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| 02:10.48 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 02:45.11 | xcom | wow so wiet |
| 02:45.37 | lanning | I'm hunting wabbits. |
| 02:50.17 | hebber | shhhhhhhh |
| 02:52.39 | *** join/#asterisk BoRiS (~raiden@S010660a44cdcb910.wp.shawcable.net) |
| 02:59.45 | BoRiS | hI guys, Have problems compiling asterisk 11. Keep getting a bunch of errors when compiling like multiple definition of `CRYPTO_set_locking_callback', multiple definition of `ERR_free_strings'. Any ideas? ( http://www.pastebin.ca/2369566 ) |
| 03:07.25 | xcom | lanning: lol |
| 03:10.56 | BoRiS | ? |
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| 04:00.48 | BeeBuu | hi,all |
| 04:01.50 | BeeBuu | anyone tell me is there any limit on send command to asterisk by http ? |
| 04:01.57 | BeeBuu | please? |
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| 04:19.26 | igcewieling | the http interface to Asterisk is not very commonly used. You may have to check the source code. |
| 04:19.52 | igcewieling | BoRiS: remove ffmpeg from your system, and rebuild from scratch, do a make distclean first |
| 04:19.58 | [TK]D-Fender | 42 <- |
| 04:21.57 | igcewieling | BoRiS: and did you file a bug report on Jira like I suggested the other day? |
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| 05:09.15 | BoRiS | igcewieling: I removed ffmpeg(and ffmpeg-devel), re-checked out asterisk 11 from svn and tried rebuilding it. Still the same error. |
| 05:09.29 | BoRiS | igcewieling: I did.... https://issues.asterisk.org/jira/browse/ASTERISK-21720 |
| 05:09.30 | LieutPants | [ASTERISK-21720] [Status: Triage] Asterisk 11 cannot complie - https://issues.asterisk.org/jira/browse/ASTERISK-21720 |
| 05:09.44 | igcewieling | BoRiS: also remove libavcodec |
| 05:11.17 | igcewieling | if that doesn't work, wait for the bug report to get some activity |
| 05:11.20 | BoRiS | That library was apart of ffmpeg which has been removed. |
| 05:11.44 | igcewieling | when you re-checked out did you remove the directory first. |
| 05:12.20 | BoRiS | yeah, I rm -rf asterisk-11 directory and recheckout the latest asterisk-11 from svn. |
| 05:12.21 | LieutPants | [ASTERISK-11] [Status: Closed] AGI channel_status failure - https://issues.asterisk.org/jira/browse/ASTERISK-11 |
| 05:14.16 | BoRiS | The error almost looks like something is main/libasteriskssl.c is also being defined in openssl (1.0.1e) libcrypto. |
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| 05:52.33 | ChrisInSydney | g'day all |
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| 07:50.19 | krotos | good morning guy |
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| 08:31.07 | hrolf | Hi #asterisk |
| 08:31.50 | hrolf | What is the '-' mean in _64XX? |
| 08:31.55 | hrolf | In extensions.conf |
| 08:32.01 | hrolf | '_' |
| 08:33.21 | ectospasm | '_' tells the interpreter that this is a pattern |
| 08:33.34 | hrolf | ectospasm: Okay. |
| 08:33.37 | hrolf | Thanks. |
| 08:33.57 | ectospasm | without the '_', it will try to match a literal 64XX, which will be difficult to do for most technologies. |
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| 08:54.28 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 08:55.58 | Tuju | hi, i have a remote asterisk "register" thingy and some settings in [<linename>] section in my sip.conf and it's visible in sip show peers list. how can i use it as trunk? |
| 09:01.15 | ectospasm | Tuju: Dial(SIP/trunkname/${EXTEN})... |
| 09:01.48 | Tuju | ectospasm: exten => _9XXXX.,n,Dial(SIP/fooprovider/${EXTEN:1},60) i found that from my extensions |
| 09:02.15 | Tuju | do those have to be in the same 'context' as my phones? |
| 09:02.30 | ectospasm | yes. |
| 09:02.37 | Tuju | my 'sip show registry' shows nothing. |
| 09:02.55 | Tuju | regardless that 'sip show peers' lists that line just fine. |
| 09:02.58 | ectospasm | The phones make outbound calls within the confines of "context=<e.g. from-internal>" |
| 09:03.10 | ectospasm | Are you registering to the peer? |
| 09:03.15 | ectospasm | Or is the peer registering to you? |
| 09:03.23 | Tuju | are those the 'register' lines in sip.conf? |
| 09:03.34 | Tuju | i think i'm register towards them. :) |
| 09:03.35 | ectospasm | do you have a register => line in the [general] section of sip.conf? |
| 09:03.39 | Tuju | yes |
| 09:03.56 | Tuju | but i recall that they had @ char in username and that caused some gray hairs. |
| 09:04.59 | ectospasm | sip show registry will only show current registration status for things that register to your local Asterisk instance, I think |
| 09:05.07 | Tuju | egister => +372654321:password@proxy.example.com is the line i have. |
| 09:05.22 | Tuju | ectospasm: ack, makes sense. |
| 09:05.27 | Tuju | so i need to get that line working. |
| 09:05.37 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.78) |
| 09:06.01 | Tuju | i think the actual line would be correctly something like this: |
| 09:06.11 | Tuju | egister => +372654321@example.com:password@proxy.example.com |
| 09:06.28 | Tuju | that is, 'username':'password'@proxy.example.com |
| 09:06.40 | Tuju | but it doesn't like the @ char there. |
| 09:07.18 | ectospasm | it probably doesn't like the + |
| 09:07.30 | ectospasm | the @ symbol is expected for most register statements. |
| 09:09.46 | Tuju | i try that |
| 09:10.18 | Tuju | hmm...sorry, not that + is not there. it starts directly with 372, i worte that from my musclemem. |
| 09:12.28 | ectospasm | Tuju: do you know if Asterisk is actually sending the REGISTER requests? |
| 09:13.03 | Tuju | i set the: sip set debug peer <linename> on and try to figure out that |
| 09:13.08 | Tuju | but i'm not seeing any. |
| 09:13.42 | Tuju | SIP/2.0 403 Forbidden |
| 09:13.45 | Tuju | came now |
| 09:13.54 | Tuju | apparently it doesn't like the syntax |
| 09:14.47 | Tuju | how does it go, lines are so that asterisk is the 'core' and phones are 'terminals', those are specified in [<linename>] sections, right? |
| 09:15.16 | Tuju | i understood that i - regardless of above - need to have such 'line' section for my upstream ISP too. |
| 09:15.17 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 09:15.26 | Tuju | and also, the register line. |
| 09:15.44 | Tuju | it would be logical that not to have that [<linename>] section for trunks. |
| 09:16.36 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 09:16.40 | Tuju | that is: in my mind, i would differientate the upstream trunks and downstream lines with different settings. |
| 09:16.56 | Tuju | and that 'register =>' line makes perfectly sense. |
| 09:17.34 | *** part/#asterisk uosiu (~uosiu@hajdamowicz.info) |
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| 09:17.56 | ectospasm | "lines" doesn't make sense with SIP |
| 09:17.58 | *** join/#asterisk tallest_red (~CNZ@ip98-169-197-57.dc.dc.cox.net) |
| 09:18.24 | bulkorok | hi |
| 09:18.26 | ectospasm | Unless you know what you're doing, only have one peer definition of "friend" |
| 09:18.33 | Tuju | apparently they don't. that's why it's so hard to wind your thoughts around this new concept. |
| 09:18.48 | Tuju | http://www.voiptalk.org/products/asterisk-voiptalk-sip-trunk-registration-using-outbound-proxy-setup i've more or less similar config. |
| 09:18.52 | Tuju | bulkorok: hi |
| 09:20.03 | Tuju | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf ha! there is @ char |
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| 09:23.49 | ectospasm | er, I meant |
| 09:24.16 | ectospasm | Unless you know what you're doing, only have one peer definition for each trunk/endpoint, with type=friend |
| 09:24.57 | Tuju | all i have here are friends. |
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| 09:37.52 | Tuju | http://foorum.elion.ee/foorum/viewtopic.php?f=43&t=20789 that is actually the isp i'm trying to use |
| 09:38.17 | Tuju | i've a feeling that at some point they have sanded some gears intentionally. |
| 09:39.10 | Tuju | ectospasm: does the 'register' line anyhow depend on related [line] ? |
| 09:39.41 | ectospasm | it should, I don't remember (and don't have any SIP trunks to look at right now) |
| 09:42.07 | ectospasm | Tuju: instead of proxy.example.com, try sip_proxy, matching the peer definition below [sip_proxy] |
| 09:43.21 | Tuju | ack. btw, i now commented out the whole 'register =>' line and keep getting the same 403 Forbidden error :-( |
| 09:43.29 | Tuju | so it's not related to that line at all. |
| 09:43.37 | ectospasm | it must not like that password |
| 09:43.38 | Tuju | regardless of that, i see the damn line in sip show peers. |
| 09:43.45 | ectospasm | ...or your account/username |
| 09:43.47 | Tuju | with OK |
| 09:47.11 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
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| 09:55.03 | ectospasm | do you see the REGISTER requests in sip debug? < Tuju |
| 09:56.44 | Tuju | i'm not sure. |
| 09:57.07 | Tuju | i've set the debug only to that peer, but i keep getting debug messages from other phones too |
| 09:57.20 | Tuju | http://pastie.org/7742080 this is something certainly from that problem line. |
| 09:57.44 | Tuju | that 'tuju.fi' is of course 'example.com' :) |
| 09:58.22 | Tuju | is that Cseq the line that shows what it was trying to do? |
| 09:58.34 | *** join/#asterisk polysics (~Adium@host27-146-dynamic.24-79-r.retail.telecomitalia.it) |
| 09:58.39 | polysics | hello |
| 09:58.55 | polysics | anyone can point me to ConfBridge docs for 1.8, please? |
| 09:59.19 | polysics | I would like to find out what "admin mode" means and stuff like that. |
| 09:59.48 | ectospasm | polysics: wiki.asterisk.org? |
| 10:00.06 | polysics | been digging on there but there's no explanations :-) |
| 10:00.10 | polysics | https://wiki.asterisk.org/wiki/display/AST/Application_ConfBridge |
| 10:00.11 | Tuju | Really destroying SIP dialog '4604cad11ac8043b6030f0006546e9b3@example.com' Method: OPTIONS |
| 10:00.29 | ectospasm | Tuju: that's for an OPTIONS request, as it clearly states. |
| 10:00.52 | Tuju | that's the part when sip protocol tries to negotiate all connection parameters? |
| 10:00.52 | ectospasm | Tuju: it may be easier to capture the traffic with tcpdump/wireshark |
| 10:01.00 | Tuju | yep |
| 10:01.47 | polysics | it looks like the only way is to indirectly do so by hanging up the channel |
| 10:02.43 | ectospasm | um, no |
| 10:03.07 | ectospasm | Try a sip reload, after changing one of the SIP peer parameters. |
