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| 02:16.30 | bisco_ | can someone explain to me why the queues that i've added in queues.conf are not showing up in the web interface please? i've verified that they exist with >queue show |
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| 02:18.55 | bisco_ | i've also reloaded the module and even gone as far as restarting the server |
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| 02:21.03 | [TK]D-Fender | What "web interface"? |
| 02:22.09 | bisco_ | the asterisk admin interface (or freepbx interface???) |
| 02:22.41 | bisco_ | yeah...looking now should've probably said the freepbx interface...sorry |
| 02:22.44 | [TK]D-Fender | Idf you read those config file they explicitly tell you to NOT modify them by hand |
| 02:22.55 | [TK]D-Fender | thFreePBX does not read the config files... it WRITES THEM. |
| 02:22.59 | [TK]D-Fender | One-way relationship |
| 02:23.01 | bisco_ | queues.conf doesn't say that |
| 02:23.35 | [TK]D-Fender | You do not do this by hand and expect the GUI to take it in. TO the contrarary most of those files get blown away and regenerated based on GUI data when you hit "Apply" |
| 02:24.08 | bisco_ | is there any way to migrate configs when moving servers? |
| 02:24.24 | bisco_ | upgrading a super old system to a new machine with the latest release |
| 02:24.28 | [TK]D-Fender | FreePBX has a backup & restore feature |
| 02:24.36 | [TK]D-Fender | bothe servers have to be on the same version to do this |
| 02:25.26 | bisco_ | that's what i thought...the old server is running 11.12.0 and the new one is 13.7.1 |
| 02:27.45 | [TK]D-Fender | upgade the old |
| 02:27.59 | [TK]D-Fender | or reinstall the new one with an older version and work it up |
| 02:28.24 | bisco_ | hadn't thought about that...thanks much for the info...wish me luck |
| 02:28.45 | [TK]D-Fender | Don't forget the goat. |
| 02:28.51 | bisco_ | haha |
| 02:28.52 | [TK]D-Fender | Animal sacrifice my be required |
| 02:28.52 | bisco_ | thx |
| 02:29.00 | bisco_ | will bring two just in case :P |
| 02:29.00 | [TK]D-Fender | And check in with #freepbx |
| 02:29.04 | [TK]D-Fender | it isn't supported here |
| 02:29.08 | bisco_ | will do |
| 02:29.13 | bisco_ | cheers |
| 02:29.14 | [TK]D-Fender | #1 |
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| 02:30.32 | MajesticFudgie | Just remembered something |
| 02:31.08 | MajesticFudgie | Can people append an extension to my phone number when they're calling in from the telco |
| 02:31.22 | MajesticFudgie | Or would I have to handle that with extensions |
| 02:31.57 | ChannelZ | Extensions |
| 02:32.20 | MajesticFudgie | okie |
| 02:32.21 | ChannelZ | Or buy a separate DID |
| 02:32.22 | [TK]D-Fender | Depends what you mean by that\ |
| 02:32.25 | [TK]D-Fender | ^^^ |
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| 02:33.10 | MajesticFudgie | Well I'm with BT on a residential line, ideally someone inputs my number I have with BT followed by say 6002 and it'd hit an extension for 6002 |
| 02:33.23 | [TK]D-Fender | what KIND of line? |
| 02:33.29 | MajesticFudgie | I'm going to assume even if that's possible, residential lines aren't going to like it |
| 02:33.34 | MajesticFudgie | What kinds are there? |
| 02:33.39 | [TK]D-Fender | "residential" isn't a kind |
| 02:33.52 | MajesticFudgie | What kinds are there? |
| 02:33.55 | [TK]D-Fender | what TECHNOLOGY? What SIGNALING? |
| 02:34.14 | MajesticFudgie | Not entirely sure, sec |
| 02:34.43 | [TK]D-Fender | Europe has both ANALOG and ISDN-BRI popularly. |
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| 02:35.16 | MajesticFudgie | I'm in the UK, idk if the UK sticks to one primarily for residential customers |
| 02:35.21 | MajesticFudgie | or one for everyone |
| 02:35.54 | MajesticFudgie | I have no clue how to check |
| 02:36.03 | [TK]D-Fender | It's either dumb analog... or digital ISDN-BRI |
| 02:36.28 | MajesticFudgie | Possibly digital? My router/modem connects through the same port using a split of some sort |
| 02:36.43 | MajesticFudgie | and I get pretty good speeds, So I'd assume there's some digital stuff going on |
| 02:36.47 | [TK]D-Fender | So far odds on you using ADSL over ANALOG lines |
| 02:37.07 | [TK]D-Fender | ADSL is a digital signal, but the VOICE service on the underlying line would be ANALOG |
| 02:37.10 | MajesticFudgie | Best way to check what I have? |
| 02:37.15 | MajesticFudgie | yeah |
| 02:37.41 | [TK]D-Fender | Unless its a DRY LINE which only carries ADSL and you ctually have a modem device that acts as a VoIP ATA |
| 02:37.56 | [TK]D-Fender | Best way is to have a clue what you have and how it's wired. |
| 02:38.10 | [TK]D-Fender | Do you have phones plugged in PARALLEL with your modem? |
| 02:38.14 | [TK]D-Fender | Or are they all BEHIND it? |
| 02:38.23 | [TK]D-Fender | What does the MANUAL for it say? |
| 02:38.31 | [TK]D-Fender | You need to get a clue about how phones work where you are |
| 02:38.34 | [TK]D-Fender | and what you have. |
| 02:38.40 | [TK]D-Fender | and that means .... |
| 02:38.46 | [TK]D-Fender | <PROTECTED> |
| 02:38.49 | MajesticFudgie | In parallel, |
| 02:39.00 | MajesticFudgie | Both go into a little plastic splitter into the wall socket |
| 02:39.09 | [TK]D-Fender | Ream them. Learn what you have and what thaat means. Look at your phone bill. Your contract. telco website. |
| 02:39.10 | [TK]D-Fender | Google |
