? |
| 18:04.59 | [TK]D-Fender | You are not describing a NEED or how anything is expected to FILL it, or what alternatives you are considering. |
| 18:05.18 | [TK]D-Fender | You also haven't described what each side HAS to begin with |
| 18:05.41 | [TK]D-Fender | You have not described anything even vaguely close to a coherent picture |
| 18:06.37 | jamesaxl | [TK]D-Fender: call centers of course, connecting two groups that are not belong to the same Area |
| 18:06.55 | [TK]D-Fender | You are talking in broken pieces |
| 18:07.14 | fauxalliance | bridging the gap between tagolog / hindi one ACD call at a time. |
| 18:07.19 | fauxalliance | ^^call centres |
| 18:08.15 | jamesaxl | [TK]D-Fender: I need good quality between areas that is all |
| 18:10.28 | jamesaxl | [TK]D-Fender: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html I read this , but I asked, maybe someone has experience with |
| 18:10.46 | fauxalliance | IAX/2 over VPN |
| 18:10.49 | fauxalliance | done |
| 18:10.52 | jamesaxl | fauxalliance: la france STP silence. |
| 18:11.12 | [TK]D-Fender | <jamesaxl> [TK]D-Fender: I need good quality between areas that is all <- VPN has NO benefit for "quality" |
| 18:11.25 | fauxalliance | nor does extra routers |
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| 18:11.56 | jamesaxl | [TK]D-Fender: then, the speed of internet that I will need |
| 18:12.17 | [TK]D-Fender | again, that isn't even a complete sentence |
| 18:12.22 | fauxalliance | STP? |
| 18:12.26 | fauxalliance | oh |
| 18:12.28 | fauxalliance | derp |
| 18:12.39 | fauxalliance | no one is ever polite with me. |
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| 18:13.11 | [TK]D-Fender | I love Stone Temple Pilots, but they aren't here now... |
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| 18:13.41 | jamesaxl | [TK]D-Fender: I do not want have trouble with calls, then I should just increase internet speed |
| 18:13.49 | fauxalliance | NO |
| 18:13.54 | fauxalliance | REDUCE LATENCY |
| 18:14.09 | fauxalliance | speed is barely relevent. |
| 18:14.33 | fauxalliance | geschwindigkeitsbegrenzung |
| 18:14.35 | fauxalliance | ! |
| 18:19.40 | jamesaxl | cresl1n: thank you |
| 18:20.18 | [TK]D-Fender | ... |
| 18:20.50 | fauxalliance | ...to be continued. |
| 18:21.41 | [TK]D-Fender | I made the mistake of thinking were were actually conversing. |
| 18:21.46 | [TK]D-Fender | I was apparently mistaken |
| 18:22.47 | fauxalliance | jeez, hard enough getting verbose log output, let alone a complete sentence in any language. |
| 18:22.52 | jamesaxl | [TK]D-Fender: I got what I need :) thank you for mocking me |
| 18:23.11 | fauxalliance | seul, lol. |
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| 18:27.30 | [TK]D-Fender | Or proper credit |
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| 19:05.31 | RovingWriter | https://pastebin.com/dAjHYLSR |
| 19:05.43 | RovingWriter | any ideas on this? worked before, now doesn't. |
| 19:10.55 | RovingWriter | nevermind. for some reason, it was owned by root, not asterisk |
| 19:11.20 | RovingWriter | worked before, don't know if it got changed after or whatever, but, thats the issue |
| 19:23.40 | fauxalliance | RovingWriter: you have a lock file holding you hostage |
| 19:24.14 | fauxalliance | possibly because of permissions issues |
| 19:24.31 | fauxalliance | need more descriptive example of the problem |
| 19:25.11 | fauxalliance | i.e. when i do X the log says Y,TF |
| 19:26.04 | fauxalliance | as a nurse, I will check the bowel movement for color and consistency, but, whats the fucking symptom. |
| 19:28.02 | fauxalliance | like the doc that order an ekg during / after every poop, then gave the patient a shit load of laxitives, and said Dr. soandso is on call.. Im out. |
| 19:28.04 | voipmonk | What's species of worm is inside the excrement |
| 19:28.24 | fauxalliance | playthelemthes |
| 19:29.36 | fauxalliance | eyes the wall of foreign bodies removed from chldren, and winks at darwin. |
| 19:30.29 | fauxalliance | sadly, poison control tells me more seniors with dementia have died from eating tide pods than children... |
| 19:30.38 | fauxalliance | 6:3 |
| 19:38.47 | jamesaxl | fauxalliance: have you managed to stream with asterisk using icecast ogg? |
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| 21:23.50 | Dovid | Hi. I seem to have this issue in 13.8. Any pointers? https://issues.asterisk.org/jira/browse/ASTERISK-17124 |
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| 22:02.18 | Samot | Get on the latest release of 13 |
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