| 10:03.48 | polysics | ectospasm: you mean, kicking an user out of a conference? |
| 10:05.01 | polysics | I'd be happy with something as small as an explanation fof "admin mode" :-) |
| 10:05.12 | Tuju | ectospasm: http://forums.whirlpool.net.au/archive/740107 i removed the qualify and got rid of that 403 Forbidden. |
| 10:07.49 | ectospasm | Tuju: was the 403 Forbidden for the OPTIONS? That is harmless, you should have left it |
| 10:08.19 | Tuju | yes, well if it doesn't work. |
| 10:08.29 | Tuju | but the register line doesn't work. |
| 10:09.26 | ectospasm | How do you know? Do you see any REGISTER packets being sent? |
| 10:09.38 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.78) |
| 10:10.35 | polysics | so, no one has EVER used ConfBridge? :-D |
| 10:13.02 | Tuju | ectospasm: no, i just set the tcpdump up |
| 10:13.40 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 10:16.34 | *** join/#asterisk bungle_ (5181467c@gateway/web/freenode/ip.81.129.70.124) |
| 10:16.49 | bungle_ | hello |
| 10:19.43 | bungle_ | ive spent about a week trying to get to grips with asterisk and read many posts - but i cant figure out how to setup a trunk to SPA3102 - i have the asterisk gui installed but am also happy to work in cli - can anyone give me some basics - good links etc? |
| 10:20.52 | ectospasm | ~thebook |
| 10:20.53 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:20.59 | ectospasm | bungle_: ^ |
| 10:21.38 | polysics | +1 on the book ,it is great |
| 10:22.13 | bungle_ | great - many thanks |
| 10:24.02 | polysics | sorry for reposting, but does anyone know how admin mode in ConfBridge 1.8 works, please? |
| 10:24.13 | bungle_ | how can i tell if dahdi is installed - i dont recall doing this - but installation sections all seem to say install dahdi first - is there a way to check if its installed? |
| 10:24.40 | ectospasm | bungle_: dahdi show channels in CLI |
| 10:24.52 | ectospasm | or dahdi_hardware in Linux shell |
| 10:25.07 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.smb.curriegrad2004.ca) |
| 10:27.21 | bungle_ | dahdi show channels - says no such command - so looks as though its not installed |
| 10:28.26 | ectospasm | bungle_: all that means is that chan_dahdi.so is not loaded |
| 10:28.42 | ectospasm | if you don't have DAHDI installed, chan_dahdi.so won't be built |
| 10:29.09 | ectospasm | unless you have a PSTN adapter driven by DAHDI, you don't need to worry about it |
| 10:29.30 | bungle_ | i see - yes its not there - so no dahdi |
| 10:30.27 | Tuju | amazing how those isp's keeping hiding simple things like username, password and proxy strings. |
| 10:30.28 | bungle_ | ahh ok - i have an SPA3102 - ive managed to have it register and forward calls (kind of) |
| 10:30.32 | Tuju | how-hard-can-it-be? |
| 10:31.00 | Tuju | i've found 5-10 different settings (of course they keep changing 'em along the years) |
| 10:31.19 | bungle_ | so i wont need dahdi to set up a trunk to the SPA3102? |
| 10:32.02 | ectospasm | what do you mean "trunk?" I think you're misusing the word |
| 10:32.14 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 10:33.01 | *** join/#asterisk sekil (~sekil@196.219.197.5) |
| 10:33.23 | ectospasm | SPA3102 is a SIP endpoint, yes? |
| 10:33.34 | ectospasm | just define an appropriate peer in sip.conf |
| 10:35.21 | bungle_ | yes - im getting confused between the terms in the gui - i basically want to set up a 'channel' or trunk? between the SPA and asterisk so that incoming calls on the pstn are handled by asterisk etc. I managed to have the SPA register with asterisk as a device - defined in sip.conf but nothing.... ahh ok... it was ok how i set it up then? |
| 10:36.02 | bungle_ | omg my head spins with this - it must be the single hardest thing ive ever tried to set up :) |
| 10:38.09 | bungle_ | i managed to set it up as a sip endpoint and direct all incoming calls the next problem was how to handle calls - ie. if i rejected the call on the softphone the pstn did not hang up etc.i assume this was in the dial plans - and would like to use the gui to help me out with those for now - but the SPA does not appear as a user etc.... |
| 10:38.16 | ectospasm | AsteriskGUI is unmaintained |
| 10:38.20 | ectospasm | Unless you mean FreePBX |
| 10:38.28 | ectospasm | and then, you should go to #freepbx |
| 10:39.09 | bungle_ | no i am using the asterisk 2 gui |
| 10:39.20 | ectospasm | wtf is that? |
| 10:40.42 | bungle_ | lol - i though thats what it was called lol - its not...sorry - GUI-version : SVN--r5219M |
| 10:40.54 | izbushka | hi |
| 10:41.21 | bungle_ | hi |
| 10:41.59 | *** join/#asterisk carrar (~tim@osburn.com) |
| 10:43.11 | ectospasm | bungle_: is that the AsteriskGUI, from Digium? If so, that is no longer maintained and you should NOT use it. |
| 10:47.19 | bungle_ | yes it is - i didnt know that :( thanks so its been replaced by freepbx? |
| 10:48.24 | ectospasm | not really, it lost the GUI battle to FreePBX |
| 10:51.00 | izbushka | is it ok to Playback() a probably nonexistent file or I should better check for it existence first? |
| 10:52.03 | bungle_ | wow - ok decided to start afresh just wiped all asterisks folders - what would you recomend as way forward - im trying to set up asterisk on a NAS device that has linux running on it |
| 10:52.58 | bungle_ | it comes with a 1.4.22.1b version of asterisks ready to install |
| 10:55.11 | ectospasm | Asterisk 1.4 is EOL (End Of Life) |
| 10:55.17 | ectospasm | better go with something more recent |
| 10:55.39 | ectospasm | do you have shell access to this NAS? |
| 10:55.44 | bungle_ | yes |
| 10:56.54 | ectospasm | bungle_: can you get a complete build environment on it? If so, just build Asterisk from source |
| 10:58.23 | bungle_ | yes if you mean ./config make etc. but im kind of out of depth and have a hard time following errors - when they inevitably happen - but you think its worth it? |
| 11:04.50 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 11:09.22 | ectospasm | yes. You won't get support for Asterisk 1.4 here |
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| 11:24.55 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 11:57.32 | polysics | re-re-post: anyone can please explain how the "admin" feature works in Confbridge on 1.8? Can an admin boot a person from the room from DTMF or does it require AMI? |
| 12:00.20 | polysics | undocumented functionality ftl :-) |
| 12:03.22 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:06.15 | kaldemar | polysics: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 12:09.24 | kaldemar | polysics: for the DTMF, see [sample_admin_menu] in the sample confbridge.conf |
| 12:15.18 | *** join/#asterisk davlefouAMD (~david@197.15.91.132) |
| 12:18.38 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:20.23 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
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| 12:40.19 | polysics | kaldemar: does 1.8 ConfBridge do the sam things, more or less? |
| 12:48.05 | polysics | kaldemar: oh, so there is no "stock" way to kick a person unless you build a menu for it first? |
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| 13:02.20 | Tuju | ectospasm: how come i can list one line as OK in sip show peers, but if i call to it, i'm said that 'it's not available' |
| 13:02.41 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 13:03.04 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:03.04 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:03.37 | Tuju | ectospasm: if i put the same line into nokia n9, it works. |
| 13:04.47 | [TK]D-Fender | Tuju, You should probably be showing us the complete call with SIP DEBUG enabled alonw with the dump of "sip show peers" and "sip show peer X" |
| 13:04.51 | [TK]D-Fender | ~pb |
| 13:04.52 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:04.55 | [TK]D-Fender | Tuju, ^^^ |
| 13:06.47 | Tuju | [TK]D-Fender: i try to collect some debug, there is just so much of it |
| 13:10.51 | *** join/#asterisk tedstriker (~textual@host-135-196-33-208.lines.viateldsl.com) |
| 13:13.12 | igcewieling | Tuju: if you are using FreePBX try the #FreePBX channel. |
| 13:13.26 | Tuju | ack. well i'm not. |
| 13:13.34 | Tuju | why would i be here then? |
| 13:13.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.79) |
| 13:14.03 | igcewieling | Tuju: some people think #asterisk is 2nd tier (or even 1st tier) support for #FreePBX |
| 13:14.46 | jmetro | what are tiers? #asterisk is my support for everything IT |
| 13:14.57 | igcewieling | We call those people "idiots". I only mention it because you said there was a lot debug, and FreePBX is known for its complex dialplan and lots of CLI output 8-) |
| 13:15.36 | igcewieling | pushes jmetro in front of the 3:10 to Yuma |
| 13:16.33 | igcewieling | ....er..I mean Hey there jmetro! |
| 13:17.29 | *** join/#asterisk Sacrimi (~goury@85.93.149.62) |
| 13:17.37 | jmetro | :3 |
| 13:17.41 | Sacrimi | ello |
| 13:17.51 | Sacrimi | i have the problem: |
| 13:18.09 | Tuju | what is the command that shows is a line registered or not? |
| 13:18.12 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 13:18.18 | jmetro | sip show peers like "expected peername" |
| 13:18.21 | [TK]D-Fender | Tuju, "sip show peer X" |
| 13:19.04 | Sacrimi | ringing group settings have no effect on order for incomming calls |
| 13:19.07 | Sacrimi | =( |
| 13:19.20 | Tuju | <PROTECTED> |
| 13:19.21 | Sacrimi | im using freepbx with centos |
| 13:19.26 | [TK]D-Fender | Sacrimi, Your terminology is unclear... |
| 13:19.31 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-edfdqvbfwgvmlobk) |
| 13:19.31 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:19.43 | [TK]D-Fender | Tuju, ADDRESS <- |
| 13:19.45 | Tuju | [TK]D-Fender: why you have those brakets in your name, it's hard to write 'em with other than us layout. |
| 13:19.52 | Tuju | hmmm |
| 13:19.53 | [TK]D-Fender | Sacrimi, #freepbx <-------- |
| 13:20.03 | Sacrimi | i must make it call 1->2->3->4->5 but it calls in wrong order |
| 13:20.18 | Sacrimi | okay |
| 13:20.21 | jmetro | Tuju: if you type [ and hit your autocomplete button, TK is the only one with brackets. |
| 13:20.22 | Tuju | [TK]D-Fender: Addr->IP ? |
| 13:20.23 | igcewieling | Tuju: see what I mean. LOL! |
| 13:20.29 | [TK]D-Fender | Sacrimi, Change channels. This isn't the place for FreePBX support |
| 13:20.35 | Tuju | jmetro: yes, that [ is the hard one. |
| 13:20.55 | Tuju | i've estonian layout, i need to use altgr+8 |
| 13:21.01 | [TK]D-Fender | <[TK]D-Fender> Sacrimi, #freepbx <-------- |
