00:00.29 | MatBoy | are there companies providing area based telphony using Asterisk ? |
00:00.46 | Strom_M | "area based telephony:? |
00:00.48 | BBHoss | area based? |
00:00.56 | Corydon76-dig | You mean CLECs? |
00:00.56 | [TK]D-Fender | .... nvm :) |
00:01.01 | Strom_M | catsex? |
00:01.13 | [TK]D-Fender | asciipr0n |
00:01.51 | MatBoy | Strom_M, no that customers will still have their normal telephone number that they used to have at home |
00:02.00 | Corydon76-dig | I know at least one CLEC is using Asterisk to route their customers' long distance calls... considering I wrote their routing logic |
00:02.10 | BBHoss | you mean porting? |
00:02.21 | Strom_M | MatBoy: so what you're asking, really, is whether there are ITSPs that support number portability |
00:02.30 | Strom_M | and yes, there are plenty |
00:03.02 | Corydon76-dig | So yes, there's a traditional CLEC using Asterisk in their POTS service plan |
00:03.10 | MatBoy | Strom_M, no the real question is... how stable it is comparing to PortaOne for an example. I think it's very stable, but also good on load ? |
00:03.13 | Corydon76-dig | at least one |
00:03.36 | Corydon76-dig | PortaOne is a billing platform |
00:04.14 | Corydon76-dig | AFAIK, they don't even have an interface to the PSTN |
00:04.37 | MatBoy | Corydon76-dig, no, it's a full platform, indeed with billing, that supports Cisco |
00:04.51 | MatBoy | Corydon76-dig, Portaone has sip-servers |
00:05.08 | Corydon76-dig | MatBoy: in other words, they're using a 3rd party vendor to support PSTN connections |
00:05.08 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
00:05.29 | Corydon76-dig | MatBoy: Asterisk has native interfaces to both sides |
00:05.30 | MatBoy | Corydon76-dig, true, and I was looking if Asterisk can do the same |
00:06.04 | Corydon76-dig | MatBoy: You're comparing apples and oranges. Asterisk is a telephony platform. PortaOne is a billing platform. |
00:06.16 | Corydon76-dig | You certainly can interconnect the two |
00:07.04 | MatBoy | yeah, OK, let's say can I build with Asterisk, also on billing, what I can do with Portaone ? |
00:07.21 | Corydon76-dig | I dunno, ask their sales dept |
00:07.43 | Corydon76-dig | I'm sure they can set you up with some very expensive equipment |
00:07.47 | BBHoss | MatBoy: you ought to rephrase that question |
00:08.27 | Corydon76-dig | You can build a billing platform on top of Asterisk, but Asterisk does not have a native billing app |
00:08.35 | MatBoy | Corydon76-dig, PortaOne costs about 70K without HW |
00:08.41 | MatBoy | full SMP license |
00:08.53 | Corydon76-dig | Asterisk is free without HW |
00:09.08 | MatBoy | Corydon76-dig, indeed, but you need billing for calling :) |
00:09.09 | BBHoss | and the hardware is cheap |
00:09.15 | Corydon76-dig | $70k is more than I make in a year |
00:09.16 | MatBoy | yeah indeed |
00:09.27 | BBHoss | you can use a2billing, although ive not used it personally |
00:09.59 | MatBoy | Nice to investigate this, I have to check in how far I can clutser Asterisk :) |
00:10.03 | MatBoy | would be awsome to do |
00:10.13 | BBHoss | and you have CDR, so really all it takes is a program that can parse the files |
00:10.19 | BBHoss | or pull from mysql |
00:10.37 | *** join/#asterisk PaulAviles (n=salinas9@dsl-7-36.cofs.net) |
00:10.52 | PaulAviles | any cisco 79xx users? |
00:10.53 | MatBoy | BBHoss, indeed, that's all it takes |
00:11.07 | BBHoss | looks like a2billing has grown a lot |
00:11.12 | BBHoss | http://trac.asterisk2billing.org/cgi-bin/trac.cgi |
00:12.36 | MatBoy | BBHoss, I was just looking at it, looks nice |
00:13.29 | MatBoy | BBHoss, I think you can start a good business with it |
00:13.34 | BBHoss | yeah |
00:13.56 | BBHoss | not sure how clustering works exactly, havent ever needed that big of a system before |
00:14.41 | MatBoy | BBHoss, ok, but I know platforms that are doing at least 1 million minutes per month |
00:14.41 | MatBoy | so |
00:14.51 | MatBoy | clustering might be nice I think |
00:15.05 | BBHoss | why do you want to cluster, for failover or for performance |
00:15.39 | MatBoy | BBHoss, both actually, but if a system can hold it... failover |
00:16.32 | *** join/#asterisk anthm (n=anthm@mbe0736d0.tmodns.net) |
00:16.32 | *** mode/#asterisk [+o anthm] by ChanServ |
00:16.56 | MatBoy | BBHoss, keep quiet, an op arrived :P |
00:17.25 | BBHoss | looks like there is something out there called biocluster, but then again i've never used it |
00:17.46 | MatBoy | BBHoss, but your other unused idea was also quite nice, so let me look :) |
00:17.57 | BBHoss | heh |
00:19.22 | MatBoy | BBHoss, and again... seems to be nice |
00:23.35 | BBHoss | MatBoy: a2billing has a demo you can try out |
00:24.02 | MatBoy | BBHoss, yep I was already in, looks perfect |
00:29.55 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
00:30.17 | PaulAviles | any cisco 79xx users? |
00:32.25 | *** join/#asterisk nny (n=Scott@64.20.149.250.dyn-e-pool22.pool.hargray.net) |
00:32.28 | *** join/#asterisk captiancrash (n=captianc@c-68-53-165-155.hsd1.ky.comcast.net) |
00:32.59 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
00:33.12 | nny | so.. what if someone wanted to tear trixbox out of a system, and use the real mc coy. Is there a howto for excorsing the beast? |
00:33.22 | [TK]D-Fender | nny, format. |
00:33.23 | nny | exorcising* |
00:33.27 | nny | [TK]D-Fender: lol yeah |
00:33.34 | [TK]D-Fender | nny, grab some sane distro and wipe it clean |
00:33.35 | nny | [TK]D-Fender: we are/were except one is in Panama |
00:33.43 | nny | [TK]D-Fender: yeah thats the plan |
00:34.01 | nny | [TK]D-Fender: just spent the better part of the day discovering who did what wrong |
00:34.06 | nny | long story |
00:34.09 | [TK]D-Fender | nny, Or you could just stop freePBX from loading. Thats the only really evil part. |
00:34.25 | nny | original plan was to wipe and re setup |
00:34.37 | nny | yeah how doe sfreepbx load? not in /etc/init.d/ |
00:34.42 | nny | i noticed this is centos |
00:35.17 | nny | cause that would save me having to walk customer through installing the initial OS |
00:37.00 | nny | and kill it dead |
00:39.23 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
00:39.55 | nny | http://www.freepbx.org/2006/09/28/un-trixbox-your-trixbox/ |
00:58.48 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
00:58.58 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
00:58.58 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
01:02.19 | *** part/#asterisk nny (n=Scott@64.20.149.250.dyn-e-pool22.pool.hargray.net) |
01:02.20 | BBHoss | nny: i read somewhere i guide to changing an OS wholly over ssh |
01:03.30 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.135) |
01:08.37 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
01:08.53 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
01:09.19 | *** join/#asterisk mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
01:09.19 | *** mode/#asterisk [+o mog] by ChanServ |
01:13.38 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
01:16.57 | *** join/#asterisk cybrside (n=cybrside@ppp-70-253-88-208.dsl.austtx.swbell.net) |
01:17.48 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
01:24.14 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
01:27.10 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
01:29.19 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
01:37.28 | Mackes | Whats that? |
01:39.00 | BBHoss | what? |
01:39.01 | MatBoy | BBHoss, ow ? |
01:39.06 | MatBoy | where di you read it |
01:39.16 | BBHoss | oh |
01:39.21 | BBHoss | lemme find it |
01:39.28 | MatBoy | BBHoss, not a VM ? |
01:39.47 | BBHoss | it was like for dedicated hosting providers that dont let you use ubuntu, only debian and such, hang on |
01:39.50 | BBHoss | but yes, no vm |
01:40.04 | MatBoy | ow kewl |
01:40.06 | MatBoy | hehe |
01:40.24 | MatBoy | but what if they don't provide you ssh ? make a good hack script than :P |
01:41.01 | BBHoss | yeah you could do it through perl, but you really need root access |
01:41.34 | BBHoss | http://www.goudkov.com/public/articles/changing_distro.jsp |
01:42.18 | MatBoy | BBHoss, ah as I expected... you actually "move" it |
01:42.57 | BBHoss | it would have solved his problem though |
01:43.11 | MatBoy | kewl |
01:43.19 | MatBoy | let us try it in a vm :D |
01:43.36 | BBHoss | OR you could install it on your own box or a vm, then make a dd image, then apply that dd image to the server |
01:44.33 | MatBoy | yep indeed |
01:44.38 | MatBoy | ok, now I really need to sleep |
01:44.43 | MatBoy | 3 am here almost |
01:45.46 | MatBoy | sleep well |
01:45.49 | MatBoy | BBHoss, thanks ! |
01:45.52 | MatBoy | for everything |
01:45.59 | MatBoy | bbl |
01:46.00 | BBHoss | np, good luck |
01:46.08 | BBHoss | report back on that clustering stuff |
01:46.19 | *** join/#asterisk ming_zym (n=ming_zym@124.14.234.227) |
01:46.19 | MatBoy | yep will be done next year I think |
01:46.25 | BBHoss | heh |
01:46.30 | MatBoy | maybe next month |
01:46.33 | MatBoy | dunno yet |
01:58.45 | *** join/#asterisk jero (n=jerome@modemcable169.212-70-69.mc.videotron.ca) |
01:59.52 | jero | hi |
02:01.24 | BBHoss | sup |
02:02.01 | jero | i'm looking for a way to trap a phone registration in chan_sip.. To run a program at the time a phone registers |
02:02.05 | ManxPower | If clustering was easy there would be many people doing it. |
02:02.22 | ManxPower | jero: be prepared to do extensive coding in chan_sip.c |
02:02.37 | *** join/#asterisk PepOSX (n=pepOSX@190.72.149.231) |
02:02.37 | jero | ManxPower: extensive ? |
02:02.40 | mosty | is there a way to forward a call between two asterisk servers that will preserve channel variables? |
02:02.51 | ManxPower | jero: extensive = much = many |
02:03.09 | mosty | ie so channel variables on the first server are available on the second server? |
02:03.12 | jero | ManxPower: you think id better parse logs in realtime ? |
02:03.55 | Mackes | I dont think so. |
02:05.07 | jero | i maybe should use SER ? i'm implementing a distributed architecture where a wireless phone can register in multiple offices on the same SSID / using the same credentials .. and have its calls re-routed to the correct site. |
02:05.35 | BBHoss | oh shit! here we go again!!! |
02:06.09 | Mackes | Jero, I might be able to help... what are you using... what is your plan? |
02:06.11 | BBHoss | jero: are you trying to do this for a single project? |
02:07.57 | jero | i'm using nokia cellular+wifi phones.. + anything required. the only missing part is catching register. if the feature is not in chan_sip, we'll either add it, or use SER if more appropriated |
02:08.15 | BBHoss | because with SIP-DECT, you can carry a handset across the world, and have it ring the same extension |
02:08.27 | BBHoss | but since your using cell, that could be a problem |
02:08.29 | Mackes | If you are interested, I would be happy to tell you about our setup |
02:08.49 | jero | BBHoss: its a sip + cell phone. it does both |
02:09.20 | jero | Mackes: of course, any info will be helpful |
02:09.51 | *** join/#asterisk GuyOCanada (i=GuyOCana@75.155.220.205) |
02:09.54 | GuyOCanada | Hello |
02:09.59 | BBHoss | sup |
02:10.01 | Mackes | We have one Asterisk server that handles all calls across 21 locations |
02:10.07 | Mackes | 31 locations |
02:10.17 | [TK]D-Fender | BBHoss, ;) |
02:10.23 | Mackes | They are all tied together with T1's |
02:10.25 | BBHoss | heh |
02:10.31 | jero | oh okay |
02:10.33 | Mackes | with me so far |
02:10.35 | jero | brb |
02:10.46 | GuyOCanada | is there a rpm package for asterisk-1.4.14 |
02:10.54 | Mackes | do you have something similar? |
02:11.00 | BBHoss | GuyOCanada: its best to build from source with * |
02:11.16 | BBHoss | there may be something in yum, but its probably old as dirt |
02:11.28 | Mackes | Jero? |
02:11.33 | Mackes | Oh ok |
02:11.35 | *** join/#asterisk mihinomenest (n=argh@66.255.220.17) |
02:11.55 | BBHoss | GuyOCanada: http://www.voip-info.org/tiki-index.php?page=Asterisk%20RPM may help |
02:11.55 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
02:12.18 | GuyOCanada | BBHoss: I want to install asterisk asterisk-1.4.14 on one of my servers which will only handle voip calls no hardware do i still need to install the packages (libpri zaptel etc.) or would it be ok if i just install asterisk-1.4.14 from source? |
02:12.34 | BBHoss | GuyOCanada: depends on what you want to do with asterisk |
02:12.49 | BBHoss | GuyOCanada: some * features require the zaptel dummy module (ztdummy) |
02:13.18 | GuyOCanada | how do you compile the ztdummy module? |
02:13.20 | BBHoss | MeetMe is the biggest one, i think there are a few more |
02:13.26 | BBHoss | compile zaptel |
02:14.18 | bkw_ | no comment |
02:14.26 | [TK]D-Fender | GuyOCanada, first go download THE BOOK. It will tell you how to do all of this. |
02:14.27 | [TK]D-Fender | ~book |
02:14.28 | jbot | 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 |
02:14.29 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
02:14.38 | bkw_ | I use that book for a monitor stand |
02:14.40 | bkw_ | its all its good for |
02:14.48 | bkw_ | poor rain forest |
02:14.50 | BBHoss | shit, here we go |
02:15.07 | MrTelephone | toiletpaper.pdf |
02:15.12 | bkw_ | MOVING ON |
02:15.14 | bkw_ | NEXT!!! |
02:15.32 | Mackes | YOu dont like the book? |
02:15.38 | [TK]D-Fender | bkw_, No, its much thicker than rev 1..... it is now a suitable tactical defense solution! |
02:15.39 | Mackes | Who doesnt like that book |
02:15.46 | bkw_ | my name is in the book |
02:15.54 | Mackes | its THE book |
02:15.54 | BBHoss | somone ought to go through voip-info.org take all the good info, clean it up, and put it in its own wiki |
02:16.08 | bkw_ | Mackes: I don't.. it gives false hope to some |
02:16.19 | Mackes | how is that? |
02:16.19 | GuyOCanada | ok another question |
02:16.30 | GuyOCanada | anyone running asterisk on a box with plesk control pannel installed? |
02:16.41 | JT | BBHoss: you mean sort of like copyright infringement? |
02:16.50 | *** join/#asterisk Qb3rt (n=eric@modemcable156.182-80-70.mc.videotron.ca) |
02:17.00 | bkw_ | yah copyright infringement? |
02:17.20 | BBHoss | doubt it |
02:17.20 | BBHoss | oh is it copyrighted? |
02:17.23 | JT | .... |
02:17.28 | JT | you've got to be joking |
02:17.36 | JT | it's a PUBLISHED O'REILY BOOK |
02:17.37 | bkw_ | oh lord help us |
02:17.39 | JT | yes it has copyright |
02:17.50 | BBHoss | no im talking about the voip-info wiki, not the book |
02:17.56 | bkw_ | it too is copyrite |
02:17.58 | bkw_ | er right |
02:18.05 | bkw_ | lord help ya |
02:18.17 | BBHoss | many wikis use GNU |
02:18.27 | JT | voip-info, commpartners owns the copyright to everyone's work there, apparently |
02:18.39 | bkw_ | GNU is still a copyright |
02:18.49 | Mackes | Well, I think its a great book. |
02:18.51 | bkw_ | someone needs to get the clue stick out |
02:18.58 | bkw_ | Mackes: I don't |
02:19.00 | Qb3rt | what is the minimum cpu and memory needed if i want to setup one server with 20 lines (3 talkings and the rest wainting with music on hold) All calls recorded... And a PRI on it?? |
02:19.05 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
02:20.00 | Mackes | Is there a better book, I would love to read another? BTW, who has it given false hope, and why? |
02:20.07 | [TK]D-Fender | GuyOCanada, I've never heard of it in my years here, and I can't see any reference to an Asterisk module for it anywhere. |
02:20.36 | [TK]D-Fender | Qb3rt, decent P4 + 1 gig should be fine. |
02:20.41 | Mackes | PLesk is a web management pannel.... kinda like Webmin |
02:20.43 | GuyOCanada | [TK]D-Fender: plesk is a produfct of swsoft www.swsoft.com its a control panel for hosting environments |
02:20.46 | Mackes | Closed Source |
02:20.53 | JT | yes i think we all know what plesk is |
02:20.53 | BBHoss | How does GNU FDL keep you from copying and modifying it, as long as you apply the license? |
02:20.54 | [TK]D-Fender | GuyOCanada, I know, I'm on their site |
02:21.13 | JT | BBHoss: it's still a goddamn type of copyright |