| 02:39.30 | [TK]D-Fender | So far I'd be putting money on "analog" |
| 02:39.33 | [TK]D-Fender | Go prove it. |
| 02:39.51 | [TK]D-Fender | Do you not know if you are holding an analog handset in your hands VS an ISDN-BRI? |
| 02:39.53 | [TK]D-Fender | o look. |
| 02:39.57 | MajesticFudgie | Trying to, Their website seems to be lacking in technical specs |
| 02:40.00 | [TK]D-Fender | Your phone also has a manual no doubt |
| 02:40.09 | [TK]D-Fender | how about ANY of these? |
| 02:40.18 | MajesticFudgie | I lack manuals for them |
| 02:40.35 | [TK]D-Fender | take a PICTURE and show us. |
| 02:40.53 | [TK]D-Fender | with the PLUG obviously |
| 02:40.54 | MajesticFudgie | Router wise I use https://en.wikipedia.org/wiki/BT_Home_Hub I'll check the phones model |
| 02:41.05 | MajesticFudgie | Ik theres an RJ11 socket involved with the phone |
| 02:41.07 | MajesticFudgie | brb |
| 02:42.10 | [TK]D-Fender | then it's ANALOG |
| 02:42.23 | [TK]D-Fender | And as we see lower down, it's an ADSL modem |
| 02:42.27 | [TK]D-Fender | so you have a dumb analog line |
| 02:42.41 | [TK]D-Fender | Last I checked you don't get "extra digits" over analog. |
| 02:42.53 | [TK]D-Fender | I know you can over ISDN (esp in mainland EU) |
| 02:43.29 | [TK]D-Fender | If you get more digits that would be post answer and would be in-line DTMF which means the server would have to start processing while listening for more |
| 02:43.35 | [TK]D-Fender | If they are in fact delivered |
| 02:43.45 | [TK]D-Fender | Go learn how your phone service works |
| 02:43.52 | [TK]D-Fender | the rest should announe itself. |
| 02:44.03 | [TK]D-Fender | And on that note... I'm out for a while.... |
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| 02:48.11 | MajesticFudgie | I think I knocked something out |
| 02:48.23 | MajesticFudgie | Last thing I saw was my response regarding a DSL socket |
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| 03:24.43 | MajesticFudgie | [TK]D-Fender I assume I'll need to add in a way to allow someone to enter an extension when they call, e.g. a 5-10 second delay before the call goes through so they can enter a extension? What do you recommend? |
| 03:25.30 | V5_12R\X | MajesticFudgie you'll need to add some logic for a keypress IVR. |
| 03:25.55 | MajesticFudgie | fun |
| 03:26.27 | V5_12R\X | MajesticFudgie yeah thats where all the magic happens |
| 03:26.27 | MajesticFudgie | Right now I have it so inbound calls from the outside would automatically be answered and then dial specific devices |
| 03:26.56 | MajesticFudgie | hm |
| 03:27.13 | MajesticFudgie | Whats the expected system? |
| 03:27.29 | MajesticFudgie | E.g. someone calls up and wants to enter an extension, whats the common setup |
| 03:28.50 | V5_12R\X | Someone calls, the phone system answers and plays an instruction message, when waits n seconds for a DTMF sequence. |
| 03:29.07 | MajesticFudgie | ah |
| 03:29.19 | MajesticFudgie | I assume after n seconds It can go ahead and dial through to reception |
| 03:31.12 | V5_12R\X | MajesticFudgie sure, whatever you want. I just described the most common, basic IVR setup that I think fits what you're asking for. |
| 03:34.19 | MajesticFudgie | Can you point in the right direction to do that |
| 03:34.39 | MajesticFudgie | Or is it as simple as wait()? |
| 03:34.50 | MajesticFudgie | and then any tones typed in skip to any defined extensions? |
| 03:37.12 | V5_12R\X | MajesticFudgie http://pastebin.com/0vtBgsv2 |
| 03:37.28 | V5_12R\X | MajesticFudgie no it's a little more complicated than that. You |
| 03:37.54 | MajesticFudgie | ah |
| 03:37.58 | V5_12R\X | MajesticFudgie you're going to need to look at the docs on Asterisk extension.conf "language" if you want to call it that |
| 03:38.39 | V5_12R\X | MajesticFudgie the example I gave you is pretty stripped down, as in it doesnt do any pattern matching or anything. |
| 03:38.51 | MajesticFudgie | This is gonna be fun to do, I'll probably leave such a feature out for now and have a seperate outside number for myself lol |
| 03:39.25 | V5_12R\X | MajesticFudgie honestly if you're interested in getting up and running quick you should probably just install freepbx |
| 03:39.37 | MajesticFudgie | True, I've avoided stuff like that atm |
| 03:39.46 | MajesticFudgie | Trying to learn without having guis do it for me and such |
| 03:40.42 | MajesticFudgie | I've read them |
| 03:40.50 | MajesticFudgie | wc |
| 03:40.51 | V5_12R\X | MajesticFudgie sure, but there's a very steep learning curve here. Asterisk assumes you already have a pretty good working knowledge of telephony/Internet telephony, then pile asterisk on top |
| 03:42.28 | MajesticFudgie | yeah I lack all of that, maybe a general idea of how it works but I'm here to learn |
| 03:42.51 | MajesticFudgie | I like to tackle large tasks |
| 03:44.23 | MajesticFudgie | Whats the purpose of the counter? Moreso how does it work? I see it goes to greeting if the counters 2? |
| 03:45.56 | MajesticFudgie | Oh, is it a repeating counter? e.g. the start extesion repeats twice then goes to greeting? |
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| 03:51.10 | V5_12R\X | MajesticFudgie basically it's so people aren't caught in an endless loop. After the third try it goes to a designated extension |
| 03:51.24 | MajesticFudgie | yeah, whats making it loop? |
| 03:51.34 | MajesticFudgie | Or does Asterisk look the start extension its self |
