| 00:03.37 | *** join/#asterisk b14ck (~b14ck@173.219.15.98) |
| 00:10.46 | NEEDINGHELP123 | guys |
| 00:10.47 | NEEDINGHELP123 | anyone here? |
| 00:11.27 | ChannelZ | nope |
| 00:11.30 | ChannelZ | just us trannies |
| 00:11.47 | NEEDINGHELP123 | ha |
| 00:11.49 | NEEDINGHELP123 | ++++ ooh323_indicate 18 on ooh323c_o_3 |
| 00:11.50 | NEEDINGHELP123 | --- setup_rtp_connection |
| 00:11.50 | NEEDINGHELP123 | --- find_call |
| 00:11.50 | NEEDINGHELP123 | +++ find_call |
| 00:11.50 | NEEDINGHELP123 | +++ setup_rtp_connection |
| 00:11.50 | NEEDINGHELP123 | --- ooh323_hangup |
| 00:11.50 | NEEDINGHELP123 | <PROTECTED> |
| 00:11.51 | NEEDINGHELP123 | +++ ooh323_hangup |
| 00:11.51 | NEEDINGHELP123 | --- close_rtp_connection |
| 00:11.52 | NEEDINGHELP123 | --- find_call |
| 00:11.52 | NEEDINGHELP123 | +++ find_call |
| 00:11.53 | NEEDINGHELP123 | +++ close_rtp_connection |
| 00:11.53 | NEEDINGHELP123 | --- onCallCleared ooh323c_o_2 |
| 00:11.54 | NEEDINGHELP123 | --- find_call |
| 00:12.00 | ChannelZ | yeah nobody is going to help you when you flood the channel like that |
| 00:12.47 | NEEDINGHELP123 | i'm not flooding i'm posting the msg |
| 00:12.53 | ChannelZ | ~pb |
| 00:12.53 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 00:16.27 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 00:34.43 | leifmadsen | NEEDINGHELP123: it's still called "flooding" |
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| 01:50.40 | cmendes0101 | Im trying to stream audio out of a call by using EAGI. I'm using a method I found on the internet that calls this script by EAGI http://asterisk.pastey.net/138873 |
| 01:51.16 | cmendes0101 | Its not working to well. Is there a way I could take out the ezstream portion to replace with something else to test. Like if I output to a file could I play that file back? |
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| 02:14.17 | *** mode/#asterisk [+o bkruse] by ChanServ |
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| 02:24.33 | TJNII | cat /dev/fd/3? Wouldn't that be some open file for the bash process? What kind of chicanery is that script doing? |
| 02:25.28 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
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| 02:31.37 | *** join/#asterisk aster1sk (~aster1sk@69-165-175-216.dsl.teksavvy.com) |
| 02:38.14 | aster1sk | If you had unlimited development resources for a from-scratch web based Asterisk manager panel - what would you want to see? |
| 02:41.19 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 02:42.09 | *** part/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 02:44.11 | xheliox | A developer behind the project that doesn't have to ask questions like that. |
| 02:44.13 | chuckf | I'd like to see something that works in all cases from the novice user to the most advanced. I want something that autocompletes what I intend to do and not what I actually put into the field. |
| 02:47.16 | aster1sk | I agree with both of you, though in my defence I believe a dev must probe to see what the community wants. |
| 02:47.58 | aster1sk | ... in order to create a successful project. |
| 02:48.56 | aster1sk | I believe that on the fly validation is key, meaning that there are checks in place to determine conflicting configurations. |
| 02:49.32 | aster1sk | I also believe that there should be a basic / advances user interface for those who are looking for total control. |
| 02:50.56 | aster1sk | i just built a stock minimal os in a VM with * 1.6.2 to spec - I want to write a secure web UI as a replacement for the alternatives. |
| 02:50.57 | *** join/#asterisk comradeb14ck (~b14ck@173.219.15.98) |
| 02:52.41 | aster1sk | This is probably the wrong channel to be asking these kinds of questions, most of us run vanilla * and have no need for a ui. |
| 02:53.26 | [TK]D-Fender | aster1sk: No necessarily |
| 02:53.49 | aster1sk | [TK]D-Fender: elaborate. |
| 02:54.06 | [TK]D-Fender | aster1sk: Yes, we appreciate the raw control over the core, but LOTS of people here run GUI installs as well. So far there aren't a lot of good free ones. |
| 02:54.35 | [TK]D-Fender | aster1sk: FreePBX isnt multi-tennent, and has other failings (which are being worked on for 3.0 but thats far off and not for * yet) |
| 02:54.44 | mmlj4 | anyhow, define secure |
| 02:54.47 | [TK]D-Fender | aster1sk: AsteriskGUI is well ... dead |
| 02:55.19 | aster1sk | Well first of all the web daemon should not be run as root. |
| 02:55.22 | aster1sk | or asterisk |
| 02:55.49 | aster1sk | secondly, auth should be both session / ip based. |
| 02:56.11 | aster1sk | I also think that openssl would be nice, even with an unsigned cert. |
| 02:56.16 | mmlj4 | mine runs as user apache... I don't see a problem |
| 02:56.31 | mmlj4 | you want SSL? trivial |
| 02:56.49 | aster1sk | Oh that is fine, I am talking about the dists that run * as a priv user |
| 02:56.55 | *** join/#asterisk grabes (~grabes@70.15.27.211.res-cmts.sm.ptd.net) |
| 02:57.31 | [TK]D-Fender | aster1sk: Off direct functionality the others don't. yes the invisible perks like "stock" settings that are more secure are a bonus, but if you want to stand out, then you need to work different. |
| 02:57.57 | aster1sk | I was thinking about a drag/drop IVR |
| 02:58.07 | mmlj4 | for HTTP auth... well, turn it on and limit as you please |
| 02:58.26 | mmlj4 | my point is... what you're wanting isn't that groundbreaking |
| 02:58.53 | aster1sk | no sh1t, you focused on the security comment. |
| 02:59.24 | [TK]D-Fender | aster1sk: And you just mentioned about the first NON-security thing to do :) |
| 02:59.32 | mmlj4 | any what "ooh, asterisk, I wonder what that is... will it run on windows?" luser cars about security? |
| 02:59.33 | [TK]D-Fender | astaAfter I prodded :) |
| 03:00.16 | aster1sk | Heh true, well to be honest I am not really interested in writing support for anything but *nix / asterisk |
| 03:00.28 | aster1sk | but if the demand is there I may roll out a M$ version. |
| 03:00.53 | aster1sk | Let me outline my goals and see if this will convince you... |
| 03:01.00 | mmlj4 | <aster1sk> but if the demand is there I may roll out a M$ version. |
| 03:01.03 | mmlj4 | you just lost me |
| 03:01.11 | aster1sk | M$ == windows. |
| 03:01.35 | mmlj4 | i mean you just told us you don't really understand security at all |
| 03:01.35 | russellb | 1) Write GUI. 2) ??? 3) PROFIT |
| 03:01.56 | aster1sk | No profit whatsoever. |
| 03:01.56 | chuckf | but a web interface should be OS neutral |
| 03:02.15 | aster1sk | chuckf: false. |
| 03:02.33 | russellb | orly |
| 03:02.39 | chuckf | what do you mean false? Of course a web interface should work for all OS's |
| 03:02.42 | aster1sk | Server side. |
| 03:03.15 | aster1sk | I am saying I wouldn't write a port for 3cx or whatever it is called. |
| 03:04.04 | aster1sk | The app will only dump configs for * on a *nix box, the web ui will surely work cross platform, but the app is intended to configure *nix hosts. |
| 03:04.37 | aster1sk | I think that is where we got messed up. |
| 03:04.41 | chuckf | is there a * that runs on MS Windows? |
| 03:04.58 | aster1sk | I believe on old version however dev was discontinued. |
| 03:05.11 | pabelanger-lap | chuckf: http://www.asteriskwin32.com/ |
| 03:06.56 | aster1sk | none of you have probably never seen my any of my projects but everything I do out side of work is totally open source. Take a look at enumplus.org or geekhut.org/projects/asterisk-stickies/ |