02:21.23 | GuyOCanada | but it forks linux installations |
02:21.31 | Qb3rt | Already got that and its not going well... not enought memory and the cpu is really loaded |
02:21.32 | BBHoss | ok, but you follow the license, you can use it, correct? |
02:22.03 | JT | BBHoss: everything on voip-info is property of commpartners |
02:22.14 | BBHoss | yeah i was speaking in a general sense |
02:22.15 | JT | there is no gnu or anything |
02:22.19 | GuyOCanada | for some reason freepbx or other web gui's do not work on a plesk installed server |
02:22.21 | [TK]D-Fender | Qb3rt, well you haven't mentioned anything like what card, what codecs, etc. |
02:22.51 | [TK]D-Fender | GuyOCanada, Ask THEM. GUI's are not supported here. |
02:23.03 | ManxPower | http://www.voip-info.org/terms_of_service.html |
02:23.07 | Mackes | Check the port number it uses |
02:23.19 | BBHoss | ManxPower: yes i saw that, too bad |
02:23.20 | GuyOCanada | [TK]D-Fender: i am not asking for support of a GUI i just asked if someone was using it |
02:23.21 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
02:23.21 | Mackes | it might be attempting to use a port that is in use |
02:23.32 | Qb3rt | i dont have these details right now... but ill get back here when i have them... thanks for you help :D |
02:23.39 | ManxPower | BBHoss: if you don't like it, don't use it. |
02:23.51 | BBHoss | ManxPower: what are you talking about |
02:23.55 | ManxPower | GuyOCanada: nobody here uses GUIs |
02:23.58 | *** join/#asterisk CrashHD (n=crashhd@67-107-9-130.starstream.net) |
02:24.04 | BBHoss | ManxPower: i never said i didnt like it |
02:24.13 | Mackes | I think Webmin is very helpful |
02:24.16 | [TK]D-Fender | GuyOCanada, Well I've answered that. There has been no mention of Plesk here. |
02:24.25 | JT | helpful if you can't use ssh |
02:24.34 | BBHoss | Mackes: webmin has a tendency to trash installations |
02:24.36 | Mackes | Yep |
02:24.40 | ManxPower | Sorry, I assumed that when you said "ManxPower: yes i saw that, too bad" you were expressing a dislike for the policy. |
02:24.44 | Mackes | Really? How so? |
02:24.52 | GuyOCanada | ManxPower: sorry |
02:25.23 | BBHoss | ManxPower: no, it just means that i can't use it :) |
02:25.23 | Mackes | I guess if you have made custom changes and Webmin rewrites your configs |
02:25.39 | BBHoss | ManxPower: however it is odd that a user-created page somehow becomes their copyright |
02:25.57 | BBHoss | compared to other wikis ive seen |
02:25.59 | Corydon76-dig | I suspect it means something different |
02:26.20 | Corydon76-dig | They don't want you to put up pages that are specific to any one company |
02:26.35 | Corydon76-dig | rather, general information that is useful to everybody |
02:26.43 | BBHoss | Mackes: well if it gets the version wrong and writes a config file wrong, or changes it someway |
02:27.06 | Corydon76-dig | Yes, they have to word it strongly to ensure that they can go after people who abuse the service |
02:27.17 | Mackes | Yeah.. That would be a bumper. |
02:27.34 | BBHoss | Corydon76-dig: so you think they just dont want people profiting off of it? |
02:27.38 | Mackes | However I have installed it on Serveral Distros without an issue. |
02:28.04 | Mackes | Has anyone worked with openVZ |
02:28.54 | Corydon76-dig | BBHoss: I think they just don't want to be an involuntary host to information that promotes a single company |
02:29.08 | Corydon76-dig | BBHoss: that's always a danger when running a Wiki |
02:29.22 | BBHoss | yeah, its probably best for me to email them |
02:29.39 | Mackes | Why the discussion about the Wiki... Do you have an issue with them? |
02:29.51 | JT | as if anyone ever gets a reply from commpartners |
02:29.56 | JT | Mackes: clearly there are issues |
02:30.07 | BBHoss | well, the design is ugly |
02:30.17 | Corydon76-dig | You can very easily set up multiple disparate "sites" on a single Wiki... It's all in how the pages are linked together |
02:30.18 | BBHoss | the pages don't work |
02:30.24 | BBHoss | the examples/info are bad |
02:30.30 | rob0 | And its mother dresses it funny. |
02:30.38 | BBHoss | rob0: indeed |
02:30.43 | JT | the info being bad is not a fault of the sponsor |
02:31.07 | BBHoss | yeah i know, but if im going to clean up all the info, id rather have all the good docs in one place |
02:31.22 | Mackes | Hmmm... Wow, tonight we have knocked the best book on Asterisk and the most detailed Wiki on Asterisk.... |
02:31.34 | Mackes | fantastix |
02:31.42 | JT | Mackes: yes, such is the real world |
02:31.58 | Corydon76-dig | What's wrong with TFOT2? |
02:31.59 | BBHoss | i didnt trash the book, but a lot of the stuff on the wiki sucks |
02:32.01 | JT | Mackes: and it is a VERY WELL KNOWN fact that the wiki has tonnes of incorrect or outdate info on it |
02:32.05 | BBHoss | and its very unorganized |
02:32.28 | JT | unorganised isn't a word |
02:33.55 | MrTelephone | what the hell is TFOT2 |
02:34.00 | BBHoss | the book |
02:34.02 | jero | Mackes: back |
02:34.05 | [TK]D-Fender | JT : Apparently it IS... http://dictionary.reference.com/browse/unorganised |
02:34.06 | JT | the second edition of the book |
02:34.24 | MrTelephone | i want to get back into online gaming |
02:34.31 | Mackes | I am sooo thankful to the community at large. Asterisk and the folks who contribute to Wikis help me do my job, which in turn feeds my family--- and ---- and---- its free. Microsofts, Cisco, Alcatel, Dell, HP...... All of there Tech support Docs sections are difficult. Why knock the one that is a direct result of our community... especially here! |
02:35.04 | JT | [TK]D-Fender: only americans would add such an incorrect word to the dictionary ;) |
02:35.04 | MrTelephone | asterisk is a good project but its becoming a little bloated |
02:35.05 | BBHoss | wanting to do something better isnt knocking it |
02:35.30 | BBHoss | JT: what would u use? |
02:35.42 | JT | Mackes: things will never improve if everyone pretends everything is perfect |
02:35.48 | JT | BBHoss: the word is "disorganised" |
02:36.06 | BBHoss | JT: ok, that word is not used often in america |
02:36.25 | JT | i guess that's because they speak american, and not english |
02:36.34 | BBHoss | JT: yep |
02:36.42 | JT | it's a horrible abomination :P |
02:37.34 | Mackes | But bitching about it doesn't help? Lots of Wikis are available and they are all a mess. Why, because they are community driven. Does anyone have a BETTER site, that is free, and current? |
02:37.53 | BBHoss | Mackes: wikipedia is the way a wiki should be |
02:38.04 | JT | even wikipedia has issues :P |
02:38.11 | BBHoss | noone is perfect |
02:38.22 | MrTelephone | JT is perfect |
02:38.53 | Mackes | man |
02:39.03 | JT | the asterisk book is pretty good |
02:39.14 | Mackes | I think so as well |
02:39.15 | JT | although i don't like how it hypes stuff up |
02:39.29 | JT | especially the whole asterisk being the future thing |
02:39.30 | *** join/#asterisk [N00B] (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
02:39.36 | JT | clearly very biased in that respect ;) |
02:39.42 | Mackes | I have all of the Cisco and Alcatel/Lucent books on Voip |
02:39.55 | Mackes | and they are very difficult to read and ue |
02:39.56 | Mackes | use |
02:40.32 | Mackes | The * book is direct, and a great reference |
02:40.38 | asdx | i agree |
02:40.56 | MrTelephone | all the problems I have in asterisk are all packet loss issues |
02:41.00 | [TK]D-Fender | JT : If they called the "suggestions of physics" nobody would be following them and we'd all be flying around! Its called MARKETING! |
02:41.04 | MrTelephone | and its not even asterisks' fault |
02:41.06 | *** join/#asterisk iamamoron (n=t@202.137.121.84) |
02:41.11 | iamamoron | hi there |
02:41.31 | JT | asterisk is lacking in a lot of areas, but is fine in others |
02:41.34 | iamamoron | i have an existing pabx panasonic kx-td816 |
02:41.39 | JT | it's certainly no be all and and all |
02:41.50 | iamamoron | and there is an existing telephone line connected to it |
02:42.17 | iamamoron | what i want is the asterisk all cater all incoming calls and pass it to the kx-td816 |
02:42.22 | iamamoron | would it be possible? |
02:42.30 | BBHoss | iamamoron: yes |
02:42.32 | iamamoron | by using digium tmd400p |
02:42.43 | BBHoss | iamamoron: just get 1 fxo and 1 fxs |
02:43.04 | BBHoss | iamamoron: but you will be limited in what you can do with asterisk |
02:43.42 | Mackes | is the kx-td816 use digital handsets? |
02:44.01 | Mackes | if so, and FXS/FXO isnt going to work |
02:44.03 | [TK]D-Fender | iamamoron, What exactly do you want * to do for you? |
02:44.22 | Mackes | you might need to create a tie line using a PRI |
02:44.38 | BBHoss | if you just want to play a message then pass the call to the panasonic, then it will do it |
02:44.50 | JT | Mackes: he wants to do pbx intercept, it is irrelevant if the pbx has digital handsets or not |
02:44.58 | ManxPower | Mackes: you are assuming that the Panasonic supports T-1/PRI |
02:45.13 | iamamoron | from pstn asterisk will get all calls |
02:45.18 | iamamoron | and outbound calls |
02:45.26 | Mackes | Yep |
02:45.37 | iamamoron | now an IVR is in asterisk |
02:45.49 | JT | Mackes: there is no reason it can't be done over FXO/FXS ports |
02:45.53 | BBHoss | yeah its possible, but it aint gonna be easy |
02:45.57 | JT | not as nice as digital obviously |
02:46.01 | Mackes | OK |
02:46.03 | Mackes | So |
02:46.17 | iamamoron | say if user at pstn press 1 it will go to extension 222 at panasonicj kx-td816 |
02:46.18 | Mackes | Call comes in, Asterisk Grabs it, says a few things |
02:46.27 | JT | i do pbx intercept for BRI lines |
02:46.29 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
02:46.34 | Mackes | and then passes it to the kx-td816 |
02:46.38 | JT | except i changed the telco lines to PRI after a bit |
02:46.49 | iamamoron | any ideas? |
02:46.57 | Mackes | It will do that if you only have POTS with FXO/FXS |
02:47.06 | Mackes | However, that is all you will be able to do |
02:47.11 | iamamoron | yes i have |
02:47.21 | iamamoron | what should i do? |
02:47.32 | Mackes | Remove the kx-td816 |
02:47.34 | iamamoron | i have a x100p here |
02:47.36 | JT | you can make asterisk give voip calling capabilities to the panasonic |
02:47.39 | BBHoss | can you plug a fax machine into that PBX? |
02:47.43 | JT | or recording capabilities |
02:47.45 | BBHoss | hehehehe |
02:47.46 | Mackes | How many extentions do you need? |
02:48.07 | iamamoron | you said all i need is 1 fxo and 1 fxs |
02:48.14 | iamamoron | i have a x100p here |
02:48.20 | iamamoron | what should i do now |
02:48.21 | Mackes | That will give you 1 channel |
02:48.25 | Mackes | just one |
02:48.25 | JT | the x100p is junk |
02:48.30 | Mackes | one caller at a time |
02:48.30 | JT | and you need an fxs port |
02:48.31 | BBHoss | iamamoron: get a better card first |
02:48.38 | Mackes | is that what you want |
02:48.38 | JT | perhaps get a linksys SPA-3102 |
02:48.48 | iamamoron | for trial only |
02:48.53 | Mackes | I mean, why would you have a kx-td816 |
02:48.55 | iamamoron | i know x100p sucks |
02:49.00 | Mackes | if you only handled one call |
02:49.01 | JT | you still need an fxs port for that trial, iamamoron |
02:49.03 | iamamoron | but i want to test it first before i buy |
02:49.07 | Mackes | ahhh |
02:49.10 | BBHoss | iamamoron: avoid the x100p like the plague |
02:49.18 | Mackes | Are you using POTS on your kx-td816 |
02:49.26 | iamamoron | kx-td816 is obsolete |
02:49.31 | GuyOCanada | linksys SPA-3102 can you use that remotely? |
02:49.33 | MrTelephone | i bought a linksys 8-port sipura |
02:49.34 | iamamoron | Mackes; yes |
02:49.36 | *** join/#asterisk BiG^DoG (n=BiG^DoG@c-67-162-233-20.hsd1.de.comcast.net) |
02:49.38 | iamamoron | POTS |
02:49.45 | JT | GuyOCanada: sure |
02:49.50 | MrTelephone | avoid anything analogue if you can |
02:50.07 | iamamoron | no i cant coz i need to interop |
02:50.14 | Mackes | The kx-td816 has BRI interface |
02:50.16 | iamamoron | Mackes? |
02:50.19 | Mackes | yep |
02:50.26 | BiG^DoG | I'm going crazy... Is there any way to get call waiting to work with my SPA3102 and asterisk? |
02:50.26 | iamamoron | can i simulate using x100p? |
02:50.33 | Mackes | ok |
02:50.40 | JT | BiG^DoG: what do you mean call waiting? |
02:50.41 | BBHoss | iamamoron: you would be better off trashing the kx-td816 and getting a new system with polycom or another brand ip phone |
02:50.44 | GuyOCanada | so I can have spa-3102 at my home and connect it using IAX2 or SIP to my asterisk box at work and answer calls from another sip location or call using my home phone from another sip location? |
02:50.48 | JT | iamamoron: NO, you need an FXO AND and FXS port |
02:50.50 | BBHoss | then get a tdm400p or tdm800p to do POTS |
02:50.52 | Mackes | How many stations does your kx-td816 support |
02:51.08 | iamamoron | JT: yes i have |
02:51.12 | obnauticus | BBHoss, do you know how to setup a Cisco 7960? |
02:51.15 | iamamoron | my x100P has fxo and fxs |
02:51.19 | BiG^DoG | JT: I have call waiting on my PSTN line and i want to be able to pick up that second call |
02:51.22 | JT | Telco POTS <---> Asterisk <---> Panasonic |
02:51.26 | JT | iamamoron: rubbish |
02:51.30 | iamamoron | yes |
02:51.31 | JT | iamamoron: NONE have FXS |
02:51.32 | BBHoss | obnauticus: sorry, never played with ci$co before |
02:51.32 | JT | EVER |
02:51.36 | iamamoron | thats the setup i want |
02:51.37 | Mackes | http://www.prodcat.panasonic.com/shop/NewDesign/ModelTemplate.asp?ModelId=16638&show_all=false&product_exists=True&active=1&ModelNo=KX-TD816&CategoryId= |
02:51.48 | JT | iamamoron: let me repeat ths to be crystal clear |
02:51.57 | JT | iamamoron: the X100P does NOT HAVE AN FXS PORT |
02:51.59 | rob0 | The other RJ11 is a passthrough port. |
02:52.05 | Mackes | It looks as if oour KX has 8 stations. |
02:52.06 | JT | it is physically IMPOSSIBLE |
02:52.07 | iamamoron | Mackes: fxs --> panasonic dco? |
02:52.20 | BBHoss | but dont you set it up as fxsks is the zapata config :) |
02:52.25 | BBHoss | :} |
02:52.29 | Mackes | I would replace them all with Asterisk and new phones... its the best way |
02:52.34 | iamamoron | JT: mine has |
02:52.35 | JT | the X100P has NO 90VAC @ 20Hz ringing generator, NO -48VDC line battery feeding |
02:52.37 | BBHoss | Mackes: indeed |
02:52.37 | iamamoron | fxs |
02:52.42 | JT | iamamoron: absolute garbage |
02:52.51 | JT | iamamoron: there is no way it is an X100P |
02:52.57 | Mackes | fxs --> panasonic dco? ????? |
02:52.58 | JT | if it has FXS |
02:53.05 | BBHoss | the panasonic system will just cripple your system |
02:53.12 | rob0 | Mine has 2 RJ11 jacks as well. The second one passes through to another phone/device. |
02:53.27 | Mackes | Really, I know you want to test first, but BBHoss is right, the Panasonic will just hold you back |
02:53.28 | iamamoron | cdo is where i connect my pstn going to panasonic |
02:53.38 | iamamoron | some kind of fxs |
02:53.46 | JT | iamamoron: why won't you listen to us? |
02:53.46 | BBHoss | sweet, i'm right! |
02:53.58 | iamamoron | JT: |
02:53.59 | iamamoron | ok |
02:54.03 | iamamoron | i will listen to you |
02:54.06 | JT | iamamoron: the X100P DOES NOT, absolutely NOT have an FXS port EVER |
02:54.08 | Mackes | Here is the specs on the Pan: |
02:54.08 | iamamoron | what should be done |
02:54.09 | rob0 | Who's going to pick up on the straight line first? |
02:54.12 | Mackes | http://www.prodcat.panasonic.com/shop/NewDesign/ModelTemplate.asp?ModelId=16638&show_all=false&product_exists=True&active=1&ModelNo=KX-TD816&CategoryId= |
02:54.20 | rob0 | Get something with FXS. :) |
02:54.29 | BBHoss | the CO ports on the panasonic system are FXS devices, the ports at your phone company are FXO ports |
02:54.32 | JT | iamamoron: the cheapest option is to buy an SPA-3102 |
02:54.39 | BBHoss | its really damn confusing |
02:54.43 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
02:54.47 | iamamoron | and then |
02:54.52 | Mackes | Yep. Get a SPA 3102 |
02:54.55 | iamamoron | how can i integrate that to my panasonic? |
