| 00:00.57 | pontis | how i use minicom? |
| 00:01.06 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 00:02.41 | mmlj4 | open a terminal and type "man minicom" |
| 00:03.00 | mmlj4 | screen also works |
| 00:04.04 | pontis | i´m using raspberry |
| 00:04.18 | mmlj4 | and that matters to me how? |
| 00:05.19 | pontis | how i open terminal here? |
| 00:05.39 | pontis | sorry i´m recent to all this! |
| 00:06.00 | mmlj4 | new, not recent |
| 00:06.41 | pontis | ya, thanks |
| 00:10.35 | *** join/#asterisk andrewyager (~andrewyag@CPE-144-132-193-27.nsw.bigpond.net.au) |
| 00:11.27 | pontis | i do i know what usb port is? |
| 00:12.40 | pontis | how do i know what usb port is? |
| 00:12.43 | pontis | sorry |
| 00:13.28 | WIMPy | You have to try. |
| 00:16.36 | pontis | it appears this dongle is incompatible at all! |
| 00:17.38 | *** join/#asterisk andrewyager (~andrewyag@CPE-144-132-193-27.nsw.bigpond.net.au) |
| 00:18.05 | igcewieling | I find that trying to make random hardware work with Asterisk fails 100% of the time. |
| 00:22.47 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 00:29.03 | pontis | sameone know about compability of huawei k3772? |
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| 00:50.24 | mmlj4 | pontis: buy supported hardware, problem solved |
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| 00:55.10 | pontis | ok, i see. thank´s |
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| 01:20.38 | Cubber | is googlevoice support going away with the hangout changes? |
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| 01:55.46 | Katty | ASTERISK, etc. |
| 02:10.42 | WIMPy | I think you mean /etc/asterisk/. |
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| 04:34.09 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 06:52.37 | linocisco | which software on PC can call US and canada's landlines and mobiles free like magicjack? |
| 06:52.53 | linocisco | which software on PC can call US and canada's landlines and mobiles free like magicjack on smartphones? |
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| 07:15.19 | BlackDex | Hello there... |
| 07:15.40 | BlackDex | Is it possible to globally add a SIP header to all internal SIP calls? |
| 07:16.06 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 07:16.17 | gavimobile | kaldemar: have you awakened kind sir!!! |
| 07:16.18 | BlackDex | We have 60XX as our internal range, and i don't want to add SIPAddHeader to every single exten => 60?? |
| 07:18.57 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 07:18.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 07:19.15 | *** join/#asterisk bombev (~User@PPPoE-Static-40-132.UnicsBG.Net) |
| 07:20.10 | linocisco | infobot |
| 07:20.19 | linocisco | ~infobot |
| 07:23.44 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:24.42 | ChannelZ | What do you need to add a header for? |
| 07:24.56 | BlackDex | ChannelZ: I think i have found it :) |
| 07:25.07 | BlackDex | I want a different ringtone for internal sip calls |
| 07:26.36 | BlackDex | I found that when i add exten => _60XX,1,SIPAddHeader.... and then for each internal line which needs/supports it just change the first 1 to a 2 :) it works |
| 07:27.03 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:28.32 | kaldemar | gavimobile: i bet that was not what you really want to know. :) |
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| 07:29.39 | gavimobile | you must have some esp or something! |
| 07:29.47 | gavimobile | tarror card reader |
| 07:30.28 | gavimobile | you got a minute? it should be an easy one for ya. |
| 07:30.37 | kaldemar | ~ask |
| 07:30.38 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:32.22 | gavimobile | to refreshen your memory, yesterday you helped me with an issue of a trunk using different servers to send me the call. after a lot of research, setting allowguests=yes solved the problem, I don't want to do this. the other solution is adding a bunch of hosts in my sip.conf, 1 host for each server. |
| 07:34.05 | gavimobile | my question is, is there a more efficent way to do this? like change host=*callcentric.com to a wild card, or can I at least make a template for hosts like the templates we use for peers? |
| 07:36.01 | gavimobile | kaldemar: ^ |
| 07:36.55 | ChannelZ | They don't have a single hostname calls come from? |
| 07:37.54 | gavimobile | ChannelZ: nope :-( |
| 07:38.03 | gavimobile | and im not settings allowguests=yes |
| 07:38.18 | ChannelZ | According to their guides it's just "callcentric.com" |
| 07:38.37 | gavimobile | ChannelZ: their guide is wrong |
| 07:38.49 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 07:38.50 | ChannelZ | Sounds like a company worth giving money to. |
| 07:38.57 | gavimobile | haha |
| 07:40.05 | gavimobile | ChannelZ: well I just need a free did for testing purposes |
| 07:40.05 | gavimobile | I already have one with ipcomm |
| 07:40.05 | gavimobile | which is great |
| 07:44.09 | gavimobile | ChannelZ: do you know if I can create a template for hosts? or only for peers? |
| 07:44.54 | ChannelZ | A peer can only have one host |
| 07:46.01 | kaldemar | gavimobile: allowguest=yes just hides the problem, it doesn't solve it. you'll have to make multiple peers. |
| 07:46.04 | ChannelZ | Unfortunately you are stuck with creating a peer for each IP, or matching all the calls generically and doing control through some other means (your firewall) |
| 07:46.28 | kaldemar | gavimobile: sure you can make a template and then multiple entries with just the host line. |
| 07:46.36 | gavimobile | kaldemar: as I mentioned above, I don't want allowguest=yes ! I agree with you |
| 07:46.52 | gavimobile | kaldemar: sweet, ill try it |
| 07:46.59 | ChannelZ | You can make a template with all your peer settings except host, yes, but you still have to make a peer for each one |