| 03:07.05 | chuckf | it looks like the last updae was 2/2008. You cannot consider that a valid option |
| 03:07.37 | mmlj4 | lemme guess... you write in python? |
| 03:08.07 | aster1sk | PHP actually, but language doesn't really matter to an experienced developer. |
| 03:08.40 | russellb | what about doing it in bash |
| 03:08.41 | TJNII | Oh god. |
| 03:08.44 | mmlj4 | Wrong. Java. I win. |
| 03:08.47 | TJNII | Is that you, DrClue? |
| 03:09.04 | chuckf | assembly for the win! |
| 03:09.12 | aster1sk | Obfuscated perl FTW |
| 03:09.30 | aster1sk | guhh you beat me to it lol |
| 03:09.44 | TJNII | Obfuscated perl? Isn't that an oxymoron? |
| 03:09.46 | TJNII | ducks |
| 03:09.50 | aster1sk | Heh |
| 03:10.23 | russellb | oxymoron? you mean redundant? |
| 03:10.55 | aster1sk | russellb: ++ |
| 03:11.21 | aster1sk | buys beers for all |
| 03:11.53 | TJNII | russellb: Not the attack vector I was expecting. You are, however, correct. |
| 03:12.16 | chuckf | calls his AA sponsor telling of someone trying to sabotage his sobriety |
| 03:13.26 | russellb | ~dance |
| 03:13.27 | infobot | <(*.*<) <(*.*)> \(*.*)/ (>*.*)> |
| 03:16.21 | aster1sk | i love the simplicity of askozia, I would like to build upon the stability / ease of use with escalated control. |
| 03:17.21 | carrar | Is that Disco dan dancing? |
| 03:18.26 | *** part/#asterisk grabes (~grabes@70.15.27.211.res-cmts.sm.ptd.net) |
| 03:20.09 | pabelanger-lap | aster1sk: askozia was build on m0n0wall, so you should be fine. |
| 03:20.18 | pabelanger-lap | s/build/built |
| 03:23.22 | aster1sk | sed |
| 03:23.33 | russellb | awk |
| 03:23.36 | aster1sk | true - but now is a t2 buils |
| 03:23.40 | mmlj4 | newbie |
| 03:23.46 | aster1sk | I was thinking minimal debian. |
| 03:25.27 | aster1sk | heh guess a $few_beers->leet_status($status) |
| 03:25.29 | aster1sk | ss |
| 03:25.48 | aster1sk | guhh I am going away, time for real life heh. |
| 03:30.04 | [TK]D-Fender | Reality is for people who can't handle drugs... |
| 03:31.57 | aster1sk | [TK]D-Fender: how bout booze? |
| 03:32.06 | [TK]D-Fender | aster1sk: that counts |
| 03:32.09 | [TK]D-Fender | counts |
| 03:32.29 | aster1sk | well that explains why I am not making sense then. |
| 03:33.31 | aster1sk | you know what's got me drinkin so heavy... those damn 7961's with the 7914 sidecar. |
| 03:34.45 | aster1sk | built sccp-b on there and had to provision the device but the boss wouldn't give me router creds [though i doubt the router had 66 to begin with] |
| 03:35.40 | aster1sk | so I had to use my sketchy dev box with win 7 to load this skinny firmware.... i told him to buy aastra / polycom but he insisted the reception phone be the 7961 for asthetics. |
| 03:36.33 | *** part/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 03:36.39 | *** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 03:36.39 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:37.58 | aster1sk | now I would have been fine with SIP image on the 7961 but this bloody sidecar is not supported with sip, after a few hours I was able to get the thing runnig but had to catch up on other work that caused me to stay two hours later. |
| 03:41.18 | Gershwin | open up the 7961 and break something internally |
| 03:42.00 | Gershwin | because the broken phone means that you essentially "wasted your time"... tell your boss not to buy another POS cisco or he can set it up |
| 03:46.52 | aster1sk | ^^ good advice |
| 03:50.19 | [TK]D-Fender | "We can make it look like an accident" |
| 03:55.50 | *** join/#asterisk DaveCanoe (~Dave@strike.eicat.ca) |
| 03:56.47 | aster1sk | well once configured I wasn't so upset |
| 03:56.58 | aster1sk | but cisco's sure are a pain |
| 03:57.38 | TJNII | Again with the redundancies from you. |
| 04:04.49 | aster1sk | This channel isn't as friendly as I remember. |
| 04:05.05 | aster1sk | Well I am out of cigarettes, later fellas, |
| 04:11.33 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 04:11.44 | xheliox | rolls his eyes |
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| 05:27.11 | Mango | Anyone aware of a Canadian VoIP provider that went bankrupt this week? Apparently they stopped routing calls at 3:30PM Eastern on Wednesday. |
| 05:28.56 | coppice | it must be VoIP bankruptcy week |
| 05:29.37 | Mango | oh? |
| 05:30.37 | DogBoy | oh my word |
| 05:30.38 | coppice | Howlertech, the G.729 codec guys, and someone else I forgot also passed away this week |
| 05:30.49 | Mango | Rats :( |
| 05:38.23 | *** join/#asterisk sawgood (~sawgood@173-13-158-17-sfba.hfc.comcastbusiness.net) |
| 05:40.24 | drmessano | Glad I invested my money in their floating codec |
| 05:40.27 | drmessano | not |
| 05:40.43 | drmessano | well, floating codec license |
| 05:41.09 | drmessano | Another reason why cloud licensing sucks |
| 05:46.10 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 05:50.41 | *** join/#asterisk inwx (~inwx@190.2.33.129) |
| 06:01.30 | coppice | their non-floating licences will be just as screwed, as soon as anything in a system is changed |
| 06:04.13 | coppice | floating licences are a well liked idea by people who want to implement failover, but that's the very area where floating licences are most problematic |
| 06:14.51 | drmessano | Yeah |
| 06:20.18 | drmessano | Guess paying the extra 2 bucks for Digium G729 licenses was worth it |
| 06:20.27 | drmessano | Unless 1.8 sucks ass and Digium goes down in flames |
| 06:20.40 | drmessano | MAKE IT COUNT, GUYS |
| 06:24.31 | coppice | digium needs to streamline its versions. there are too many balls in the air |
| 06:25.32 | drmessano | I think that's where the new-old release model comes in |
| 06:26.18 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.94) |
| 06:27.55 | drmessano | It's a shame there's so many installs out there being admined by unqualified individuals, because it's going to be a while before 1.4 and the three 1.6.x releases go away..meaning those balls are gonna hang around for a while |
| 06:28.09 | drmessano | I left off 1.2 in there too |
| 06:28.31 | coppice | the three 1.6.x "releases" was total mismanagement |
| 06:31.09 | drmessano | I think they had the right idea with the faster releases, just wish the "LTS" would have kicked in sooner and maybe we would have 1.4 LTS out there, a 1.6, 1.8, and 1.10 release behind us, and be talking about a 1.12 LTS release now. |
| 06:33.34 | drmessano | I really didn't have a problem with the 1.6.x releases other than it confusing the crap out of newbs and fly-by-night "admins" who didn't "get it". |
| 06:34.21 | drmessano | You spent almost as much time figuring out which branch they were on as you did fixing their issue |
| 06:35.26 | drmessano | or the complaints that 1.6.2.x shouldn't be so different from 1.6.0.x, though that was entirely the idea. |
| 07:08.51 | *** join/#asterisk timholum_ (~chatzilla@72-160-218-212.dyn.centurytel.net) |
| 07:09.10 | timholum_ | has anyone sucsessfuly gotten the gtalk module to work? |
| 07:10.50 | *** join/#asterisk digitalml (digitalml@wsip-24-234-120-155.lv.lv.cox.net) |
| 07:13.07 | timholum_ | it appers that the my jabber and gtalk modules are not enabled, but I compiled? any idea's |
| 07:15.41 | timholum_ | I have autoload=yes in modules.conf and in my modules folder I have chan_gtalk.so and res_jabber.so, and they are not in an noload command in the modules.conf |
| 07:21.16 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 07:21.57 | timholum_ | is anyone online? |
| 07:25.48 | WIMPy | timholum_: No, the internet is broken. |