02:55.08 | BBHoss | iamamoron: trash the panoshit |
02:55.12 | iamamoron | i want to preserve all the telephone |
02:55.18 | BBHoss | iamamoron: or pass it off on ebay |
02:55.21 | JT | iamamoron: the 3102's FXS ports plugs into the Panasonic's FXO port. the 3102's FXO port connect to the telco |
02:55.34 | BBHoss | iamamoron: how many stations/phones are you using? |
02:55.43 | iamamoron | now 16 |
02:55.57 | iamamoron | quite many i cant just throw it away |
02:55.59 | Mackes | sorry man |
02:56.05 | Mackes | I understand |
02:56.21 | Mackes | Is this your companies system? |
02:56.29 | iamamoron | yeap |
02:56.33 | BBHoss | $1400 for new IP hones (IP 320 Polycom) |
02:56.33 | Mackes | Ok |
02:56.37 | Mackes | Well, |
02:56.44 | ManxPower | iamamoron: Generally if you can't replace your PBX, don't bother trying to try tieing Asterisk into the existing PBX |
02:56.49 | Mackes | Build yourself an Asterisk System on the side. |
02:56.49 | iamamoron | i am planning to have asterisk to be my IVR |
02:56.58 | Mackes | dont tie it in to your pan |
02:57.03 | JT | ManxPower: i disagree, there are times when PBX intercept is useful |
02:57.05 | ManxPower | iamamoron: you'll still have issues when you use analog |
02:57.16 | JT | iamamoron: forget about an IVR on analogye |
02:57.19 | iamamoron | JT? |
02:57.20 | BBHoss | JT: this is a rather small system though |
02:57.22 | ManxPower | JT: only if you have PRI or E&M./Wink 8-) |
02:57.23 | Mackes | What will the Intercept do for you? |
02:57.46 | JT | iamamoron: analogue has terrible call status signalling |
02:57.46 | Mackes | it only will have ONE FXS to route to |
02:57.55 | JT | iamamoron: IVRs need good signalling |
02:58.06 | BBHoss | anybody ever buy from ip phone warehous? they have the 320s for $83.99 |
02:58.12 | Mackes | I have |
02:58.15 | Mackes | Very goo |
02:58.15 | JT | iamamoron: you might be able to get away with polarity reversal from the telco |
02:58.18 | Mackes | d place |
02:58.23 | BBHoss | damn, they are cheap |
02:58.37 | Mackes | Just got a Snom 370 and an hitachi Wifi from them |
02:58.57 | iamamoron | so if i buy SPA 3102? |
02:59.24 | BBHoss | iamamoron: please dont punish us by trying to do what you're trying to do |
02:59.37 | iamamoron | JT? |
02:59.41 | iamamoron | i am talking to JT |
02:59.58 | BBHoss | this is a chat room |
03:00.15 | obnauticus | Does anyone here have knowledge of how to update firmware on a Cisco 79xx (preferably 7960) IP Phone? |
03:00.19 | ManxPower | iamamoron: expect to have lines stuck offhook if you insist on using analog |
03:00.44 | iamamoron | because i am PSTN here |
03:00.53 | iamamoron | BBHoss: dont use us |
03:00.56 | BBHoss | iamamoron: JT even said ivr was a bad idea |
03:00.58 | iamamoron | yes it is a chat room |
03:01.05 | iamamoron | but i am talking to JT |
03:01.13 | ManxPower | obnauticus: that would be on the Cisco web site. |
03:01.14 | iamamoron | you are not JT |
03:01.21 | BBHoss | moron :) |
03:02.06 | JT | iamamoron: then you would link the 3102 to asterisk via sip |
03:02.29 | JT | iamamoron: you need polarity reversal on disconnect from your telco to run an ivr |
03:02.31 | obnauticus | ManxPower, I understand thatr but they have a bug in their newest SIP firmware apperentally. |
03:02.42 | BBHoss | obnauticus: as always :) |
03:02.53 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
03:02.53 | ManxPower | obnauticus: The release notes and firmware upgrade instrucitons have a bug? |
03:02.53 | obnauticus | yeh |
03:02.59 | obnauticus | lol |
03:03.10 | obnauticus | http://www.voip-info.org/wiki/index.php?page=Firmware+issues+on+7940+-+7960 <-- that is what I'm reading |
03:03.15 | obnauticus | and it looks like I'm getting that issue |
03:03.17 | ManxPower | you asked "how to update", well "how to update" would be answered "on cisco web site" |
03:03.18 | obnauticus | or something like that. |
03:03.28 | obnauticus | My bad :/ |
03:04.16 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
03:04.26 | ManxPower | obnauticus: here you go http://www.cisco.com/en/US/products/hw/phones/ps379/prod_installation_guides_list.html |
03:04.34 | BiG^DoG | so no help on my spa3102 call waiting problem? :( |
03:05.12 | JT | BiG^DoG: best to forget about call waiting |
03:05.41 | ManxPower | BiG^DoG: What is your "call waiting problem"? |
03:05.53 | [TK]D-Fender | ManxPower, he wants to flash the FXO port |
03:05.59 | BiG^DoG | yes |
03:06.01 | ManxPower | obnauticus: Actually, a link to the correct upgrade/release notes will be included with the firmware you have. |
03:06.09 | BiG^DoG | or find a suitable alternative |
03:06.11 | BBHoss | iamamoron: use your asterisk box to recieve calls from the telephone company |
03:06.14 | ManxPower | BiG^DoG: It is not possible to use the telco call waiting with SIPura products |
03:06.21 | [TK]D-Fender | BiG^DoG, Which I don't know if *'s SIP has a way of passing, nor does an app exist for that channel |
03:06.25 | BiG^DoG | workarounds/alternatives? |
03:06.34 | obnauticus | ManxPower, no it's not |
03:06.42 | [TK]D-Fender | BiG^DoG, you CAN do this on a zaptel interface. |
03:06.45 | obnauticus | It's the SIP firmware :/ open to the public. |
03:06.45 | ManxPower | BiG^DoG: If there were workarounds/anternatives then it would not be "impossible" |
03:07.14 | ManxPower | obnauticus: I have no idea what you mean. The firmware should include release notes,. Are you saying that the firmware does NOT include release notes? |
03:07.44 | obnauticus | no i'll find the,m |
03:07.45 | BiG^DoG | could I ditch the spa3102 and get another ATA? |
03:07.56 | Mackes | Cisco |
03:07.58 | JT | BiG^DoG: no, a physical card |
03:08.02 | Mackes | What do you want to know |
03:08.04 | ManxPower | BiG^DoG: I am not aware of ANY ATA that supports what you want to do |
03:08.10 | JT | BiG^DoG: can't signa; hookflash over SIP |
03:08.24 | ManxPower | obnauticus: that is the "m" |
03:08.37 | ManxPower | obnauticus: odd. |
03:08.59 | [TK]D-Fender | BiG^DoG, * cann't tell any SIP device to "flash". Your only current option is to get a Zaptel interface. |
03:09.48 | Mackes | obnauticus, did you find what you wanted? |
03:10.00 | MrTelephone | is that the guy who ignored me yesterday |
03:10.02 | MrTelephone | fruitcake |
03:10.16 | BiG^DoG | I thought the most recent sipura firmware allowed a double hook flash to pass through to pstn |
03:10.18 | obnauticus | Still looking this is interesting :/ |
03:10.23 | JT | which fruitcake? |
03:10.35 | MrTelephone | obnauticus |
03:10.42 | MrTelephone | christmas fruitcake :P |
03:10.43 | [TK]D-Fender | BiG^DoG, * can't SEND one in the first place. |
03:10.43 | JT | BiG^DoG: maybe from the fxs port |
03:10.56 | BBHoss | i hate fruitcakes |
03:11.15 | BiG^DoG | could it be done from the fxs port? |
03:11.24 | Mackes | Here you go |
03:11.27 | [TK]D-Fender | JT : Shouldn't. The FXS & FXO ports are INDEPENDANT. So I guess that'd only apply if those 2 ports are bridged and KNOW IT. |
03:11.27 | Mackes | ftp://ftp.cisco.com/pub/voice/ip-phone/sip-7960/ |
03:11.41 | Mackes | and |
03:11.44 | Mackes | http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP |
03:11.57 | obnauticus | http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/relnote/phnrn82s.htm#wp1149999 |
03:12.16 | JT | Mackes: how about the firmware for a Cisco 12SP+? :) |
03:12.19 | GuyOCanada | what cisco phone would you recommend me |
03:12.24 | obnauticus | none. |
03:12.26 | BBHoss | JT: heh |
03:12.26 | obnauticus | Polycom |
03:12.37 | Mackes | Do you have a Cisco account. |
03:12.42 | JT | no |
03:12.44 | obnauticus | No. |
03:12.50 | Mackes | Because if not, you are going to need my link |
03:12.52 | ManxPower | BiG^DoG: no matter how many times you ask that question, the answer will be the same. |
03:13.01 | JT | the 12SP+ fills a niche that no other ip phones does |
03:13.02 | Mackes | ftp://ftp.cisco.com/pub/voice/ip-phone/sip-7960/ |
03:13.03 | MrTelephone | mackes do you have a cisco account? |
03:13.05 | BiG^DoG | it was a different question |
03:13.09 | Mackes | yep |
03:13.15 | Mackes | But you dont need it |
03:13.17 | Mackes | ftp://ftp.cisco.com/pub/voice/ip-phone/sip-7960/ |
03:13.22 | GuyOCanada | ok which Polycom do you recommend |
03:13.28 | JT | the 12SP+ are indestructable |
03:13.28 | Mackes | That is the magic man |
03:13.37 | MrTelephone | can you download me the newest ubr7200 image :P |
03:13.43 | JT | we deploy a dozen of them in the middle of the bush every year |
03:13.53 | Mackes | 12SP+ ??? What is that? |
03:14.09 | JT | beige coloured ip phone |
03:14.13 | JT | does H.323 or SCCP |
03:14.22 | Mackes | ohhh. No SIP? |
03:14.25 | JT | no |
03:14.45 | ManxPower | MrTelephone: Sure! You want a copy of MS Office and CD Key as well? |
03:14.45 | BBHoss | GuyOCanada: depends on application |
03:14.55 | Mackes | I am looking for an indestructible Sip Phone |
03:14.58 | ManxPower | I can give you the latest pirated music too! |
03:14.58 | BBHoss | GuyOCanada: the IP330s work good for general purpose |
03:15.17 | Mackes | The Cisco SIP Image is availible to the public right now |
03:15.22 | JT | indestructable H.323 phone, good enough ;) |
03:15.27 | GuyOCanada | I have only one place I need a real ip phone for the use with asterisk |
03:15.34 | obnauticus | ManxPower, it's weird because it as the ip 10.0.0.95, then when it continues on to try and get the CM list it changes to 10.0.1.71 |
03:15.36 | JT | Mackes: can you get firmware or documentation for the 12SP+ series? |
03:15.40 | obnauticus | And it starts timing out |
03:15.41 | GuyOCanada | all my stuff is routed over voip and there are no hard phones everyplace is using softphones |
03:15.54 | Mackes | they are Ciscos? |
03:15.59 | obnauticus | Then it says opening 10.0.0.110 and in my TFTP window it's getting a TIMEOUT waiting for ACK Blcok #1 |
03:16.10 | JT | Mackes: yes, i already mentioned that :) |
03:16.25 | JT | they're pre 7000 series |
03:17.17 | Mackes | That is an Old Ass looking phone |
03:17.35 | JT | they're actual strong, unlike the latest ip telephones |
03:17.37 | Mackes | Sorry, my login info is at the office |
03:17.41 | Qwell | 12SP+ == relabeled Selsius |
03:17.49 | Qwell | (Selsius Phone - get it?) |
03:18.02 | JT | never heard of selsius |
03:18.06 | [TK]D-Fender | Qwell, freezes solid randomly :p |
03:18.19 | Qwell | the 12SP+? heh |
03:18.25 | Qwell | I can lock mine up pretty easily.. |
03:18.37 | Mackes | You have one? |
03:18.41 | Qwell | actually, the 30VIP is *easy* to lock up.. just try to set 30 speeddials |
03:18.42 | JT | i have one |
03:18.47 | JT | i haven't used it yet |
03:18.56 | Mackes | Hey Qwell.... how did you get Mod status |
03:19.05 | Mackes | That would be helpful in this group |
03:19.08 | JT | Mackes: work for digium? |
03:19.11 | Qwell | Mackes: /whois me |
03:19.23 | *** join/#asterisk wothinn (n=Allfathe@vs1.svartalfheim.net) |
03:19.36 | Mackes | Haaa Cool |
03:20.03 | Mackes | Qwell, How do you feel about Cisco Sip, Polycom, and Snom and Aastra |
03:20.09 | Qwell | google images has failed me |
03:20.13 | Qwell | ~phones |
03:20.13 | jbot | extra, extra, read all about it, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream ... |
03:20.27 | Mackes | We have had a long disagrement about such things |
03:20.35 | Mackes | yep ,phones |
03:22.08 | GuyOCanada | I have a question i know its not asterisk specific but when i run menuconfig to select what i want to install with asterisk on the codecs list it says speex and when i am on it it gives me XXX (Depends on speex(E)) what does that E mean? |
03:24.51 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
03:26.44 | Nugget | it means you need to install the speex libraries on your machine before you can compile that support in to asterisk. |
03:27.08 | Nugget | http://www.speex.org/downloads/ |
03:27.38 | GuyOCanada | Nugget: I already installed it using yum install speex and libspeex |
03:28.04 | Nugget | you probably need speex-devel or something along those lines |
03:28.45 | GuyOCanada | speex-devel is also installed |
03:28.55 | Nugget | I dunno then |
03:29.31 | Nugget | have you re-run ./configure since doing that? |
03:29.43 | GuyOCanada | yes |
03:29.53 | GuyOCanada | make clean ; ./configure |
03:30.27 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
03:31.55 | *** join/#asterisk implicit (n=implicit@ip68-101-122-30.oc.oc.cox.net) |
03:34.47 | *** join/#asterisk dkwiebe (n=chatzill@h66-112-187-16.mcsnet.ca) |
03:35.03 | dkwiebe | good evening. |
03:35.50 | dkwiebe | I was on here a while back looking for help setting up a couple of analog DID trunks. I received help but I was unable to get it to work. Anyways, I'm just following up. |
03:36.08 | dkwiebe | We got it working by using a couple of Audiocodes Analog DID Trunk gateways. |
03:36.17 | dkwiebe | They were priced reasonably and work well. |
03:36.36 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
03:38.15 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
03:42.24 | [TK]D-Fender | dkwiebe, How is it signalled? Flash? Immediate DTMF? |
03:43.03 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
03:43.25 | dkwiebe | They called it a "wink" which I believe is the same as a flash. |
03:44.12 | jameswf-home | wink is often a revpol |
03:44.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
03:45.12 | dkwiebe | Somebody here, was it you [TK]D-Fender? suggested some dialplan code. I tried it but Telus was not very helpful and they insisted that our side was marking the lines as "busy" |
03:45.14 | [TK]D-Fender | dkwiebe, So you answer, issue the wink, they pass fixed length dtmf and then hand off the channel? |
03:45.18 | coppice | a wink is in the opposite direction from a flash, so one is a pulse of reversal and the other a pulse of open loop, the intended effect is similar |
03:45.54 | dkwiebe | that is the theory. They're acting exactly like an extension of our pbx except that they want a wink when they pick up. |
03:46.27 | [TK]D-Fender | dkwiebe, nifty |
03:46.41 | [TK]D-Fender | coppice, thanks for that little tidbit. Something to remember. |
03:46.48 | dkwiebe | coppice: Thanks, I'll remember that. |
03:47.09 | dkwiebe | Do you know if it's possible to get asterisk to "wink" from the dialplan? |
03:47.38 | jero | wink ? |
03:47.47 | jameswf-home | no wink is signalling |
03:47.59 | [TK]D-Fender | dkwiebe, not to send a signal to the Audiocodes to do that... |
03:48.04 | jameswf-home | the hardware does it |
03:48.37 | dkwiebe | If I thought we'd ever need it again I'd open a bounty. :-) |
03:48.44 | dkwiebe | maybe I should anyways... |
03:49.09 | dkwiebe | the audiocodes takes care of it all |
03:49.16 | dkwiebe | ok. |
03:49.36 | dkwiebe | They're actually pretty cool little boxes, trivial to configure and they just work. |
03:49.57 | *** join/#asterisk izaak (n=izaak@modemcable132.248-130-66.mc.videotron.ca) |
03:51.36 | *** part/#asterisk jero (n=jerome@modemcable169.212-70-69.mc.videotron.ca) |
03:51.37 | [TK]D-Fender | dkwiebe, "trivial" is a term I can't say I've ever heard applied to AudioCodes configuration. Cryptic, convoluted, obscene are more common :) |
03:51.49 | dkwiebe | I'm sorry |
03:52.00 | dkwiebe | my mistake. |
03:52.06 | izaak | Anyone wanna shout their recommendation for a fan-less system to drive a tdm400? i'm thinking of the intel D201GLY2. the wrap alix 1c also looks interesting, but i'm unsure if an amd lx 500mhz processor could handle it. |
03:52.10 | dkwiebe | not audiocodes. They're brutal to configure |
03:52.39 | dkwiebe | I'm going to have to look up these boxes as my mind has gone blank. |
03:53.05 | dkwiebe | lol, that's better. "Multitech Analog DID Trunk Adapter" |
03:53.22 | dkwiebe | "trivial" isn't a term I'd use after spending the last 2 days fighting with them. |
03:53.27 | dkwiebe | them = audiocodes |
03:53.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
03:54.04 | [TK]D-Fender | dkwiebe, For a second I was thinking you must be from Bizarro-World from whence they came ;) |
03:54.25 | dkwiebe | lol, yeah I bet. |