| 03:56.38 | V5_12R\X | Look at WaitExten's docs |
| 03:56.58 | MajesticFudgie | ah |
| 03:57.00 | MajesticFudgie | Looking now |
| 03:57.21 | MajesticFudgie | Whats the best way to execute the start extension with a software phone |
| 04:00.03 | MajesticFudgie | bbl |
| 04:00.12 | V5_12R\X | MajesticFudgie use a pattern matching context that uses a Goto statement to call that context and extension. Or you could just change the "start" to be a pattern match, which is the simplest answer. |
| 04:00.27 | MajesticFudgie | yeah |
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| 04:01.08 | MajesticFudgie | I'll play around with it tomorrow, thanks |
| 04:01.57 | Lope | I'm logged into the asterisk CLI in one terminal window, monitoring in verbose mode. Elsewhere I have a script that runs `asterisk -rx 'dialplan reload'` this makes a huge amount of text fly through my verbose terminal that I don't want to see. So I would like to find a way to silence the interactive terminal for the duration of the external command. |
| 04:02.17 | V5_12R\X | Lope core set verbose off |
| 04:02.42 | Lope | I've tried all of these options, without success: -rx -rxm -mrx -mcx -mcqx |
| 04:03.20 | V5_12R\X | Lope oh you were talking about the script... redirect it's output to /dev/null |
| 04:03.24 | Lope | V5_12R\X: how can I stack commands in one call? 'core set verbose off\ndialplan reload\ncore set verbose 3' ? |
| 04:03.34 | Lope | No the script is not outputting anything much. |
| 04:03.43 | Lope | It's my interactive terminal that's receiving all the text. |
| 04:03.54 | V5_12R\X | core set verbose off |
| 04:03.58 | V5_12R\X | core set debug off |
| 04:04.02 | Lope | V5_12R\X: how can I stack commands in one call? 'core set verbose off\ndialplan reload\ncore set verbose 3' ? |
| 04:04.10 | V5_12R\X | Lope you cant |
| 04:04.27 | Lope | ok |
| 04:05.36 | Lope | core set verbose off doesn't work. |
| 04:05.48 | Lope | it doesn't affect the interactive terminal when run from another terminal. |
| 04:05.54 | V5_12R\X | Lope I dont know what to tell you then on that |
| 04:08.20 | Lope | Yeah it seems like it's not possible. |
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| 05:27.37 | Lope | does anyone have any sox tips to increase recorded conversation volume. Simply doing normalize is not very good, because there can be one "pah!" sound that has a high amplitude, which limits the amount of amplification that will be given to the entire sound file. |
| 05:27.52 | Lope | I'd like to be able to do some kind of levelling? |
| 05:30.51 | [TK]D-Fender | "off" is not a valid value. |
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| 05:39.17 | Lope | TKD: noted. |
| 05:46.17 | ChannelZ | Look at the 'compand' option of sox for doing level compression with gain |
| 05:51.33 | V5_12R\X | So I have a systemd asterisk unit file that I'd like to contribute but not quite sure how to do it. I read the patch contrib guide (https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process) but that didn't cover config files |
| 05:52.38 | V5_12R\X | Lope look at liquidsoap. It's a godsend for stuff like this. |
| 05:52.58 | V5_12R\X | You can set up a liquidsoap script as well, and run it as a daemon. |
| 05:56.55 | V5_12R\X | Lope it's in the debian jessie main repos with a whole bunch of plugins for different formats, etc |
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| 09:47.20 | file | V5_12R\X, everything goes through gerrit that is going into the tree |
| 09:48.15 | V5_12R\X | @file hey so I submitted an issue for an improvement and when I get approved I'll be submitting the file with it. Does that work? |
| 09:48.29 | file | sure, although it'll go faster if you do it through gerrit yourself |
| 09:49.00 | V5_12R\X | file how would I go about doing that? Doesn't there need to be a review process on this? |
| 09:49.10 | file | the review process is on gerrit |
| 09:49.13 | V5_12R\X | ah ok |
| 09:49.54 | file | few people have access to directly commit and never should - all changes go up for review on gerrit and if accepted then they are submitted there |
| 09:51.12 | V5_12R\X | ok so how would that work? Just pull master, add the file, commit and push? |
| 09:51.43 | file | https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage |
| 09:55.02 | V5_12R\X | file sweet thanks. Should I close the issue then? |
| 09:55.25 | file | nope, everything should have a JIRA issue - it helps the release process |
| 09:56.42 | V5_12R\X | gotcha so I will make sure that I include the JIRA issue in the review then. Very nice. |
| 09:57.02 | file | yes, there's a commit messages format you should use |
| 09:57.18 | file | https://wiki.asterisk.org/wiki/display/AST/Commit+Messages |
| 09:57.19 | V5_12R\X | file I saw that, thanks a lot for the link! |
| 09:57.47 | file | it ties it all together and then during the release it pulls in all the right stuff for the changelog/release summary |
| 09:57.58 | file | and it also means the roadmap on JIRA reflects the issues that were solved in a release |
| 09:58.52 | V5_12R\X | cool. This will be my first contribution |
| 10:00.17 | file | the first of many we hope! |
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| 10:30.50 | FarhaadN | who can help me? i want use sipml5 and https and asterisk on centos os,any of document asterisk i read,i can register with extention ,but i cant make a call,when simulate call with 7777,asterisk log show me call stablished and annoncment played,but i cant hear anythings |