| 08:32.40 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 08:32.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 08:32.44 | *** join/#asterisk geeksteve (~geeksteve@emh-nat.poundbury.com) |
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| 09:18.16 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 09:18.16 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 09:18.42 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
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| 09:22.17 | gavimobile | says im connected |
| 09:25.57 | msaraiva | Still working here., |
| 09:26.00 | msaraiva | Just tested. |
| 15:01.43 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:01.43 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:02.30 | seik0 | igcewieling: ok, it's quite cool |
| 15:03.05 | igcewieling | seik0: the CLI help is one of the primary sources of Asterisk's documentation. Learn to use it. |
| 15:06.33 | seik0 | igcewieling, for example, it's said, that Originate command with Async parameter generates OriginateResponce event, which I can't see through "manager show commands" |
| 15:07.08 | seik0 | "manager show commands" shows actions only? |
| 15:07.20 | seik0 | and how to look at events? |
| 15:09.27 | seik0 | "manager show command Originate" doesn't say, that it generates any events |
| 15:09.33 | seik0 | in some questions the only primary source - sources of Asterisk ) |
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| 15:34.28 | igcewieling | seik0: most people use an AGI library for listening for manager events |
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| 15:52.55 | DBordello | Does anybody have any recommendations for an E911 provider? |
| 15:56.26 | carrar | Been happy with Connexon |
| 15:56.26 | carrar | http://www1.911enable.com |
| 15:56.26 | igcewieling | DBordello: easiest thing to do is use a SIP provider which uses an e911 service provider. Verizon SIP and Level 3 and Vitelity all do this, I'm sure others do as well. |
| 15:56.47 | DBordello | This is actually completely Asterisk unrelated (but I figured it'd pass). My parents are using google voice via an Obi at home. It just came to my attention google voice does not route 911 |
| 15:56.59 | DBordello | So I am looking for a 3rd party SIP provider just for E911 |
| 15:57.24 | DBordello | carrar, that looks interesting |
| 15:57.49 | DBordello | Callcentric seems reasonable, and inexpensive ($1.50/mo) |
| 15:58.26 | DBordello | But obviously going with the cheapest/shadies 911 provider defeats the whole purpose |
| 16:00.12 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 16:00.34 | *** join/#asterisk Foxi352 (~Foxi352@v-172-221.access.restena.lu) |
| 16:00.36 | carrar | They can't afford a POTS line? |
| 16:01.08 | carrar | You wanna RISK your parents life on the internet? |
| 16:01.23 | DBordello | Actually, perhpas they are still live for 911 |
| 16:05.13 | igcewieling | carrar: People will jump through hoops to spend more money on a complicated error prone setup. |
| 16:09.10 | carrar | DBordello, you can also search for a 10 digit Emergency number and try dialing that through google |
| 16:09.10 | carrar | local to your area |
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| 16:18.12 | *** join/#asterisk cybrNaut (cybrNaut@unaffiliated/cybrnaut) |
| 16:18.55 | igcewieling | carrar: that is about as helpful as telling someone who needs to go on a diet to "just eat dirt" |
| 16:21.18 | carrar | sometimes you gotta life a finger and do some work |
| 16:21.25 | carrar | lift even |
| 16:22.04 | carrar | 10 Digit Emergency numbers are dependant on where you live |
| 16:22.11 | carrar | I can't read minds |
| 16:22.23 | Greenlight | I'd have guessed most people have mobiles phones that they'd use for emergency calls |
| 16:37.10 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:37.10 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:40.23 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:40.23 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:40.39 | drmessano | There is a significant difference between the 10-digit number for your local Dispatch and calls being processed through e911 |
| 16:41.09 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:41.43 | DBordello | Does anybody see any problems with using CallCentric with E911 then routing 911 through them? |
| 16:43.44 | drmessano | I use Flowroute.. I can't afford CallCentrics rates |
| 16:44.04 | drmessano | $.02/min.. is this 1999? |
| 16:44.44 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:45.40 | *** join/#asterisk cmendes0101| (~cmendes01@72.1.46.254) |
| 16:47.11 | DBordello | drmessano, I don't plan on routing any calls through them |
| 16:47.14 | DBordello | Just E911 ;) |
| 16:47.46 | drmessano | So why not just use Flowroute? |
| 16:47.58 | *** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 16:48.00 | drmessano | Then you have a backup with decent rates |
| 16:49.02 | DBordello | It looks like they require a DID for E911 |
| 16:49.16 | igcewieling | drmessano: he has that google voice fetish which has been going around. |
| 16:49.18 | drmessano | Well yeah, you gotta have something tied to it |
| 16:49.35 | igcewieling | Hopefully it will run its course soon and sanity will return |
| 16:49.47 | drmessano | Callcentric will terminate e911 without a DID involved? |
| 16:49.48 | DBordello | mmmm, google voice |
| 16:49.52 | DBordello | drmessano, yup |
| 16:49.55 | drmessano | ROFL |
| 16:49.59 | drmessano | Oh thats just peachy |
| 16:50.09 | drmessano | Ripe For Abuse(TM) |
| 16:51.15 | drmessano | I guess since Callcentric has more of a Residential lean, they just handle any e911 calls based on the account into |
| 16:51.18 | drmessano | Kinda silly, but ok |
| 16:51.40 | drmessano | igcewieling, Google Voice will be useless soon enough. Don't worry |
| 16:51.41 | igcewieling | my low usage phone is so cheap I don't even think of the cost anymore. |