03:56.20 | MrTelephone | obnauticus has me on ignore from yesterday and I could help him |
03:56.39 | MrTelephone | <obnauticus> Okay, I have a cisco 7960 IP Phone, I am trying to upgrade the firmware to the SIP firmware, and now it's in some weird loop. |
03:56.40 | [TK]D-Fender | MrTelephone, some people you just can't reach! |
03:56.43 | MrTelephone | haha |
03:56.49 | MrTelephone | no i said something offensive to him yesterday |
03:57.42 | MrTelephone | knocked up was a pretty good movie |
04:00.15 | Qwell | MrTelephone: ripped it last night :p |
04:00.20 | Qwell | erm, yeah |
04:00.24 | Qwell | made a backup copy |
04:00.34 | Qwell | ...in case hollywood video ever needs it |
04:01.56 | MrTelephone | hahha |
04:02.12 | MrTelephone | hey i paid for a pack of dvds and some royalties went to hollywood |
04:02.19 | MrTelephone | so basically i own a percentage of anything they make |
04:03.06 | MrTelephone | qwell, that problem i was having with the sip clients had something to do with packet loss.. but it may happen again :( |
04:08.17 | coppice | Its a sad day on IRC when at least one idiot doesn't add you to their ignore list or call you a retard |
04:09.19 | MrTelephone | coppice your mean sometimes |
04:09.39 | MrTelephone | :( |
04:09.44 | MrTelephone | but i didn't ignore you |
04:09.58 | MrTelephone | i havn't been ctcp flooded in 12 years |
04:10.53 | MrTelephone | qwell is working on skinny? |
04:10.59 | Qwell | LIES |
04:11.13 | MrTelephone | get those wirelss voip phones working already will ya |
04:11.13 | coppice | he's just mildly overweight |
04:11.21 | [TK]D-Fender | Qwell, it's not just a river in Egypt! ;P |
04:11.24 | Qwell | mildly, hell |
04:11.31 | Qwell | MrTelephone: they work just fine |
04:11.37 | Qwell | MrTelephone: send me one, and I'll keep it updated |
04:13.31 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
04:13.37 | UserReg_CL | hi, helpme please... |
04:13.46 | UserReg_CL | (alguien habla espaol?) |
04:14.01 | coppice | 不是 |
04:15.59 | MrTelephone | do you know how to timestamp graphs made by rrdtool? |
04:17.11 | UserReg_CL | call to extension fail for one context (number internal) |
04:17.19 | GuyOCanada | what does the module embedding options do? |
04:18.07 | *** join/#asterisk hawky (n=geoff@c-71-231-188-226.hsd1.or.comcast.net) |
04:18.36 | MrTelephone | do you guys get ftp bruteforce attacks alot? |
04:18.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
04:19.25 | coppice | no, our FTPs get lured into a false sense of security by the Sirens |
04:19.48 | asdx | UserReg_CL: si, pero soy un newbie con asterisk |
04:20.16 | UserReg_CL | asdx: tengo un problemita... quiza puedas darme una mano... |
04:20.23 | asdx | UserReg_CL: ok |
04:20.49 | UserReg_CL | asdx: no puedo efectuar una llamada a una extension :( |
04:20.54 | asdx | s/newbie/principiante |
04:22.30 | MrTelephone | the guy keeps trying to login as Administrator |
04:22.56 | UserReg_CL | ~book |
04:22.56 | jbot | 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 |
04:22.59 | asdx | UserReg_CL: postea tus configuraciones en pastebin y copia/pega aca, estoy seguro que te van a poder ayudar los que tienen mas experiencia, yo soy muy nuevo con asterisk |
04:23.00 | MrTelephone | Nov 16 23:22:37 bonnie proftpd[4413]: localhost.localdomain (::ffff:88.191.14.100[::ffff:88.191.14.100]) - no such user 'Administrator' |
04:23.05 | MrTelephone | over and over |
04:23.23 | UserReg_CL | asdx: gracias |
04:23.30 | asdx | UserReg_CL: de nada |
04:24.21 | asdx | UserReg_CL: yo estoy probando conectarme a un ITSP |
04:24.21 | coppice | MrTelephone: sounds like its a windows attack |
04:25.58 | UserReg_CL | asdx: que es eso ? |
04:26.04 | [TK]D-Fender | MrTelephone, nothing says "I Love You" like an iptables drop rule ;) |
04:26.23 | MrTelephone | you read my mind |
04:26.41 | coppice | I know spanish is the third most spoken language, but I wonder why almost anything on here that is not in English is in Spanish? |
04:26.45 | [TK]D-Fender | MrTelephone, Get off my channel... this is a licensed frequency! |
04:26.51 | MrTelephone | hehe |
04:26.55 | asdx | UserReg_CL: proveedor de voip, para hacer llamadas locales/internacionales |
04:27.08 | asdx | UserReg_CL: a PSTN |
04:27.22 | [TK]D-Fender | coppice, there is a nearly derogatory sounding socio/political explanations for that.... |
04:27.29 | UserReg_CL | asdx: yo tengo una targeta Digium conectada a una trama E1 por la que salgo a pstn |
04:27.41 | asdx | UserReg_CL: que bueno |
04:27.41 | MrTelephone | i added a 5minute crontab folder in /etc/ and added the appropriate line to /etc/crontab to run-parts in the new folder.. restarted cron using /etc/init.d/cron restart.. doesn't work :( |
04:27.51 | [TK]D-Fender | UserReg_CL / asdx : Hey |
04:27.55 | [TK]D-Fender | ~asteriskspanish |
04:27.56 | jbot | [~asteriskspanish] Asterisk Community in Spanish, just visit http://www.asterisk-la.org -=- IRC channel #asterisk-es |
04:27.57 | [TK]D-Fender | ^^^^^^^^^^ |
04:28.09 | asdx | lol sorry |
04:28.13 | [TK]D-Fender | :p |
04:28.36 | [TK]D-Fender | If you want help you're more able to understand, why not use the resources that are out there... |
04:28.48 | UserReg_CL | sorry.. |
04:29.15 | [TK]D-Fender | UserReg_CL, I'm not actually annoyed or anything, but we all feel like we're missing out :) |
04:30.08 | asdx | :P |
04:30.09 | coppice | the americans here just feel anyone not writing in english must be terrorists plotting against them. :-) |
04:31.36 | [TK]D-Fender | Al-aqueba, jihad! Jihad! |
04:31.44 | [TK]D-Fender | .... I mean.... pass the mayo please! |
04:31.51 | [TK]D-Fender | *cough* |
04:33.19 | UserReg_CL | I don't know how to call from extensions to other extension, have configuration: http://pastebin.com/m6a05de80 |
04:34.15 | [TK]D-Fender | UserReg_CL, exten => 9XX,1,Dial(SIP/${EXTEN}) <- tis doesn't work because you are missing the "_" in front of your PATTERN <--- |
04:34.30 | [TK]D-Fender | UserReg_CL, should look like THIS : exten => _9XX,1,Dial(SIP/${EXTEN}) |
04:34.36 | [TK]D-Fender | UserReg_CL, Now go fix the rest |
04:34.51 | [TK]D-Fender | UserReg_CL, and remove the quotes from : include => "internos" |
04:35.02 | [TK]D-Fender | NEXT@!@ |
04:35.06 | [TK]D-Fender | (c) BKW |
04:35.07 | UserReg_CL | Thank, sorry |
04:35.14 | [TK]D-Fender | UserReg_CL, You're welcome |
04:35.17 | *** join/#asterisk n7okn (n=n7okn@ip68-109-169-42.ph.ph.cox.net) |
04:37.25 | UserReg_CL | now can call to 951 from 950 ? |
04:38.09 | obnauticus | F*cking hell this Cisco 7960 is almost Too good of a learning experience (bad) lol. |
04:38.58 | n7okn | I have a question. I'm a newbie and I found an alternative conference app for Asterisk I'd like to try... mainly for the experience of installing an Asterisk app. It has a .c extension. I tried the make and configure way but no good. What is it, and how do I compile it? |
04:40.01 | *** join/#asterisk ManxPower (n=manxpowe@55.sub-70-197-188.myvzw.com) |
04:40.41 | [TK]D-Fender | UserReg_CL, go try |
04:40.41 | n7okn | I tried to configure a 7960 and had to really scratch my head. I think I'll stick with Linksys LOL |
04:40.47 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
04:40.52 | UserReg_CL | yes |
04:40.58 | grandpapadot | Anyone see why this won't evaluate? exten => _8XX,n,ExecIf($[${LEN(${DNID})}!=3]|Set|GROUP()=${CONTEXT}) |
04:41.06 | obnauticus | n7okin what about change the uhh |
04:41.09 | obnauticus | Firmware on the thing? |
04:41.14 | grandpapadot | If I just do "=3" it works, but "!=3" doesn't work. |
04:42.47 | n7okn | ??? |
04:43.24 | n7okn | u mean on the 7960? not my phone. |
04:43.54 | UserReg_CL | I return... reboot machine :) |
04:46.45 | n7okn | still looking for .c script install path. |
04:50.29 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
04:50.38 | obnauticus | Oh n7okn im doing it right now |
04:50.40 | obnauticus | i think I got it |
04:50.48 | obnauticus | i have to do an incremental upgrade from their first release. |
04:51.01 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
04:51.05 | UserReg_CL | hi :) |
04:51.16 | UserReg_CL | one softphone know ? |
04:51.31 | UserReg_CL | ~software |
04:51.32 | jbot | from memory, software is doing something that hardware should |
04:51.38 | obnauticus | ~softphone |
04:51.39 | jbot | something that should be drug out into the street and shot |
04:51.46 | obnauticus | ~phone |
04:51.46 | jbot | extra, extra, read all about it, phone is warbling while I'm updating the flash from blob...it's amusing now, it's like it knows I'm erasing its brain. Mwa ha ha ha. |
04:51.54 | obnauticus | god damn it |
04:51.55 | obnauticus | :| |
04:51.59 | GuyOCanada | can you recommend me a good softphone |
04:52.12 | JT | none of them are good |
04:52.15 | JT | some are ok though |
04:52.32 | UserReg_CL | yes need one softphone for windows |
04:52.40 | GuyOCanada | i need one too |
04:52.45 | coppice | what they fail to tell you is all IP phones are soft, and none are very good :-) |
04:52.46 | obnauticus | Xlite is decent. |
04:52.48 | GuyOCanada | untill i have finished my testing and decide which phone to buy |
04:53.06 | JT | lol, decent, sort of ok is more my take on slite |
04:53.08 | JT | xlite |
04:53.12 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
04:53.37 | GuyOCanada | I used SJphone before but not so good |
04:53.38 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
04:53.39 | asdx | does xlite uses alsa on linux? |
04:53.52 | obnauticus | probably |
04:54.17 | UserReg_CL | need one small |
04:54.43 | dlynes | !seen flauto |
04:54.49 | [TK]D-Fender | Zoiper is pretty decent. One of the few with Transfer. Or go try Ekiga. |
04:55.41 | UserReg_CL | thank |
04:56.39 | ManxPower | Xlite is like one of those Japanese gadgets that do a million things, but the user interface is so cluttered, complicate, and confusing that you can't actually do much with it. |
04:58.01 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
04:58.26 | UserReg_CL | mmm |
04:58.37 | GuyOCanada | xlite has no spywares in it does itr |
04:59.15 | ManxPower | GuyOCanada: no, it does not as far as I know |
05:01.29 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
05:01.51 | [TK]D-Fender | eek |
05:02.30 | dlynes | ~seen flauto |
05:02.33 | jbot | flauto <n=zhao@71.194.141.225> was last seen on IRC in channel #asterisk, 13d 11h 25m 35s ago, saying: 'and email'. |
05:02.45 | UserReg_CL | Grr not connect Zoiper |
05:03.29 | obnauticus | anyone here with Cisco Phone Experience: `Phone Unprovisioned' recent firmware upgrade to SIP 3.4 |
05:06.39 | Nugget | I get that whenever the phones aren't getting the tftp server address from my dhcp server |
05:07.50 | UserReg_CL | bearercapability notauth |
05:08.37 | obnauticus | Nugget how do I tell my DHCP server what the TFTP is |
05:08.41 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
05:08.44 | *** join/#asterisk mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
05:08.53 | mackes | hey |
05:10.38 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
05:11.32 | [TK]D-Fender | obnauticus, depends which dhcp server you're using. |
05:11.42 | obnauticus | Whatever pfsense default is :/ |
05:12.05 | [TK]D-Fender | obnauticus, Odds are if you don't even know what you have, you shouldn't be using it :) |
05:12.47 | De_Mon | obnauticus read the documentation of said DHCP server, duh. |
05:13.10 | obnauticus | im using my primary monitor right now otherwise i would |
05:13.29 | BBHoss | obnauticus: u have to set option 66 |
05:13.39 | coppice | [TK]D-Fender: that sounds like a line from a bad country and western song |
05:13.44 | UserReg_CL | not work zoiper :( |
05:14.21 | [TK]D-Fender | coppice, you say that... as though there were any other kind ;) |
05:14.26 | coppice | it was only for emphasis |
05:15.27 | asdx | [Nov 17 05:14:25] NOTICE[6407]: chan_iax2.c:5258 register_verify: Peer 'teliax' is not dynamic (from 190.52.158.12) |
05:15.30 | asdx | what does that means |
05:15.37 | BBHoss | obnauticus: you can edit the dhcpd file to includ this |
05:15.52 | obnauticus | k |
05:15.53 | BBHoss | option tftp-server-name"YOURIP"; |
05:15.57 | BBHoss | then |
05:16.16 | BBHoss | put a tab after tftp-server-name |
05:16.38 | BBHoss | http://cvstrac.pfsense.com/tktview?tn=1026 |
05:17.46 | coppice | testing T.38 with 2M bytes FAX pages allows so much time for tea and chatting on IRC :-) |
05:17.58 | obnauticus | BBHoss where does it say this file is |
05:18.04 | UserReg_CL | need 2 xlite in same windows but not work |
05:18.09 | BBHoss | obnauticus: i think that was wrong |
05:18.18 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
05:18.38 | BBHoss | obnauticus: it just uses dhcpd though, you could look up how to set it in there |
05:19.49 | obnauticus | found it BBHoss |
05:20.07 | BBHoss | im pretty sure its just freebsd's version of dhcpd |
05:20.24 | [TK]D-Fender | UserReg_CL, you'll have to configure one to use a different port. And even then it'll fight over sound card resources. |
05:20.29 | Mavvie | freebsd doesn't have their own dhcpd, it's the ISC DHCPD you are using. |
05:20.30 | GuyOCanada | I successfully connect to asterisk but i can not hear any response |
05:20.49 | UserReg_CL | thank |
05:21.14 | mackes | neaaaaaaaaaaaattttttoooo |
05:21.32 | UserReg_CL | thank friends.. |
05:21.37 | UserReg_CL | good night |
05:22.11 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-207-168.rgv.res.rr.com) |
05:22.17 | asdx | UserReg_CL: night |
05:22.23 | GuyOCanada | http://pastebin.com/m1e0937e5 |
05:22.44 | UserReg_CL | asdx: Buenas noches ... |
05:22.55 | TrentCreek | I know Fender knows this one...reject calls from PayPhones |
05:23.00 | TrentCreek | BUENOS NACHOS |
05:23.13 | asdx | UserReg_CL: igualmente |
05:23.19 | TrentCreek | nop |
05:23.19 | asdx | TrentCreek: lol |
05:24.08 | TrentCreek | diesperto |
05:24.41 | TrentCreek | everyone wake up |
05:24.45 | mackes | hey |
05:25.04 | mackes | Does anything ever happen in the other asterisk channels? |
05:25.07 | TrentCreek | good another victim |
05:25.15 | TrentCreek | during the day in the US |
05:25.15 | mackes | Oh no |
05:25.26 | mackes | Im not new to this happy bunch |
05:25.29 | TrentCreek | I needs to know how to reject payphone calls |
05:27.50 | GuyOCanada | what is the default codec that is assigned? |
05:29.32 | TrentCreek | that would be in extensions.conf |
05:30.49 | [TK]D-Fender | GuyOCanada, default is whatever YOU set in your channel drivers conf file. |
05:32.43 | TrentCreek | oh thereis meester geetar |
05:33.26 | MrTelephone | IF ${CLIENTID}=PAYPHONE Hangup() |
05:33.32 | coppice | his real name is probably Les Paul or Mr Fernendes |
05:33.34 | MrTelephone | i wish |
05:33.47 | TrentCreek | hey!! thanks Mr Obvious! |
05:34.03 | TrentCreek | or Esteban! |
05:34.10 | MrTelephone | your welcome skywalker |
05:34.28 | GuyOCanada | 150 ms delay thats too much for voip isnt it? |
05:34.45 | MrTelephone | trentcreek, see if your telco delivers ANI2 codes |
05:34.58 | MrTelephone | im 150ms and it sounds good |
05:35.15 | MrTelephone | packetloss and jitter are the killers |
05:35.28 | MrTelephone | sat phones are like 300-600ms |
05:35.47 | GuyOCanada | well i hear my sound like 1 sec after i talk on the echo test |
05:35.49 | TrentCreek | "You know thousand of youngers long to express theior hearts with the gift of Music! |
05:35.57 | GuyOCanada | but the ping latency is 150 ms |
05:36.10 | MrTelephone | but the echo test is already delayed |
05:36.28 | GuyOCanada | it is? |
05:36.37 | MrTelephone | did you try it on your local network? |
05:36.47 | ectospasm | asterisk-gui needs work... I couldn't figure out how to get it to go to my voice menu (instead of the demo) without hand editing extensions.conf... |
05:36.50 | GuyOCanada | no i work remote |
05:37.03 | MrTelephone | 150ms should be good |
05:37.34 | GuyOCanada | but the lady talking when i call stops talking for a sec or two sometimes |
05:37.37 | TrentCreek | oh Mr telephone... |
05:37.57 | MrTelephone | you get used to it |
05:38.15 | MrTelephone | its even worse if your not tied into the pstn at your server |