| 07:26.16 | timholum_ | WIMPy :), I ment in the channel that is still here :) |
| 07:26.22 | digitalml | nooooooooooooooooooo, not the internet |
| 07:26.24 | digitalml | i cant live without it |
| 07:26.26 | digitalml | :( |
| 07:26.38 | timholum_ | I need a new job if it is broken :) |
| 07:27.00 | WIMPy | digitalml: Maybe you can get some UUCP replacement. |
| 07:28.11 | timholum_ | :) |
| 07:28.39 | timholum_ | that would suck, I dont think logmein supports uucp :) |
| 07:29.48 | WIMPy | wonders why that could matter |
| 07:31.02 | timholum_ | 90% of my work is done on windows machines remotly ( using logmein ) |
| 07:31.51 | digitalml | just threw up a little in his mouth |
| 07:31.55 | WIMPy | sends his condolence |
| 07:32.30 | timholum_ | :) I like linux a lot better, but I have to go where the work is :) |
| 07:38.25 | timholum_ | hmm, when I try to do load chan_gtalk.so it tells me " loader.c:382 load_dynamic_module: Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory" yet I do have libiksemel.so.3 in /usr/local/lib/libiksemel.so.3 |
| 07:38.37 | timholum_ | is there a different directory that I have to load that into? |
| 07:42.12 | *** join/#asterisk bmg505 (~leon@196-209-99-210.dynamic.isadsl.co.za) |
| 07:43.05 | ChannelZ | ldconfig ? |
| 07:46.27 | *** join/#asterisk timholum1 (~chatzilla@75-121-147-5.dyn.centurytel.net) |
| 07:46.44 | timholum1 | sorry, my flaked out there for a min |
| 07:47.16 | timholum1 | ChannelZ: I just did ldconfig then restarted asterisk and I get the same error |
| 07:47.19 | WIMPy | So the internat actually IS broken? |
| 07:47.31 | WIMPy | scnr |
| 07:48.16 | timholum1 | :) my internet does that every few days, I am looking for a new ISP |
| 07:49.02 | timholum1 | it is worse when it rains, so I think there is a leek in one of the pedistils on my block |
| 07:51.41 | digitalml | so guys, setting up a small asterisk install at home and was wondering what a decent cheapo router would be to do QOS... I was thinking this: http://www.guideband.com/index.php/featured-products/v2920.html |
| 07:51.56 | digitalml | cheapo < $300 |
| 07:53.08 | WIMPy | Can't tell abot the specific product, but the Vigor series seem to perform very well. |
| 07:54.12 | digitalml | cant find a single review on the 2920 though |
| 07:54.14 | digitalml | :/ |
| 07:57.46 | ChannelZ | OoOoo, ASCII-art Asterisk logo for 1.8 |
| 07:58.09 | *** join/#asterisk timholum1 (~chatzilla@98-125-79-250.dyn.centurytel.net) |
| 08:06.06 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 08:22.46 | *** join/#asterisk garymc (~chatzilla@host86-162-166-186.range86-162.btcentralplus.com) |
| 08:23.46 | garymc | yo anyone around. |
| 08:24.29 | Kyosh | sometimes |
| 08:24.53 | garymc | I upgraded to 1.6 yesterday. Broke my working system. I must have had a few addons that need upgrading too. Is there anyway of me knowing or finding out what addons I had? I upgraded from 1.4 |
| 08:25.33 | Kyosh | i would have stayed at 1.4.2 |
| 08:25.47 | garymc | i need a feature that only 1.6 has |
| 08:25.54 | WIMPy | It should have moaned about them when installing. |
| 08:26.00 | Kyosh | hmm |
| 08:26.28 | Kyosh | http://www.asterisk.org/asterisk-versions |
| 08:26.30 | Kyosh | interesting |
| 08:26.35 | WIMPy | make install again and see if it moans about old modules. |
| 08:26.41 | Kyosh | 1.4 will outlive 1.6 |
| 08:27.12 | Kyosh | cept 1.6.2 |
| 08:27.18 | WIMPy | sees five days more for 1.6 there. |
| 08:27.45 | Kyosh | thats what i said |
| 08:27.47 | Kyosh | cept 1.6.2 |
| 08:28.04 | garymc | i got 1.6.2 sorry |
| 08:28.14 | garymc | i think letme check again :S |
| 08:28.27 | Kyosh | no i wont letyou |
| 08:29.00 | Kyosh | what happens when you teach your customer how to manage an asterisk box? |
| 08:29.48 | garymc | 1.6.2.9 |
| 08:29.49 | WIMPy | You long for an app_eliza? |
| 08:29.57 | Kyosh | they get a $3500 bill from verizon because the dumbasses added an extension with a 3 digit numeric secret which is accessible over the internet |
| 08:30.03 | garymc | I wont be teaching no body |
| 08:30.34 | garymc | im not good enough |
| 08:30.57 | garymc | Kyosh ouch |
| 08:31.10 | garymc | someone using chatlines for free :) |
| 08:31.25 | Kyosh | the best part is, they tried to blame me |
| 08:31.45 | Kyosh | oh no, they got hacked and someone was vishing all over the world, especially to africa |
| 08:31.55 | garymc | nice |
| 08:31.59 | Kyosh | ya |
| 08:32.31 | garymc | so do you know how to find out what plugins I had and what im now missing? |
| 08:32.32 | Kyosh | so i checked their log files and found that the ext in question was added this year. i installed asterisk 1-1/2 years ago with all configs at that time |
| 08:32.49 | Kyosh | nopes, i have no clue which app your missing |
| 08:32.58 | garymc | how could I find out? |
| 08:33.09 | Kyosh | no clue |
| 08:33.10 | WIMPy | garymc: make install again and see if it moans about old modules. |
| 08:33.21 | garymc | no it didnt moan |
| 08:33.43 | garymc | it went clean as a whistle but removed my Sangoma card |
| 08:33.56 | garymc | well the card disapeared and had to get new drivers for it |
| 08:34.02 | Kyosh | ack! |
| 08:34.11 | garymc | was a ball ache to get working so I wont be doing that again |
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| 10:58.04 | FILLVAIO3 | Hi guys, does anybody know wy wav files playing but streaming does not work in musiconhold? |
| 10:59.13 | tzafrir | FILLVAIO3, what do you mean by "does not work"? What have you set up? What have you expected to happen? What hapened? |
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| 11:01.58 | FILLVAIO3 | i have radio stream - http://www.stb.net.ru/radio/chanel/asx/90.asx, but in asterisk i can use mpg123 or streamplayer with <ip> <port> parameters |
| 11:03.35 | tzafrir | FILLVAIO3, what Linux program can you use to play it? |
| 11:04.30 | FILLVAIO3 | application=/usr/bin/mpg123 -q -r 8000 -f 8192 -s --mono http://www.stb.net.ru/radio/chanel/asx/90.asx |
| 11:05.02 | tzafrir | MIME type of that page: video/x-ms-asf |
| 11:05.22 | tzafrir | Look into mplayer |
| 11:07.25 | FILLVAIO3 | in debian? |
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| 12:43.05 | _zoom_ | guys have faced a problem of passing g729 over openvpn? |
| 12:44.08 | [TK]D-Fender | _zoom_: Why should OpenVPN care about the packets you pass in it? |
| 12:45.34 | _zoom_ | [TK]D-Fender: dont know, but when i use g711 it worked very good, but comes to .729 openvpn link goes down |
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| 12:48.30 | [TK]D-Fender | _zoom_: makes no sense. Its both basic UDP at the same packetization rate, just slightly smaller packets |
| 12:49.01 | pabelanger-lap | _zoom_: do you have g729 liceneses? |
| 12:49.13 | _zoom_ | [TK]D-Fender: yeah for one channel |
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| 12:50.57 | pabelanger-lap | we would need to see a debug log from asterisk, why the call is rejected |
| 12:53.02 | [TK]D-Fender | pabelanger-lap: He never said it was dropped or rejected by * |
| 12:53.12 | [TK]D-Fender | paHe said OpenVPN goes down. |
| 12:53.55 | _zoom_ | exactly |
| 12:54.25 | pabelanger-lap | Then debug openvpn |
| 12:54.53 | _zoom_ | it dropped when callee send 200 OK -accept- |
| 12:55.33 | [TK]D-Fender | _zoom_: And you're sure that OpenVPN itself is what goes down? |
| 12:55.35 | _zoom_ | this has nothing to do with firewalls or IDS a cross internet right? |
| 12:55.55 | pabelanger-lap | _zoom_: post a SIP Debug |