| 16:51.52 | igcewieling | drmessano: the cure is within reach! |
| 16:52.16 | DBordello | The Obi crowd heavily relies on GV. They don't seem to be as concerned about the changes |
| 16:52.36 | drmessano | DBordello, they use the same protocol.. It's the same issue |
| 16:52.40 | drmessano | We discussed that before |
| 16:52.50 | [TK]D-Fender | DBordello: Obliviousness / denial FTW |
| 16:52.50 | drmessano | They think Obi-Wan is going to save them |
| 16:52.52 | drmessano | But they cant |
| 16:53.05 | DBordello | likely true |
| 16:53.11 | drmessano | Not unless the Obi-Hai is going to run X and a Google Hangout client |
| 16:53.31 | DBordello | they aren't convinced the XMPP interface is going away |
| 16:53.39 | DBordello | but *shrug*, I know nothing about it |
| 16:53.42 | drmessano | Oh.. thats hilarious |
| 16:53.50 | drmessano | It already HAS gone away |
| 16:54.19 | drmessano | Google Hangout cannot be accessed via XMPP |
| 16:54.33 | drmessano | and Google Voice is moving to Hangouts. That's all public knowledge |
| 16:56.15 | drmessano | There's no theory here.. The roadmap is in place. |
| 16:56.23 | DBordello | ah, okay |
| 16:56.36 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 16:56.43 | carrar | GOOGLE ROADMAP: Offer it free, get people hooked, drop offering |
| 16:56.57 | drmessano | Indeed |
| 16:58.08 | [TK]D-Fender | "First hit's free" |
| 16:58.18 | DBordello | maybe I'll just switch them to flow route anyways |
| 16:59.10 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 16:59.10 | *** mode/#asterisk [+o file] by ChanServ |
| 16:59.17 | drmessano | DBordello, I don't really understand those that see this as theory or conjecture. The Google Voice blog clearly states Google Voice is going to move to Hangouts.. and those that have moved from GTalk to Hangouts within GMail need to wait a little longer before getting that functionality back. Then there's the absolute that XMPP has been cut off from Hangouts due to serious issues making it all work |
| 16:59.39 | drmessano | I guess we need a reply in infobot with the links lol |
| 17:01.24 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 17:01.35 | drmessano | But there is also a theory/conjecute on another motivation.. which absolutely feeds the bear.. XMPP access to Google Voice gives Microsoft a path to Lync <> GV integration via a popular open protocol. They already had a spat over Outlook.com having access to GTalk and shut that door |
| 17:01.46 | _abc_ | Hi all. Does anyone know which one, if any, of the free sip phones which run on linux supports MWI with asterisk? |
| 17:02.08 | drmessano | So theres no reason to think Google isn't also motivated to disallow Lync <> GV integration |
| 17:02.16 | DBordello | drmessano, you clearly have more knowledge on this matter than I do, so I'll trust you :) |
| 17:02.27 | _abc_ | I have subscribemwi=yes |
| 17:02.33 | DBordello | Did outlook.com lose GTalk access? |
| 17:02.38 | drmessano | Yes |
| 17:02.39 | _abc_ | in sip.conf but my client currently linphone does not indicate anyhting |
| 17:02.50 | drmessano | It was cut off hours after it was announced |
| 17:02.55 | drmessano | Google said "Hell no, back off" |
| 17:03.02 | drmessano | Very publicly, I might add |
| 17:03.03 | [TK]D-Fender | _abc_: Show us |
| 17:03.05 | [TK]D-Fender | ~pb |
| 17:03.05 | 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. |
| 17:03.06 | [TK]D-Fender | ^^^ |
| 17:03.20 | _abc_ | [TK]D-Fender: show what? |
| 17:03.42 | _abc_ | I have a perfectly working config, it sends mwi to the client, the client I currently use which is linphone does not have a mwi indicator |
| 17:03.45 | [TK]D-Fender | _abc_: Debug and ocnfig to backup your problem clearly.... |
| 17:03.52 | _abc_ | ?! |
| 17:04.01 | [TK]D-Fender | _abc_: Sorry, I think I missed something there. |
| 17:04.07 | _abc_ | I am not going to fix linphone man |
| 17:04.18 | [TK]D-Fender | _abc_: Yeah, it was more a statement of incompatability with the client.. |
| 17:04.24 | _abc_ | I was asking which free sip phone on linux supports sip notification mwi |
| 17:04.37 | DBordello | drmessano, that is hillarious. |
| 17:04.44 | [TK]D-Fender | _abc_: Have you tried jsut letting * broadcast it instead of expecting a subscribe? |
| 17:04.45 | DBordello | I still connect to GTalk through XMPP |
| 17:04.58 | [TK]D-Fender | _abc_: (which is the norm) |
| 17:05.01 | _abc_ | What is Lync <> GV? |
| 17:05.11 | WIMPy | _abc_: I think i've seen it on zoiper. |
| 17:05.11 | _abc_ | Google Voice and Lync is what? |
| 17:05.27 | _abc_ | [TK]D-Fender: no, but pretty sure linphone has no mwy |
| 17:05.30 | _abc_ | *mwi |
| 17:05.38 | _abc_ | it even shows undecoded packet when it comes in |
| 17:05.44 | _abc_ | so that is the mwi :) |
| 17:05.59 | drmessano | Google Voice is googles VoIP service, and Lync is Microsofts IM/VoIP Enterprise application |
| 17:06.02 | [TK]D-Fender | _abc_: Well... there goes that! |
| 17:06.09 | *** join/#asterisk Olobola (~Olobola@c-71-198-191-166.hsd1.ca.comcast.net) |
| 17:06.17 | drmessano | We have Lync here, and its horrible |
| 17:06.26 | [TK]D-Fender | _abc_: Yeah I can't say which ones support for sure beyond Counterpath's clients.... |
| 17:06.40 | _abc_ | I actually says 'Not Impleneted' in the status bar but it also says that for other packets too |
| 17:06.42 | drmessano | SOOOOOO BUY ME A STICK OF RAM SO I CAN RUN PLZ KTHX LAGGGGGGGGGGY |
| 17:07.31 | _abc_ | drmessano: does it expect to run on a LN2 cooled supercluser with gigabit deidicated fiber? |
| 17:07.43 | _abc_ | -i |
| 17:08.01 | drmessano | _abc_, i'm talking about just the desktop client. I've no idea what the ass-end requires |
| 17:08.15 | _abc_ | I was talking about the desktop client too |
| 17:08.22 | drmessano | Oh |
| 17:08.29 | drmessano | Well, then yes |
| 17:08.30 | drmessano | lol |