05:38.15 | TrentCreek | I realise I need to reject before it hangs up..I dont want to be wracking up 50 cents payphoen carges |
05:38.35 | TrentCreek | i mean reject before it answers |
05:38.53 | MrTelephone | i read one article where if you disconnect before 10 seconds its not considered a call |
05:38.57 | MrTelephone | that was for collect though |
05:39.08 | GuyOCanada | grr i think its a codec problem but not sure |
05:39.42 | MrTelephone | theres 1000ms in a second |
05:39.45 | TrentCreek | I wonder if I can put that statement before the asnwer statement |
05:39.53 | mackes | Do any of you use FWD or Sipphone? |
05:40.09 | MrTelephone | trentcreek, you don't have to answer() incoming on a t1 |
05:40.18 | MrTelephone | ? |
05:40.46 | MrTelephone | GuyOcanada, why do you think that? |
05:40.54 | GuyOCanada | MrTelephone yes there is but still it is annoying when the auto attendant drops for a second |
05:41.08 | TrentCreek | not operating on T1 |
05:41.10 | MrTelephone | at the beginning of the call or in the middle of the call? |
05:41.28 | GuyOCanada | con grat ula tions , con figu rate d |
05:41.32 | GuyOCanada | its talking like that |
05:41.41 | [TK]D-Fender | GuyOCanada, 150 can be just fine. What codec are you using, and what is the bandwidth of your connection? |
05:41.50 | GuyOCanada | if you need more te c hni cal info rma |
05:41.52 | MrTelephone | thats scray |
05:41.57 | MrTelephone | thats packetloss or jitter |
05:42.05 | GuyOCanada | well |
05:42.11 | GuyOCanada | the server runs on a 100mbit connection |
05:42.32 | GuyOCanada | the clients runs on a 1.5 mbit dsl connection |
05:42.38 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
05:42.38 | *** mode/#asterisk [+o russellb_] by ChanServ |
05:42.41 | GuyOCanada | i have not changed any config files so its the default codec |
05:43.28 | MrTelephone | if you access your voicemail on it, does it sound shitty? |
05:43.30 | [TK]D-Fender | GuyOCanada, Never say "default". Go look in your configs and find out what you're doing. |
05:43.33 | MrTelephone | or just when you phone another client? |
05:43.47 | MrTelephone | never say default. hah thats a new one |
05:43.49 | MrTelephone | haha |
05:43.52 | GuyOCanada | let me check |
05:44.07 | MrTelephone | ./nick MrDefault |
05:44.26 | GuyOCanada | quality 3 |
05:44.32 | GuyOCanada | complexity 2 |
05:44.42 | GuyOCanada | enhancement true |
05:44.43 | MrTelephone | is that iax? |
05:44.45 | [TK]D-Fender | GuyOCanada, in ASTERISK. |
05:44.55 | [TK]D-Fender | GuyOCanada, That was meaningless |
05:44.59 | MrTelephone | i never seen those figures before |
05:45.06 | MrTelephone | i guess i have to start drinking |
05:45.21 | GuyOCanada | :) i was looking in to the codecs.conf |
05:45.26 | [TK]D-Fender | MrTelephone, Necrophilliac? |
05:45.44 | [TK]D-Fender | GuyOCanada, You are clealy clueless. Go stop and read THE BOOK. |
05:45.45 | [TK]D-Fender | ~book |
05:45.46 | jbot | 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 |
05:45.50 | MrTelephone | Necrophilliac=ulaw |
05:46.13 | coppice | Go stop - what kind of assistance is that? |
05:46.34 | GuyOCanada | I already have the book open |
05:46.35 | MrTelephone | i hope my grandmother doesn't fall down in front of that guy |
05:46.49 | GuyOCanada | Setting Up the Dialplan for Some Test Calls thats where i am |
05:47.17 | MrTelephone | test play your site opz without bitttorent open |
05:47.34 | [TK]D-Fender | GuyOCanada, you set codecs in your channel driver's conf file. if thats SIP, then look in sip.conf |
05:48.06 | MrTelephone | why can't asterisk configure itself? who the hell made this program.. who didn't think of that |
05:48.13 | MrTelephone | i should be on the board |
05:48.22 | MrTelephone | :P |
05:49.25 | GuyOCanada | bittorent and voip please not in the same building |
05:49.50 | MrTelephone | that breaking up tho |
05:49.54 | MrTelephone | i have that on my cable network |
05:50.02 | [TK]D-Fender | MrTelephone, You asked for it... |
05:50.29 | MrTelephone | i went out today and changed a bunch of connectors on the poles and got my errors down to %0.001 |
05:50.39 | MrTelephone | and things are clear again |
05:56.40 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
05:58.43 | russellb_ | the release of res_telepathy is expected tomorrow |
05:59.18 | *** topic/#asterisk by russellb_ -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.14 (2007/11/16), *-Addons 1.4.4 (2007/10/16), Zaptel 1.4.6 (2007/10/18), Libpri 1.4.2 (2007/10/16) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
05:59.38 | TrentCreek | "You know thousand of youngers long to express their hearts with the gift of Music" |
05:59.56 | TrentCreek | "Hello, this is Fender and I have important news" |
06:00.41 | TrentCreek | "I just saved a bundle by switching my Geetar Insurance to GEICO" |
06:00.51 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
06:01.18 | [TK]D-Fender | russellb_ : But then again... we knew that already ;) |
06:01.43 | De_Mon | I didn't know that, you must be testing before release |
06:01.46 | [TK]D-Fender | TrentCreek, and my nick isn't named for the guitar maker... |
06:01.53 | MrTelephone | i rootshelled a nasa.gov computer once |
06:01.57 | MrTelephone | they were running IRIX |
06:02.05 | TrentCreek | oh.... |
06:02.13 | MrTelephone | is that another name for MAC/OS |
06:02.20 | coppice | "Hello, this is Fender and I have important news. That guy in the movie Robots was an imposter" |
06:02.26 | TrentCreek | I just saves a bund by switching my Classic Car insurance to GEICO |
06:02.43 | De_Mon | where does your name come from [TK]D-Fender ? |
06:03.04 | TrentCreek | from Les Paul as Mr Obvious said |
06:03.40 | MrTelephone | whose mr obvious |
06:03.46 | De_Mon | whos Les Paul? |
06:03.56 | De_Mon | s/s/se/ |
06:04.02 | MrTelephone | he was the guy who played kermit the frog on sesame street |
06:04.13 | coppice | De_Mon: he plays a Fender, and has so far only learned to play in D |
06:04.29 | De_Mon | okay... |
06:04.34 | [TK]D-Fender | coppice, how ASStute :p |
06:04.36 | ectospasm | I'm confused as well |
06:04.57 | De_Mon | Muppet humor I suppose |
06:05.13 | [TK]D-Fender | De_Mon, its from my preffered role while playing tribes 1 CTF years ago. It stuck with me through my AHL that followed |
06:05.13 | MrTelephone | i had a friend who burned out on guitar hardcore |
06:05.24 | MrTelephone | he downloaded the song wipeout and memorized it in 7 minutes |
06:05.25 | TrentCreek | is an American jazz guitarist and inventor. He is a pioneer in the development of the solid-body electric guitar which "made the sound of rock and roll possible."[1] His many recording innovations include overdubbing, delay effects such as "sound on sound" and tape delay, phasing effects and multitrack recording. |
06:05.37 | De_Mon | That, makes much more sense |
06:05.51 | [TK]D-Fender | AHL clan* |
06:05.53 | MrTelephone | you thought it was a lesbian didn't you? |
06:06.21 | TrentCreek | Burnt out on a geetar? Its kept Les Paul going for 92 years |
06:07.20 | De_Mon | dont let D`mon getchoo down fight back with our black powwah |
06:07.47 | MrTelephone | I had to look up IIRC |
06:08.04 | TrentCreek | Here's Mr Obvious |
06:08.05 | TrentCreek | http://riz.vox.com/library/audio/6a00cd96fcbade4cd500d4142b13ea6a47.html |
06:08.21 | coppice | I don't think the Les Paul guitar exactly made rock and roll possible. This guitar http://www.stevehowe.com is older than the Les Paul |
06:08.26 | MrTelephone | another 3 years and we'll all be writing in 4 and 5 letter words capitalized |
06:13.13 | De_Mon | IDTS |
06:13.28 | MrTelephone | overn cleaner? |
06:13.46 | De_Mon | they will be acronyms in that case |
06:14.00 | MrTelephone | DMN R PPL GUNA USE ABREVS ALL THE TIME |
06:14.04 | MrTelephone | heheh |
06:14.15 | MrTelephone | whose listening to mr obvious? |
06:14.24 | *** join/#asterisk bintut (n=chatzill@cm246.gamma178.maxonline.com.sg) |
06:14.28 | De_Mon | oh, that |
06:14.30 | TrentCreek | everyone! |
06:14.41 | TrentCreek | He helps me a lot |
06:14.47 | De_Mon | MrTelephone have you watched Californication? |
06:14.58 | MrTelephone | he thinks oven cleaner is made for cleaning out a vagina |
06:15.06 | MrTelephone | i don't think so |
06:15.13 | MrTelephone | i watched the californian fires |
06:15.21 | TrentCreek | Don't forget that EasyOff for your wife! |
06:15.28 | bintut | hello all.. i'm wondering why the output of "sip show peers" inside the asterisk shell tells me that one of my users is using port 10064? any idea? i'm running asterisk-1.4.13 here |
06:15.36 | De_Mon | david ducovney(sp) is a writer and he bitches about that cyber talk in one episode |
06:15.46 | MrTelephone | bintut, because thats what they are using |
06:15.59 | MrTelephone | bintue, you should phone MrObvious |
06:16.03 | bintut | MrTelephone: you mean, on the sip softphone side? |
06:16.05 | MrTelephone | yeah |
06:16.11 | bintut | i see.. |
06:16.18 | MrTelephone | the outbound/return port of the client |
06:16.29 | bintut | i have to inform the user then to use 5060 |
06:16.37 | MrTelephone | how can you |
06:16.43 | MrTelephone | maybe they have a router and its randomly picking a port |
06:17.06 | TrentCreek | Yes..Mr Obvious is giving excellent advice! http://riz.vox.com/library/audio/6a00cd96fcbade4cd500d4142d73793c7f.html |
06:18.32 | De_Mon | I've got peers behind nat on all sorts of random ports |
06:18.48 | MrTelephone | bintut, that is not the ports on your server |
06:18.54 | MrTelephone | its on the clients side |
06:18.58 | MrTelephone | why does it matter |
06:19.02 | MrTelephone | me too |
06:19.10 | bintut | MrTelephone: yeah. thanks.. :) |
06:19.30 | De_Mon | maybe you should share the problem instead of jump to the wrong solution |
06:20.03 | GuyOCanada | how do you setup a DNS SRV record for sip? |
06:22.05 | MrTelephone | i even have a customer on port 65536 for pete sakes |
06:22.23 | MrTelephone | google dns srv asterisk |
06:25.02 | MrTelephone | IIRC = Interactive Illinois Report Card |
06:29.26 | bintut | gtg now.. thanks all.. thanks MrTelephone.. :) |
06:30.58 | MrTelephone | i love the linux convos about mounting |
06:31.21 | [TK]D-Fender | ~sex |
06:31.22 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
06:34.33 | MrTelephone | hahaha |
06:34.35 | MrTelephone | funny |
06:35.45 | TrentCreek | My command: exten => _X.,3,Dial(SIP/phone1@phone1,10) is not calling the device..darn it |
06:36.08 | MrTelephone | <edman007> Jester_, i just mounted an FS inside itself, and it works :) and its ext3 |
06:36.08 | MrTelephone | nerd |
06:36.20 | GuyOCanada | which is the least bandwidth using codec? |
06:36.21 | TrentCreek | app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
06:36.27 | MrTelephone | guy0canada, g729 |
06:36.38 | MrTelephone | trent, show sip peers |
06:36.46 | MrTelephone | trent, are they registered? |
06:36.51 | [TK]D-Fender | G.723 |
06:36.54 | [TK]D-Fender | ^ |
06:37.14 | MrTelephone | is g723 supported by most phones? |
06:37.20 | MrTelephone | might as well use gsm |
06:37.29 | TrentCreek | ooops..no such command |
06:37.51 | TrentCreek | ohhh |
06:37.58 | MrTelephone | sip show peers |
06:39.00 | TrentCreek | oh...I see now...thanks..I go the wrong name |
06:39.34 | MrTelephone | when is asterisk going to add background noise i wonder |
06:39.49 | TrentCreek | next version |
06:39.59 | MrTelephone | 1.6? |
06:40.42 | TrentCreek | alreayd near 1.6? I am still on 1.4.11 |
06:40.49 | MrTelephone | im using 1.2 |
06:40.52 | GuyOCanada | background noise? |
06:41.03 | *** join/#asterisk Defraz (n=t0tal@63.228.246.245) |
06:41.24 | MrTelephone | so you don't hear complete silence when someone stops talking |
06:41.26 | TrentCreek | last I saw it was at 1.4.13 |
06:41.34 | TrentCreek | maybe a month ago? |
06:43.19 | MrTelephone | yeah i think so |
06:43.24 | MrTelephone | im too scared to try it |
06:45.17 | [TK]D-Fender | 1.4.14 |
06:46.06 | asdx | i need a phone number of a pstn line to try this |
06:46.45 | MrTelephone | where the hell is sqrt on wincalc |
06:46.52 | MrTelephone | is it 1/x^2 |
06:47.15 | [TK]D-Fender | MrTelephone, install something better :) |
06:49.02 | MrTelephone | piece of crap |
06:49.10 | MrTelephone | 1/x^-2 |
06:49.15 | MrTelephone | ? |
06:51.35 | [TK]D-Fender | MrTelephone, stop whining and REPLACE IT already... |
06:52.47 | GuyOCanada | ; a call in the case of a phone disappearing from the net, |
06:52.47 | GuyOCanada | ; like a powerloss or grandma tripping over a cable. |
06:52.54 | GuyOCanada | grandma tripping over a cable (LOL) |
06:53.13 | MrTelephone | oh its X^.5 |
06:53.14 | TrentCreek | staple them in the wall |
06:53.17 | MrTelephone | to get square root |
06:53.24 | TrentCreek | anyone seen the goodie ads for next week? |
06:54.11 | MrTelephone | no but neo beat the machines |
06:54.37 | TrentCreek | www.bfads.net |
06:54.42 | TrentCreek | not for long |
06:55.19 | asdx | i already hear tone |
06:55.32 | asdx | does that means that i'm connected to pstn? |
06:55.36 | asdx | when i dial |
06:56.10 | TrentCreek | are you uising a SIP device? |
06:56.23 | asdx | IAX2 |
06:56.51 | TrentCreek | i think it may provide the dial tone for you then it merely dials out on that line |
06:57.05 | TrentCreek | after you enter the number |
06:57.22 | asdx | ok |
06:59.22 | TrentCreek | because you may dial an extension rather than a phone number so it needs to know what you want to do before it starts doing it |
07:01.46 | asdx | i see |
07:05.27 | [TK]D-Fender | <PROTECTED> |
07:07.40 | asdx | cool |
07:13.10 | asdx | when i'm making the call, do i have to use Answer() or just Dial()? |
07:14.16 | [TK]D-Fender | asdx, Just Dial |
07:14.25 | asdx | [TK]D-Fender: ok |
07:15.32 | [TK]D-Fender | ok, bed time. Later all |
07:24.02 | asdx | i'm dialing some numbers but i don't get answer |
07:24.16 | asdx | i hear tone though |
07:28.14 | *** join/#asterisk BeeBuu (n=chatzill@218.13.66.237) |
07:30.14 | BeeBuu | i had installed asterisk 1.2.13,can i play wav files without install addons? |
07:36.57 | GuyOCanada | how can you authenticate a user? |
07:39.24 | TrentCreek | with SIP? |
07:40.22 | GuyOCanada | yes |
07:40.33 | GuyOCanada | but what i want to do is |
07:40.42 | GuyOCanada | I have an incoming line which is a tollfree number |
07:41.07 | GuyOCanada | when i call to that number from a payphone i want to be able to call someone using an outbound connection |
07:41.19 | GuyOCanada | so i need a method to authenticate the caller |
07:44.30 | BBHoss | look up DISA |
07:49.51 | GuyOCanada | so if i do "exten => 555,1,DISA(134|payphone-dialout)" and call my tollfree number and enter 555 when im asked for the extension and then right after it enter 134 i will be redirected to the payphone-dilaout context? |
07:51.54 | GuyOCanada | is it possible to send video using asterisk (a media file that is saved on the system?) |
07:52.42 | tzafrir_home | BeeBuu, sure. As long as those files are 8000Hz, mono, 16 bits per sample |
07:55.38 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
08:01.56 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
08:02.13 | asdx | i can dial and everything but i never get an answer from the other side |
08:02.18 | asdx | why could that be? |
08:03.28 | *** join/#asterisk BeeBuu (n=chatzill@218.13.97.55) |
08:03.47 | BeeBuu | any one help me please |
08:04.04 | BeeBuu | i can't heard any music when set music on |
08:04.48 | BeeBuu | there is "-- Started music on hold,class 'default' on channel 'sip/998-09d3aa80' " |
08:04.51 | DarkDlx | <tzafrir_home> BeeBuu, sure. As long as those files are 8000Hz, mono, 16 bits per sample |
08:04.53 | BeeBuu | in CLI> |
08:04.57 | tzafrir_home | BeeBuu, where exactly do you try to hear it? |
08:05.05 | GuyOCanada | alarm receiver? |
08:05.14 | tzafrir_home | Is there anything else you can hear there? |
08:05.20 | BeeBuu | nothing... |
08:05.24 | BeeBuu | silent.... |
08:05.36 | tzafrir_home | BeeBuu, please help us help you |
08:05.55 | BeeBuu | tzafrir_home: so what i need to do now? |