| 13:21.03 | Tuju | twist my wrist |
| 13:21.06 | Sacrimi | what if i ask you to help me determine where is ringing policy determined by config files? |
| 13:21.18 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:21.34 | igcewieling | Sacrimi: The ringing policy is determined by the config FreePBX generates. |
| 13:21.36 | Sacrimi | cant grep anything useful in /etc/asterisk |
| 13:21.36 | [TK]D-Fender | Sacrimi, You don't go there unless you're a dev typically |
| 13:21.40 | jmetro | "ringing policy" might mean "dialplan" which would normally be "extensions.conf" but is most likely elsewhere because of how freepbx mangles the configs |
| 13:21.59 | Tuju | Addr->IP : 217.159.187.4:5060 does that mean that it's registered? |
| 13:22.09 | Tuju | that's from sip show peers |
| 13:22.18 | [TK]D-Fender | Tuju, LOOKS like it |
| 13:22.26 | [TK]D-Fender | Tuju, then again we don't see you looking at a CALL |
| 13:22.27 | jmetro | Tuju: typically having an IP does, yes, but dialing it is the real test too |
| 13:22.33 | Tuju | well, if i call to that line, it says it's unavailable. |
| 13:22.43 | [TK]D-Fender | Tuju, And everything less... is typically junk until then |
| 13:22.49 | Tuju | ack |
| 13:22.51 | [TK]D-Fender | "it says" means nothing to me. |
| 13:23.07 | Tuju | well, it was quite devasting for me :) |
| 13:23.14 | Tuju | i took it seriously. |
| 13:23.16 | igcewieling | Tuju: seeing the pastebin will tell us most of what we need. |
| 13:23.23 | [TK]D-Fender | Tuju, Try #psychology :) |
| 13:23.27 | Tuju | i try to dig one |
| 13:23.32 | jmetro | Tuju: the second you give d-fender a full sip debug of you making a call to the phone, he will solve the issue. |
| 13:23.55 | *** join/#asterisk suge (~SoOJ@unaffiliated/suge) |
| 13:24.18 | igcewieling | [TK]D-Fender is one of the Old Ones, like Cthulhu or Baal. |
| 13:24.31 | jmetro | Mephisto, the lord of pain. |
| 13:24.32 | igcewieling | He knows all, he sees all, and he'll tell all. |
| 13:24.46 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 13:25.06 | *** join/#asterisk roswell (roswell@62.69.14.137) |
| 13:25.07 | Tuju | http://pastie.org/7742777 i think this is a working dump packet |
| 13:25.21 | Tuju | there is, rport for example. |
| 13:25.31 | Tuju | nokia n9 doesn't show much how it uses the line. |
| 13:25.42 | [TK]D-Fender | Tuju, ASTERISK SIP debug with verbose 10 |
| 13:25.46 | igcewieling | Tuju: We don't see any dialplan lines there and 1 packet doesn't help. |
| 13:25.47 | [TK]D-Fender | Tuju, No substitutes... |
| 13:26.02 | [TK]D-Fender | Tuju, and that is a REGISTRATION, not a CALL |
| 13:26.06 | Tuju | sip set debug on ? |
| 13:26.09 | [TK]D-Fender | Tuju, Strike one.... |
| 13:26.13 | [TK]D-Fender | Tuju, Yes |
| 13:26.14 | Tuju | i'm going to drown to it. |
| 13:26.23 | jmetro | sip set debug on , core set verbose 999, core set debug 999 |
| 13:26.25 | [TK]D-Fender | Tuju, Breath it in. It's good for you |
| 13:27.24 | *** join/#asterisk BoRiS (~raiden@S010660a44cdcb910.wp.shawcable.net) |
| 13:27.51 | igcewieling | Tuju: you don't have to parse the data, [TK]D-Fender does. |
| 13:27.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:27.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:30.59 | Tuju | that's one problem, it doesn't seem to be sending those REGISTER packets from asterisk. |
| 13:31.12 | Tuju | if i set qualify=yes, it sends some OPTIONS ones. |
| 13:32.07 | Tuju | register => 372654321@elion.ee:123456@proxy.elion.ee i have this line in sip.conf |
| 13:32.46 | igcewieling | jmetro: why do you think people ask for help, we tell them what we need (cli output of a failed call) and then they blather about all sorts of stuff and don't provide the pastebin? |
| 13:33.00 | igcewieling | Odd, don't you think? |
| 13:35.35 | [TK]D-Fender | Tuju, We aren't looking for a registration. We are looking for THE CALL THAT FAILED |
| 13:36.21 | Tuju | [TK]D-Fender: there is nothing to look at it as i said. |
| 13:36.34 | Tuju | proxy in telco side doesn't know that my asterisk exists. |
| 13:36.38 | Tuju | no packets. |
| 13:36.41 | [TK]D-Fender | tutYou said it said it failed. That means it SAID SOMETHING |
| 13:36.52 | [TK]D-Fender | Tuju, So YOUR server said nothing |
| 13:36.57 | Tuju | YES, it says with VOICE, 'it's not available'. |
| 13:37.04 | [TK]D-Fender | Tuju, You need a better description next time |
| 13:37.06 | Tuju | my server says nothing. |
| 13:37.30 | Tuju | tcpdump doesn't show any packets from that proxy. |
| 13:37.39 | [TK]D-Fender | <[TK]D-Fender> "it says" means nothing to me. |
| 13:38.06 | [TK]D-Fender | Tuju, Then enable SIP debug, so "sip reload" and show us the attempt |
| 13:38.12 | jmetro | igcewieling: Language barriers for one, and B, people have their own idea of the problem and are usually trying to be helpful but as problem solvers, we are used to just finding our own problem gathering our own info and solving it. |
| 13:38.22 | jmetro | igcewieling: the hostility probably doesnt help. |
| 13:38.26 | Tuju | [TK]D-Fender: it's on, there is nothing to show from there. |
| 13:38.33 | igcewieling | Tuju: here is an example of what we are looking for. http://pastebin.ca/2370044 see all those sip packets and dialplan lines? That is what we are looking for. |
| 13:38.38 | Tuju | just flood of packetes from other devices. |
| 13:38.43 | [TK]D-Fender | Tuju, SIP RELOAD AND WATCH |
| 13:39.02 | igcewieling | jmetro: worse than trying to train a cat |
| 13:39.22 | roswell | shall we take a deep breath and Tuju, try asking again? ) |
| 13:39.32 | roswell | from a scratch |
| 13:39.34 | igcewieling | jmetro: and I have, in fact, trained my cat. |
| 13:39.51 | Tuju | it's not that i'm sitting with unix systems first time my life. |
| 13:40.12 | anonymouz666 | [TK]D-Fender: do you already configured audiocodes FXO gateway? |
| 13:40.25 | [TK]D-Fender | anonymouz666, Once in 2006 |
| 13:40.39 | Tuju | igcewieling: i understand that you want to see those dumps but there are non of those as there are no ip packets between those hosts, right? |
| 13:40.45 | Tuju | none |
| 13:40.58 | Tuju | or where those dumps would pop out then? |
| 13:41.19 | Tuju | those are, ip packets carrying sip packets, parsed in asterisk. |
| 13:41.35 | jmetro | Tuju: honestly i would just set everything to highest debug, start your log, and try to call the phone, and post the entire giant thing for them |
| 13:41.39 | igcewieling | Tuju: then explain to us again what EXACTLY the problem is? |
| 13:41.39 | Tuju | my tcpdump is silent. no packets. |
| 13:41.41 | jmetro | in a paste |
| 13:41.47 | jmetro | not tcpdump |
| 13:41.50 | jmetro | asterisk cli |
| 13:42.10 | Tuju | igcewieling: asterisk is not registering my telco. |
| 13:42.29 | Tuju | regardless that i've the 'register' line and related [<line>] section. |
| 13:42.41 | Tuju | and i can get it register with nokia n9 |
| 13:42.42 | igcewieling | I was thinking something like "when I try calling extension X from extension y it doesn't work" or "when I try calling a PSTN number via my ITSP from extension X it doesn't work" |
| 13:43.02 | Tuju | well don't think, read what i've been writing here. pun intended. |
| 13:43.32 | roswell | Tuju, which linux distro you run your asterisk on? |
| 13:43.44 | Tuju | it's not very constructive to be unpolite and keep shouting me when i've nothing to give you. |
| 13:43.46 | igcewieling | Tuju: if the only packet you see is an outgoing register than you either have a NAT issue or a networking issue. |
| 13:43.54 | Tuju | roswell: this is centos6 |
| 13:44.08 | Tuju | i've quite many, 10-15 phones working just fine. |
| 13:44.14 | igcewieling | even if your provider was refusing you, you would get a packet back with a 401 Unauthorized |
| 13:44.16 | Tuju | this is just first trunk i'm trying to set up. |
| 13:44.34 | Tuju | igcewieling: imo my side asterisk is not even trying to register. |
| 13:44.51 | Tuju | and imo i don't need nat as both boxes use native addresses. |
| 13:44.57 | Tuju | or do i? |
| 13:45.02 | roswell | Tuju, do you have wireshark installed? |
| 13:45.17 | roswell | not on the asterisk server, of course |
| 13:45.23 | Tuju | roswell: well, i'm not even in the same country where that centos and asterisk is. :) |
| 13:45.29 | igcewieling | Tuju: when you do a "sip reload" asterisk should try to register again |
| 13:45.36 | Tuju | igcewieling: i try again |
| 13:45.56 | Tuju | igcewieling: nothing, not a single packet. |
| 13:45.58 | roswell | Tuju, actually it's not a big trouble, unless you're offline ) |
| 13:46.08 | roswell | so, any wireshark around? |
| 13:46.13 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 13:46.16 | Tuju | roswell: i could dump into file and read it into shark yes. |
| 13:46.18 | igcewieling | also what does "sip show registry" show? |
| 13:46.32 | Tuju | igcewieling: nothing, that was the startingpoint of my problems. |
| 13:46.38 | Tuju | sip show peers did show OK |
| 13:46.49 | igcewieling | then pastebin your sip.conf, masking ONLY the secret= line. |
| 13:46.51 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 13:47.05 | igcewieling | sip show peers has nothing whatsoever to do with registering to a remote server from asterisk |
| 13:47.06 | Tuju | but call lady in the central switch said that "device is not available" |
| 13:47.31 | igcewieling | Tuju: it would be easier to debug if you were trying to call out from asterisk, not trying to call into asterisk |
| 13:48.02 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 13:48.23 | *** join/#asterisk jkroon (~jkroon@41.17.28.157) |
| 13:48.36 | Tuju | http://pastie.org/7742884 this is the line config |
| 13:48.41 | Tuju | changed number and pwd |
| 13:48.51 | roswell | Tuju, great. let's dump *all* traffic down this way: on asterisk end, enter 'tcpdump -n -i any -w /somewhere/to/dump' , issue 'core restart when convenient' onto asterisk cli, wait for 20 seconds, then Ctrl+C tcpdump |
| 13:49.01 | [TK]D-Fender | Tuju, that has NOTHING to do with a registration |
| 13:49.24 | roswell | oh i meat *wait for 20 seconds* when cli goes offline |
| 13:49.32 | Tuju | register => 372654321@elion.ee:123456@proxy.elion.ee is the other line. |
| 13:49.50 | igcewieling | can you ping proxy.elion.ee from the asterisk server? |
| 13:49.56 | Tuju | roswell: wait a second.... |
| 13:50.09 | roswell | igcewieling, nice guess btw ) |
| 13:50.36 | Tuju | igcewieling: nope, but that could mean that they block icmp echo-reply's etc. |