| 17:08.33 | _abc_ | I have laptops which are rejected by Vista for Aero for 'lack of performance' and they run Compiz at light speed |
| 17:08.45 | _abc_ | Compiz is heavyer on gui eye candy than all of aero |
| 17:09.03 | _abc_ | 3d raytraced affine transforms in real time fgs |
| 17:09.32 | drmessano | I have a 5 yr old laptop with 4GB RAM and Win7 on it, and Lync makes it feel like my old K62-550 |
| 17:09.44 | _abc_ | ol |
| 17:09.46 | _abc_ | lol |
| 17:13.29 | carrar | drmessano, whats sucking about Lync? |
| 17:13.37 | DBordello | They just rolled out Lync here on campus, it seems to be well received |
| 17:15.20 | [tpn]Cuz | we're messing with lync too |
| 17:15.53 | drmessano | If you love Windows and Office, needing a bunch of RAM to even operate, Outlook/Exchange Mail and all that Microsoft has to offer as a package, Lync is great and fits in nicely with the bundle |
| 17:17.10 | drmessano | It's just a typical Microsoft product. It's heavy on the eye candy, does about 25% of what it could/should do, is heavy on system resources, and costs a lot to deploy |
| 17:18.04 | drmessano | It's a polished up version of Windows Messenger that runs on your own backend and interfaces with Active Directory. Yay? |
| 17:18.44 | tzanger | drmessano: I haven't found a good substitute for Office. OO is *terrible* |
| 17:19.06 | DBordello | I am a big fan of Office as well. |
| 17:19.24 | drmessano | I use LibreOffice all day long |
| 17:19.29 | tzanger | oh I'm not a big fan. drmessano is right, it's a pig, but it's fast and nothing beats excel for quickly graphing data |
| 17:19.57 | DBordello | agreed |
| 17:20.56 | DBordello | Although I haven't used OO in years |
| 17:21.01 | DBordello | It was awful during 1.x |
| 17:21.13 | drmessano | OO is dead.. Long live LibreOffice |
| 17:22.01 | drmessano | I think just in the short time that LibreOffice has existed they've already made mass improvment in the code |
| 17:23.15 | [tpn]Cuz | I'm not a big fan of anything, I use what I'm asked to use, and if I'm given the choice, I try to choose what i believe is the best tool for the job. In an office where everyone uses Outlook & Exchange, I'm not going to try to pet the cat backwards by getting Mutt to work. |
| 17:24.17 | tzanger | drmessano: I use OO/LO interchangeably. LO is a pig when you try to graph large (say 6000+ rows) of data. I *do* like LO/OO's word processor better than Word, but I don't word process nearly as oftena s I look at data sets |
| 17:24.19 | drmessano | [tpn]Cuz, that's why I have one machine with Win7, Outlook, my AD Admin Tools, etc and then my Ubuntu box where I can actually get things done |
| 17:24.45 | tzanger | "pet the cat backwards" hahaha |
| 17:25.29 | drmessano | tzanger, I think we're in opposite places.. I use Word Processing extensively, and my only use for Excel is restructuring data and dumping it into other places |
| 17:25.49 | drmessano | So LO's Calc works great.. and it opens faster |
| 17:26.07 | [tpn]Cuz | drmessano: I just work in Windows |
| 17:27.19 | tzanger | drmessano: is the 32000 row limit still there? |
| 17:28.03 | DBordello | tzanger, anything that large, excel is probably the wrong tool anyways |
| 17:28.23 | DBordello | MATLAB is superior to Excel for plotting |
| 17:28.36 | drmessano | tzanger, I had to google.. I have never come close. Apparently it's not.. they have a changelog for 3.3 that notes 1 million rows in Calc |
| 17:28.45 | tzanger | DBordello: nah, it works *fantastically* well. It may very well be a misapplication but when it comes to grabbing a huge csv file, filtering/picking/manipulating the data and graphing it... it's awesome |
| 17:29.00 | tzanger | drmessano: I may have to give LO another go |
| 17:29.11 | DBordello | tzanger, perhaps :) I use excel a lot as well |
| 17:30.48 | drmessano | tzanger, like I said, they've made HUGE changes since LO forked. Really shows just how much progress was being impeded.. I was skeptical that LO could exist/succeed as even OpenOffice was a bit of red-headed stepchild, but I think they've transformed it so much that OpenOffice is a distant, progress-restricted memory |
| 17:30.58 | *** join/#asterisk serafie (~erin@nat/digium/x-uzswuyormuocvwpp) |
| 17:31.57 | *** join/#asterisk Methose (~Methose@MAIL.VERTISYS.COM) |
| 17:31.58 | drmessano | I remember them having a booth at SELF 2011, and they were talking about the BIG changes they had in store.. and I thought "HA, a fork with a roadmap promising big things. Thats SOOO original" |
| 17:32.02 | drmessano | But yeah |
| 17:33.02 | igcewieling | ~ook |
| 17:33.04 | igcewieling | ~book |
| 17:33.04 | 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 |
| 17:35.10 | Methose | I have a signel phone that after being factory defaulted, and reconfigured, will not connect to the pbx. From everything that I can see on the Polycom webGUI, the settings are identical with a working phone (with the exception of the extension). Could someone point me to a log where I might be able to see why the phone does not come online? It get's an IP via DHCP, and had dialtone, but... |
| 17:35.11 | Methose | ...cannot dial outbound and does not appear online. (forgive me I'm newish) |
| 17:38.23 | ChannelZ | The console should reveal all |
| 17:38.32 | ChannelZ | asterisk -rvvv |
| 17:39.14 | ChannelZ | If it's registering but badly, you'll see it. If it's not registering, you'll obviously see nothing and probably the phone is confused or misconfigured about whom it's supposed to be talking to |
| 17:40.40 | *** join/#asterisk SuPrSluG (~SuPrSluG@rrcs-50-75-185-122.nys.biz.rr.com) |
| 17:41.18 | Methose | ok, let me reboot her and see what it shows, thank you ChannelZ |
| 17:47.22 | Methose | uhg, so I've rebooted it twice now, an I do not see any entry that seems related in the console. is there a log file, where I might see if it's a bad password for the registration or soemthing of that nature? |