08:06.04 | tzafrir_home | ah, a SIP phone |
08:06.27 | tzafrir_home | can you hear a simple sound file in it? |
08:06.37 | tzafrir_home | something with Playback? |
08:06.45 | BeeBuu | i had set mode=files and directory=/var/lib/asterisk/mohmp3 |
08:06.57 | tzafrir_home | Can you hear anything from Asterisk? |
08:06.58 | BeeBuu | no,just silent.... |
08:07.05 | BeeBuu | nothing at all. |
08:07.43 | BeeBuu | there is "-- Started music on hold,class 'default' on channel 'sip/998-09d3aa80' " in CLI> |
08:08.09 | BeeBuu | so ,that's mean music playing,right? |
08:08.44 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
08:11.14 | tzafrir_home | BeeBuu, not from on-hold music. Can you call some test extension? echo test? |
08:11.23 | tzafrir_home | play a file? |
08:12.02 | BeeBuu | exten=>_99X,n,SetMusicOnHold(default) |
08:12.29 | BeeBuu | exten=>_99X,n,waitMusicOnHold(20) |
08:12.44 | BeeBuu | that's all about musiconhold |
08:13.39 | BeeBuu | my headphone is OK |
08:18.34 | tzafrir_home | BeeBuu, how can you tell that? |
08:19.16 | tzafrir_home | let's see.... |
08:19.28 | BeeBuu | i can hear the dial sound |
08:19.51 | tzafrir_home | the dialtone comes from the handset itself, not from Asterisk |
08:20.23 | BeeBuu | i can hear anything play with other application.. |
08:20.43 | tzafrir_home | fine |
08:21.06 | tzafrir_home | so, what is the output of: 'moh show classes' in the CLI? |
08:21.16 | tzafrir_home | please pastebin it if it is longer than 3 lines |
08:21.57 | BeeBuu | No such command |
08:26.10 | BeeBuu | "moh classes show " show this http://pastebin.comd1f99926 |
08:27.20 | BeeBuu | o,god,it's OK now |
08:27.31 | BeeBuu | thanks tzafrir_home |
08:27.55 | *** join/#asterisk porche (n=porche@81.215.122.108) |
08:28.01 | porche | hi everybody |
08:28.53 | porche | i am looking for some reasonably prices + reliable voip termination service, like gafachi (but cannot use gafachi currently), which one do you suggest? |
08:32.57 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
08:33.01 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
08:36.20 | tzafrir_home | BeeBuu, something is wrong with that URL |
08:36.33 | porche | tzafrir hi |
08:36.57 | tzafrir_home | ah, problem fixed, I see. So was this a matter of reload? |
08:38.12 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
08:38.15 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
08:38.56 | BeeBuu | tzafrir_home: you are right :-P |
08:39.00 | BeeBuu | must reload |
08:43.12 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
08:55.47 | *** join/#asterisk cypherdelic (n=cypher@p5B27EA05.dip.t-dialin.net) |
09:04.37 | *** part/#asterisk porche (n=porche@81.215.122.108) |
09:11.55 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
09:11.57 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
09:16.55 | *** join/#asterisk cypherdelic (n=cypher@p5B27EA05.dip.t-dialin.net) |
09:20.38 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.142.153) |
09:21.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
09:26.59 | *** join/#asterisk guillote_GNU (n=guillote@host101.200-82-48.telecom.net.ar) |
09:33.59 | *** join/#asterisk JimmyDee (n=James@97-86-161-253.static.stls.mo.charter.com) |
09:34.41 | JimmyDee | morning everyone, I have a problem, my http.conf is set to 8080 and my manager.conf 8081 and neither port is open on nmap, what gives? |
09:38.54 | GuyOCanada | JimmyDee |
09:39.05 | GuyOCanada | what os? |
09:39.56 | DarKnesS_WolF | any idea about a decent IAX client opensource ? |
09:40.30 | GuyOCanada | JimmyDee: service iptables stop and try again |
09:48.15 | JimmyDee | iptables is stopped, no joy |
09:55.14 | CBU[^_^]M`` | my SPA3102 wont register :( |
10:07.24 | GuyOCanada | Anyone using Windows Mobile 6 SIP client with asterisk? |
10:09.36 | tzafrir_home | DarkDlx, basically two to look at: kiax and iaxcomm |
10:10.00 | tzafrir_home | DarKnesS_WolF, that is. |
10:15.05 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
10:16.34 | DarKnesS_WolF | tzafrir_home: hehe long time didn't see u too ;) |
10:16.49 | DarKnesS_WolF | tzafrir_home: i tryped zoiper but sometime the asterisk server loses teh DTMF |
10:17.02 | DarKnesS_WolF | kiax nahhh i'm not big fan of qt and kde in gernal will try iaxcomm it is ugly but works |
10:17.08 | tzafrir_home | Losing DTMFs? on IAX? |
10:17.21 | DarKnesS_WolF | tzafrir_home: yes |
10:17.26 | DarKnesS_WolF | i'm using a2billing |
10:17.30 | tzafrir_home | in what direction? |
10:17.38 | DarKnesS_WolF | so when i enter teh cardnumber sometime it loses |
10:17.54 | DarKnesS_WolF | zoiper ---> IAX ---> |
10:17.55 | DarKnesS_WolF | * |
10:18.11 | obnauticus | Can someone here tell me what is wrong with this: |
10:18.11 | obnauticus | exten => *5,1,Dial(sip/*5) |
10:18.18 | tzafrir_home | very strange. |
10:18.19 | obnauticus | (retorical question) |
10:18.37 | obnauticus | When I call *5 from an other line it's not ringing it. |
10:18.44 | obnauticus | an other device* |
10:19.18 | tzafrir_home | Try a more robust test: exten => _123.,1,SayDigits(${EXTEN:3}) |
10:19.52 | tzafrir_home | "123" is an arbitrary prefix to allow you to combine this in your dialplan |
10:20.12 | tzafrir_home | just dial a number and listen to what Asterisk thinks it has recieved |
10:20.42 | obnauticus | <--- SIP read from 10.0.0.110:51546 ---> |
10:20.42 | obnauticus | ACK sip:123*5@10.0.0.109 SIP/2.0 |
10:20.54 | obnauticus | It still doesn't ring SIP/*5 |
10:20.58 | obnauticus | but i can Origionate a call. |
10:21.03 | obnauticus | originate SIP/*2 application Dial sip/*5 |
10:21.07 | obnauticus | That works :| |
10:25.30 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
10:26.12 | obnauticus | K i got it tzafrir_home |
10:27.52 | tzafrir_home | obnauticus, I agree that using '*5' as a sip peer name is quite strange |
10:28.12 | obnauticus | I'm trying to orgonize peers with a * in their name :/ |
10:28.19 | obnauticus | or.... ya |
10:28.21 | obnauticus | you know what I mean |
10:28.28 | obnauticus | services with ##<serv_#>## |
10:28.32 | obnauticus | and etc. |
10:35.32 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
10:35.45 | *** join/#asterisk apardo (i=apardo@68.64.220.87.dynamic.jazztel.es) |
10:42.00 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
10:46.34 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
10:49.21 | *** join/#asterisk porche (n=porche@81.215.122.108) |
10:49.29 | porche | hi all |
10:49.51 | porche | can someone suggest me a good termination service? |
10:57.06 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-127-67.oc.oc.cox.net) |
10:57.28 | Shaun2222 | how's asterisk work on 64bit linux distros, any problems? |
10:58.36 | DarKnesS_WolF | Shaun2222: i don't think so i never did but i don't think there is should be a problem |
10:59.36 | tzafrir_home | Shaun2222, it is regularily used |
11:05.40 | *** join/#asterisk _feqma (n=paul@pool-72-65-41-105.bflony.east.verizon.net) |
11:07.36 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
11:14.42 | *** join/#asterisk Greek-Boy (n=Greek-Bo@41.221.58.2) |
11:17.22 | Siya | Anyone here with any hints why a Linksys ADSL router might prevent me from connecting to my * server via SIP? |
11:17.51 | Siya | Other places/routers/links with NAT work fine |
11:18.02 | Siya | So I'm suspecting the Linksys |
11:18.10 | Siya | or the ISP blocking SIP |
11:30.14 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
11:33.03 | *** join/#asterisk bintut (n=chatzill@cm246.gamma178.maxonline.com.sg) |
11:42.02 | *** join/#asterisk Bl0w_M0nk (n=ak@24-159-239-196.dhcp.mdsn.wi.charter.com) |
11:42.10 | *** part/#asterisk Bl0w_M0nk (n=ak@24-159-239-196.dhcp.mdsn.wi.charter.com) |
11:45.42 | *** join/#asterisk feqma (n=chatzill@pool-72-65-41-105.bflony.east.verizon.net) |
12:27.30 | *** join/#asterisk macTijn (i=martijn@rachel.insecure.nl) |
12:37.25 | *** part/#asterisk porche (n=porche@81.215.122.108) |
12:44.43 | feqma | \quit |
12:45.41 | CBU[^_^]M`` | hello |
12:45.44 | CBU[^_^]M`` | anyone here? |
13:02.33 | *** join/#asterisk MatBoy (n=Matt@wiljewelwetenhe.xs4all.nl) |
13:03.02 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
13:06.59 | ussrback | Hi all |
13:07.07 | ussrback | I have error [Nov 17 16:48:16] NOTICE[3580]: src/chan_h323.c:1885 reload_config: Unable to load config ooh323.conf, OOH323 disabled |
13:07.08 | ussrback | Loaded chan_ooh323 => (Objective Systems H323 Channel) |
13:07.20 | ussrback | how can i fix it |
13:07.27 | ussrback | i have installec chan_ooh323 |
13:14.57 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
13:23.29 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
13:26.45 | *** join/#asterisk waKKu (n=worth@unaffiliated/wakku) |
13:28.10 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
13:42.49 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
13:44.04 | *** join/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com) |
13:49.38 | rob0 | I'm back in the USSR ... you don't know how lucky you are, BOY. |
13:50.12 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
13:50.44 | grandpapadot | Hi all. Using Asterisk Static-Realtime in 1.2.x, can I have my extensions.conf in a db, then still call external files via #include? |
13:58.02 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
13:58.19 | *** part/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
14:10.46 | grandpapadot | Is there any way to set context specific "global" variables? |
14:11.42 | deeperror | CONTEXT_VAR? |
14:19.22 | *** join/#asterisk BiG^DoG (n=BiG^DoG@c-67-162-233-20.hsd1.de.comcast.net) |
14:20.05 | BiG^DoG | is there a list of good voip providers? I'm considering switching my analog phone number to voip |
14:20.25 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
14:20.37 | deeperror | BiG^DoG: I use VoicePulse |
14:23.11 | *** join/#asterisk Trionnis (i=lordkuri@s233-51-251.nap.wideopenwest.com) |
14:23.23 | BiG^DoG | satisfied with them? |
14:23.57 | deeperror | used them for a while |
14:24.03 | deeperror | not had many issues |
14:24.26 | deeperror | also ran 10 agents in a callcenter on them for a while |
14:24.40 | *** join/#asterisk disa-help (n=phobosd@shell.intarwebnetorg.com) |
14:24.51 | deeperror | also used callcentric not too bad |
14:25.06 | BiG^DoG | my big question is what happens if my asterisk box dies and I'm away for a day or two? |
14:25.22 | deeperror | haha |
14:25.29 | deeperror | your * box doesn't die |
14:25.38 | Trionnis | I would think the answer to that would be kinda obvious :) |
14:25.53 | BiG^DoG | I didn't know if they offered some kind of voicemail service |
14:25.59 | BiG^DoG | as an add on |
14:26.23 | deeperror | they might if you signup for regular service but not on connect |
14:26.38 | disa-help | anyone know much about this error msg? |
14:26.40 | disa-help | -- Executing NoOp("SIP/5553-08201b48", "Dial failed due to CONGESTION - failing through to other trunks") in new stack |
14:26.55 | disa-help | inbound calls are getting all circuits busy too -- but i'm not sure if it's an asterisk tag |
14:27.16 | deeperror | firewall? |
14:27.28 | disa-help | negative |
14:27.35 | deeperror | iptables? |
14:27.42 | disa-help | nope :-/ |
14:27.46 | disa-help | zaptel config the same |
14:27.51 | disa-help | PRI provider claiming it's us |
14:27.55 | disa-help | you familiar with asterisk tags? |
14:28.03 | disa-help | maybe you can call to check? hehe |
14:28.09 | disa-help | i honestly can't remember what they sound like.. |
14:28.23 | [TK]D-Fender | disa-help, means you're running a GUI that isn't supported here. |
14:28.40 | disa-help | GAH |
14:28.42 | disa-help | ALWAYS GETTIN ME |
14:28.43 | disa-help | lol |
14:28.46 | Trionnis | uh oh, now you have Andrew after you |
14:28.50 | BiG^DoG | deeperror: so I don't sign up for the $24.99 plan? I get the IAX termination, right? |
14:29.02 | disa-help | [TK]D-Fender: that, or it's somtehing on the provider side |
14:29.09 | deeperror | for asterisk yea |
14:29.15 | BiG^DoG | k |
14:29.18 | deeperror | they call it connect |
14:29.26 | deeperror | you can shop around though |
14:29.30 | deeperror | there might be a better deal out there |
14:29.47 | Trionnis | how many minutes are you talking about? |
14:29.49 | BiG^DoG | that's what I was looking for... I know if I'm looking for a webhosting company, I can go to a website that ranks the top hosting companies |
14:29.52 | Trionnis | rough guess |
14:29.54 | BiG^DoG | is there a simliar thing for IAX termination? |
14:30.04 | [TK]D-Fender | disa-help, that message doesn't actually mean ANYTHING. we don't see your dial, link status, or configs. |
14:30.39 | disa-help | true, but i find it hard to believe it's my pbx's problem if nothing has changed |
14:30.50 | Trionnis | how many minutes a month? |
14:31.05 | BiG^DoG | inbound or outbound? |
14:31.10 | Trionnis | either or both |
14:31.12 | BiG^DoG | it's a residential thing so typical "home use" |
14:31.15 | Trionnis | ah |
14:31.16 | BiG^DoG | I haven't actually measured |
14:31.17 | [TK]D-Fender | disa-help, pastebin the full CLI output of your failed call at verbose 10 and do "pri debug" first so we can see whats going on |
14:31.17 | Trionnis | n/m |
14:31.28 | Trionnis | I thought it was for a call center... must have misread |
14:31.49 | deeperror | i was just stating that i have ran some cc activity thru VP with no issues |
14:32.01 | Trionnis | you can get really good pricing from voipjet if you push enough minutes through them :) |
14:32.37 | deeperror | http://www.digium.com/en/ecosystem/partners/partners.php |
14:34.03 | deeperror | my old man put 100 bucks on a pre pay account...told him he will never use that much on voip that would be like 5 years of calls haha |
14:34.48 | BiG^DoG | so $50 for startup would last a while? |
14:35.03 | deeperror | if you get did will cost 11/month |
14:35.09 | deeperror | he just has it for outbound calls |
14:35.42 | deeperror | but you would probably want did |
14:35.52 | deeperror | could even port your number to them |
14:36.27 | BiG^DoG | and is there a per minute charge on DID? |
14:36.50 | deeperror | did per month per minute outbound |
14:37.51 | Trionnis | some places do per minute, some do flat rate |
14:37.55 | Trionnis | most are flat rate |
14:38.17 | BiG^DoG | per minute outbound and there's no local area? every outbound call has a per minute? |
14:38.37 | Trionnis | most I've found are like that |
14:38.54 | deeperror | depends on how many minutes you run. if your not using the flat rate worth of minutes better to not pay if your making a lot of calls flat rate |
14:40.05 | BiG^DoG | got it... if I'm making more than $25 worth of outbound a month, go with the flat rate... Otherwise, stick with per minute |
14:40.42 | Trionnis | $25/mo if calls...... wow... |
14:40.54 | Trionnis | I don't think I've ever seen one of our phone bills that cheap |
14:40.55 | Trionnis | ;) |
14:41.13 | BiG^DoG | is $25 a month a lot? |
14:41.27 | BiG^DoG | remember, I'm talking residential, not commercial |
14:41.33 | Trionnis | I know |
14:41.54 | Trionnis | depending on the per minute, that can be a good amount of calls |
14:42.06 | Trionnis | at home one of the DID providers I use is sipmedia |
14:42.07 | [TK]D-Fender | other calculation is that if you need multiple simultaneous channels, the per-min option often lets you have more than 2 calls unlike "residential" style fixed channel services |
14:42.17 | Trionnis | $5/mo for a DID with 500 minutes outbound |
14:42.29 | Trionnis | I've never gone over |
14:42.55 | Trionnis | my bad, the 500 outbound is $9.95/mo, the $5 plan is with 100 outbound |
14:42.58 | BiG^DoG | I was just looking over the voicepulse website and it says they don't offer callerid name on did... is that fairly common? |
14:43.10 | Trionnis | most voip providers don't offer that |
14:43.25 | Trionnis | they don't have the CNAM equipment to do the lookups |
14:43.34 | deeperror | I setup a script and database and make custom names for callers |
14:43.39 | Trionnis | that works too |
14:43.42 | BiG^DoG | gotcha |