| 13:51.09 | igcewieling | Tuju: can you do a "host proxy.elion.ee" or an "nslookup proxy.elion.ee" |
| 13:51.10 | Tuju | shouldn't that register line be enough to connect into trunk? |
| 13:51.24 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.6.230) |
| 13:51.29 | Tuju | igcewieling: sure, it's proxy.elion.ee has address 217.159.187.4 |
| 13:51.56 | Tuju | and that damn isp has like five zillion different proxy names in instructions. |
| 13:52.08 | Tuju | but i saw that nokia n9 uses that one and it works. |
| 13:52.19 | igcewieling | I wish you the best of luck. |
| 13:52.24 | Tuju | :D |
| 13:52.45 | [TK]D-Fender | Tuju, Three is also a difference between proxy IP's and target server IP's. |
| 13:52.48 | [TK]D-Fender | There* |
| 13:53.05 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.124.84) |
| 13:53.09 | Tuju | you mean that it could redirect it in initial registration? |
| 13:54.14 | [TK]D-Fender | It could reach the proxy and not know where to forward it to. |
| 13:54.34 | igcewieling | Tuju: it is far easier to test if you try dialing out from your asterisk box. You do not need to be registered to do that. |
| 13:54.49 | Tuju | i need to move one phone into same context |
| 13:54.57 | Tuju | i have never done it. :) |
| 13:55.26 | *** join/#asterisk andrewyager (~andrewyag@1.144.72.51) |
| 13:55.30 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 13:55.37 | *** join/#asterisk vlad_st__ (~vlad_star@109.188.125.102) |
| 13:55.48 | roswell | registration, in terms of asterisk, is needed only to accept incoming calls. do you have some according [peer] section? |
| 13:56.22 | roswell | however... whatever... |
| 13:56.43 | Tuju | how do i route all numbers into that trunk? |
| 13:57.13 | roswell | exten => _X!,Dial(trunk) |
| 13:57.54 | Tuju | ack |
| 13:58.13 | Tuju | 's socks are spinnning |
| 13:58.48 | roswell | incomplete, though... exten => _X!,Dial(trunk/${EXTEN}) ; this shall be more politically correct |
| 13:59.29 | [TK]D-Fender | roswell, horrible pattern.... and still invalid syntax |
| 13:59.43 | Tuju | ....rejected because extension not found in context 'incoming'. |
| 14:00.01 | roswell | right, [TK]D-Fender |
| 14:00.03 | [TK]D-Fender | <roswell> registration, in terms of asterisk, is needed only to accept incoming calls. do you have some according [peer] section? <- not really accurate |
| 14:00.46 | roswell | would you correct me? |
| 14:00.53 | [TK]D-Fender | Registration is when you need to auth to a server before the care to send you calls, and/or when they also need to know what IP to send the calls to |
| 14:01.09 | Tuju | still the same. |
| 14:01.48 | Tuju | exten => _X!,Dial(372654321/${EXTEN}) is that correct? |
| 14:01.54 | roswell | Tuju, no |
| 14:01.56 | Tuju | context is right, it triggers it |
| 14:03.00 | roswell | what's it trigger i wonder |
| 14:03.05 | Tuju | exten => _X!,Dial(trunk/${372654321}) is that correct? |
| 14:03.14 | Tuju | nope, it can't be |
| 14:03.24 | roswell | Tuju, exten => _X!,1,Dial(trunk,${EXTEN}) ; [TK]D-Fender was right i've forgot about precedence |
| 14:03.29 | [TK]D-Fender | PRIORITY <---------------------------------------------------------------- |
| 14:03.49 | Tuju | and i replace 'trunk' with my line number? |
| 14:04.47 | Tuju | http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx exten => _9XXXX.,n,Dial(SIP/fooprovider/${EXTEN:1},60) |
| 14:04.53 | roswell | Tuju, with a sip peer or friend id |
| 14:05.18 | roswell | preceded with SIP/ |
| 14:05.39 | [TK]D-Fender | Tuju, exten => _XXX.,1,Dial(SIP/372654321/${EXTEN}) |
| 14:06.49 | roswell | fyi, [TK]D-Fender , _XXX. is a bit overflew in this specific case, don't you find? |
| 14:07.29 | *** join/#asterisk tedstriker (~tedstrike@host-135-196-33-208.lines.viateldsl.com) |
| 14:07.35 | roswell | afair he wanted to redirect any number to an extension, which _X! does well |
| 14:07.42 | [TK]D-Fender | roswell, Do you think "22" is something valid to dial to the PSTN anywhere? |
| 14:08.22 | [TK]D-Fender | roswell, "any number" is vague crap that bits you in the ass and usually there are exceptions .... like wanting to do INTERNAL THINGS |
| 14:08.40 | roswell | ain't he doing some internal things? |
| 14:09.00 | [TK]D-Fender | roswell, Don't use stupid patterns. Ever. |
| 14:09.12 | roswell | sighs |
| 14:09.58 | roswell | according to yours, 2222 would be valid either |
| 14:10.27 | [TK]D-Fender | exten => _XX.,1,Dial(SIP/372654321/${EXTEN}) |
| 14:10.30 | [TK]D-Fender | There, better |
| 14:10.50 | roswell | ok let's quit bitching ) |
| 14:11.04 | [TK]D-Fender | That does cover several technically valid standard #'s (depending on geography, etc) |
| 14:11.07 | roswell | i've only applied the pattern to this case |
| 14:12.44 | Tuju | for some reason, my context is empty. |
| 14:12.58 | Tuju | regardless that i've those under [incoming] in extensions.conf |
| 14:13.56 | roswell | Tuju, what's say context= in [general] section of sip.conf? |
| 14:14.27 | Tuju | context = default ; Default context for incoming calls |
| 14:14.45 | Tuju | but i've those two lines with context=incoming |
| 14:14.58 | Tuju | dialplan show is full of some ael crap. |
| 14:15.09 | Tuju | i tried to clean 'em out already, but still a lot |
| 14:15.23 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 14:18.50 | Tuju | how do i disable pbx_ael.so ? |
| 14:19.20 | jmetro | remove the module from your module list? |
| 14:20.14 | mmlj4 | or rename the file |
| 14:26.18 | Tuju | i looked the list but grep didn't find it |
| 14:26.19 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 14:26.21 | Tuju | rejected because extension not found in context 'incoming'. |
| 14:26.33 | Tuju | so the line is not registered i presume. |
| 14:29.17 | jmetro | that error means that you have an extension that doesnt match your dialplan |
| 14:29.37 | jmetro | like if i dial 1234 in my system it saids extension 1234 not found in incoming |
| 14:33.00 | Tuju | so it has nothing to do with line then. |
| 14:33.06 | jmetro | right |
| 14:33.15 | jmetro | check your dialplan for the extension it said it was missing |
| 14:34.25 | Tuju | ha! Received response: "Forbidden" from '"Juha Tuomala" <sip:372654321@elion.ee>;tag=as7ca297e6' |
| 14:34.33 | Tuju | i'm not welcome - i see. |
| 14:34.37 | Tuju | now the sip debug |
| 14:38.50 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:38.50 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:40.15 | *** join/#asterisk tamiel (~tamiel@ip-38.net-81-220-93.rev.numericable.fr) |
| 14:41.31 | Tuju | http://pastie.org/7743220 is that an attempt to call to number 300 ? |
| 14:41.51 | Tuju | no, to 900972597604759 |
| 14:41.54 | jmetro | From: 300<sip:300@88.114.107.83>;tag=4458854b To: 900972597604759<sip:900972597604759@88.114.107.83> |
| 14:42.04 | jmetro | from 300 to [that long number] |
| 14:42.16 | Tuju | some asswipe is trying to make proxy to call somewhere. |
| 14:43.25 | Tuju | jmetro: is that attempt authenticated into proxy, can it be seen from that? |
| 14:44.18 | Tuju | jmetro: how can i prevent that non-registered can't initiate invites? |
| 14:44.39 | jmetro | send them into deadend context |
| 14:44.47 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:45.14 | Tuju | would it help if all my own lines would be in own context? |
| 14:45.22 | Tuju | my friend once said that he woke up when some wanker was running a script against that phone. :) |
| 14:45.29 | jmetro | set default to deadend and set your registered ones to Incoming, yes. |
| 14:46.34 | Tuju | ack, i'll do that. |
| 14:46.52 | Tuju | but - howcome those attempts come from local isp's ip-address space? |
| 14:47.19 | Tuju | is such cisco desktop phone somehow crackable that someone could use it as a proxy for those attempts? |
| 14:48.11 | Tuju | they all seem to have same Contact that implies that this one device is always involved. |
| 14:48.54 | jmetro | Cisco phones had a huge vulnerability recently |
| 14:49.11 | jmetro | you could turn any cisco phone into an always-on microphone and duplicate the hack onto all the phones in the network in about 5s |
| 14:50.34 | Tuju | that's sweet. |
| 14:50.51 | Tuju | i need to let him know to turn that crap off. |
| 14:58.54 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.7.224) |
| 14:59.28 | *** join/#asterisk SpiderMon (~SpiderMon@68.152.22.22) |
| 15:00.18 | beardy | jmetro: In which image? The SIP image too? |
| 15:00.25 | *** join/#asterisk butthurtface (~Butthurtf@38.122.108.2) |
| 15:02.18 | *** join/#asterisk Free99 (~Free99@ool-4350dd5c.dyn.optonline.net) |
| 15:05.40 | Tuju | http://pastebin.com/UdQDDp2m there is that error i've been hunting all day. |
| 15:06.20 | jmetro | http://arstechnica.com/security/2013/01/hack-turns-the-cisco-phone-on-your-desk-into-a-remote-bugging-device/ |
| 15:07.00 | Tuju | maybe it's related that it has java crap inside. |
| 15:07.51 | Tuju | maybe i don't tell my friend that it can be turned into remote mic. :) |
| 15:10.22 | Tuju | jmetro: about that context thing yet, if i move my terminals into own context, is there an easy way to still receive calls like emails without making any settings beforehand between two domains? |
| 15:12.43 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:13.23 | jmetro | i have 3 context, dead end, incoming, and internal[also handles outbound] |
| 15:13.37 | jmetro | your domain is the only thing in your incoming context |
| 15:13.42 | jmetro | er no..not your domain |
| 15:13.46 | jmetro | your...itsp |
| 15:14.06 | jmetro | all your phones go in internal/outbound |
| 15:15.08 | Tuju | are those contexts somehow chained? |
| 15:15.13 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 15:15.40 | jmetro | itsp says "call for [your phone number] |
| 15:16.01 | jmetro | my extensions.conf goes [my phone number] -> [company1-extensions] |
| 15:16.16 | jmetro | company1-extensions is an AA, or dialing, etc.. |
| 15:16.50 | jmetro | company1-internal is outbound and rules for dialing internal extensions [like 100 to reach ext 100, or 200 to open the front door, etc.] |
| 15:16.57 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.138) |
| 15:17.11 | Tuju | ack, i need to dive into that later with more thoughts. |
| 15:17.16 | jmetro | so eventually incoming can hit a rule that says Dial(100 @ company1-internal) |
| 15:17.23 | Tuju | i'm looking into that error now. hairy. |
| 15:17.33 | *** join/#asterisk tedstriker (~tedstrike@host-135-196-33-208.lines.viateldsl.com) |
| 15:17.39 | Tuju | copies above into file. |