| 12:55.59 | [TK]D-Fender | ^^^^ |
| 12:56.43 | _zoom_ | yes, cause i got pinging replies from google.com |
| 12:58.40 | pabelanger-lap | I doubt openvpn is actually dropping, maybe a codecs issue |
| 13:02.17 | tzafrir | FILLVAIO3, in Debian? maybe look into http://debian-multimedia.org/ |
| 13:07.29 | FILLVAIO3 | i have sh script, where mplayer command [/usr/bin/mplayer "mms://87.242.72.62/relaxfm?WMBitrate=41600&WMContentBitrate=41600" -really-quiet -quiet -shuffle -ao pcm -format 0x2000 -channels 1 -af resample=8000 -ao pcm:file=$PIPE | cat $PIPE], but playing go too slow and with distortion. does any body know why? |
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| 14:05.16 | Goshen | ok, I got a Polycom 330, installed AsteriskNOW, Updated Freepbx, how do I get my phone talking to asterisk? :) |
| 14:06.44 | [TK]D-Fender | Goshen: www.polycom.com <- go download the admin guide |
| 14:07.20 | Goshen | thanks |
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| 14:08.55 | Goshen | Sweet, 421 pages! :) |
| 14:10.32 | NEEDINGHELP123 | Hi Guys |
| 14:10.38 | NEEDINGHELP123 | please advise |
| 14:10.45 | NEEDINGHELP123 | "no channel type registered for oh323" |
| 14:10.54 | NEEDINGHELP123 | even though the h323 module is loaded |
| 14:11.02 | NEEDINGHELP123 | I get he error above |
| 14:11.20 | NEEDINGHELP123 | the* |
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| 14:12.11 | [TK]D-Fender | NEEDINGHELP123: then use h323, not oh323. Ther are MULTIPLE H.3233 channel drivers for * |
| 14:12.45 | NEEDINGHELP123 | tried that aswell |
| 14:14.32 | [TK]D-Fender | NEEDINGHELP123: there is also ooh323 |
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| 14:55.58 | Goshen | how do I increase the verose level? set verbose doesn't work |
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| 14:57.46 | LemensTS | is there a cli cmd to show the variables of a call |
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| 15:44.40 | sokoow | hi all |
| 15:45.30 | sokoow | I need some help with receiving calls on 7941G, anybody would help please? |
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| 15:53.49 | sokoow | I need some help with receiving calls on 7941G, anybody would help please? |
| 15:54.52 | halindrome | I am using FreePBX (2.7.0.4) configured with a sipgate account and a gv account for incoming and outgoing calls. It was working great. Then I stupidly applied some centos upgrades. now when I make a call it seems to take about 30 seconds for anything to happen - like it is trying something and waiting for it to timeout. Does anyone know where I can look to figure out where it is getting wedged? |
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| 16:04.42 | pabelanger-lap | Goshen: core set verbose 15 |
| 16:05.11 | pabelanger-lap | ~ask |
| 16:05.12 | 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. |
| 16:05.14 | pabelanger-lap | sokoow: ^^^ |
| 16:05.35 | pabelanger-lap | halindrome: #freepbx |
| 16:07.04 | sokoow | pabelanger-lap ;) |
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| 16:16.46 | halindrome | pabelanger-lap: okay thanks |
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| 17:20.12 | stanmancan | Anybody have any issues installing on Ubuntu through aptitude ? |
| 17:20.47 | ChannelZ | just builds the source.. easy-peasy |
| 17:21.15 | stanmancan | I'm getting errors, mostly because of dahdi |
| 17:21.51 | stanmancan | sudo aptitude install asterisk asterisk-1.6.2 asterisk-config asterisk-sounds-extra asterisk-sounds-main |
| 17:22.11 | ChannelZ | what do you mean errors because of dahdi then |
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| 17:22.25 | stanmancan | https://gist.github.com/bf2c97ea6925f0d659ad |
| 17:22.34 | stanmancan | Do I need to install all/any of the recommended? |
| 17:22.43 | stanmancan | I've already installed linux-source |
| 17:23.27 | ChannelZ | not unless you want to build dahdi yourself from source |
| 17:23.53 | ChannelZ | but if you're going to do that you might as well build asterisk yourself too and not put on half these things you probably don't even need |
| 17:24.26 | stanmancan | I just figured using the package manager would be easier to do updates and stuff later? |
| 17:24.46 | stanmancan | Any good tutorials on 1.6/1.4 on building from source? |
| 17:24.57 | ChannelZ | shrugs - depends on what release stream you're on and if the package manager keeps it up to date |
| 17:25.02 | ChannelZ | ./configure |
| 17:25.04 | ChannelZ | make |
| 17:25.10 | ChannelZ | make install |
| 17:25.18 | chuckf | stanmancan: the package manager will be easier for updates, as the Ubuntu project updates things. But they may be slower than what you want |
| 17:25.18 | ChannelZ | Tutorial complete |
| 17:25.46 | stanmancan | I really don't expect to be updating that much TBH |
| 17:26.01 | ChannelZ | I know some people came here complaining of * in Ubuntu 10.4 awhile ago but I don't remember what the issue was |
| 17:26.05 | stanmancan | But I _am_ rather new in linux, I can make my way around it no problem but I don't have much experience building things |
| 17:27.17 | chuckf | stanmancan: will your asterisk install be on a dedicated box? |
| 17:30.22 | pabelanger-lap | stanmancan: You should beable to just do $ apt-get install asterisk, and it will resolve all dependancies. |
| 17:30.51 | pabelanger-lap | however, they do use some custom patches |
| 17:32.35 | stanmancan | Yea |
| 17:32.37 | stanmancan | well, on a VPS |
| 17:32.46 | stanmancan | runns my webserver and stuff too |
| 17:33.35 | xheliox | ducks |
| 17:33.54 | xheliox | I smell a "running Asterisk on a virtual guest" discussion coming on. |
| 17:34.24 | stanmancan | https://gist.github.com/6a45279201e3bc574263 |
| 17:34.43 | ChannelZ | I smell tacos |
| 17:35.03 | drmessano | TACOS |
| 17:35.19 | xheliox | Well at least we know how to excite drmessano. |
| 17:35.26 | stanmancan | Whats wrong with running on a vps? |
| 17:35.32 | drmessano | Everything |
| 17:35.38 | drmessano | Well, nothing |
| 17:35.39 | drmessano | Depends |
| 17:35.45 | xheliox | lol - that about sums it up. |
| 17:36.25 | ChannelZ | stanmancan: chances are you don't even need dahdi |
| 17:36.42 | stanmancan | dahdi's just for call conferencing right? |
| 17:37.00 | xheliox | stanmancan: Certain applications within Asterisk need a solid timing source. Some virtual machines are better at allowing that than others. |
| 17:37.33 | stanmancan | Should I just build from source? |
| 17:41.17 | digitalml | so guys, setting up a small asterisk install at home and was wondering what a decent cheapo router would be to do QOS... I was thinking this: http://www.guideband.com/index.php/featured-products/v2920.html |
| 17:41.51 | stanmancan | Got an old spare computer around? |
| 17:45.27 | stanmancan | if so you could drop a couple $40 intel NIC's in and make your own |
| 17:46.02 | digitalml | how do i build my own? |
| 17:46.11 | stanmancan | pfsense |
| 17:46.35 | stanmancan | there's a few other options, but that's what I've used |
| 17:47.07 | stanmancan | http://www.pfsense.org/index.php?option=com_content&task=view&id=52&Itemid=49 |
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| 17:55.12 | digitalml | the sip limitation on the NAT feature seems like a problem |
| 17:56.05 | stanmancan | There's a way around it but I can't recall exactly what |
| 17:56.14 | stanmancan | involves pinging every ____ |