14:43.46 | BiG^DoG | that was what I was going to say |
14:43.56 | BiG^DoG | * can take the cid and do a database lookup and insert it |
14:44.20 | deeperror | even wrote a windows service that will pop open web browser to edit that db info on inbound calls |
14:44.46 | Trionnis | wow |
14:44.50 | deeperror | buttons for callback, edit info |
14:44.56 | Trionnis | that's a lot more effort than I'd put into it ;) |
14:45.15 | BiG^DoG | thanks for the good information... the dead end last night on call waiting put the final nail in my analog coffin I think |
14:45.19 | deeperror | well i play around to learn new things and sometimes toys evolve haha |
14:46.02 | Trionnis | speaking of playing around... anyone have about 200-250 channels sitting on the PSTN they want to let me tie up for about 30 minutes? ;) |
14:46.11 | Trionnis | need to do some capacity testing :) |
14:51.35 | *** join/#asterisk cypherdelic (n=cypher@p5B27EA05.dip.t-dialin.net) |
14:54.30 | *** join/#asterisk UserReg_CL (n=dede@200.113.130.111) |
14:54.36 | UserReg_CL | hi, good day !!! |
14:54.55 | UserReg_CL | hi TK |
14:56.28 | UserReg_CL | one question... where is defined Trunks ? |
14:59.46 | BiG^DoG | is broadvoice any good? |
15:00.11 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
15:02.20 | Greek-Boy | who has tried Five 9's? |
15:07.39 | *** join/#asterisk irule (n=irule@200.53.61.4) |
15:11.51 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
15:13.59 | *** join/#asterisk mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
15:14.28 | mackes | Good Morning everyone? |
15:15.30 | brookshire | hi |
15:15.50 | mackes | Hey |
15:19.04 | *** part/#asterisk IOscanner (n=IOscanne@cpe-76-187-195-124.tx.res.rr.com) |
15:22.34 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
15:26.12 | disa-help | fixed |
15:26.17 | disa-help | had to unplug/replug the damn thing ;p |
15:26.25 | disa-help | zaptel cards are pissy sometimes i guess |
15:26.25 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
15:26.25 | *** mode/#asterisk [+o russellb_] by ChanServ |
15:32.26 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
15:39.48 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
15:40.05 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:40.10 | *** mode/#asterisk [+o blitzrage] by ChanServ |
15:41.21 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
15:53.58 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
15:55.10 | jameswf-home | jbot: you suck |
15:55.11 | jbot | and very well I might add |
15:55.26 | Greek-Boy | jbot: stuff you |
15:55.27 | jbot | ACTION grab's you, fills them up with stuffing, and sticks them in the oven |
15:55.45 | Greek-Boy | jbot: u not clever |
15:55.48 | jameswf-home | jbot: drob table; |
15:56.03 | Greek-Boy | jbot: you idiot |
15:56.04 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
15:56.25 | unixdog | what is asterisk and how will it better my life ? |
15:56.26 | jameswf-home | jbot: dropdatabase; |
15:56.26 | jbot | So you think you could kill me, did you. You thought you could come in here, in my own home, and take me down without a fight. I would have given you a chance, really. But now, you have unleashed hell. May god have mercy on your soul. |
15:56.31 | unixdog | how will it make me rich |
15:56.32 | jameswf-home | lol |
15:56.57 | coppice | unixdog: its not about wealth. its about sex |
15:56.58 | jameswf-home | it wont if you wanna get rich go to M$ |
15:57.35 | riddlebox | lol |
15:58.13 | jameswf-home | I get Laid every other friday thankls to asterisk, but seriously its more likely the paycheck that makes my wife give it up |
15:58.52 | riddlebox | lol, jameswf-home so you sell and install asterisk servers? |
15:59.03 | Greek-Boy | yeah women just love money! |
15:59.20 | jameswf-home | no I develop and support for an asterisk hardware mfg |
15:59.36 | Greek-Boy | jameswf: which one? |
15:59.42 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
15:59.43 | jameswf-home | not in china not sangoma |
15:59.56 | riddlebox | jameswf-home, cool, I wish I could work on linux/asterisk full time |
16:00.13 | Greek-Boy | u lucky man |
16:00.15 | jameswf-home | move to az i will try to get you a job lol |
16:01.27 | riddlebox | jameswf-home, I did my Avaya IP Office training in AZ, I would love to live there |
16:01.56 | Greek-Boy | riddlebox: isn't it too hot? where do you live now? |
16:02.04 | riddlebox | Greek-Boy, Illinois |
16:02.10 | jameswf-home | In our office you can use windows but people will make fun of you so most have some flavor of linux or mac |
16:02.56 | riddlebox | I would love an environment like that, I always complain how we work on Avaya systems like Communication Manager which runs redhat but the admin software in windows only |
16:03.27 | jameswf-home | When I started in telephony I installed toshibas... it was ass |
16:04.23 | riddlebox | I work on a Toshiba Perception e --it was installed in 1983, I was 3 haha |
16:04.33 | unixdog | MS is for loosers |
16:04.42 | unixdog | who have to real kill sets |
16:05.15 | unixdog | who are appliance users |
16:05.33 | unixdog | and I was making a bad joke |
16:05.41 | unixdog | < ==== formerly darwin 35 |
16:05.51 | jameswf-home | well all money hungry r tards should use MS as they have the same goals |
16:05.56 | unixdog | <==== he who ported asterisk to freebsd |
16:06.25 | jameswf-home | didnt you tell me your not a programmer unixdog |
16:06.35 | unixdog | MS has one goal make users stupid so they have to pay more money for support |
16:06.55 | jameswf-home | *cough Tricboc *cough |
16:07.01 | unixdog | I am not but back whan I ported it . it was not hard |
16:07.15 | riddlebox | lol |
16:07.15 | unixdog | 9trashbox |
16:08.11 | jameswf-home | jbot kerry |
16:08.12 | jbot | it has been said that kerry is something you eat |
16:08.42 | coppice | jbot pussy |
16:08.43 | jbot | Read: coppice |
16:08.58 | riddlebox | haha |
16:08.59 | jameswf-home | jbot coppice |
16:09.00 | jbot | well, coppice is rather like underwood, only different, or the faxing master |
16:09.13 | riddlebox | jbot riddlebox |
16:09.23 | jameswf-home | jbot faxing |
16:09.23 | jbot | well, faxing is 8% knowledge, 5% skill, 11% luck, and 76% voodoo |
16:09.29 | riddlebox | dang |
16:09.43 | coppice | jbot voodoo |
16:09.44 | jbot | [voodoo] black magic, stay away from it. It is also a chipset, ask me about 3dfx. |
16:09.54 | Qwell | ~3dfx |
16:09.54 | jbot | 3dfx is probably the name of the company who makes the legendary voodoo 3D acceleration cards (see http://www.3dfx.com). There is also /dev/3dfx which is needed for glide support under X <=3.3.6 (see device3dfx-source package). It is not needed for X4, use tdfx instead. now owned by nVida.... time to start buying Matrox :), or at #3dfx |
16:09.54 | coppice | jbot T.38 |
16:09.55 | jbot | hmm... t.38 is see t38 |
16:10.05 | jameswf-home | jbot sex |
16:10.06 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
16:10.07 | riddlebox | is there any sip phones that will allow you to just start pressing buttons to dial without hitting speaker or picking up the handset? |
16:10.11 | coppice | jbot t38 |
16:10.11 | jbot | it has been said that t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
16:10.58 | coppice | jbot foip |
16:10.59 | jbot | from memory, foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject |
16:11.53 | jameswf-home | jbot: you |
16:11.54 | jbot | jbot is probably a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
16:12.20 | coppice | jbot [TK]D-Fender |
16:12.20 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
16:13.04 | jameswf-home | jbot: wtf sex |
16:13.14 | Qwell | ~botabuse |
16:13.14 | jbot | i guess botabuse is fun |
16:13.24 | coppice | jbot virgin |
16:13.24 | jbot | I'm sexless |
16:13.29 | jameswf-home | jbot: botsnack |
16:13.29 | jbot | aw, gee, jameswf-home |
16:13.31 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
16:14.33 | coppice | 100 page FAX tests are boriiiiinnnngggg zzzzzzzzz....... |
16:15.16 | jameswf-home | try 200 |
16:15.40 | coppice | nah. I'm using 100..... but over and over all day |
16:16.01 | coppice | and a 2M byte single FAX page |
16:16.26 | Greek-Boy | anyone here use astpp? |
16:16.46 | riddlebox | coppice, you are testing faxing over IP? |
16:16.56 | coppice | yes |
16:17.32 | riddlebox | through a provider? or just through asterisk? |
16:17.32 | Greek-Boy | coppice: Have you tested T.37 and if yes which solution/package do you recommend? |
16:19.22 | coppice | I don't think anyone has done a precise implementation of T.37 for use with free faxing - e.g. hylafax, spandsp, etc - but there are several email-to-fax solutions that come pretty close |
16:19.54 | Greek-Boy | well |
16:19.57 | tzanger | coppice: sounds like what I do to listen to radio stations on satellite |
16:20.05 | Greek-Boy | I just want to be able to send faxed and receive them from asterisk via e-mail |
16:20.15 | tzanger | the receiver wants a video signal so I have to fake one by sending the same video frame over and over and precise intervals |
16:20.42 | coppice | don't let Catch Curve catch you doing that :-) |
16:20.54 | tzanger | catch curve? |
16:21.04 | coppice | J2 by another name |
16:21.10 | jameswf-home | jbot: poo |
16:21.10 | jbot | poo is smelly |
16:21.26 | coppice | jbot J2 |
16:21.55 | coppice | he doesn't know any of the interesting stuff |
16:22.03 | jameswf-home | jbot: f |
16:22.03 | jbot | ACTION gives jameswf-home a big [kiss,hug] |
16:22.10 | jameswf-home | lol |
16:22.33 | riddlebox | is there a way to convert the voicemail email to ogg or something else besides wav? before it is emailed? |
16:23.52 | coppice | jbot J2 is the Devil's appointment reaper of FAX users. |
16:23.53 | jbot | coppice: okay |
16:25.28 | jameswf-home | jbot: j2 |
16:25.28 | jbot | [j2] j squared |
16:25.44 | coppice | jbot J2 |
16:25.45 | jbot | j2 is probably j squared |
16:25.59 | coppice | jbot J2 is the Devil's appointment reaper of FAX users. |
16:25.59 | jbot | ...but j2 is already something else... |
16:26.17 | coppice | jbot J2 is also the Devil's appointment reaper of FAX users. |
16:26.17 | jbot | coppice: okay |
16:26.37 | coppice | jbot J2 is also -1 |
16:26.38 | jbot | coppice: okay |
16:26.47 | coppice | jbot J2 |
16:26.48 | jbot | i heard j2 is -1 |
16:26.57 | endre | i heard u like mudkips |
16:27.01 | coppice | jbot J2 |
16:27.12 | coppice | jbot J2 |
16:27.26 | endre | jobd raid |
16:27.29 | endre | jbod |
16:27.35 | coppice | now he's sulking |
16:27.45 | jameswf-home | <PROTECTED> |
16:27.46 | jbot | jameswf-home: okay |
16:27.50 | jameswf-home | jbot: j2 |
16:27.51 | jbot | i guess j2 is the Devil's appointment reaper of FAX users. |
16:29.01 | coppice | jbot Catch Curve is see leech |
16:29.01 | jbot | ACTION lures Curve is see leech into the crab motel -- Curve is see leech checks in, but won't be checking out... |
16:29.36 | jameswf-home | jbot: wiki sangoma |
16:30.19 | jameswf-home | jbot: wiki bamf |
16:30.32 | coppice | jbot wiki j2 |
16:30.50 | jameswf-home | jbot: bamf |
16:30.51 | jbot | extra, extra, read all about it, bamf is to disappear with a poof. |
16:30.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
16:31.20 | coppice | jbot wiki catch curve |
16:31.31 | jameswf-home | jbot: bamf is also bad a$$ mother fu**er |
16:31.31 | jbot | okay, jameswf-home |
16:31.32 | De_Mon | what is th purpose of the Verbose app? |
16:31.47 | tzanger | De_Mon: to display information only if verbose level is > some value |
16:31.57 | riddlebox | I guess I may as well upgrade my system this morning |
16:32.00 | tzanger | De_Mon: think of a conditional noop |
16:32.06 | De_Mon | yeah, but on the CLI I see the message regardless, do I have something else turned on thats show that to me? |
16:32.29 | jameswf-home | adjust logger.conf |
16:32.33 | *** join/#asterisk mamep (i=fallen@helios.edu.uoc.gr) |
16:32.42 | De_Mon | does the verbose stuff goto a log file somewhere that the execution of the priority doesn't go to |
16:32.53 | coppice | 100 pages in a FAX TIFF file |
16:32.54 | coppice | Yo ho ho, and a flavour of rum |
16:32.59 | mamep | hello, i'm trying to connect to cisco callmanager using ooh323 but i can't find a guide how to do it..anyone can help? |
16:33.18 | De_Mon | jameswf-home is it debug thats show the priority execution, or...? |
16:33.21 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
16:33.22 | blitzrage | mamep: you'll have to ask specific questions |
16:33.48 | mamep | blitzrage : i need a little bit assistance with ooh323 |
16:34.02 | mamep | is it possible to route my calls through cisco's callmanager? |
16:34.16 | blitzrage | mamep: right -- but you should ask specific questions with examples of what is not working -- if you need someone to guide you through the whole install, that is more of a job for a consultant |
16:34.32 | *** join/#asterisk kotyagin (n=knkbox@ppp85-140-239-38.pppoe.mtu-net.ru) |
16:34.36 | mamep | blitzrage : ok let me ask then.. |
16:34.53 | mamep | first of all ooh323 module is loaded automatically in asterisk? |
16:36.05 | disa-help | hey guys.. |
16:36.07 | disa-help | show queues |
16:36.08 | disa-help | (dynamic) (In use) |
16:36.11 | disa-help | for an extension |
16:36.15 | disa-help | what does 'in use' mean? |
16:36.35 | disa-help | i'm assuming it means that the extension is busy? |
16:37.02 | kotyagin | Hi, all !!! Is there any programmers who able to discuss about VAD/CNG ? |
16:38.01 | tzafrir_home | jbot, tell kotyagin about ask |
16:43.04 | nestAr | jbot is a hata |
16:43.09 | mamep | how can i add ooh323 peer with username and pass? |
16:43.12 | kotyagin | Ok. I'll be more exact... Is asterisk really generates CNG frames with ast_rtp_sendcng function from rtp.c ??? |
16:43.14 | nestAr | :) |
16:48.52 | mamep | how can i add ooh323 peer with username and pass?? |
16:50.20 | jameswf-home | <butthead> hu hu he said ooh323 peer hu hu </butthead> |
16:53.25 | jameswf-home | jbot: ping |
16:53.25 | jbot | pong |
16:54.08 | jameswf-home | jbot: pong |
16:54.09 | jbot | PING! |
16:54.20 | jameswf-home | jbot: porn |
16:54.21 | jbot | Porn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type. |
16:57.19 | riddlebox | lol |
16:57.59 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
17:00.26 | endre | lol |
17:06.09 | riddlebox | when I make a call out of my TDM card, it takes like 5 seconds or so before the call starts to ring |
17:06.23 | riddlebox | is there a way to speed that up? |
17:08.03 | [TK]D-Fender | riddlebox, Not really. Don't forget, its pulling the line , dialing, and then waiting as well. in effect you waited in dialing a number to #, and wait again as it send it out your analog line |
17:08.23 | endre | riddlebox: switch off echocancel training |
17:08.32 | tzafrir_home | riddlebox, callerid settings, I guess |
17:08.46 | endre | tzafrir_home: he said calling OUT |
17:09.11 | tzafrir_home | dialplan, I guess |
17:09.23 | tzafrir_home | echo training is not something visible to Asterisk |
17:09.34 | tzafrir_home | and not 5 seconds |
17:10.57 | mamep | WARNING[4451]: channel.c:3393 ast_channel_make_compatible: No path to translate from OOH323/ucnet-99f8(256) to SIP/51030-b6f01c20(4) |
17:11.48 | mamep | what's this? |
17:13.41 | DarkRift | Is there a macro variable that returns the caller extension ? |
17:14.01 | DarkRift | Or somewhere I can find help on the available macro's variable |
17:15.50 | De_Mon | DarkRift your question relveals much about your lack of knowledge, seek the book and learn from it |
17:15.58 | De_Mon | ~book |
17:15.58 | jbot | 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 |
17:16.18 | DarkRift | thx |
17:16.32 | *** join/#asterisk cesar_CR (n=cesar@201.195.35.62) |
17:17.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
17:17.58 | [TK]D-Fender | mamep, means you don't have any free G.729 licences and * can't transcode |
17:18.11 | mamep | yeah changed to gsm now |
17:18.15 | mamep | but having some other problems |
17:18.55 | mamep | http://pastebin.ca/778505 |
17:19.32 | jameswf-home | someone should work on adding aix to gaim (pidgin) |