| 15:18.14 | Tuju | what line has the name and 'realm' or domain that is used in authentication? |
| 15:18.24 | Tuju | Contact: ? |
| 15:18.40 | jmetro | i dont know what that is |
| 15:19.01 | Tuju | cause at least that i had wrong if that is used for auth |
| 15:19.17 | Tuju | if it is From: that is used, then it was correct. |
| 15:19.35 | jmetro | are you still debugging your auth to your itsp>? |
| 15:19.47 | Tuju | yes, it still doesn't work. |
| 15:19.52 | jmetro | because i bet if you called them they could tell you what they are seeing and what you should have |
| 15:20.17 | Tuju | well, that's the point - it's teliasonera these days. they hate this service. |
| 15:20.19 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:20.19 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:20.22 | Tuju | and don't want to help. |
| 15:20.36 | Tuju | it's eating their core business. |
| 15:21.23 | jmetro | they still offer customer service htough |
| 15:21.29 | *** join/#asterisk navaismo (~navaismo@189.241.51.199) |
| 15:21.30 | jmetro | you might have to fight to reach a guy but the guy will help you |
| 15:22.39 | Tuju | i have talked them before. |
| 15:22.55 | Tuju | they nowdays charge every minute you talk to them. |
| 15:23.07 | Tuju | it all smells and far. |
| 15:23.11 | jmetro | for service you are already buying? |
| 15:23.25 | jmetro | if you are transmitting the right username / password its not your fault |
| 15:25.08 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 15:26.07 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 15:27.29 | Tuju | i mean if i ask their help, they ask money. |
| 15:28.08 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 15:28.14 | Tuju | i mean if i ask their help, they ask money. |
| 15:28.29 | jmetro | well youre not asking for help, youre asking for them to fix their problem =) |
| 15:31.27 | Qwell | umm, if your ITSP isn't helping you, maybe it's time to find a new one? |
| 15:34.57 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:35.32 | Tuju | Qwell: yeah, like there were that many choices here. |
| 15:35.48 | Tuju | the one that has landlines, gives sip line for free. |
| 15:36.06 | Tuju | imo all that bundling should be banned. |
| 15:36.16 | *** join/#asterisk PickledParrot (~aj@79.135.102.10) |
| 15:37.08 | Tuju | <PROTECTED> |
| 15:38.53 | Tuju | food |
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| 16:22.42 | *** join/#asterisk gbit86_ (~gbit86@204.11.31.54) |
| 16:23.31 | gbit86_ | I need some assistance with a Switchvox by Digium. It is supposed to be based on Asterisk, but unfortunately it is a bit closed, so it is making my development with its API somewhat difficult |
| 16:24.02 | igcewieling | Switchvox support is provided by Digium support. We can't support it here since it is such a closed system |
| 16:24.03 | gbit86_ | Right now I am simply trying to authenticate against it via Digest Authentication and using Nodejs |
| 16:24.13 | mmlj4 | you have a device, in other words? |
| 16:24.32 | mmlj4 | but their "support" is only going to be helpful to you if you pay them |
| 16:24.33 | gbit86_ | ah… well then I guess I am just screwed because I have contacted them 3 times now and they don't want to support their own system.. this is great |
| 16:25.01 | gbit86_ | well we have silver support I believe it is and well it is a waste.. if you don't buy gold you have nothing. |
| 16:25.08 | mmlj4 | if you want to develop stuff, use vanilla asterisk, not some packaged system |
| 16:25.25 | gbit86_ | agreed.. but I wasn't part of this decision as is often the case |
| 16:25.30 | _Corey_ | gbit86_: Contact the partner who sold you the solution... they may have additional support capabilites |
| 16:25.54 | gbit86_ | lol.. if only you knew the partner they purchased it from |
| 16:26.10 | gbit86_ | normally that would have been a good suggestion though |
| 16:26.19 | _Corey_ | lol, I understand |
| 16:26.46 | _Corey_ | Just ask for another partner in your area, I'm sure Digium can accommodate |
| 16:26.48 | gbit86_ | screw them.. I guess I am just going to enable root access to this device |
| 16:27.30 | gbit86_ | there is a way to do that, I read about someone that did it awhile back but I didn't think I would need to pursue that option |
| 16:28.20 | _Corey_ | gbit86_: You'll be violating the license agreement if you go that route... |
| 16:28.34 | gbit86_ | I'm used to that |
| 16:28.45 | igcewieling | mmlj4: Switchvox is not some packaged asterisk, it is a closed, custom, commercial Astersik |
| 16:28.46 | gbit86_ | but I also get shit done |
| 16:29.09 | mmlj4 | worse than I thought, then |
| 16:29.21 | igcewieling | totally unsupportable by regular asterisk people |
| 16:29.40 | jmetro | from what i know, it looks like its the only asterisk with a real working GUI too |
| 16:30.05 | mmlj4 | I haven't used any GUI tool, so I can't say |
| 16:30.36 | gbit86_ | this is the 2nd closed solution the company I work for has gone with, the other was based on freeswitch |
| 16:31.54 | gbit86_ | in both cases these companies like to act like glare issues on the PRI don't exist and were very incompetent when it came to fixing it, where as in asterisk it would have been a simple parameter to flip and it is a fairly well known issue online |
| 16:32.10 | gbit86_ | needless to say I am very unimpressed with these closed solutions |
| 16:37.08 | Tuju | igcewieling: you saw the debug listing? |
| 16:37.26 | igcewieling | Tuju: I cannot help you further |
| 16:37.48 | Tuju | howcome? you all pressed so fiercly for that trace. |
| 16:37.52 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 16:38.10 | igcewieling | Tuju: then I gave up and now have real work to do. |
| 16:39.15 | Tuju | i understood that it would have been obvious small thing once that debug is provided. |
| 16:39.20 | navaismo | ignore list?? |
| 16:39.46 | jmetro | navaismo: how did uninstalling the ffvmpeg thing screw up your box? |
| 16:40.35 | igcewieling | *** navaismo has been added to Ignore List |
| 16:40.37 | navaismo | no, uninstalling ffmpeg-devel fix the issue |
| 16:40.40 | igcewieling | navaismo: your request is complete. |
| 16:40.43 | igcewieling | 8-) |
| 16:40.45 | navaismo | ¬¬ |
| 16:40.57 | jmetro | navaismo: so i was right =D |
| 16:40.58 | igcewieling | navaismo: I'd not add you to my ignore list. |
| 16:42.25 | gbit86_ | so I am going to download the free home version of the switchvox first and see if I can just spin up a vm. I should be able to understand how digest authentication is being hashed from there and get the answers to all of my questions. |
| 16:42.36 | navaismo | jmetro, I was testing a webrtc freepbx module with sipml5, and I updated my asterisk version, my webrtc2sip gateway version and do alot compiling with ffmpeg codecs and stuff then suddenly asterisk wont compile and the system have issues because i compiled fmmpeg and didnt remove the package from repos |
| 16:42.39 | butthurtface | Any of y'all ever run into this problem with Playback? pbx.c:3680 pbx_extension_helper: No application 'Playback ' for extension |
| 16:42.41 | navaismo | so basically silly me.... |
| 16:43.15 | _Corey_ | gbit86_: I think the API may be nonexistant on the home version... Could be wrong, but I'd check before investing too much time |
| 16:43.40 | *** join/#asterisk gbit86__ (~gbit86@204.11.31.54) |
| 16:43.41 | navaismo | butthurtface, seems like playback is not compiled/loaded |
| 16:43.55 | igcewieling | gbit86_: SIP Digest should be VERY similar to HTTP Digest auth, you might check for information on that. |
| 16:44.21 | igcewieling | butthurtface: stop putting extra spaces in your exten lines |
| 16:44.29 | navaismo | hehe |
| 16:44.43 | butthurtface | I wondered that too but Playback(intro) works fine when the system picks up the call. It's just one line it's having trouble too. |
| 16:44.45 | igcewieling | There is no application Playback<space> in Asterisk. |
| 16:44.47 | navaismo | i always used to do that, its a grammar rule |
| 16:46.23 | butthurtface | Ohhhhhh I see. |
| 16:46.26 | butthurtface | Let me try to see that. |
| 16:46.31 | butthurtface | Watch, I'm an idiot- this will work now. |
| 16:46.42 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:46.48 | butthurtface | bahaha yep. I'm an idiot. |
| 16:47.09 | butthurtface | Oh man I didn't even see the blank space in 'Playback ' |
| 16:47.36 | igcewieling | butthurtface: some of us have been using Asterisk for a VERY long time. |
| 16:48.37 | jmetro | im so used to coding in really terrible languages [c++] that i never use spaces anymore |
| 16:48.43 | butthurtface | Haha |
| 16:49.48 | bungle_ | hi, can someone help me understand some basic concepts with asterisk? im trying to setup an SPA3102 - given up installing things via gui and starting out again with a fresh asterisk installation |
| 16:50.24 | navaismo | bungle_, ask specific questions please |
| 16:50.52 | Qwell | ~book |
| 16:50.53 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:52.07 | *** join/#asterisk gbit86_ (~gbit86@204.11.31.54) |
| 16:52.56 | Tuju | does the protocol allow making INVITE before REGISTER? |
| 16:53.14 | navaismo | i choose yes |
| 16:53.29 | igcewieling | Tuju: the ONLY thing registration does is inform the far server of your current IP address. It has nothing to do with ANYTHING else. |
| 16:53.53 | igcewieling | none of our endpoints register |
| 16:54.10 | Tuju | if i compare my n9 session to telco, it does first REGISTER with few pkgs and it works. |
| 16:54.15 | Tuju | then it makes call and it works. |
| 16:54.30 | bungle_ | i think i can set the SPA3102 up as a sip endpoint using sip.conf and then asterisk will handle all the calls using extensions.conf? |
| 16:54.30 | Tuju | asterisk has this register => line, but it never REIGSTERs |
| 16:54.42 | igcewieling | bungle_: that information is in the Asterisk book |
| 16:55.50 | joesuffceren | any ideas on querying CDR (mysql) to get highest number of simultaneous DAHDI calls in a given time period? |
| 16:56.29 | talntid | that query would be pretty intensive |
| 16:56.31 | bungle_ | thanks igcew - yes, im trying to get to grips with it - theres so many new terms etc im trying to understand that im getting totally confused and wanted to get a basic overview again - i dont think the digium gui helpd because different terms used for users - trunks extensions etc. |