| 10:48.50 | TandyUK | FarhaadN: sounds liek a problem with a firewall blockign the media stream |
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| 12:00.14 | FarhaadN | TandyUK: firewall for what? my server install on esx and firewall not install |
| 12:00.27 | FarhaadN | iptables empty |
| 12:00.36 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:05.08 | *** join/#asterisk CeBe1 (~CeBe@89.204.138.179) |
| 12:16.42 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:21.07 | *** join/#asterisk Draecos (~Draecos@203-121-194-197.e-wire.net.au) |
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| 12:39.26 | *** part/#asterisk FarhaadN (~Farhad@82.99.206.194) |
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| 12:54.20 | Lope | Does asterisk need to write to any of the conf files in /etc/asterisk. is there anything bad about me chowning them root:asterisk? |
| 12:54.59 | [TK]D-Fender | yes |
| 12:55.01 | Lope | with chmod 740? (asterisk read only)? |
| 12:55.16 | Lope | TKD can you please elaborate? |
| 12:55.40 | [TK]D-Fender | voicemail.conf & users.conf are R/W by *, etc |
| 12:56.03 | Lope | is there a list of files that * needs to write to? |
| 12:56.07 | [TK]D-Fender | technically even extensions.conf can be |
| 12:56.29 | Lope | That's very strange. Perhaps I should just leave the permissions alone on that one. |
| 12:56.40 | Lope | (the whole /etc/asterisk dir) |
| 12:56.41 | [TK]D-Fender | No, I doubt there is a list but there are other things you should probably prioritized before you start playing whack-a-mole with this and screw something up. |
| 12:56.48 | Lope | just seems like a vulnerability. |
| 12:59.13 | [TK]D-Fender | Don't worry, there are plenty of other things you should be thinking about first |
| 13:09.34 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:3db0:9205:24f6:633f) |
| 13:09.48 | Lope | tcptls.c ast_tcptls_client_start: Unable to connect SIP socket to (my ip):5060 (and some other port). Any reason why I'd get this message? I'm not using tls? |
| 13:10.51 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-ythndnqlgwsxlbmw) |
| 13:10.51 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:13.27 | Lope | maybe it's my client trying to upgrade the connection. |
| 13:14.12 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:14.12 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:18.12 | [TK]D-Fender | You see the IP.... |
| 13:18.37 | [TK]D-Fender | Doesn't sound like this is something that requires even a "guess" when it's your own... |
| 13:25.41 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
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| 14:52.29 | Lope | I wasn't wondering who. I was wondering why. |
| 14:52.35 | Lope | night night |
| 14:54.27 | *** join/#asterisk kharwell (kharwell@nat/digium/x-nflqwswdxazmunza) |
| 15:02.40 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 15:11.23 | *** join/#asterisk lroe (~roe@unaffiliated/roe) |
| 15:27.03 | lroe | I'm looking at the various versions available and I'm reading https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 but it is unclear how disruptive migrating from 11 to 13 will be. |
| 15:27.44 | lroe | I am starting with a fresh system and trying to decide if I should stick with 11 (what's packaged in debian) or do an SSB of 13 so that I won't have to go through the pains of upgrading |
| 15:28.57 | lroe | I know from older versions that some upgrading have been onerous because of architecture and syntax changes |
| 15:29.32 | file | the UPGRADE.txt file has the stuff to be aware of when upgrading, but it ultimately depends on the previous config... if it's a fresh system then it should be fine |
| 15:29.35 | WIMPy | There really was ony one syntax change fom 1.2 to 1.4. |
| 15:30.00 | lroe | WIMPy, yea, that was the one I hit |
| 15:30.34 | lroe | the system I am replacing is currently running 1.4.12 |
| 15:30.44 | lroe | on centos 4 |
| 15:39.01 | *** join/#asterisk ipengineer (~zconkle@71.252.134.63) |
| 15:58.19 | [TK]D-Fender | 13 = big improvements and new channel drivers, etc |
| 16:16.16 | *** join/#asterisk cyford (~support@96.83.24.7) |
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| 16:25.42 | lvlinux | Hey guys, I have a small-ish setup here with about 50 phones. What's probably the best way to do relatively simple automatic phone provisioning? phoneprov.conf and users.conf, or something else? |
| 16:25.53 | lvlinux | Polycom phones |
| 16:26.26 | lvlinux | I don't really care if the dialplan is messed with as long as a PJSIP peer is created for each phone automatically. |
| 16:27.22 | lvlinux | ehh I mean PJSIP "endpoint" not "peer" :-) |
| 16:32.27 | [TK]D-Fender | that isn't usually the way this works |
| 16:32.42 | [TK]D-Fender | You don't have the phone configuring the PBX... you have the PBX configuring the phone. |
| 16:32.47 | [TK]D-Fender | Your idea is backwards |
| 16:37.58 | elyob | Can anyone suggest why my network drops when asterisk answers a call?! All computers on the network drop out for a minute. |
| 16:38.28 | [TK]D-Fender | I suggest you look at your router |
| 16:40.02 | elyob | Last time I looked, yes it had disconnected from the internet. Wasn't sure why it would do that. It's a BT Home Hub and I'm wondering about replacing it with somethihng better. |
| 16:40.59 | *** join/#asterisk todwetsprock (~florian@p5099a594.dip0.t-ipconnect.de) |