| 17:57.18 | sokoow | anybody would help with cisco + asterisk combo ? |
| 18:00.57 | stanmancan | ChannelZ: any good tutorials for building from source? |
| 18:03.17 | digitalml | so right now im using a SIP provider for outbound calls and can get 5 to 10 simultanous connections at once and pay about 1c per min. if i ever grow past this. it is possible to buy a digium card that will take in a t1 connection and give me 24 outbound lines, right? |
| 18:03.46 | xheliox | digitalml: Correct, but you don't have to use a Digium card, there are other compatible hardware. |
| 18:04.42 | *** join/#asterisk mafrac (~mafrac@77.224.248.251) |
| 18:06.44 | mafrac | Hi anybody. |
| 18:07.28 | sokoow | hi all, could somebody help with cisco phone problem please : http://pastebin.com/VUXrH5ym |
| 18:07.31 | mmlj4 | I bet nobody answers you |
| 18:07.35 | digitalml | xheliox is there anything else that gives more lines than 24 per connection? |
| 18:07.43 | digitalml | or just gotta buy more t1 lines? |
| 18:07.59 | mmlj4 | or move to europe |
| 18:08.20 | mafrac | Any spanish speaker for a private question? |
| 18:08.21 | stanmancan | well... where do you live? |
| 18:08.25 | stanmancan | T1 is actually pretty slow |
| 18:08.31 | digitalml | usa |
| 18:08.34 | stanmancan | you're paying more for the reliability than anything |
| 18:10.38 | stanmancan | I know when it's carrying voice it can do 24 connections |
| 18:10.45 | stanmancan | but for data it's only 1.544 megabits per second |
| 18:10.59 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 18:11.29 | digitalml | right but all i care about it voice |
| 18:11.47 | Kyosh | digitalml: what exactly are you looking for? |
| 18:11.49 | stanmancan | which really isn't very fast, seeing as Verizons FIOS gives you 50Mb down 20Mb up for $140/m |
| 18:12.13 | digitalml | wow wish i had fios here |
| 18:12.16 | stanmancan | Depends on your voip provider though, I would assume if you're using voip that it's going over as 'data' |
| 18:12.22 | digitalml | have cos and i get 50down 5up for 179 |
| 18:12.33 | digitalml | cos = cox |
| 18:12.39 | Kyosh | cox is a joke |
| 18:12.48 | digitalml | yah well all thats avaiable where i live |
| 18:12.48 | Kyosh | any cable company offering above 40mbps is a joke |
| 18:12.57 | Kyosh | is this for home or business? |
| 18:13.03 | digitalml | small business |
| 18:13.07 | digitalml | Kyosh i was saying |
| 18:13.08 | digitalml | right now |
| 18:13.09 | Kyosh | how many users? |
| 18:13.12 | digitalml | i use a sip provider |
| 18:13.16 | Kyosh | right |
| 18:13.21 | stanmancan | i think when they refer to T1's "24 channels of voice" they're talking about a traditional provider, not voip |
| 18:13.21 | digitalml | and get about 5 to 10 concurrent connections at once |
| 18:13.23 | digitalml | which is fine |
| 18:13.28 | digitalml | but if i ever need anything more |
| 18:13.28 | stanmancan | I could be wrong though! |
| 18:13.32 | digitalml | i would have to get a card |
| 18:13.34 | digitalml | with a t1 line |
| 18:13.38 | digitalml | for 24+ connections |
| 18:13.42 | Kyosh | digital, how many users? |
| 18:13.58 | Kyosh | seriously lets get down to the meat of it |
| 18:14.20 | Kyosh | how many users do you need to service?? |
| 18:14.29 | Kyosh | how many concurrent calls would you like to have? |
| 18:14.39 | digitalml | 3 currently which uses up the 5 sip connections i have |
| 18:14.48 | digitalml | i can up that to 10 sip |
| 18:15.02 | digitalml | which might support 6-7 users |
| 18:15.19 | digitalml | but past that i wont be able to use my sip provider |
| 18:15.19 | Kyosh | 50mbps via cable should be sufficient for a few hundred users, but of course its cable |
| 18:15.34 | digitalml | thats down |
| 18:15.36 | digitalml | not up |
| 18:15.39 | digitalml | only 5up |
| 18:15.41 | Kyosh | figure this much, 64kbps per call |
| 18:15.43 | Kyosh | fine |
| 18:15.55 | Kyosh | 78 calls on 5mbps |
| 18:16.03 | Kyosh | now do you trust cox to deliver that |
| 18:16.05 | stanmancan | most cable providers will have a business package that offers an asyncronous connection |
| 18:16.09 | Kyosh | they offer no CoS |
| 18:16.11 | digitalml | my sip porvided doesnt support more than 10 |
| 18:16.21 | Kyosh | ahh |
| 18:16.25 | Kyosh | who is the provider? |
| 18:16.30 | digitalml | voip.ms |
| 18:16.57 | stanmancan | uh, yea they do ? |
| 18:17.19 | stanmancan | Channels: |
| 18:17.19 | stanmancan | |
| 18:17.19 | stanmancan | Unlimited |
| 18:18.29 | digitalml | stanmancan: they say that but you have to 'ask' to do more than 5 simultaneous |
| 18:18.33 | digitalml | and give reasons |
| 18:19.00 | stanmancan | I've never heard of anybody having issues |
| 18:19.22 | stanmancan | even if you did "I need 24 channels to run my business" |
| 18:19.49 | digitalml | hmm ill have to try that |
| 18:20.15 | *** join/#asterisk Alagar (~Administr@122.164.36.246) |
| 18:22.06 | Kyosh | voip.ms states they can support more than 10 concurrent tho |
| 18:22.16 | Kyosh | strange |
| 18:23.10 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:25.00 | stanmancan | Soooo.... Any good tutorials on installing from source?" |
| 18:25.11 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 18:28.20 | drmessano | Some providers have a soft limit because some customers are so fucking dumb, they have their boxes get exploited, end up with 200 concurrent calls from china, and wonder why they have a $10000 bill |
| 18:29.16 | drmessano | Sorta like making your kid prove he's not dumb enough to drink the antifreeze before letting him play in the garage, even though you told him he could play anywhere he wants |
| 18:29.28 | ChannelZ | stanmancan: re: ./configure; make; make install |
| 18:30.35 | digitalml | drmessano: although with voip.ms you pay first, so a $1000 bill really couldnt happen |
| 18:32.11 | drmessano | digitalml: Same difference. If you charge up your account and let yourself get pwn3d, they still have to deal with you |
| 18:32.24 | drmessano | It's a safety net |
| 18:32.37 | digitalml | true |
| 18:33.39 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 18:34.05 | drmessano | I know several providers I used had "unlimited" termination, but all had some cap that required you to call them and prove your worth before they adjusted it |
| 18:34.17 | drmessano | Q. What is your name? |
| 18:34.19 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:34.20 | drmessano | Q. What is your quest? |
| 18:34.30 | drmessano | Q. What is your default context? |
| 18:34.32 | digitalml | yup same as voip.ms |
| 18:34.34 | drmessano | "Uh, I don't know" |
| 18:34.37 | drmessano | AHHHHHHHHHHHHHHHHHHHHHH! |
| 18:35.59 | russellb | drmessano: installing asterisk makes me like vonage right? |
| 18:36.12 | xheliox | lol |
| 18:36.12 | Kyosh | hehe |
| 18:36.16 | digitalml | haha |
| 18:36.29 | xheliox | only if you hum that silly jingle. |
| 18:36.45 | xheliox | doot doo doot doot dooo |
| 18:36.45 | digitalml | woo woo, woo woo woo |
| 18:36.47 | Kyosh | i have a customer who was compromised. dumbass office manager decided to create a new extension and use a 3 digit numeric password. |
| 18:36.51 | drmessano | russellb: app_vonage is in 1.8. Please see release notes |
| 18:36.59 | russellb | \o/ |
| 18:37.05 | Kyosh | they now have a $3500 bill from verizon |
| 18:37.16 | xheliox | app_verizon_lawsuit.so to follow. |
| 18:37.35 | russellb | Kyosh: that happens far too often :-( |
| 18:37.38 | Kyosh | more like app_verizon_bad_billing.so |