| 16:57.57 | joesuffceren | talntid: if you mean resource-intensive, that's no problem. I don't plan to run it on a regular basis. The problem I'm trying to solve is value proposition for transitioning from PRI to SIP. If there is a better way to get that historical info, I'm happy to hear it, as well |
| 16:58.46 | joesuffceren | I do have a script running now that writes the current number of concurrent DAHDI calls to a log once per minute, but I was hoping to be able to look at the last six months instead/in addition |
| 16:59.19 | joesuffceren | if you mean "pain in the butt to write and debug" then I agree. :-) |
| 17:00.40 | bungle_ | for example - outside connectivity - the SPA3102 is an ATA and the book suggests i will need dahdi? but if its connected as a sip device - i dont think i need dahdi? |
| 17:01.04 | [TK]D-Fender | bungle_, You don't ... for that |
| 17:01.12 | [TK]D-Fender | bungle_, You may need it for other things though |
| 17:02.37 | bungle_ | thanks TK, so i guess im unsure what to call the connection - ie. is it a VOIP trunk i need to setup - or simply have the SPA register as a SIP device? |
| 17:02.59 | bungle_ | i dont plan on adding any more anaolgue phones or pstn connections |
| 17:07.21 | [TK]D-Fender | bungle_, It is a SIP gateway. The term "VoIP trunk" is not great in most contexts. |
| 17:07.32 | [TK]D-Fender | bungle_, * talking to anything SIP is a peer like any other really. |
| 17:07.51 | [TK]D-Fender | bungle_, As to whether it registers to * is up to you. You can do it any way you want |
| 17:08.30 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:09.36 | bungle_ | ok,great - so the pstn line connects to * as a sip peer and then the dial plan decides what happens next? |
| 17:10.02 | [TK]D-Fender | bungle_, Yes. |
| 17:10.16 | [TK]D-Fender | bungle_, You tell the SPA to send the call to * and * does what you tell it to do |
| 17:11.34 | bungle_ | ok so for outgoing calls the diaplan on * tells the sip/pstn to call out |
| 17:12.04 | [TK]D-Fender | yes |
| 17:12.14 | bungle_ | wow i know that may sound very basic questions - but ive become swamped in information overload about channles/trunks/users/extensions etc.. |
| 17:12.35 | [TK]D-Fender | bungle_, Quick version : |
| 17:13.01 | [TK]D-Fender | SPA-3102 has 2 ports, 1 FXS, 1 FXO. The FXS is for boring phones like all the rest out there. You configure it pretty much independently of the other port |
| 17:13.15 | [TK]D-Fender | The other port (FXO) is for your lines. I has MULTIPLE routing options |
| 17:13.19 | [TK]D-Fender | This is something to watch for. |
| 17:13.39 | [TK]D-Fender | You can have IT as the caller where to go (I think it's really jsut a tone letting them dial without a real prompt). |
| 17:13.40 | [TK]D-Fender | OR... |
| 17:13.54 | [TK]D-Fender | you could just tell it "throw the call over to the registered server |
| 17:14.02 | [TK]D-Fender | Which is typically what you want to do |
| 17:14.40 | [TK]D-Fender | You send that FXO port calls almost the same way you would ring the FXS phone port on it, except for passing the # to dial out along-with |
| 17:14.45 | [TK]D-Fender | The End |
| 17:15.19 | bungle_ | i cant tell you how much that helps :-) thank you |
| 17:16.02 | [TK]D-Fender | bungle_, Getting it to toss the call over is pretty simple really |
| 17:16.19 | [TK]D-Fender | bungle_, this is a "dialplan line" on the SPA. Google up guides for the setup on them. |
| 17:16.41 | [TK]D-Fender | Pretty much a 5-minute job |
| 17:17.23 | bungle_ | hehe 5 min if you know - 4 days wading in treacle if you dont :-) |
| 17:17.33 | [TK]D-Fender | You can also have the SPA BRIDGE the FXS & FXS ports together in the event of a power failure so the connected phone rings direct. |
| 17:19.01 | bungle_ | i managed to get the SPA to register with * and forward all incoming calls - but when i answer on the softphone the caller got disconnected etc. is that handled in the dial plan? |
| 17:19.37 | [TK]D-Fender | If * got the call, and then continued to call out... and then you answer and THEN it fails then you probably have a networking issue |
| 17:19.47 | [TK]D-Fender | Or codec |
| 17:20.13 | [TK]D-Fender | bungle_, Enable sip debug "sip set debug on", "core set verbose 10" and pastebin the entire call from beginning to end. |
| 17:20.14 | [TK]D-Fender | ~pb |
| 17:20.15 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:20.16 | [TK]D-Fender | ^^^^ |
| 17:20.57 | bungle_ | ok, now i need some time to review where its up to - just done a reinstall of asterisk |
| 17:28.16 | Tuju | <PROTECTED> |
| 17:28.22 | Tuju | that's progress. |
| 17:29.47 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 17:30.03 | jmetro | ╘ o.o ╛ |
| 17:35.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.136) |
| 17:37.45 | igcewieling | Hmmm...my attempt to outwit the universe appears to have failed. |
| 17:38.50 | jmetro | igcewieling: i get that all the time trying to revive dead servers |
| 17:40.39 | igcewieling | jmetro: Anytime we announce an upgrade we suddenly get all sorts of reports of problems which are pre-existing and totally unrelated to the upgrade. |
| 17:40.49 | igcewieling | So this time we announced an upgrade and did nothing. |
| 17:40.59 | igcewieling | I suspect sales never notified their customers. |
| 17:41.08 | jmetro | =p |
| 17:41.09 | igcewieling | no problems reported. 8-( |
| 17:41.13 | jmetro | i remember that |
| 17:41.22 | igcewieling | yup. Last night was the "upgrade night" |
| 18:04.21 | bungle_ | TK - with your help i have a such a better understanding.. now have landline calling through to a softphone - only problem is that when i hang up the softphone the landline does not disconnect |
| 18:07.49 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 18:14.17 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 18:15.23 | Tuju | i'm getting close, now it only timeouts, packets appear to go right directions. |
| 18:15.29 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 18:16.55 | *** join/#asterisk jeev_ (~j@unaffiliated/jeev) |
| 18:17.41 | jmetro | bugle include a Hangup() after the dial |
| 18:17.45 | jmetro | or after the VM |
| 18:18.54 | butthurtface | Have any of you discovered a way to make Asterisk answer calls faster? It seems there is no dial tone, just a few seconds of silence prior to asterisk answering. |
| 18:19.54 | jmetro | that is your call setup time |
| 18:19.58 | jmetro | network issue |
| 18:21.25 | butthurtface | Do you think increasing bandwidth from 10meg will make a difference? |
| 18:21.48 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 18:23.59 | Tuju | is there a way to do codec transformation in asterisk? |
| 18:24.13 | navaismo | transformation? |
| 18:24.15 | leifmadsen | codec transformation? |
| 18:24.17 | Tuju | i read somewhere that my isp only accepts gsm codec |
| 18:24.17 | leifmadsen | transcoding? |
| 18:24.22 | leifmadsen | asterisk does it natively |
| 18:24.26 | leifmadsen | just enable the codecs you want |
| 18:24.29 | Tuju | yup, that sounds good. |
| 18:24.39 | Tuju | okay, so it cannot be that then. |
| 18:24.43 | leifmadsen | enable gsm on the carrier, enable whatever else you want on the other side |
| 18:27.16 | *** join/#asterisk jetlag (~jetlag@pool-71-168-242-55.cmdnnj.east.verizon.net) |
| 18:32.52 | Tuju | haa, i could call from mobile to my cisco desktop via asterisk. :D |
| 18:32.57 | Tuju | that was first time in my life. |
| 18:33.12 | Tuju | but for some reason it doesn't work otherway around. |
| 18:42.02 | *** join/#asterisk gbit86__ (~gbit86@204.11.31.54) |
| 18:45.47 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 18:46.11 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 18:46.13 | KNERD | The docs seem to be lacking in the Dial() in one area; LIMIT_WARNING_FILE...is the file location assumed to be in the /var/lib/asterisk/sounds/ or must the file path be included? |
| 18:47.52 | _Corey_ | KNERD: I'm using it on one system and have a relative path LIMIT_WARNING_FILE=custom/whatever. I think you can do a full path though. |
| 18:48.02 | *** join/#asterisk gbit86_ (~gbit86@204.11.31.54) |
| 18:48.36 | KNERD | Thanks a lot _Corey_ |
| 18:50.38 | _Corey_ | np |
| 19:08.07 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 19:14.03 | *** join/#asterisk Alagar (~helpdesk@vsusg1.vernalissystems.com) |
| 19:14.58 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
| 19:15.57 | Tuju | http://pastebin.com/wgugNZca i'm calling from 654321 to 555111, why it starts retransmitting? |
| 19:16.46 | Tuju | tcpdump shows that there comes no response packets, but when the call is being canceled, then the proxy answers. |
| 19:24.37 | [TK]D-Fender | KNERD, Same rules as every other sound file. |
| 19:24.48 | [TK]D-Fender | KNERD, Same rules as every other sound file. |
| 19:25.16 | KNERD | I am not too familiar with the rules for sound files |
| 19:25.33 | KNERD | pertaining to Asterisk |
| 19:26.27 | [TK]D-Fender | kenrelative to the sounds folder from ASTVARLIBDIR, or absolute when starting with / |
| 19:27.34 | KNERD | oh..okay thanks..but since I have the additional sound files, I would say I have to use the absolute. |
| 19:34.26 | *** join/#asterisk navaismo (~navaismo@189.241.51.199) |
| 19:36.12 | [TK]D-Fender | Those additional sounds should be under the main already... |
| 19:36.40 | igcewieling | if the files are under /var/lib/asterisk/sounds/en then you can use relative paths, if they are not, you need absolute paths. |
| 19:39.10 | KNERD | They are but there are multiple langes |
| 19:39.51 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 19:40.41 | [TK]D-Fender | selection of those is an automatic convention |
| 19:44.53 | FLeiXiuS | Under confbridge, is there a way to enable only the microphone and mute the incoming audio over the speaker? |
| 19:48.30 | KNERD | what do you mean "automatic convention"? |
| 19:48.35 | [TK]D-Fender | FLeiXiuS, Whose? Using what? When? What version? |
| 19:48.41 | *** join/#asterisk [SySteM] (~antoine@85.69.246.241) |
| 19:48.43 | [SySteM] | Hello |
| 19:48.47 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 19:48.56 | [TK]D-Fender | KNERD, languages use sub-folders that are always relative |
| 19:48.58 | [SySteM] | Anyone use asterisk with ices to stream on a icecast ? |
| 19:49.10 | FLeiXiuS | [TK]D-Fender, I have a user I want to mute what he can hear, but allow him to talk using confbridge asterisk 11.3.0 |
| 19:49.13 | KNERD | ahh...yes..thanks |
| 19:49.17 | [SySteM] | i try since 2 days to make anything working with asterisk 1.4 and ices0 and ices2.. nothing running. |