| 17:47.57 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 17:48.14 | ChannelZ | If it was a bad password you'd see the failure to authenticate on the console. |
| 17:48.30 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 17:48.37 | ChannelZ | It's probably got the wrong hostname/IP for the proxy server (asterisk) or something |
| 17:51.14 | Methose | it's getting the dhcp from the asterisk server, though I think that these phones are manually configured for the user info. I've quadruple checked all the info I can see from the webgui on the phones, so I'm wondering if there's something that I cannot see without the physical phone |
| 17:52.06 | Methose | Polycom SoundPointIP something or other. |
| 17:54.58 | *** part/#asterisk cybrNaut (cybrNaut@unaffiliated/cybrnaut) |
| 17:55.45 | ChannelZ | I dunno about them specifically but DHCP its self isn't magic as far as asterisk is concerned.. unless you are providing a tftp server or something in the DHCP response and the phone is supposed to provision its self from a config file that way |
| 17:57.38 | igcewieling | A wireshark or tcpdump will tell you if the phone is trying to talk to the server. |
| 17:57.47 | ChannelZ | If you know the IP of the phone, you can do "sip set debug ip 1.2.3.4" on the console and then do something on the phone (reboot it again perhaps) - if you see nothing on the console, then packets aren't making it to Asterisk. Whether it's because the phone is not sending them, or sending them to the wrong place, or something on the network is preventing the packets from getting there, etc. is something you'll have to determine |
| 17:57.52 | igcewieling | Then you'll know where to concentrate your efforts |
| 17:58.37 | Methose | I'm guessing there's no way to see that? I do see MAC addresses in the FreePBX gui, so perhaps it is getting it's config from tftp. |
| 17:58.52 | Methose | doing a set sip debug now, as I do have the ip of the phone. |
| 17:58.57 | *** part/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 17:59.02 | Methose | thanks again for all the help guys |
| 18:00.33 | Methose | No such command 'set sip debug ip 192.168.1.156' ( |
| 18:00.52 | ChannelZ | sip set not set sip |
| 18:01.17 | Methose | doh! it's been a long day already :( |
| 18:01.53 | *** join/#asterisk imox (~imox@24.134.18.71) |
| 18:03.54 | Methose | nice! now we're getting somewhere...I do see some info in there now |
| 18:04.37 | Methose | <--- Transmitting (NAT) to 192.168.1.156:5060 ---> |
| 18:04.38 | Methose | SIP/2.0 403 Forbidden (Bad auth) |
| 18:05.44 | Methose | so that sounds like a bad password, right? |
| 18:06.14 | ChannelZ | and/or it's not matching the peer you think it is |
| 18:09.16 | Methose | maybe you could clarify that statement a bit? the peer is the extension/user info? |
| 18:10.58 | ChannelZ | oh.. right.. freepbx.. maybe/probably? |
| 18:11.27 | tm1000 | lol |
| 18:11.34 | tm1000 | ChannelZ has now given up hope :-) |
| 18:12.38 | Methose | didn't mean to crush hopes and dreams |
| 18:12.58 | ChannelZ | Well, run out of my sphere of knowledge anyway |
| 18:13.45 | tm1000 | Methose: did you set it up manually? |
| 18:13.50 | ChannelZ | FPBX's goal is to obfuscate asterisk so I have no idea what it calls what. Apparently it calls devices 'extensions' which is not the most clear thing. |
| 18:14.50 | [TK]D-Fender | ChannelZ: Stopp alluding that FreePBX is specifically out to get you :) |
| 18:14.58 | ChannelZ | Anyway assuming the phone is trying to auth as the right 'user' as whatever entity you set that up with in FreePBX, then probably the password is wrong. Or vice-versa. |
| 18:14.59 | [TK]D-Fender | ChannelZ: You're more like "incidental damage" :) |
| 18:15.26 | Methose | I did not originally set up the system, we inherited their phone system support when we took over IT role |
| 18:15.48 | tm1000 | and somehow. after 5 lines. my question goes unanswered |
| 18:16.04 | Methose | ^^ |
| 18:16.07 | tm1000 | [TK]D-Fender: ChannelZ dislikes freepbx. I think we alllll know that by now ;-) |
| 18:16.17 | tm1000 | Methose: I meant the phone |
| 18:16.17 | carrar | ChannelZ, the NSA is out to get you |
| 18:16.24 | tm1000 | how did you get the credentials into it |
| 18:16.29 | ChannelZ | I dislike trying to help FreePBX people in an Asterisk channel |
| 18:16.35 | tm1000 | how did you even know what the password for the phone was? did you look it up |
| 18:16.45 | tm1000 | ChannelZ: I disagree with your statement |
| 18:17.01 | tm1000 | if it's a generic asterisk issue that can be taken outside the context of what a freepbx problem is then ask in here |
| 18:17.35 | ChannelZ | because the help we can offer usually gets stomped on by FPBX or we're discussing things that have no meaning to the FPBX user because they don't know what is going on underneath |
| 18:17.36 | [TK]D-Fender | ChannelZ: I agree with your statement. You clearly DO dislike trying to help FreePBX people in an Asterisk channel :) |
| 18:17.50 | tm1000 | [TK]D-Fender: +1 |
| 18:17.58 | Methose | oh, it was factory defaulted, and then it (to my knowledge) was reconfigured manually -I'm about to verify that |
| 18:18.37 | [TK]D-Fender | tm1000: Skills of great grammar have I hhmmmmMMMM!?!?!? |
| 18:18.44 | Methose | I love the back and forth that FreePBX and asterisk channels give :) one always sends me to the other |
| 18:18.59 | drmessano | I think if you don't like helping FreePBX people in an Asterisk channel you should just not help them. No need to go on and on about how much you don't want to help |
| 18:19.07 | drmessano | Just sayin |
| 18:19.18 | ChannelZ | It's not an elitism |
| 18:19.21 | Methose | I appreciate any help I can get |
| 18:19.25 | drmessano | Didn't say it was |
| 18:19.58 | ChannelZ | It's "I have no idea how to help you actually" after a 20 minute back-and-forth and THEN discovering they're using a GUI |