| 18:37.55 | Kyosh | russellb: yup i've seen it before, but just not for so much |
| 18:38.06 | Kyosh | and its only 765 minutes too |
| 18:38.15 | Kyosh | most calls were to africa strangely |
| 18:38.44 | mmlj4 | anyone use the official asterisk yum repos? |
| 18:39.30 | russellb | yes, people use it :-) |
| 18:39.52 | mmlj4 | do you use it? |
| 18:39.56 | russellb | I do not. |
| 18:40.00 | russellb | I don't actually use asterisk. |
| 18:40.01 | mmlj4 | can you tell me about your experiences, then? |
| 18:40.23 | russellb | I just know that the repo gets a lot of traffic, heh |
| 18:40.26 | russellb | and it's kept very well up to date |
| 18:40.34 | Kyosh | mmlj4: huh? |
| 18:41.07 | mmlj4 | Kyosh: I'm asking if anyone here uses the repos, and mind telling me what their experiences are with it |
| 18:41.15 | Kyosh | id love to know who these people are doing all this vishing crap |
| 18:41.23 | Kyosh | oooo |
| 18:41.26 | Kyosh | sorry |
| 18:42.08 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 18:42.19 | *** part/#asterisk hacim (~micah@debian/developer/micah) |
| 18:42.46 | *** join/#asterisk mva (mva@desktop.mva.name) |
| 18:42.49 | mva | hi there! |
| 18:42.58 | russellb | mmlj4: well, the repo is maintained by a developer at Digium that I trust. He developed all of the packages and keeps them all up to date. You should have no worries using the repo. |
| 18:43.22 | russellb | mva: hi2u2 |
| 18:43.48 | mva | i'm trying to build * from trunk, but it says, that it cannon find "../defaults.h". Which package i should install for it? |
| 18:44.00 | russellb | o.O |
| 18:44.15 | xheliox | O.o |
| 18:44.34 | mva | uhm... |
| 18:44.49 | mva | possible, make -j5 bug? |
| 18:44.51 | mva | ;) |
| 18:46.13 | russellb | yes |
| 18:46.24 | russellb | make -jN works ... sort of, but not fully |
| 18:46.31 | mva | X_x |
| 18:46.32 | russellb | sometimes you have to run it a few times to get the build to finish |
| 18:47.54 | mva | crappy :( Actualy, i'm trying to make gentoo ebuild for building * from trunk, and get this thing. As i see, it is only way to "make -j1" here... |
| 18:48.00 | mva | thx for advice ;) |
| 18:48.04 | russellb | np |
| 18:50.43 | drmessano | Yeah, or reboot exactly 3 times |
| 18:52.13 | xheliox | what if I goofed and rebooted 4 times? |
| 18:52.40 | drmessano | Thou shall not countest to 4 |
| 18:53.01 | drmessano | Thou shall not countest to 2 unless thou are proceedeth to 3 |
| 18:53.06 | drmessano | 5 is right out |
| 18:55.14 | mmlj4 | well, it seems to install and start, anyhow |
| 18:55.39 | mmlj4 | debian ships *, but that likes to core dump |
| 18:58.02 | stanmancan | so do i JUST need to download the base asterisk files to make it? |
| 18:58.29 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 19:06.35 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 19:08.24 | *** join/#asterisk nsgn (~nsgn@rrcs-24-227-246-117.sw.biz.rr.com) |
| 19:12.01 | nsgn | hello. what would be the most logical method of causing a number being dialed out to be rewritten entirely to another number. say having a customer of mine dialing my 800 number be seamlessly rewritten to actually call my local number? |
| 19:16.53 | nsgn | can this be done with outgoing dial rules? |
| 19:22.36 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 19:22.41 | nsgn | man, i cant seem to google this either. i need to basically need calls to a certain number outbound to be directed to a different number entirely than is dialed before it leaves my * box |
| 19:25.44 | *** join/#asterisk Mhaddog (~Mhaddog@nsc69.38.73-10.newsouth.net) |
| 19:25.44 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 19:26.23 | xheliox | nsgn: It's just be a basic dial rule in whatever context your phone is in. exten => 4075551212,1,Dial(Dahdi/g1/4075551313) |
| 19:26.41 | xheliox | if you dialed 4075551212, it would dial 1313 instead. |
| 19:27.10 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 19:28.13 | nsgn | and..is voip.ms' website down? someone mind checking for me? i cant get to it which is pretty rare for them |
| 19:28.49 | Kyosh | its up |
| 19:28.52 | Kyosh | check your dns |
| 19:29.50 | nsgn | darn time warner |
| 19:32.11 | stanmancan | So I just made and installed asterisk, how do i install the gui now? |
| 19:33.09 | *** join/#asterisk Beltechs (~Beltechs@netblock-68-183-48-2.dslextreme.com) |
| 19:33.32 | xheliox | pain. the awful terrible overwhelming paaaaaaaaaaiiinn.... |
| 19:33.51 | Kyosh | stan, gui? |
| 19:34.13 | stanmancan | Kyosh: I thought there was a GUI available now, maybe not |
| 19:34.14 | Kyosh | looks around in confusion |
| 19:34.20 | Kyosh | umm |
| 19:34.21 | Kyosh | no? |
| 19:34.33 | Kyosh | you want a gui for asterisk, use asterisknow |
| 19:34.41 | Kyosh | install freepbx as an overlay |
| 19:34.43 | Beltechs | Im using asterisk 1.4 and trying to use/setup AJAM. I cant find http.conf in etc/asterisk do I have to create the file? |
| 19:34.45 | stanmancan | yea but i dont' want to dedicate a whole box to it |
| 19:34.46 | xheliox | There's several web based interfaces for Asterisk, but.. I don't think there are too many people in this channel who will assist you with it. |
| 19:34.48 | Kyosh | but asterisk has no gui |
| 19:36.36 | nsgn | asterisknow's freepbx interface is decent for small purpose stuff. good for learning, occasionally convenient for quick changes, but eventually you'll wanna work up to just doing it with the configs. i started in freepbx. it's a good way to start if you need to do things quick, but i'm pretty much done installing it on boxes i do now |
| 19:36.54 | stanmancan | Any good books you guys would recommend? |
| 19:37.12 | xheliox | no actually, freepbx is an excellent way not to learn anything. |
| 19:38.08 | russellb | ~thebook |
| 19:38.09 | infobot | it has been said that thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 19:38.34 | nsgn | free pdf is pretty nice |
| 19:39.06 | xheliox | russellb: Read? C'mon, that's so 20th century. |
| 19:39.16 | xheliox | russellb: Is there some magical software that can do it all for me? |
| 19:39.30 | xheliox | Fairy dust perhaps? |
| 19:42.29 | nsgn | well..i just changed my dns and voip.ms' website still wont call up..from here or on another internet connection i'm remotely into. what the heck? |
| 19:42.40 | nsgn | i'd say they went down but someone in here said they could get to them |
| 19:42.54 | stanmancan | works fine for me |
| 19:42.58 | nsgn | annnd nevermind |
| 19:43.04 | stanmancan | http://screencast.com/t/NDI5MDdh |
| 19:43.07 | nsgn | the very second i said that it starts working when it wouldnt for the past half hour |
| 19:43.40 | nsgn | so what are yall's thoughts on the whole iNum thing? |
| 19:46.47 | *** join/#asterisk guilhermebr (~Guilherme@201-14-75-219.gnace702.dsl.brasiltelecom.net.br) |
| 19:50.55 | nsgn | are there any providers that allow free inum sip calls without purchasing other services? |
| 19:54.24 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 19:57.11 | nsgn | i'm having a hard time finding one |
| 19:59.43 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
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| 20:24.05 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:25.17 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 20:25.21 | *** join/#asterisk talntid (talntid@173.160.174.126) |
| 20:35.47 | *** join/#asterisk albertasterisk (c8740411@gateway/web/freenode/ip.200.116.4.17) |
| 20:37.05 | albertasterisk | anybody help me with a polycom ip phone? |