| 19:49.52 | *** join/#asterisk Quest (~syncsys@pool2-80-210.brain.net.pk) |
| 19:49.53 | Quest | if i have a fiber optic but I have two separate ip pools of 8 ips each (the fiber optic wire will be still one), that is 2 separate threads (thats what i have been informed). Then each thread will be having its separate bandwidth limit. is it a fact that both threads will be unaffected by each other in terms of bandwidth and traffic? That is its almost same like I am having two fiber connections? |
| 19:49.57 | Quest | <PROTECTED> |
| 19:51.25 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 19:51.33 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 19:51.36 | *** join/#asterisk gbit86_ (~gbit86@204.11.31.54) |
| 19:57.00 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 19:57.00 | igcewieling | Quest: I have no idea what you are talking about. |
| 19:57.02 | [TK]D-Fender | FLeiXiuS, when do you need to make this decision? |
| 19:57.30 | igcewieling | [SySteM]: since 1.4 is EOLd and so many people have moved to newer versions and ICES is not commonly used you may want to consider some other streaming method |
| 19:58.08 | igcewieling | [SySteM]: did you search the mailing list archives? |
| 19:58.23 | *** join/#asterisk tparcina (~tomo@cisco15.fesb.hr) |
| 19:58.26 | Quest | igcewieling, thats good for a start |
| 19:58.29 | FLeiXiuS | [TK]D-Fender, I would like for it to happen based on the extension dialed? IE, have an extension for a 'presenter' then have another extension for the listeners |
| 19:58.49 | [TK]D-Fender | FLeiXiuS, Then use func_volume to kill the audio |
| 19:58.52 | *** join/#asterisk StockdoodleGish (~cmarshall@75-145-50-25-Nashville.hfc.comcastbusiness.net) |
| 19:58.57 | Quest | igcewieling, we thought to have one fiber connection (with two threads) instead of two different fiber optic isps |
| 19:59.03 | [TK]D-Fender | FLeiXiuS, because not hearing a conference doesn't appear to be an option |
| 19:59.09 | igcewieling | Quest: Asterisk uses many threads, it has nothing to do with your internet connection or the number of IPs in the system |
| 19:59.21 | Qwell | igcewieling: He means strands of fiber. |
| 19:59.24 | FLeiXiuS | [TK]D-Fender, Good call, thanks. |
| 19:59.26 | Qwell | He has no idea what he's talking about either. |
| 19:59.30 | Quest | igcewieling, by thread i mean 2 ip pools of fiber |
| 19:59.31 | igcewieling | Quest: I assume english is not your native language? |
| 19:59.40 | Quest | igcewieling, affirmative |
| 20:00.32 | [SySteM] | igcewieling, i search on lot of forum .. |
| 20:01.10 | [SySteM] | but if modules existed, there is common way to running it no? |
| 20:01.11 | igcewieling | Quest: you cannot expect QoS on your internet connection. Your ISP is the only one who can tell you if usage of IPs in one pool will affect the bandwidth available to the other pool of IPs. Unless the ISP does something special to separate out the bandwidth, then you have ONE internet connection, regardless of how many pools of IPs you have. |
| 20:01.17 | StockdoodleGish | Novice question: using #include in extensions.conf isn't letting me include contexts in the #included files. Is this by design, or is there something I'm not doing correct? |
| 20:01.27 | Qwell | StockdoodleGish: pastebin an example |
| 20:02.23 | Quest | igcewieling, if the isp does separates the bandwidth on each pool and limits one with e.g 2mbps and the other with 1mbps . they will be totally independant? |
| 20:02.58 | igcewieling | Quest: ask your ISP, they set it up, they know. There is nothing in general networking concepts which says one way or the other. |
| 20:03.09 | Quest | hm |
| 20:03.19 | Quest | igcewieling, i just doubt that the sales man would lie |
| 20:03.34 | Quest | igcewieling, i just think* that the sales man would lie |
| 20:03.37 | StockdoodleGish | Quest: Pastebin example: http://pastebin.com/NqZaxJv5 |
| 20:03.44 | igcewieling | Quest: Q: How can you tell when a sales person is lying? A: Their lips are moving. |
| 20:04.03 | [TK]D-Fender | StockdoodleGish, Show us actual files.... |
| 20:04.08 | Qwell | StockdoodleGish: include => |
| 20:04.18 | igcewieling | Quest: ask to speak to an engineer before getting the service. |
| 20:04.22 | Quest | igcewieling, i tell that because iam in pakistan. |
| 20:04.42 | Qwell | igcewieling: "We already got the server." That's the sentence I'd put my money on. |
| 20:04.45 | [TK]D-Fender | StockdoodleGish, and indeed the syntax is incorrect |
| 20:04.47 | Qwell | service* |
| 20:04.59 | igcewieling | Quest: I am not personally familiar with any ISP in the USA which offers service like you are describing. |
| 20:05.08 | igcewieling | I can't speak for your country. |
| 20:06.41 | igcewieling | Awww, that is so cute. One of our customer service people is starting to get an "Engineer attitude". |
| 20:07.18 | igcewieling | "They SPECIFICALLY asked me to remove that (see below) – please have JEFF send me an email confirming this is what he wants before i start billing them for wasting my time." |
| 20:07.47 | butthurtface | Any of y'all know maybe why Asterisk wouldn't be detecting the # sign? |
| 20:07.58 | StockdoodleGish | Quest: Actual file: http://pastebin.com/yk8n7Lzt |
| 20:08.08 | StockdoodleGish | The => is in the original, just left it off the example |
| 20:08.28 | igcewieling | butthurtface: several reasons, but without additional description of the setup no idea which might apply |
| 20:08.28 | [TK]D-Fender | StockdoodleGish, show us your actual files |
| 20:08.53 | [TK]D-Fender | StockdoodleGish, context includes work just fine across #INCLUDE-d files |
| 20:09.02 | butthurtface | It's "Press 2 to go back, or press # to confirm" |
| 20:09.03 | Quest | igcewieling, ok |
| 20:09.10 | [TK]D-Fender | StockdoodleGish, So your files are bad, or your syntax is bad |
| 20:09.19 | butthurtface | So they press 2, works great… returns to previous prompt… Press # and it gets all funky |
| 20:09.26 | [TK]D-Fender | butthurtface, show us |
| 20:09.35 | butthurtface | Okay |
| 20:09.50 | Quest | StockdoodleGish, whats that? |
| 20:09.58 | igcewieling | butthurtface: can't think of anything obvious which would prevent Asterisk from detecting # when in an IVR in the dialplan |
| 20:10.29 | StockdoodleGish | Quest: Ignore. Mixed you up with Qwell. |
| 20:10.35 | Qwell | StockdoodleGish: Which context(s) are you having issues with? |
| 20:11.00 | *** join/#asterisk gbit86_ (~gbit86@204.11.31.54) |
| 20:11.18 | Qwell | umm |
| 20:11.18 | Qwell | line 35 |
| 20:11.20 | StockdoodleGish | Qwell: Shoot...nevermind. |
| 20:11.23 | StockdoodleGish | no underscore |
| 20:11.26 | Qwell | quite |
| 20:11.44 | StockdoodleGish | Qwell: Thanks for your time. |
| 20:11.52 | Qwell | That'll be $299.94. |
| 20:12.16 | StockdoodleGish | Qwell: Long day. I have no witty comeback. |
| 20:12.18 | *** part/#asterisk StockdoodleGish (~cmarshall@75-145-50-25-Nashville.hfc.comcastbusiness.net) |
| 20:13.14 | butthurtface | [TK]D-Fender: http://pastebin.ca/2370271 |
| 20:13.39 | [TK]D-Fender | butthurtface, QUOTES <- |
| 20:13.50 | [TK]D-Fender | butthurtface, notice to obvious difference of those 2 checks... |
| 20:14.16 | butthurtface | Yeah those quotes are new… When I remove them the issue persists. |
| 20:14.23 | Qwell | can Read() even return #? |
| 20:14.34 | [TK]D-Fender | Don't believe so... |
| 20:14.37 | butthurtface | That's what I was thinking because # is to end the listening |
| 20:14.40 | igcewieling | butthurtface: in Asterisk quotes are (usually) literal. If you have quotes on one side of an = you need it on the other side. |
| 20:14.52 | butthurtface | igcewieling: thank you for that tip. |
| 20:15.39 | igcewieling | butthurtface: # would end input, so with JUST a # then input would me empty. |
| 20:16.17 | igcewieling | butthurtface: go reread "core show application read" again carefully |
| 20:17.09 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:20.34 | butthurtface | I'm going to try it with "" input rather than #- We spent $500 on the asterisk voice lady and I don't want to spend more money on having her change her shit |
| 20:21.17 | butthurtface | Nope, didn't work with blank. That's okay. It's only going to be like $12 for 15 more words haha |
| 20:21.29 | igcewieling | butthurtface: did you remember quotes on both sides of = |
| 20:21.36 | butthurtface | I didm;t use quotes |
| 20:21.47 | *** join/#asterisk madhatt (~madhatt@23.31.65.29) |
| 20:21.56 | igcewieling | butthurtface: you have to if you want to check for empty. |
| 20:22.20 | igcewieling | GotoIf($["${refconfirm} "= ""] could be written as GotoIf($[X${refconfirm}X = XX] QUOTES ARE LITERAL |
| 20:22.36 | madhatt | hey everyone. can anyone help me. I need to think of an easy "test" for a backup tech I'm interviewing tomorrow… you know, "break" my pbx in some fashion that should be super easy for any * tech to find…. I can think of several things but wanted to ask the community |
| 20:22.51 | igcewieling | well GotoIf($["${refconfirm}" = ""] of course |
| 20:23.14 | Qwell | madhatt: Ask him to tell you every step of a line of dialplan leifmadsen writes for you. |
| 20:23.18 | butthurtface | That actually might work. |
| 20:23.27 | igcewieling | madhatt: add a space after an application name in your dialplan |
| 20:23.29 | butthurtface | I'm going to try that right now igcewieling |
| 20:24.19 | madhatt | hrmm… good ideas… |
| 20:24.29 | igcewieling | butthurtface: you are checking for quote ${refconfirm} quote = quote quote. The quotes are PART of the comparison |
| 20:24.47 | igcewieling | madhatt: change one of your 1NXXNXXXXXX or NXXNXXXXXX to NXXNXXXXX |
| 20:25.11 | igcewieling | Matthias: make an extension with a missing priority 1 |
| 20:25.24 | butthurtface | igcewieling: That didn't work. We're going to either replace the "Pound" with "1" so it can just be a 1/2 prompt. |
| 20:25.37 | butthurtface | Easier to rework the audio to get her to say what we like. |
| 20:25.39 | igcewieling | madhatt: think back to all the mistakes you made. |
| 20:25.55 | madhatt | igcewieling: great! I've made so many mistakes |
| 20:26.13 | Qwell | madhatt: Make him come in here, and answer the first question he sees. |
| 20:26.47 | madhatt | shit, I run our 100+ employee * pbx (and several others) and I'm a novice by you all's standards! |
| 20:26.50 | igcewieling | Qwell: remind me to never interview with you. 8-| |
| 20:26.51 | madhatt | (i'm sure) |
| 20:31.25 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 20:33.20 | Tuju | i can't get that outbound trunk to work. |