| 16:42.51 | MajesticFudgie | Hm, I dont seem to have the pjsip command |
| 16:42.59 | MajesticFudgie | How would I get pjsip working? |
| 16:45.04 | MajesticFudgie | Using Asterisk 11.7.0 I got via apt-get on Ubuntu |
| 16:45.44 | file | PJSIP does not exist in Asterisk 11 |
| 16:45.52 | Chainsaw | Also, aren't we on 11.22.0 by now? |
| 16:45.57 | file | yes. |
| 16:46.50 | MajesticFudgie | Oh |
| 16:46.56 | MajesticFudgie | What version would I need? |
| 16:47.02 | file | Asterisk 13. |
| 16:47.05 | MajesticFudgie | okie |
| 16:47.14 | MajesticFudgie | Unless 11 has support for a SIP Provider? |
| 16:47.19 | MajesticFudgie | Or is it only via pjsip |
| 16:47.22 | file | it does... |
| 16:47.27 | file | chan_sip speaks SIP. |
| 16:47.40 | MajesticFudgie | ah |
| 16:49.50 | MajesticFudgie | I have a feeling I'm going to want to upgrade to Asterisk 13 |
| 16:53.43 | [TK]D-Fender | being behind isn't usually a great thing |
| 16:55.22 | file | 11 goes bye bye... in the future |
| 16:55.59 | file | October |
| 16:56.26 | file | well, security fix only |
| 16:59.04 | *** join/#asterisk theron_ (~theron@2620:10d:c091:200::2:e050) |
| 17:00.48 | [TK]D-Fender | When just plain broken stuff will remain broken .... get the hell out of Dodge |
| 17:01.14 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:08.01 | *** join/#asterisk areski (~areski@80.174.128.27.dyn.user.ono.com) |
| 17:09.37 | MajesticFudgie | I keep getting configure: error: *** uuid support not found |
| 17:09.50 | MajesticFudgie | when doing the configure command |
| 17:10.21 | [TK]D-Fender | https://www.google.ca/#q=asterisk+install+uuid |
| 17:10.25 | MajesticFudgie | I did |
| 17:10.26 | *** join/#asterisk AAmit (~Amit@219.91.218.90) |
| 17:10.41 | MajesticFudgie | I get told to do contrib/scripts/install_prereq install |
| 17:10.48 | MajesticFudgie | Which does't change the result |
| 17:10.48 | *** join/#asterisk robink (~quassel@unaffilated/robink) |
| 17:10.58 | lvlinux | [TK]D-Fender: I'm fully aware of how it works---I didn't mean the "phone" configuring Asterisk, I meant I don't need any dialplan extensions or voicemail entries automatically created, where a provisioned phone can be dialed without an extension entry being manually put into the dialplan. I just need the endpoint config'ed for each phone that ASTERISK provides provisioning info to, not vice versa. |
| 17:12.02 | [TK]D-Fender | lvlinux, You could jsut make your own provisioning files and GREP/SED/AWK them based on * configs or some other means |
| 17:12.11 | [TK]D-Fender | tthese are just dumb text files. |
| 17:12.36 | [TK]D-Fender | * does have one weak provisoning based thing which did kind work for polycom.... but it seems largely abandoned. |
| 17:12.43 | [TK]D-Fender | and works in it's singular way |
| 17:13.17 | *** join/#asterisk jge (~jge@unaffiliated/bitfury) |
| 17:15.12 | lvlinux | K. Yeah that may be the best way. |
| 17:16.50 | lvlinux | I thought about setting up a database based solution with Asterisk realtime and aotomatic config file creation but probably would end up being more trouble than it's worth, and it goes against my tendency to have complete control over the system. |
| 17:17.52 | [TK]D-Fender | Easy enough to text-parse sip.conf, etc looking for a custom value you set and have a shell-script SED some templated configs for them |
| 17:18.52 | lvlinux | Yup |
| 17:18.53 | *** join/#asterisk ipengineer (~zconkle@71.252.134.63) |
| 17:23.20 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 17:25.15 | MajesticFudgie | configure: error: *** Asterisk now uses SQLite3 for the internal Asterisk database. |
| 17:25.25 | MajesticFudgie | I have SQLite 3 installed o-o |
| 17:26.00 | *** join/#asterisk theron (~theron@2620:10d:c091:200::2:ea5a) |
| 17:27.53 | [TK]D-Fender | Not the proper library clearly... |
| 17:28.12 | MajesticFudgie | Yes, 100% says its not that |
| 17:28.13 | MajesticFudgie | ._. |
| 17:28.50 | MajesticFudgie | Slowly getting through it |
| 17:29.07 | MajesticFudgie | the configure script fails continually each time I've sorted something else |
| 17:29.34 | MajesticFudgie | I think that sorted that error, had to install SQLLite3 Headers on ubuntu |
| 17:29.38 | [TK]D-Fender | The * WIKI has very clear instructions for various distros... |
| 17:31.05 | *** join/#asterisk raspberrypifan (~raspberry@2604:2000:6016:be00:3db0:9205:24f6:633f) |
| 17:36.44 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 17:36.44 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 18:13.41 | jwpierce3 | within sip.conf [general]; the "register => username:password@sip.blah.com" is for calls comming into asterisk correct? |
| 18:14.32 | jwpierce3 | which then routes through context=incoming right? |
| 18:14.44 | MajesticFudgie | checking if external Speex devkit is installed... aconfigure: error: Unable to use external Speex library. If Speex development files are not available in the default locations, use CFLAGS and LDFLAGS env var to set the include/lib paths |
| 18:14.49 | MajesticFudgie | How do I resolve this? |
| 18:15.56 | jwpierce3 | MajesticFudgie, sounds like speex is not installed. what distro |
| 18:16.04 | *** join/#asterisk KerioMorgan (~Adium@gw-us.kerio.com) |
| 18:17.50 | MajesticFudgie | Ubuntu |
| 18:17.55 | [TK]D-Fender | jwpierce3, No. |
| 18:17.59 | MajesticFudgie | I've installed 'speex' its self |
| 18:18.09 | [TK]D-Fender | ~sipregister |