| 20:37.08 | albertasterisk | please |
| 20:38.42 | talntid | What's going on? :O) |
| 20:38.52 | [TK]D-Fender | albertasterisk: Ask a specific question and you might get a specific answer... |
| 20:38.53 | albertasterisk | i forgot the admin password |
| 20:39.04 | [TK]D-Fender | albertasterisk: 456 <- |
| 20:39.08 | talntid | Default username is "Polycom" - password is "456" |
| 20:39.14 | albertasterisk | no it does not |
| 20:39.22 | nsgn | there's a hard reset for it if you specified one other than default and forgot it |
| 20:39.30 | albertasterisk | no other person has changed this pass |
| 20:39.30 | nsgn | ensure the P in polycom is upper case |
| 20:39.33 | albertasterisk | and i dont know |
| 20:39.36 | albertasterisk | it |
| 20:39.47 | Goshen | albertasterisk, I had to do that today, here is the link http://forum.voxilla.com/polycom-voip-support-forum/master-reset-polycom-ip-430-a-28712.html |
| 20:39.52 | nsgn | it is Polycom with uppercase P and 456. if it is not you need to hard reset the phone |
| 20:40.12 | nsgn | otherwise your browser may be doing something weird or (highly unlikely) the phone's software is damaged |
| 20:40.19 | Goshen | Now I jsut need to figure out how to set up this Polycom 330 to connect to my Asterisk box I just installed |
| 20:40.36 | nsgn | Goshen, that should be a 10 second deal |
| 20:40.39 | albertasterisk | what is the hard reser please |
| 20:40.44 | albertasterisk | *reset |
| 20:40.46 | nsgn | albertasterisk, varies by model. google |
| 20:40.49 | Goshen | albertasterisk, I gave you the link |
| 20:40.55 | albertasterisk | soundpoint 500 |
| 20:41.07 | nsgn | oh well, i'm starving. later |
| 20:41.10 | albertasterisk | ok i will chek the link thanks goshen |
| 20:41.29 | Goshen | nsgn, have a cookbook? or link? someone told me to read the manual this morning, and I have been its great, buts it is 421 pages! |
| 20:42.01 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 20:42.08 | [TK]D-Fender | albertasterisk: the default is Polycom:456 |
| 20:42.39 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 20:42.39 | [TK]D-Fender | albertasterisk: For the WEB interface that is. for phone functions, its just "456" |
| 20:43.30 | albertasterisk | i try with 456 and it does not work |
| 20:43.39 | [TK]D-Fender | albertasterisk: Try WHERE? |
| 20:43.57 | Goshen | I have a set to factory defaults Polycom 330, and fesh installed and updated Asterisknow box(centos) now how do I get the polycom to connect, do I need to set up an ftp server and put the config files on it? |
| 20:44.16 | [TK]D-Fender | Goshen: No, but it's highly recommended |
| 20:44.31 | Goshen | have a link to a sample config? |
| 20:44.50 | [TK]D-Fender | People configuring Polycom phones via the web interface or directly on the phone should be dragged out and shot. Survivors should be shot AGAIN. |
| 20:45.16 | talntid | Goshen, you put the files in an accessable FTP (or http) server, and the phone connects and asks for [macaddress].cfg |
| 20:45.19 | [TK]D-Fender | Goshen: Samples are in the firmware ZIP |
| 20:45.39 | talntid | Sample files: http://www.voip-info.org/wiki/view/Polycom+Phones#SIP32andBootROM42 |
| 20:45.43 | albertasterisk | [TK]D-Fender i press 4 6 8 * and next: 456, |
| 20:46.29 | albertasterisk | also i press 4 6 8* and next: MAc Address |
| 20:47.00 | albertasterisk | and it does not work |
| 20:47.35 | talntid | albertasterisk, we have all given you plenty of information to solve your issue |
| 20:49.47 | albertasterisk | I've tried it all worse I have not succeeded in restoring Administrator Passwords |
| 20:53.54 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 20:56.20 | *** join/#asterisk philipp64|laptop (~chatzilla@mail.redfish-solutions.com) |
| 20:58.42 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 21:17.19 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 21:20.30 | *** join/#asterisk albertasterisk (c8740411@gateway/web/freenode/ip.200.116.4.17) |
| 21:23.23 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 21:23.59 | albertasterisk | please someone help em, I'm in Colombia needs your help |
| 21:25.06 | pabelanger-lap | Bogata is a nice place |
| 21:25.06 | albertasterisk | I could not know how to recover an administrator password Polycon SoundPoint 500 |
| 21:25.14 | albertasterisk | jejeje xd |
| 21:25.18 | albertasterisk | cartagena is better |
| 21:25.39 | albertasterisk | anyway |
| 21:25.55 | albertasterisk | pabelanger dou you know about ip phones polycom? |
| 21:26.09 | pabelanger-lap | Seriously, if you are close to Andres Carnes in Bogota, go! Awesome place |
| 21:26.15 | pabelanger-lap | 456? |
| 21:26.26 | albertasterisk | polycom soundpoint 500 |
| 21:26.41 | pabelanger-lap | No, that is the default password '456' |
| 21:26.47 | pabelanger-lap | if I remember right |
| 21:27.04 | albertasterisk | the default password is 456 but someone changed ah need to delete or change |
| 21:27.39 | pabelanger-lap | factory reset then |
| 21:28.01 | TJNII | Oh if only someone had posted a forum link to instructions..... |
| 21:28.05 | albertasterisk | please tell me how to do a reset factoyr please |
| 21:28.28 | albertasterisk | not what you say in that link does not work |
| 21:28.29 | TJNII | Someone named Goshen .... When you asked an hour ago..... |
| 21:28.52 | albertasterisk | ahhh |
| 21:28.52 | albertasterisk | :( |
| 21:29.07 | TJNII | So what does it do? You're just crying "It doesn't work!" We're not psychic, you need to give details. |
| 21:29.09 | TJNII | Don't pm me |
| 21:29.15 | albertasterisk | ok |
| 21:29.30 | albertasterisk | ok wait |
| 21:30.25 | xheliox | sprinkles magic fairy dust on albertasterisk's phone |
| 21:30.54 | mmlj4 | you know someone pointed you to a webpage showing you how to reset the phone |
| 21:31.34 | mmlj4 | maybe you should as your system administrator to reset the phone for you |
| 21:31.58 | albertasterisk | he intentado presionando 4 6 8 * y me pidio la contraseña del administrador, y como no la tengo ingrese la direccion MAC, luego se reinicio, y trato deentrar al >SETUP y em pide una contraseña y le doy 456 que es la que tare por defecto y no funciona |
| 21:32.05 | mmlj4 | ugh, why am I so grumpy today? |
| 21:32.16 | albertasterisk | sorry |
| 21:32.17 | albertasterisk | I tried pressing 4 6 8 * and I asked for an administrator password, and as I have not enter the MAC address, then reboot, and try to deentrar> SETUP em for a password and I 456 that is the task default and does not work |
| 21:32.23 | xheliox | mmlj4: This channel can have that affect on you. |
| 21:32.29 | albertasterisk | <PROTECTED> |
| 21:33.03 | digitalml | is it possible to have a dial plan connect a caller on the line that originated from an asterisk outbound call connect them to a differnt external phone number and not an internal extension? |
| 21:33.18 | TJNII | Based on the forum post you need to use 468* to get to the password prompt, then enter the mac address with lowercase characters as the password. Does that do anything? |
| 21:34.14 | mmlj4 | ok, I'll be nice and hopefully contructive: it's a sad fact that any newbie can post on a forum, which means that many times forums are the worst places to look for help |
| 21:34.46 | mmlj4 | seek out experts and clued users instead of newbies |
| 21:35.12 | albertasterisk | ok TJNII wait givem a second please |
| 21:35.25 | TJNII | hahaha. Yea, when I post in a forum it is usually because I've exhausted all other option that I know of. I love it when some newbie replies with the first google result. |