| 18:20.30 | Methose | so chat on. I think there must be a config issue between the phone and the PBX, either set by FreePBX or on the station. |
| 18:21.48 | carrar | ChannelZ, need to negotiate acceptable protocols before handshaking |
| 18:21.48 | carrar | a=FreePBX isn't one of those |
| 18:21.48 | carrar | or somethign |
| 18:21.48 | _Corey_ | loves a good SDP joke |
| 18:21.52 | carrar | heh |
| 18:22.36 | ChannelZ | The best thing about UDP jokes is you don't care whether anyone gets them. |
| 18:22.42 | drmessano | HA.. |
| 18:22.45 | carrar | so true! |
| 18:23.20 | _Corey_ | :) |
| 18:27.02 | tm1000 | Methose: if you have freepbx problems go to freepbx. |
| 18:27.12 | tm1000 | if we assume they are asterisk related we will send you here |
| 18:27.14 | ChannelZ | in any event, for the time being, it sounds like Methose has a username/password problem |
| 18:27.19 | tm1000 | we have a good relationship with all in here |
| 18:27.25 | tm1000 | I just like to give ChannelZ a hard time :-) |
| 18:32.28 | *** join/#asterisk navaismo (~navaismo@189.241.51.79) |
| 18:32.34 | Methose | ok, changing channels |
| 18:33.07 | tm1000 | Methose: ummmm. whats the password of the ext and is it the same as on the phone |
| 18:33.10 | tm1000 | thats all you need to check |
| 18:58.13 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 18:59.41 | *** join/#asterisk kresp0 (~kresp0@18.Red-83-42-125.dynamicIP.rima-tde.net) |
| 19:05.13 | igcewieling | ChannelZ: sometimes I think we should not care that someone is running FreePBX and direct them to edit extensions.conf and sip.conf. If they are wasting our time, why not waste theirs in return? |
| 19:06.22 | Robotman321 | then you get a rap for being trolls.. |
| 19:06.52 | *** join/#asterisk [tpn]Cuz (~ccuzner@198.41.29.45) |
| 20:11.04 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:11.04 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 20:11.39 | n0rth | trying being the operative word igce |
| 20:12.26 | gavimobile | igcewieling: yea I see that |
| 20:12.29 | gavimobile | sorry n0rth |
| 20:12.44 | gavimobile | handling a few things at once here but that not your problem ;-) |
| 20:13.03 | igcewieling | gavimobile: let us know when you have time to concentrate on the issue |
| 20:13.19 | gavimobile | n0rth: when you say externip, you mean in sip.conf? this is the same that it was and is |
| 20:15.08 | n0rth | what version did you upgrade from? |
| 20:15.08 | gavimobile | well my issue seems more frequent now 3 out of 3 |
| 20:15.08 | gavimobile | n0rth: I upgraded from 1.8 |
| 20:15.08 | gavimobile | to 11 |
| 20:15.08 | n0rth | what does rtp debug show? |
| 20:15.08 | n0rth | you got media streams? |
| 20:18.22 | igcewieling | n0rth: sip debug should show if Asterisk is sending out the public ip in the SDP |
| 20:18.22 | igcewieling | gavimobile: what is nat= set to |
| 20:18.22 | gavimobile | igcewieling: ohhh sh*t I did change nat |
| 20:18.22 | gavimobile | I feel stupid |
| 20:18.22 | n0rth | good |
| 20:18.22 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 20:18.22 | n0rth | as long as you never make that mistake again were all good ;) |
| 20:18.22 | gavimobile | let me try, but that will fix my problem |
| 20:18.22 | gavimobile | haha n0rth |
| 20:18.22 | igcewieling | gavimobile: unless you are running a recent Asterisk 11, using the recommended settings will break NAT |
| 20:18.22 | igcewieling | EXACTLY what version of Asterisk 11 are you running? |
| 20:19.44 | n0rth | gavimobile: Im not sure I would upgrade to asterisk 11 on my production server from 1.8 yet |
| 20:20.14 | n0rth | my fun box has 11.3 tho |
| 20:20.25 | igcewieling | n0rth: we use 11 on a few of our production servers, but not all of them |
| 20:20.27 | gavimobile | ok got some more info. nat was comented because it was displaying a bunch of junk ill talk about in a few minuts in my log |
| 20:20.50 | gavimobile | asterisk 11 says nat=yes is depreciated and I should use nat=force_rport |
| 20:21.05 | igcewieling | gavimobile: I'm not going to ask again. (4:18:07 PM) igcewieling: EXACTLY what version of Asterisk 11 are you running? |
| 20:21.33 | gavimobile | but when I went to test now nat=force_rport didn't seem to work. only nat=yes worked |
| 20:21.35 | n0rth | igcewieling: i like 11 a lot , but 1.8 is rock solid |
| 20:21.48 | gavimobile | sorry igcewieling im reading your posts now |
| 20:21.56 | gavimobile | how can I check the version of 11 im running? |
| 20:22.12 | n0rth | XD |
| 20:22.15 | igcewieling | Early versions of Asterisk 11 had a bug where NAT would become disabled if you followed the documented settings for nat= in Asterisk 11. I don't know if you are using one of those or not since you don't know what version of Asterisk you are running |
| 20:22.26 | igcewieling | gavimobile" asterisk -rvvv" |
| 20:22.37 | gavimobile | igcewieling: this is the current version I downloaded today *-current |
| 20:22.39 | gavimobile | one sec |
| 20:22.44 | n0rth | 11.4? |
| 20:22.53 | gavimobile | 11.4.0 |
| 20:22.58 | gavimobile | n0rth: yes |
| 20:23.01 | n0rth | im using 11.3 |
| 20:23.12 | gavimobile | n0rth: the chan_motif is cool |
| 20:23.14 | gavimobile | :-p |
| 20:23.27 | igcewieling | Unfortunatly, I don't have the patience today to help another high maintainence person needing help, so I wish you the best of luck and hope someone can help you. |
| 20:23.28 | n0rth | yeah free calls with google voice |
| 20:24.21 | gavimobile | n0rth: I had the calls working with 1.8 but it just stoped working today |
| 20:24.39 | gavimobile | igcewieling: sorry if I caused you any inconvience... you are fast though |
| 20:26.10 | gavimobile | so what do you think about nat, should I just leave it as yes |
| 20:26.19 | gavimobile | I hate when my log looks dirty |
| 20:27.29 | gavimobile | I think I may have misread it |