| 18:18.09 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 18:18.10 | MajesticFudgie | Which hasn't changed anything |
| 18:18.10 | [TK]D-Fender | ^^^ |
| 18:18.32 | MajesticFudgie | inb4 I need its headers |
| 18:19.25 | jwpierce3 | [TK]D-Fender, tell your provider what IP address & EXTEN to send INCOMING calls to |
| 18:19.33 | jwpierce3 | seems like incoming to me |
| 18:20.06 | [TK]D-Fender | Thet tells them WHERE to reach you |
| 18:20.18 | [TK]D-Fender | it has nothing to with MATCHING a call that actually arrives. |
| 18:20.29 | jwpierce3 | I understand that |
| 18:20.41 | [TK]D-Fender | <jwpierce3> which then routes through context=incoming right? <- so NOT this |
| 18:20.43 | jwpierce3 | that's done in extensions.conf |
| 18:20.46 | [TK]D-Fender | no |
| 18:21.00 | [TK]D-Fender | The call still has to get matched by sip.conf |
| 18:21.22 | [TK]D-Fender | it doesn't magically and automatically fall into the dialplan |
| 18:22.24 | jwpierce3 | what "option/options" would drop or reroute calls away from the dialplan? |
| 18:22.51 | jwpierce3 | alwaysauthreject=yes? |
| 18:23.21 | [TK]D-Fender | WHICH calls? |
| 18:23.43 | jwpierce3 | great question. :) |
| 18:24.07 | [TK]D-Fender | jwpierce3> alwaysauthreject=yes? <- tthis is a 1st response change for security obfuscation. This is not normally a "hard" rejection, but it commonly a healthy thing to have |
| 18:24.19 | [TK]D-Fender | by rarely an end-of-the-line |
| 18:25.15 | V5_12R\X | MajesticFudgie use install the install_prereq script for dependency resolution, it makes it easier. |
| 18:25.49 | MajesticFudgie | kk |
| 18:26.02 | MajesticFudgie | Then bootstrap.sh, configure.sh ? |
| 18:26.08 | jwpierce3 | my trunk provider is flowroute. I have one account and many DID's. I am filtering those via dialplan to each server I setup for each of my customers. Is this the best way to do this? |
| 18:26.22 | MajesticFudgie | I'd want to do ./configure --with-pjproject-bundled wouldnt i? |
| 18:26.32 | V5_12R\X | MajesticFudgie I would think so |
| 18:26.38 | MajesticFudgie | kk ty |
| 18:27.38 | V5_12R\X | jwpierce3 Flowroute has a sip configuration example for asterisk on their site. From there you are simply matching the pattern on the user part of the SIP URI on SIP INVITE's that you receive from flowroute. |
| 18:27.39 | jwpierce3 | MajesticFudgie, so ubuntu has a ppa that doesn't have all the options you want compilied into it right? just curious? |
| 18:27.47 | MajesticFudgie | Idk |
| 18:27.59 | MajesticFudgie | I just want pjsip to work, thats all I know at this point lol |
| 18:28.14 | V5_12R\X | I does, but it's a much older version of asterisk. |
| 18:28.49 | MajesticFudgie | Asterisk 11.7 |
| 18:29.01 | V5_12R\X | MajesticFudgie use install_prereq, install everything and update. Compile using the --with-pjproject-bundled flag. |
| 18:29.02 | MajesticFudgie | and no change |
| 18:29.21 | MajesticFudgie | I have no clue what I'm missing |
| 18:29.36 | [TK]D-Fender | a clue |
| 18:29.51 | [TK]D-Fender | <MajesticFudgie> Asterisk 11.7 <-- NO PJSIP IN 11. YOU WERE TOLD ALREADY |
| 18:29.58 | MajesticFudgie | I KNOW |
| 18:30.10 | *** join/#asterisk obelixBE (~obelix@2a02:1811:c52c:1700:583a:11d3:ea42:8dd6) |
| 18:30.11 | MajesticFudgie | I was pointing out the Ubuntu PPA has Asterisk 11.7 |
| 18:30.12 | [TK]D-Fender | Go to the WIKI. |
| 18:30.20 | V5_12R\X | MajesticFudgie clone the git repo from http://github.com/asterisk |
| 18:30.24 | [TK]D-Fender | Follow the instructions for installing on Ubuntu |
| 18:31.04 | MajesticFudgie | I cant find any on the wiki |
| 18:31.16 | jwpierce3 | funtoo/gentoo is sooo much easier than fooling with release distros :) |
| 18:31.33 | MajesticFudgie | I've been trying to use this guide http://www.ipcomms.net/sample-device-configurations/41-asterisk/181-install-asterisk-13-on-ubuntu-debian |
| 18:31.51 | MajesticFudgie | and even that guide sucks |
| 18:32.32 | obelixBE | why you don't follow the 'book'? |
| 18:32.39 | MajesticFudgie | What book |
| 18:34.39 | obelixBE | http://asterisk-service.com/downloads/Asterisk-%20The%20Definitive%20Guide,%204th%20Edition.pdf |
| 18:36.11 | MajesticFudgie | I'm not having an issue getting asterisk compiled and installed. I'm having the issue of repeating it with PJSIP |
| 18:36.34 | obelixBE | ow sorry :) i misread :) |
| 18:36.40 | MajesticFudgie | It works fine without PJSIP, just trying to install with pjproject it just throws the speex error |
| 18:36.44 | MajesticFudgie | It's alright lol |
| 18:36.58 | MajesticFudgie | It's as if speex isnt installed , it is, headers as well |
| 18:37.00 | obelixBE | i don't have experience with pjsip |
| 18:38.21 | MajesticFudgie | I've been trying to follow this https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled |
| 18:38.30 | MajesticFudgie | I might just try doing pjproject manually |
| 18:40.15 | obelixBE | and why do you want to use pjsip? it works good with sip |
| 18:40.51 | MajesticFudgie | Trying to use SIPGate with Asterisk for incoming and outgoing calls to the outside world |
| 18:41.16 | MajesticFudgie | and the only way I know is using pjsip via the config they provide |
| 18:41.23 | MajesticFudgie | See http://basichelp.sipgate.co.uk/hc/en-gb/articles/206404129-Asterisk-PBX |
| 18:43.08 | obelixBE | i take a look |
| 18:44.52 | [TK]D-Fender | And if you google "sipgate asterisk guide" you see HUNDREDS of others using chan_sip |