| 21:36.12 | albertasterisk | then you can help me or not? |
| 21:36.32 | mmlj4 | dude, we've tried helping you for over an hour |
| 21:36.52 | albertasterisk | ok |
| 21:37.03 | TJNII | I probably can't help you. However, if you TRY THE SUGGESTIONS MADE and REPLY with WHAT HAPPENS people may be able to actually diagnose your problem. |
| 21:37.08 | mmlj4 | if your phone is not broken, then the passwords easily reset |
| 21:37.27 | *** join/#asterisk philipp64|laptop (~chatzilla@mail.redfish-solutions.com) |
| 21:37.40 | albertasterisk | sorry, I think the translation from English to Spanish is highly ambiguous as there is communication error, anyway much help pro sgracias |
| 21:38.26 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 21:41.14 | albertasterisk | I really appreciate your help very much pro if at any time there was a communication misunderstanding, I think it is because wing translation, greetings from Colombia, who have a great day! Proponer una traducción mejor Gracias por proponer una traducción al Traductor de Google. Sugiere una traducción mejor: Idiomas disponibles para traducción: afrikaans albanés alemán árabe armenio azerbaijani bielorruso búlgaro c |
| 21:41.36 | albertasterisk | jeje xd |
| 21:42.19 | albertasterisk | byebye |
| 21:42.28 | digitalml | wtf was that |
| 21:45.08 | TJNII | Another newbie in over his head + a language barrier. |
| 21:47.08 | *** join/#asterisk [netman] (~netman@83.54.34.28) |
| 21:49.23 | Goshen | Should my polycom 330 be a friend or peer? |
| 22:03.36 | p3nguin_ | peer |
| 22:03.58 | p3nguin_ | Phones are almost always fine as peers. |
| 22:04.32 | xheliox | But it's always nice to have more friends. |
| 22:07.30 | *** join/#asterisk devdvd (~twister19@70-14-57-205.pools.spcsdns.net) |
| 22:09.35 | devdvd | hey all, using asterisk 1.6.2.10 and have my queue members in a mysql database. the issue is im using a linear strategy but when asterisk pulls the members from the table, it orders by the interface field ASC, i actually want it to order by the uniqueid field ASC |
| 22:09.57 | devdvd | is there somewhere in the configs that i can cahnge this that im just not seeing or will i need to modify the source |
| 22:16.52 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
| 22:31.16 | Alton35 | I can help people who speak spanish if they'll message me. |
| 22:31.33 | Alton35 | That is, if I can help them with their particular problem. I guess at the very least I could translate. |
| 22:32.10 | *** join/#asterisk devdvd (~twister19@173-31-175-216.client.mchsi.com) |
| 22:37.10 | traxx | hi. does anybody know of any CCBS functionality that works with asterisk ? |
| 22:48.36 | russellb | traxx: Yes. |
| 22:48.40 | russellb | It is supported in Asterisk 1.8. |
| 22:48.40 | digitalml | is it possible to have a dial plan connect a caller on the line that originated from an asterisk outbound call connect them to a differnt external phone number and not an internal extension? |
| 22:48.49 | russellb | traxx: testing would be great :-) |
| 22:50.03 | russellb | digitalml: yes |
| 22:50.12 | russellb | it's just as easy as connecting them to an internal extension |
| 22:50.28 | russellb | asterisk knows nothing about "internal" vs. "external" |
| 22:51.43 | digitalml | hmm ok |
| 22:51.51 | digitalml | but will the call still be through my sip trunk |
| 22:51.54 | digitalml | using my sip mins? |
| 22:52.24 | russellb | yes |
| 22:52.35 | devdvd | yes, at that point you would be using double your minutes cuz you would have 2 simultanious calls at once |
| 22:52.54 | devdvd | another caveat is you need to make sure your provider will even allow you to have 2 concurrent calls on the same trunk |
| 22:53.00 | devdvd | else you will have to get another trunk |
| 22:53.02 | digitalml | they do |
| 22:53.24 | digitalml | i already have concurrent outbound calls on the same trunk |
| 22:53.30 | digitalml | but i dont currently forward anyone |
| 22:53.57 | russellb | the provider might let you redirect the call instead of having it go through your box |
| 22:54.06 | russellb | try using the Transfer() application in the dialplan and see what happens |
| 22:55.22 | digitalml | cool thanks, ill look into it |
| 22:56.15 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
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| 23:05.35 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 23:14.11 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 23:14.14 | drfreeze | Hi |
| 23:14.36 | drfreeze | Anyone know how to get callerid name and num to display on a polycom phone? |
| 23:15.11 | drfreeze | I see lots of posts on how people only got name to display because they were running old firmware |
| 23:15.43 | drfreeze | I'm running 3.2.3 bootrom on a 550 |
| 23:16.35 | drfreeze | and sip.ld ver 2.1.0.2708 |
| 23:29.21 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 23:29.29 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 23:30.13 | *** join/#asterisk philipp64|laptop (~chatzilla@mail.redfish-solutions.com) |
| 23:30.56 | *** join/#asterisk lancey (~root@213.137.58.10) |
| 23:32.07 | lancey | hi all. i'm having problems with res_config_mysql on * 1.6.2.9. the bindings seem to be configured okay, the module does connect to mysql, but when i do a simple 'realtime load ... |
| 23:32.31 | lancey | i get no rows matching your criteria... though there are rows matching |
| 23:32.52 | lancey | and, as a result, nothing using realtime could work. any hints to what to look at? |
| 23:33.14 | digitalml | so right now i have my .call file calling a cetain context in my dial plan. but i now need to pass some arguments to that context. how can i do this? do I have to use AGI? |
| 23:34.30 | lancey | digitalml: use SetVar in the callfile |
| 23:34.52 | digitalml | ok ill look into that, thanks |
| 23:35.01 | lancey | digitalml: 'Setvar: variable=value' |
| 23:35.58 | digitalml | and inside the context how do i read that? |
| 23:36.18 | lancey | digitalml: ${variable} |
| 23:36.54 | digitalml | i can do $() outside an AGI file? eg: directly inside the extensions.conf? |
| 23:37.12 | lancey | digitalml: sure. what asterisk version are you running? |
| 23:37.34 | lancey | digitalml: its ${} - the brackets are curly |
| 23:37.44 | digitalml | 1.6 |
| 23:37.59 | lancey | digitalml: yup, you are perfectly fine with ${variable} |
| 23:38.19 | digitalml | sweet, ill give it a go |
| 23:38.20 | digitalml | thanks |
| 23:38.34 | lancey | ur welcome |
| 23:52.14 | digitalml | anyone have a good sugestion for a decent qos router < 300 |
| 23:53.16 | lancey | digitalml: depends on what you mean by decent, and what bandwidth it's gonna push, number of conns, etc. |
| 23:53.52 | lancey | depends on the currency of the 300 too ;) |
| 23:54.00 | digitalml | 300 usd |
| 23:55.11 | lancey | digitalml: we use lots of cisco 1711/12, but that's mainly because of their secure vpn features. they do qos fine though |
| 23:55.26 | digitalml | yah dont really need vpn |
| 23:55.29 | lancey | and they fit in the bill (refurbished) |
| 23:56.42 | lancey | hmmm. i think something is fucked up with my res_config_mysql generally. it does not give me help for 'realtime mysql cache' |
| 23:57.57 | lancey | digitalml: a mikrotik will also do the work. |
| 23:58.21 | digitalml | someoen suggested this: http://www.draytek.com/user/PdInfoDetail.php?Id=114#PdInfo |
| 23:58.26 | digitalml | ever hear of it? |