| 20:27.56 | phyrexianslug | Anybody know dahdi with T1 cards and Rhino channel banks? |
| 20:28.01 | gavimobile | <PROTECTED> |
| 20:34.52 | n0rth | not sure on that gavimobile |
| 20:34.52 | gavimobile | n0rth: im playing around with it now |
| 20:34.52 | gavimobile | maybe I can get figure it out |
| 20:34.52 | n0rth | yeah just look at debug |
| 20:34.54 | gavimobile | will do! hope I didn't tic off igcewieling, he sounds really knowledable with * |
| 20:34.56 | n0rth | yeah you prob did |
| 20:35.38 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 20:37.18 | gavimobile | well that makes me feel much better now |
| 20:54.00 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:54.00 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 20:54.08 | ChannelZ | fondles infobot |
| 20:54.47 | phyrexianslug | Problem is I changed it already, everything says it should be in em_w, asterisk won't load the module thinking the line is in the wrong mode |
| 20:54.49 | igcewieling | phyrexianslug: I thought your issue was already solved. |
| 20:55.12 | igcewieling | "module unload chan_dahdi.so" and "module load chan_dahdi.so" |
| 20:55.15 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 20:55.50 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 20:55.51 | igcewieling | phyrexianslug: what does dahdi_cfg show you? |
| 20:55.55 | phyrexianslug | unload reports module's not loaded, load reports the error. |
| 20:56.18 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 20:56.33 | phyrexianslug | Output of dahdi_cfg -vvv -> http://pastebin.com/1wA1H7h4 |
| 20:57.12 | igcewieling | you have your card configured for PRI |
| 20:58.02 | pzn | asterisk started not stablishing any call. phones just dial and ring, but the audio does not appear. "messages" file shows "app_dial.c: Unable to create channel of type 'sip' (cause 20 - Unknown)", nothing has changed in /etc/asterisk/* in the last 30 days. the problem occours with all phones to all extensions. version is 1.8.10. any hints? |
| 20:59.55 | igcewieling | pzn: that error message indicates the phones are not registered |
| 20:59.55 | pzn | igcewieling, "sip show peers" shows all them OK, not lagged. |
| 20:59.55 | n0rth | igcewieling: you work for digium? |
| 20:59.55 | igcewieling | n0rth: no. |
| 20:59.58 | pzn | igcewieling, wireshark shows the REGISTER/ACK packages going every 2 minutes (that is the timeout I put on the phones) |
| 21:00.06 | igcewieling | pzn: I guess standard troublehooting -- pastebin the CLI output of a failed call and hope someone can help. |
| 21:00.17 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 21:02.54 | danfromuk | Is it possible to SIP DEBUG on a per peer basis? |
| 21:02.58 | *** join/#asterisk navaismo (~navaismo@189.241.51.79) |
| 21:03.32 | phyrexianslug | sip set debug peer [X] ? |
| 21:03.39 | igcewieling | danfromuk: yes. |
| 21:04.20 | phyrexianslug | to anyone that I was asking earlier, thanks much. problem was out of band. :P |
| 21:04.22 | pzn | calls stopped working. http://pastebin.com/Lh1HyvPL can anyone help? |
| 21:04.28 | danfromuk | I never spotted that one. Thanks! |
| 21:05.13 | danfromuk | Doesnt appear to work. |
| 21:05.29 | igcewieling | pzn: 1) that call worked. 2) pastebin the output of 503 |
| 21:05.41 | navaismo | Hi there i have this odd issue, I have a license for g729 codec, so i did the same procedure as always, and when i try to load the module it said: Error loading module 'codec_g729a.so': /usr/lib/asterisk/modules/codec_g729a.so: cannot open shared object file: Permission denied |
| 21:05.42 | navaismo | <PROTECTED> |
| 21:05.54 | phyrexianslug | @danfromuk : sip set debug ip ? |
| 21:06.04 | phyrexianslug | if it's not a real "peer" |
| 21:06.22 | n0rth | navaismo: when you moved the module over did you make sure it was owned by asterisk? |
| 21:06.28 | danfromuk | It is a real peer as defined in sip.conf. |
| 21:06.30 | n0rth | or the user that asterisk is running ? |
| 21:06.37 | navaismo | but if a stop asterisk and run as asterisk -vvvvcg it load. |
| 21:06.51 | danfromuk | 'sip set debug on' works fine for those all peers. |
| 21:06.52 | pzn | igcewieling, 503 does not exist. it existed some months ago. |
| 21:06.55 | navaismo | n0rth, yup tried with chown and same efect |
| 21:07.03 | igcewieling | pzn: then why are you dialing it? |
| 21:07.13 | navaismo | n0rth, only running as root the asterisk -vvvcg command it load |
| 21:07.48 | pzn | igcewieling, it was the phone of one secretary that is not working anymore. should I have problems dialing it? is it a must to remove it? |
| 21:08.13 | igcewieling | pzn: Dialing a device which is not registered is the cause of your error message. |
| 21:08.39 | pzn | igcewieling, ok, I got it. but why there is silence and no audio? |
| 21:09.00 | igcewieling | pzn: audio issues are almost always NAT issues. |
| 21:09.01 | pzn | igcewieling, I'll capture sip + rtp packages to give a better debug information |
| 21:09.04 | n0rth | navaismo: what version you on? |
| 21:09.10 | navaismo | 11 |
| 21:09.15 | navaismo | 11.3.0 |
| 21:09.25 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:09.27 | igcewieling | pzn: I have no interest in slogging through sip and rtp debug. |
| 21:09.41 | n0rth | 32/64? |
| 21:09.51 | navaismo | 64 |
| 21:10.22 | igcewieling | pzn: does sip debug show the EXTERNAL IP address on all outgoing packets, including in the SDP part of the packet? |
| 21:10.24 | n0rth | you got the right version on there? |
| 21:10.54 | navaismo | n0rth, the becnhamark tool suggest the barcelona arch so i downloaded the codec_g729a-11.0_3.1.5-barcelona_64 |
| 21:10.59 | pzn | igcewieling, internal PBX. all IPs in the same 192.168 network |
| 21:11.05 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 21:11.42 | igcewieling | pzn: then you should make sure you have no nat= externip= externhost= or localnet= set |