| 18:45.53 | V5_12R\X | MajesticFudgie https://gist.github.com/TheSeanBrady/ac4aef7104540ae7bf099b3d9fd76302 |
| 18:46.43 | MajesticFudgie | I'll have a look soon |
| 18:47.34 | V5_12R\X | MajesticFudgie use pjsip. chan_sip is depricated. No sense in learning something that won't be around in a year or so |
| 18:49.38 | *** join/#asterisk monsterco (~monsterco@184.151.178.210) |
| 18:50.52 | V5_12R\X | MajesticFudgie no need to install pjsip manually, although it's rather trivial. The gist that I gave you should do it all. |
| 18:51.08 | MajesticFudgie | https://surl.im/i/58wht.png Is makemenu select |
| 18:51.19 | MajesticFudgie | What are the XXX's for? Enabled or not enabled |
| 18:52.03 | [TK]D-Fender | V5_12R\X, It's not exactly going away.... |
| 18:52.05 | V5_12R\X | MajesticFudgie enabled |
| 18:52.11 | MajesticFudgie | kk |
| 18:52.57 | V5_12R\X | [TK]D-Fender it's listed as deprecated in menuselect, therefore it's deprecated. |
| 18:53.21 | [TK]D-Fender | yes, but that doesn't mean anything as far as Asterisk's removal policy |
| 18:58.00 | *** join/#asterisk theron (~theron@2620:10d:c091:200::2:ffc1) |
| 18:58.00 | *** join/#asterisk acidfu_ (~acidfoo@modemcable002.114-70-69.static.videotron.ca) |
| 18:58.41 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-ajdkmmkhukuslyvs) |
| 19:03.21 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 19:08.08 | MajesticFudgie | WARNING[4184][C-00000006]: app_dial.c:2429 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 66 - Channel not implemented) |
| 19:08.10 | MajesticFudgie | .-. |
| 19:08.47 | MajesticFudgie | http://pastebin.com/raw/LKw8WXaC |
| 19:09.03 | *** join/#asterisk woose (~Woose@unaffiliated/woose) |
| 19:09.57 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 19:10.38 | file | chan_sip won't be removed, but the effort by Digium towards fixing stuff in it is not the same as chan_pjsip |
| 19:11.15 | MajesticFudgie | I assume my errors above indicate pjsip isn't installed/working? |
| 19:11.28 | file | it isn't installed |
| 19:11.32 | MajesticFudgie | I cri |
| 19:11.36 | V5_12R\X | MajesticFudgie correct |
| 19:11.44 | MajesticFudgie | It shows in the menu as enabled doesnt it? |
| 19:12.01 | V5_12R\X | MajesticFudgie what operating system is this again? |
| 19:12.07 | MajesticFudgie | Ubuntu |
| 19:12.15 | MajesticFudgie | Ill get the exact ver in a sec |
| 19:12.15 | V5_12R\X | MajesticFudgie version? |
| 19:12.27 | V5_12R\X | cat /etc/issues.net |
| 19:12.36 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 19:13.28 | V5_12R\X | MajesticFudgie try starting asterisk on the console. asterisk -cvvvvvv. It will verbosely display the startup process. Pastebin that please. |
| 19:13.33 | MajesticFudgie | <PROTECTED> |
| 19:13.42 | MajesticFudgie | SEc |
| 19:13.43 | V5_12R\X | MajesticFudgie ok cool. |
| 19:14.00 | V5_12R\X | MajesticFudgie of course you need to make sure that you stop the current instance first |
| 19:17.00 | MajesticFudgie | http://paste.ubuntu.com/16224657/ |
| 19:17.06 | MajesticFudgie | Inb4 my install is a trainwreck lol# |
| 19:20.01 | V5_12R\X | MajesticFudgie I dont see pjsip loading. Can you give me a paste of the output of asterisk command "modules show" and the output of ls /usr/lib/asterisk/modules please? |
| 19:20.16 | MajesticFudgie | sure |
| 19:20.31 | V5_12R\X | MajesticFudgie sorry that command is "module show" |
| 19:21.04 | MajesticFudgie | No such command 'module' (type 'core show help module' for other possible commands) o.o |
| 19:21.23 | MajesticFudgie | wierd |
| 19:21.25 | MajesticFudgie | sec |
| 19:22.18 | MajesticFudgie | http://paste.ubuntu.com/16224783/ |
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| 19:23.04 | MajesticFudgie | http://paste.ubuntu.com/16224802/ |
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| 19:24.07 | V5_12R\X | MajesticFudgie how about an "ldconfig -p | grep pj" from your linux command line |
| 19:24.39 | V5_12R\X | MajesticFudgie and you did run the prereq_install script, right? |
| 19:24.47 | MajesticFudgie | http://paste.ubuntu.com/16224830/ |
| 19:24.51 | MajesticFudgie | yup |
| 19:24.59 | MajesticFudgie | it doesnt output anything |
| 19:26.12 | V5_12R\X | MajesticFudgie looks like it didnt completely install. |
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| 19:26.13 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:26.20 | V5_12R\X | or make that it |
| 19:26.21 | V5_12R\X | or make that is |
| 19:26.23 | MajesticFudgie | Fu |
| 19:26.25 | MajesticFudgie | fun |
| 19:26.37 | MajesticFudgie | What do you suggest |
| 19:26.48 | MajesticFudgie | I've followed the gist earlier |
| 19:26.56 | MajesticFudgie | and other guides |
| 19:27.38 | file | prereq doesn't install pjproject unless you tell it to, and bundled PJSIP isn't used unless you tell it to be used |
| 19:28.07 | MajesticFudgie | ah |
| 19:28.15 | MajesticFudgie | the latter didnt work for me |
| 19:28.34 | MajesticFudgie | What do i need to pass to prereq to install it |
| 19:29.35 | file | install-unpackaged |
| 19:30.47 | V5_12R\X | http://paste.ubuntu.com/16224936/ |
| 19:31.00 | V5_12R\X | if it fails from here we'll be able to see why |
| 19:31.11 | MajesticFudgie | kk |
| 19:31.13 | file | 13.9.0-rc2 also has fixes for bundling to work on a wider assortment of distros/installs/versions/yada |
| 19:31.21 | MajesticFudgie | ill start from scratch |