| 20:33.25 | Tuju | it keeps retransmitting. |
| 20:33.32 | Tuju | inbound works fine. |
| 20:34.26 | *** join/#asterisk gbit86_ (~gbit86@204.11.31.54) |
| 20:38.31 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.141) |
| 20:49.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:20.02 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 21:25.16 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 21:26.37 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.136) |
| 21:29.43 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 21:44.10 | *** join/#asterisk tamiel (~tamiel@ip-38.net-81-220-93.rev.numericable.fr) |
| 22:03.35 | leifmadsen | Qwell: wat?! |
| 22:03.52 | Qwell | hi |
| 22:06.26 | *** join/#asterisk jagster` (~chatzilla@unaffiliated/jagster/x-9084543) |
| 22:12.21 | butthurtface | Any of you know of a reasonable way to check if a file exists? I'm using Googles Speech Recognition to ask the caller "What state is your property in" - Now I have the voice files for all 50 states, of course- if they say "Africa" the file will not exist, but how would I go about saying "I'm sorry, could you please repeat that?" rather than confirming "You said your property is in" africa.gsm does not exist. |
| 22:12.55 | jagster` | -___________________- so ive just been tasked with securing an old asterisk server and have it public facing so that people can work remotely |
| 22:13.06 | butthurtface | oh lucky you! |
| 22:13.31 | jagster` | putting it behind a border controller is a nogo since it would be a failure point for the business |
| 22:13.43 | jagster` | can anyone point me in the direction of some good readings? |
| 22:13.54 | butthurtface | iptables? |
| 22:14.14 | jagster` | yeah someone mentioned some custom brew of fail2ban |
| 22:14.36 | jagster` | its a temp fix, put this out in the wild and then upgrade to a newer, secure version of asterisk |
| 22:14.40 | butthurtface | fail2ban would be a must, but be mindful that if you receive some goofy auth errors from your users, you will need to manually remove them from the iptables system. |
| 22:15.00 | jagster` | service iptables off ;) |
| 22:15.23 | butthurtface | Honestly I would use iptables, allow anything on the internal lan to come in, and external I would be selective about it. |
| 22:15.53 | jagster` | yeah guess i could restrict it to known ip rangers |
| 22:15.55 | jagster` | ranges |
| 22:16.02 | jagster` | but how do you get your external agents to login? |
| 22:16.06 | jagster` | ie sales |
| 22:16.31 | butthurtface | You would have to make sure that those IPs are permitted after the other rules are set. |
| 22:16.59 | jagster` | but those are most likely to be dynamic ip's |
| 22:17.03 | butthurtface | Hmm. |
| 22:17.38 | butthurtface | I would say try to use their MAC address but that seems like it would be a bad idea. |
| 22:17.48 | jagster` | lol yeah mac addresses are not security |
| 22:17.56 | jagster` | they are broadcast clear text by every device |
| 22:19.45 | butthurtface | VPN isn't an option either I take it. |
| 22:21.50 | jagster` | ideally no, as its one more thing our sales guys would have to use and you know how users are with change |
| 22:22.00 | butthurtface | yeah. |
| 22:22.38 | butthurtface | Hmm… I personally don't have any other ideas but I'm like 90% on a project I'm stuck on… but I am sure out of the 150+ people in here someone might be able to offer you some sound advise. |
| 22:22.43 | jagster` | maybe an open source session border controller infront of the pbx |
| 22:22.52 | jagster` | what are you stuck on |
| 22:22.56 | butthurtface | haha |
| 22:23.20 | butthurtface | Probably going to sound stupid, but I need Asterisk to tell me whether or not a file exists. and if not, to say so. LOL |
| 22:23.35 | butthurtface | This: |
| 22:23.36 | butthurtface | Any of you know of a reasonable way to check if a file exists? I'm using Googles Speech Recognition to ask the caller "What state is your property in" - Now I have the voice files for all 50 states, of course- if they say "Africa" the file will not exist, but how would I go about saying "I'm sorry, could you please repeat that?" rather than confirming "You said your property is in" africa.gsm does not exist. |
| 22:25.33 | jagster` | hmm dunno enough about the software to ofer a suggestion |
| 22:25.58 | butthurtface | Yeah me either… The docs say to use a shell script but I'm not sure how that is going to work. |
| 22:26.38 | [TK]D-Fender | butthurtface: "core show function STAT" |
| 22:26.54 | butthurtface | D-Fender is there anything you don't know about asterisk? lol |
| 22:27.09 | igcewieling | butthurtface: "core show functions" is your friend. Get to know it, buy it a beer. go to bed with it. |
| 22:27.28 | butthurtface | Apparently it is. This actually shows me exactly what I'm looking for. |
| 22:27.42 | [TK]D-Fender | butthurtface: Sure ... but then there's the tons of useful bits I do and somehow I'm able to sleep at night.... for like 6 entire minutes as logs will attest |
| 22:28.01 | butthurtface | Couple of weeks ago a head hunter offered me $150,000 for an Asterisk Engineer role… I explained "Sorry I don't know enough to take your offer" |
| 22:28.06 | butthurtface | now I see why they get paid the big bucks. |
| 22:28.13 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 22:28.19 | igcewieling | If you read the Asterisk Book, read the docs for each function and application, you can become an expert in only a few years. |
| 22:28.30 | mmlj4 | what? under fire in afghanistan? |
| 22:28.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.93) |
| 22:28.38 | butthurtface | No. Sherman Oaks, CA |
| 22:28.45 | mmlj4 | give the dude my email address :-) |
| 22:28.51 | igcewieling | butthurtface: not much difference |
| 22:28.56 | mmlj4 | hah |
| 22:28.59 | butthurtface | LOL igcewieling TRUE that! |
| 22:29.08 | igcewieling | Actually, I was thinking Oakland, not Sherman Oaks |
| 22:29.30 | butthurtface | Sherman Oaks is better than Oakland, but Sherman Oaks is still in LA County… so yes, garbage town. |
| 22:29.33 | mmlj4 | seriously, if I could make that kind of money I'd deign to live in cali |
| 22:29.51 | butthurtface | mmlj4: I can give you the head hunters name on LinkedIn if you like. |
| 22:30.08 | butthurtface | http://www.linkedin.com/profile/view?id=39646859&authType=name&authToken=ink0&goback=%2Ermg_*1_*1_*1_*1_*1_*1_*1_*1_*1 |
| 22:30.08 | jagster` | shermanoaks is a very nice neighborhood |
| 22:30.15 | mmlj4 | mostly kidding, but thanks |
| 22:30.20 | igcewieling | for 150K/yr I might consider taking a job which makes me go into the office. |
| 22:30.34 | butthurtface | Sherman Oaks could be worse. |
| 22:30.35 | jagster` | what part of town are you in butthurtface |
| 22:30.45 | jagster` | let me guess OC |
| 22:30.47 | butthurtface | I'm in Winnetka, which is much worse than Sherman Oaks lol. |
| 22:31.01 | butthurtface | No, still in the same area as Sherman Oaks. |
| 22:31.24 | [TK]D-Fender | heads out to practice... |
| 22:31.27 | jagster` | never heard of winnetka |
| 22:31.31 | jagster` | what kinda company u work for? |
| 22:31.54 | butthurtface | I work for an adult entertainment company. |
| 22:32.06 | butthurtface | but we're branching off into mainstream... |
| 22:32.42 | jagster` | ah yes makes sense now |
| 22:32.48 | jagster` | you guys are in winnetka? |
| 22:32.56 | jagster` | i used to work @ warner center |
| 22:32.59 | butthurtface | Nah, Westlake Village - Near Thousand Oaks. |
| 22:33.07 | butthurtface | My fiancé works at the Warner Center for Health Net. |
| 22:33.15 | jagster` | tell her to ge tme a discount |
| 22:33.25 | igcewieling | butthurtface: wait, porn isn't mainstream? |
| 22:33.26 | butthurtface | LOL if only she could get herself a discount first |
| 22:33.49 | butthurtface | igcewieling: That's a good question for our VC guys LOL |
| 22:34.00 | jagster` | do you guys produce |
| 22:34.02 | jagster` | or distribute |
| 22:34.07 | jagster` | i used to work for a big .com |
| 22:34.15 | butthurtface | No our production is handled elsewhere. |
| 22:34.22 | igcewieling | if you compare the percent of people who like porn .vs. the number of people who like action flicks, I bet porn would win. |
| 22:34.30 | butthurtface | Wherever vivid has them do it. I don't participate in that. I'm just a sysadmin. |
| 22:34.46 | jagster` | when our company decided to go mainstream |
| 22:34.50 | jagster` | they opened a new business :P |
| 22:34.55 | jagster` | and put me on as sysadmin |
| 22:35.15 | butthurtface | Kind of a rough job but it has its perks. |
| 22:35.44 | butthurtface | beats the hell out of pickin' on a double row |
| 22:35.44 | jagster` | we were more of a portal |
| 22:35.51 | jagster` | kinda like ccbill |
| 22:36.56 | mmlj4 | igcewieling: you're not back on the coast, are you? |
| 22:37.07 | igcewieling | mmlj4: pensacola |
| 22:37.11 | butthurtface | Ahhh CCBill. |
| 22:37.24 | mmlj4 | ah. |
| 22:37.27 | igcewieling | easy to escape in case of a hurricane, but still on the gulf and way above sea level |
| 22:37.28 | butthurtface | I always liked iBill better… Of course they went into the shitter. |
| 22:37.32 | butthurtface | RIP iBill |
| 22:37.36 | mmlj4 | way above sea level |
| 22:37.40 | mmlj4 | I used to know what that was |
| 22:37.56 | igcewieling | mmlj4: I lived on the top of a mountain for 3 years after Katrina |
| 22:38.14 | mmlj4 | I remember about that |
| 22:38.25 | BoRiS | I have a grandstream 502 ata and whenever I make a call, I can barely hear the person on the other end even when my volume is up on my phone. What settings needs to be tweaked to increase the volume. |
| 22:38.36 | igcewieling | I don't remember how far above sea level Huntsville was, but it was enough |
| 22:38.47 | mmlj4 | aye |
| 22:38.49 | mmlj4 | later & |
| 22:39.45 | *** join/#asterisk andrewyager (~andrewyag@CPE-144-132-193-27.nsw.bigpond.net.au) |
| 22:40.14 | butthurtface | Stat/e works awesome for what I need it to do! |
| 22:40.20 | butthurtface | Thanks [TK]D-Fender. |
| 22:41.22 | *** join/#asterisk andrewya_ (~andrewyag@syd02s26-fw01.thecore.net.au) |
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| 23:12.45 | jagster` | butthurtface: we were one of the other big ones |
| 23:14.36 | butthurtface | hmmm |
| 23:14.42 | butthurtface | jettis? |
| 23:15.30 | jagster` | nah we had like hundreds of thousands of affiliate sites |
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