| 21:12.05 | igcewieling | oh and turn off direct media and upgrade your Asterisk |
| 21:12.56 | igcewieling | pzn: could the issue be your PSTN card died? |
| 21:13.41 | igcewieling | Actually what PSTN card do you have? |
| 21:15.53 | n0rth | navaismo: i would try a different version |
| 21:28.39 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 21:28.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:29.02 | *** join/#asterisk netman (~netman@154.121.20.95.dynamic.jazztel.es) |
| 21:29.12 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.156) |
| 21:29.25 | navaismo | just chowned to asterisk.asterisk original the binary has root as owner |
| 21:29.41 | navaismo | -rwxr-xr-x 1 asterisk asterisk 1.1M Jun 13 23:10 codec_g729a.so |
| 21:29.43 | pzn | igcewieling, stopping asterisk process and starting it again |
| 21:30.16 | n0rth | that looks good nava |
| 21:30.24 | n0rth | try loading the module manually from the cli |
| 21:30.46 | ChannelZ | either that or you downloaded the wrong architecture |
| 21:30.50 | ChannelZ | possibly maybe |
| 21:30.53 | navaismo | hehe n0rth i did that from that i gave you the error log |
| 21:31.30 | ChannelZ | oh but it works as root. |
| 21:31.32 | igcewieling | navaismo: it is a commercial Digium product, they provide support |
| 21:31.33 | ChannelZ | scratches his head |
| 21:31.55 | navaismo | too |
| 21:31.57 | ChannelZ | (Do any of your modules work when run not as root? Or just g729 doesnt?) |
| 21:32.06 | navaismo | just g729 |
| 21:32.22 | n0rth | google is indicating that you have the wrong binary |
| 21:32.34 | navaismo | no as root it load fine |
| 21:32.45 | n0rth | then let it run as root? |
| 21:33.17 | n0rth | what is your asterisk pid running as? |
| 21:33.23 | navaismo | asterisk |
| 21:33.50 | navaismo | i must said before that is the freepbx |
| 21:34.58 | n0rth | here is how mine is running |
| 21:35.02 | n0rth | -rwxr-xr-x 1 root root 953340 Feb 8 12:55 codec_g729a.so |
| 21:35.02 | n0rth | -rwxr-xr-x 1 root root 7568 Feb 5 20:06 format_g729.so |
| 21:35.06 | n0rth | running as root |
| 21:35.08 | n0rth | infact |
| 21:35.10 | n0rth | all my modules are |
| 21:35.40 | n0rth | but /usr/sbin/asterisk runs as asterisk |
| 21:35.57 | n0rth | safe_asterisk also as root |
| 21:37.35 | n0rth | damn this IRC is badass |
| 21:38.21 | navaismo | uh? |
| 21:39.47 | ChannelZ | You were running asterisk manually as root.. how does it start up otherwise, an init script? |
| 21:40.28 | ChannelZ | wondering if something weird is going on in there or what the heck is happening. There's always my fallback query, "is this some selinux issue?" |
| 21:41.03 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 21:41.05 | n0rth | selinux is disabled be default on freepbx? |
| 21:41.19 | navaismo | yes |
| 21:41.22 | n0rth | unless…he hacked it |
| 21:41.37 | navaismo | ChannelZ, amportal start run the safe_asterisk script as fr i know |
| 21:42.38 | igcewieling | navaismo: verify all your /etc/asterisk config files are readable by asterisk |
| 21:43.08 | navaismo | yup all owned by asterisk |
| 21:43.24 | ChannelZ | well something is not as it seems. |
| 21:43.37 | navaismo | right |
| 21:43.43 | ChannelZ | do any of your modules work!? |
| 21:43.48 | navaismo | must contact digium support Thanks |
| 21:43.52 | navaismo | yes all other stuff works fine |
| 21:44.19 | n0rth | ouch…ill never go crawling to support |
| 21:44.29 | gavimobile | what does asterisk 11 have to do with my fail2ban, I remove a droped user from iptables, but when I restart asterisk he comes back as blocked. |
| 21:44.42 | n0rth | even if I have to google all night long |
| 21:44.46 | navaismo | n0rth, i only ask for digium support when i have no more options |
| 21:44.50 | gavimobile | pretty cool, but I need to know where to control this |
| 21:44.51 | sawgood | gavimobile: fail2ban will stop the user for 2-3 minutes ... maybe more |
| 21:45.07 | gavimobile | sawgood: no its been more than that |
| 21:45.10 | ChannelZ | well that's a fail2ban issue, keeping a cache of the rules and reinstating them probably |
| 21:45.24 | navaismo | n0rth, im not like a n00b using asterisk 7 years so far many installations on g729 so thats why im going to ask |
| 21:45.31 | sawgood | gavimobile: service fail2ban stop (and) service iptables stop (then) iptables -F (then) restest |
| 21:45.45 | gavimobile | sawgood: let me try that |
| 21:45.50 | gavimobile | I did restart but not stop |
| 21:45.52 | n0rth | not calling you a n00b |
| 21:46.04 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 21:46.10 | n0rth | just saying Id rather fail than ask for help |
| 21:46.19 | n0rth | or figure something else out |
| 21:46.20 | navaismo | n0rth, periodically supporter here but sometimes ask with the company is better than waste hours |
| 23:02.05 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 23:02.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 23:02.39 | n0rth | anyone in here use freeswitch? |
| 23:05.28 | WIMPy | Wrong channel? |
| 23:07.16 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ywmgnaaktinuxgiu) |
| 23:20.05 | *** join/#asterisk netman (~netman@194.167.76.188.dynamic.jazztel.es) |
| 23:20.11 | *** join/#asterisk DEMNVT (~Adium@rmsaus7.lnk.telstra.net) |
| 23:29.32 | n0rth | just wondering, I have messed with it a little but I prefer asteris k |
| 23:29.49 | n0rth | supposedly one of the main developers moved onto freeswitch |
| 23:30.33 | n0rth | WIMPy i just read a tech blog and your name came up a lot |
| 23:31.36 | n0rth | you must be a pwner |
| 23:42.48 | WIMPy | Don't think so. What did you read? |
| 23:42.59 | n0rth | just good things |
| 23:44.22 | WIMPy | Haha |
| 23:44.36 | n0rth | stupid aastra's |
| 23:44.41 | n0rth | i hate those phones |
| 23:45.05 | WIMPy | seriousely wonders whare his name should "come up a lot". |
| 23:45.17 | n0rth | ugh |
| 23:45.17 | n0rth | hold on |