| 19:32.07 | V5_12R\X | MajesticFudgie I strongly doubt thats the issue here. I've been playing with it and it should work fine on Debian/Ubuntu using 13.8.2. My issue earlier was a missing dep, solved by prereq_install |
| 19:32.35 | V5_12R\X | MajesticFudgie no need to re-do it all, just what I sent in the last paste, after you run prereq_install |
| 19:32.54 | file | yeah, it'd be nice to know why it's not working when it should be in that environment |
| 19:33.00 | V5_12R\X | file from his ldconfig output he's probably missing the python bindings. |
| 19:33.38 | V5_12R\X | I was missing python-dev from my machine for whatever reason. I don't think that it was in my deps list previously and I mistakenly didnt update it |
| 19:33.56 | file | THAT'S IT, I'M MAKING AN ISSUE |
| 19:34.05 | V5_12R\X | lol |
| 19:34.35 | V5_12R\X | well to be fair prereq_install --test did recommend it. |
| 19:34.55 | V5_12R\X | I should've know better :) |
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| 19:36.21 | file | https://issues.asterisk.org/jira/browse/ASTERISK-25993 |
| 19:37.24 | MajesticFudgie | lol |
| 19:37.24 | file | I've been mumbling about that for a bit, but now it's an issue! |
| 19:37.33 | MajesticFudgie | yay |
| 19:37.35 | MajesticFudgie | wooo |
| 19:38.19 | robmal | If possible, please mention it was reported by someone nicknamed MajesticFudgie. |
| 19:38.21 | V5_12R\X | file I have to say that I must agree with your description |
| 19:38.32 | file | I also agree with it |
| 19:38.49 | MajesticFudgie | Be all, Some asshole named MajesticFudgie caused it |
| 19:38.50 | MajesticFudgie | lol |
| 19:39.05 | file | it wasn't caused by you, it's just crept up a few times |
| 19:39.10 | MajesticFudgie | lol |
| 19:39.27 | MajesticFudgie | You're flattering me |
| 19:39.37 | MajesticFudgie | I'm really good at breaking stuff |
| 19:39.51 | V5_12R\X | MajesticFudgie so do we have a winner after the prereq_install? |
| 19:39.57 | MajesticFudgie | No idea |
| 19:40.03 | MajesticFudgie | I had already gone from scratch |
| 19:40.09 | MajesticFudgie | I'll let you know in a few mins |
| 19:40.35 | MajesticFudgie | I think cloning from git could have solved it |
| 19:41.00 | MajesticFudgie | Rather than using a tarball |
| 19:43.19 | V5_12R\X | MajesticFudgie well if you copy it line by line I know exactly what you did and what code you used. |
| 19:44.11 | V5_12R\X | MajesticFudgie and I'm fairly positive that the issue was the python bindings, as it appears that the C lib (I think, not *that* familiar with pjproject) built but not python |
| 19:44.20 | MajesticFudgie | idk |
| 19:44.24 | MajesticFudgie | We'll see what happens |
| 19:44.35 | MajesticFudgie | Right now I've gone from scratch following the Gist to the letter |
| 19:44.52 | MajesticFudgie | If that works then using the tarball didnt work |
| 19:45.37 | V5_12R\X | ok good |
| 19:47.01 | V5_12R\X | so can anyone point me to where files would be staged to be copied during the make install process? Are there docs on make install? Never worked with make before like that |
| 19:50.19 | MajesticFudgie | So far the python bindings are fine |
| 19:52.47 | *** part/#asterisk jge (~jge@unaffiliated/bitfury) |
| 20:01.23 | MajesticFudgie | Well Im getting pjsip errors now |
| 20:01.41 | MajesticFudgie | [May 4 21:01:59] NOTICE[9276]: res_pjsip/pjsip_distributor.c:368 log_unidentified_request: Request from '<sip:thomas@192.168.2.204>' failed for '192.168.1.149:35537' (callid: -d4UCoCaPWp097aY1fkSAA..) - No matching endpoint found |
| 20:12.17 | V5_12R\X | MajesticFudgie no those are notices. As in "notice that you screwed up here" |
| 20:12.44 | V5_12R\X | MajesticFudgie now it's time for you to actually learn asterisk. See aforementioned link to the book. |
| 20:13.22 | [TK]D-Fender | Except the book knows nothing of PJSIP. |
| 20:13.33 | [TK]D-Fender | For that.... it's down to the sample config & wiki |
| 20:13.50 | V5_12R\X | [TK]D-Fender right you are |
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| 20:22.30 | V5_12R\X | So I've added my public key to Gerrit but I'm still getting denied on SSH access (cloning operation). What can I do here? |
| 20:23.42 | file | V5_12R\X, did you set a username? |
| 20:24.01 | V5_12R\X | ssh://SeanBrady@gerrit.asterisk.org:29418/asterisk is my connection string |
| 20:26.38 | V5_12R\X | file nevermind I see what you're saying I got it thanks |
| 20:27.04 | file | yeah, that's the answer every time someone says that :D |
| 20:27.13 | V5_12R\X | lol |
| 20:27.30 | V5_12R\X | sounds like something for the ~infobot |
| 20:30.19 | file | is going to edit the wiki page |
| 20:31.30 | newtonr | file, thanks |
| 20:31.51 | file | "Failure to set your username will result in clones using SSH failing, despite adding your SSH key. If you can not clone using SSH please ensure you have set your username." in a warning, which is red |
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| 20:32.53 | file | red like my bloooooood |
| 20:40.17 | V5_12R\X | yeah I actually just saw the large, red backgrounded box dead center of the page with the conspicuously large red exclamation point there. |
| 20:50.34 | V5_12R\X | Is anyone experiencing certificate issues with issues.asterisk.org? It seems it's just safari |
| 20:51.03 | file | issues.asterisk.org has a legacy requirement to request an SSL certificate, it's optional but Safari in particular likes to throw a fit |
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