17:20.00 | PaulAviles | anyone using 79xx phones? |
17:20.30 | PaulAviles | how can you customize the buttons on the bottom? |
17:20.51 | jameswf-home | a/aix/iax |
17:24.31 | *** join/#asterisk snazm (n=snazm@89.243.184.171) |
17:25.14 | snazm | Hi folks |
17:25.37 | [TK]D-Fender | mamep, looks like all your sound files are missing or can't be accessed due to auth issues |
17:27.44 | PaulAviles | all the files may be missing |
17:27.59 | PaulAviles | are you ussing h323? if not don't load the driver |
17:28.20 | snazm | I'm new to VoIP but have been consuming information on Asterisk almost solid for the last few weeks (and wow, what a system!) My boss has asked me to look at getting VoIP for our office (about 80 seats) and quotes from some companies are ridiculously high so thought I'd look at doing it myself, and I have a few questions the howto's can't answer :) |
17:28.38 | PaulAviles | shoot.. |
17:30.23 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
17:30.30 | snazm | I think I can do it myself, but there's a few consideratios. Firstly all the seats are going to get thin clients, and I'm wondering the best way to integrate VoIP with this. Is it better to have PoE telephones using the CAT5e cabling and run it as a separate system, ose use USB VoIP phones from the thin clients themselves, or simply having a headset and using the thin clients audio in/out? |
17:30.31 | jameswf-home | the cost of asterisk no matter how high usualy kicks the crap out of normal PBX's |
17:31.00 | jameswf-home | depends on the thinclient |
17:31.13 | snazm | Linux based |
17:31.16 | PaulAviles | yes, having a separe network for voip is always better and if need poe and can afford it even better |
17:31.40 | [TK]D-Fender | PoE costs very little depending how you do it. |
17:31.50 | [TK]D-Fender | snazm, Where are you located? |
17:32.00 | PaulAviles | a phone IMHO is better than any pc emulation |
17:32.08 | jameswf-home | the term "giant collision domain" gets me excited |
17:32.17 | mamep | how can add ooh323 peer with user and pass? |
17:32.24 | snazm | [TK]D-Fender: UK |
17:32.24 | [TK]D-Fender | And soft-phone + usb phone = not much cheaper and works like shit |
17:33.12 | [TK]D-Fender | snazm, Theneconomically speaking, look at linksys SPA-922/942's on Linksys/D-Link PoE switches. |
17:33.17 | [TK]D-Fender | snazm, rather affordable |
17:33.20 | PaulAviles | some soft phones are kinda nice, portsip is free and still has g729 support natively |
17:33.44 | snazm | The benefit of having the thin client based phone (whether through direct audio or a USB deskphone) is that the users extention can follow their hotdesking |
17:33.50 | snazm | (should have mentioned hotdesking) |
17:33.54 | jameswf-home | I know a school that uses linksys (due toi cost) they seem happy |
17:34.21 | PaulAviles | or you can have a phone and log in /log out and have the same |
17:34.25 | snazm | The benefit of having a separate PoE phone system is it will work even if the power goes out, with a single UPS |
17:34.42 | PaulAviles | with a BIG.. ups.. |
17:34.52 | snazm | PaulAviles: You mean login to the thin client AND the VoIP phone? |
17:35.01 | jameswf-home | what can brown do for you lol |
17:35.05 | snazm | PaulAviles: I was thinking if they were integrated, it could be automagic |
17:35.22 | PaulAviles | no.. you can have say a 7940 and have multiple profiles depending on who is sitting using the phone |
17:35.48 | jameswf-home | could also use agents |
17:36.32 | snazm | A 7940? |
17:37.02 | jameswf-home | agents happen in asterisk reguardless of the channels type |
17:37.05 | PaulAviles | 7940 or 7960 |
17:38.23 | snazm | OK, so you're suggesting that I can configure the thin client to communicate with Asterisk to tell it to route numbers to the port matching the thin clients? |
17:38.29 | snazm | Or am I getting a bit lost here? |
17:38.39 | Qwell | You guys ever seen transcoding to/from gsm just sound horrid? |
17:38.51 | Qwell | like...just...terrible |
17:39.19 | PaulAviles | mamep, have you tried register=userID:pass@host |
17:39.37 | PaulAviles | qwell, not really good quality |
17:39.54 | Qwell | I mean, gsm doesn't sound great to begin with - but this is worse than lpc10 |
17:40.06 | PaulAviles | I mean, good quality.. sorry |
17:40.19 | [TK]D-Fender | Qwell, Whats on each end? |
17:40.21 | PaulAviles | i did some testing and had no issues |
17:40.36 | Qwell | calling in on a PRI, playing a gsm prompt with Playback |
17:40.53 | PaulAviles | can you emulate the same internal with 7777? |
17:40.58 | Qwell | switching to ulaw or wav prompts sound great |
17:40.59 | PaulAviles | with the same results? |
17:41.05 | [TK]D-Fender | Qwell, EEK. Ok, I recall only 1 version of * where something went horribly wrong with the GSM codec, but thats it |
17:41.13 | PaulAviles | do you have the proper files for gsm? |
17:41.22 | Qwell | PaulAviles: it builds gsm |
17:41.54 | Qwell | it sounded like the gsm transcode is...horribly broken |
17:42.28 | Qwell | imagine the gain being set to around 50, and trying to play a file |
17:42.33 | PaulAviles | so under /var/lib/asterisk/sounds/digits you have all the .gsm files too / |
17:42.34 | Qwell | that's about how it sounds |
17:42.34 | PaulAviles | ? |
17:43.10 | unixdog | what ver of gasterisk |
17:43.18 | Qwell | 1.4.13 |
17:43.26 | unixdog | hmmm |
17:43.36 | unixdog | been runing it with no problem |
17:43.42 | [TK]D-Fender | Qwell, how does it sound on G.711 outside your PRI? |
17:43.47 | unixdog | and I have it on bsd |
17:44.02 | Qwell | [TK]D-Fender: they tested with softphones, and it was just as bad, I think |
17:44.13 | Qwell | everything else on the PRI is fine - it's just gsm |
17:44.13 | [TK]D-Fender | Qwell, ouch |
17:44.27 | unixdog | try recompileing |
17:44.41 | Qwell | did |
17:44.45 | PaulAviles | can you create an external extension to test? |
17:44.51 | [TK]D-Fender | Qwell, if it sucks on hardphones as well I'd say look at codecs.conf next |
17:45.05 | Qwell | codecs.conf? O.o |
17:45.19 | Qwell | ahh, nothing in there for gsm |
17:45.21 | [TK]D-Fender | Qwell, And then check your base sound files... maybe someone globall sox-d them to death |
17:45.29 | Qwell | I didn't know that conf existed :P |
17:45.41 | unixdog | what type of system is this on |
17:45.45 | Qwell | yeah, I reinstalled the prompts. recorded prompts sound like ass too |
17:45.49 | Qwell | xeon |
17:46.08 | Qwell | no k6opts |
17:46.12 | unixdog | what flavor of *nix |
17:46.18 | Qwell | linux, of course |
17:46.24 | unixdog | what flavor |
17:46.28 | [TK]D-Fender | SOUR :p |
17:46.29 | Corydon76-dig | Qwell: what version of gcc? |
17:46.36 | Qwell | 4.2.3 |
17:46.40 | unixdog | cent/suse |
17:46.45 | Corydon76-dig | Downgrade to gcc 4.1 |
17:46.47 | DarkRift | How can I permanently associate a mailbox witha SIP user, I mean when I call the VoicemailMain I want it to by default authenticate to his associated voicemail box, is that possible in a simple macro, or I need to build a SQL Database to make that possible ? |
17:46.47 | Qwell | well...prerelease |
17:47.05 | Corydon76-dig | There's a performance regression in gcc 4.2 |
17:47.13 | DarkRift | The voicemail box being a number, but the caller is a SIP user, which is not a number itself |
17:47.18 | Qwell | Corydon76-dig: oh? |
17:47.39 | Corydon76-dig | Qwell: trust me |
17:47.48 | Qwell | and that could make gsm sound like that? |
17:47.59 | Corydon76-dig | Recompile with gcc 4.1 and all will be well. It's not just gsm |
17:47.59 | [TK]D-Fender | DarkRift, use SetVar in your sip.conf entry for use by Voicemailmain to know which box to use |
17:48.04 | Qwell | interesting |
17:48.59 | Qwell | poking him now |
17:49.08 | DarkRift | Well, I'd need to use setvar for each users ? Let's say when you call 100 it calls me (darkrift), if darkrift call voicemailmain I want it to call the voicemail 100 automatically, using a setvar in each users entry would do that ? |
17:49.27 | Qwell | Corydon76-dig: what else have you seen this affect? |
17:49.46 | DarkRift | Got an example of what you mean ? |
17:49.51 | Corydon76-dig | Qwell: Asterisk is the only app I use where the performance is critical |
17:50.00 | Qwell | I mean, besides gsm |
17:50.31 | Corydon76-dig | I think someone said alaw sucked, as well |
17:50.40 | Qwell | didn't try alaw - wav was alright though |
17:50.55 | Qwell | so, basically, transcoding in general |
17:51.17 | Qwell | is there a way to tell what version of gcc a module was built with? |
17:51.35 | Corydon76-dig | strings, maybe? |
17:51.59 | Corydon76-dig | Nope, that didn't work |
17:54.43 | Qwell | oh well - I think that answers another problem I heard the other day too |
17:58.55 | DarkRift | Alright ! Thanks [TK]D-Fender |
17:59.51 | [TK]D-Fender | DarkRift, You're welcome |
17:59.53 | [TK]D-Fender | BBIAB |
18:02.07 | *** join/#asterisk alephcom (n=chatzill@h66-112-187-16.mcsnet.ca) |
18:10.40 | *** join/#asterisk selsky_ (n=selsky@12.sub-70-216-142.myvzw.com) |
18:12.08 | *** join/#asterisk asdx (n=diego@adsl-149-212.click.com.py) |
18:12.10 | asdx | hi |
18:12.47 | asdx | what is register => for |
18:13.46 | mvanbaak | in sip.conf ? |
18:13.56 | asdx | iax.conf |
18:14.13 | mvanbaak | to make your asterisk register to an IAX2 machine |
18:14.26 | mvanbaak | the other side has: host = dynamic |
18:14.35 | asdx | is that the same in sip.conf too? |
18:14.40 | mvanbaak | yup |
18:14.54 | asdx | oh i see |
18:15.03 | asdx | so if i set host=dynamic in my [user] |
18:15.17 | mvanbaak | the other box will need a 'register = |
18:15.19 | asdx | i will be able to connect to that user from my local client? |
18:15.55 | mvanbaak | uhhuh |
18:16.35 | *** join/#asterisk Nukemizer (n=Nukemize@15.249.sfcn.org) |
18:16.59 | mackes | what irc clients are you all using? |
18:17.02 | asdx | irssi |
18:17.21 | jameswf-home | Konversatioin and Xchat |
18:17.30 | DarkRift | mIRC |
18:17.50 | bobkare | irssi, runs in screen |
18:18.20 | mackes | neat |
18:18.24 | mackes | Mirc |
18:18.32 | asdx | the host dynamic confuses me a bit |
18:18.32 | mackes | Windows based? |
18:18.54 | jameswf-home | screem is overrated. people want to see what I am doing like it will help them understand... |
18:18.56 | DarkRift | mIRC is windows yeah |
18:19.01 | jameswf-home | *screen |
18:19.12 | mvanbaak | mackes: irssi |
18:19.33 | jameswf-home | I think xchat can be used in windoze not sure |
18:19.44 | snazm | Pidgin rocks |
18:19.51 | snazm | Works in windows too |
18:19.52 | mvanbaak | irssi can be used in winblows too |
18:19.53 | mackes | Thats two votes for irssi.. How did you ding me mvanbaak |
18:20.03 | asdx | it is the same if i use register => foo:bar@foo.org and if i use [user] host=foo.org username=foo secret=bar ? |
18:20.21 | mvanbaak | mackes: simply by putting mackes: as the first word in my line |
18:20.23 | mackes | Cool |
18:20.26 | jameswf-home | i need to get my oil changed |
18:20.33 | mackes | mvanbaak: Neat |
18:20.35 | mvanbaak | I need more beer |
18:20.39 | bobkare | jameswf-home: are you thinking about screen with multiple connected clients or something? |
18:21.02 | jameswf-home | linux program screen,, crap |
18:21.17 | jameswf-home | well not crap just useless |
18:21.40 | mackes | digium has helped me using screen |
18:21.48 | mackes | its the only time I have needed it |
18:21.56 | jameswf-home | and did you learn anything? |
18:22.21 | mackes | Hmmmm I little because I was able to ask questions while they worked |
18:22.22 | bobkare | it's genius. lets me have programs running that I can get at no matter what computer i'm sitting at |
18:22.28 | mackes | and I could see what was happing |
18:22.45 | mvanbaak | jameswf-home: screen is convenient |
18:22.50 | mackes | there is a VNC for terminal sessions now as well |
18:23.01 | jameswf-home | well typicaly if you can learn from a screen session the tech is not fast enough |
18:23.45 | mackes | hahah |
18:23.51 | mackes | I ask they slow down |
18:24.15 | mackes | If someone is in my corp network, I want them to tell me exactly what they are doing |
18:24.29 | jameswf-home | Ususaly its like i will fix it you go buy a book I will leave comments |
18:24.30 | mackes | They could be installing a rootkit |
18:24.43 | mackes | yeah |
18:24.55 | mackes | Tech support is hard |
18:25.00 | mackes | I feel for those guys |
18:25.13 | jameswf-home | well digium and us are not fonality we dont need rootkits. lol |
18:25.46 | mackes | I have only had trouble first setting up a 4 port PRI |
18:25.56 | mackes | But once I had it, I was set |
18:26.10 | mackes | so I was very happy to have the free support |
18:26.58 | jameswf-home | I was readintg the tb terms of support, they require you to install software and keep it up and if you shut it off your out of luck and $$$ verry windowsish |
18:27.25 | rob0 | But what's useless about screen? For me it's essential. |
18:27.59 | mvanbaak | I think jameswf-home just likes to have tons of xterms open |
18:28.05 | mackes | Which software is that? |
18:28.15 | jameswf-home | good 4 you, some people cant live without webmin doesnt mean it belongs |
18:28.24 | *** join/#asterisk Maan (n=Maan@155.48.255.23) |
18:28.25 | mackes | Hey.. I do have a suggestion for Digium |
18:28.36 | mackes | We use and love RPath Poundkey |
18:28.51 | mackes | But I dont want the Web Interface in the newer versions |
18:28.57 | asdx | can some of you please take a look at this: http://pastebin.com/m57964d71 <-- does this should be enough for making a call? |
18:29.06 | asdx | i hear tone when i dial test@default |
18:29.12 | asdx | but my phone doesn't ring |
18:29.32 | mackes | It would be great if they released Poundkey without the Web Interface but has the updates and 1.4 |
18:30.19 | jameswf-home | people who cant administer a system without a gui should stick to M$ |
18:30.25 | rob0 | Comparing screen and Webmin ... wow. |
18:30.27 | snazm | In Asterisk, is it possible to have it redirect a call if the end point goes dead without clearing the call down properly? |
18:30.45 | jameswf-home | sadly it is faster to admin a windows system through the cli as well |
18:30.54 | mackes | Yep |
18:31.18 | bobkare | admining windows without touching the command-line isn't possible |
18:31.24 | mackes | And the Web Interfaces really limit the admins to predefined setups |
18:31.32 | jameswf-home | screen and webmin are both useless so comparisons can be made |
18:31.47 | mackes | Webmin is a Very Good tool. |
18:31.58 | asdx | anyone? |
18:32.08 | mackes | If you have 20 Linux boxes ... It helps |
18:32.34 | jameswf-home | webmin is good for a jr admin fresh out of college, but he shouldnt be making system changes so nix that |
18:32.40 | bobkare | jikes, I haven't seen anything break configurations so completly as webmin since that horrid config util for redhat 6.something |
18:32.48 | mackes | It would really help this group if no one spoke in absolutes |
18:33.12 | mackes | man.. those are short sighted comments |
18:33.32 | tzafrir_home | asdx, you need to have a secret / md5secret in the [voipjet entry as well. I suppose you have not pasted your real md5secret there |
18:34.11 | jameswf-home | you know what webmin does is it lowers wages by giving people who should be flipping burgers the power to do rermedial tasks... |
18:34.12 | mackes | James, you must be an absolute genius. You can setup every server in linux direct from the command line from memory! |
18:34.19 | asdx | tzafrir_home: that account is free ;-) |
18:35.05 | jameswf-home | I can complete most tasks from memory, and if not there are these neat things called books |
18:35.08 | tzafrir_home | mackes, there's another aspect: the ammount of data you need to pass over the phone to support |
18:35.27 | mackes | James- I would love to know more about your background, where did you cut your teeth? |
18:35.38 | jameswf-home | people these days are too afraid to RTFM |
18:35.45 | tzafrir_home | A phone line is a communication channel with a high rate of errors |
18:36.10 | jameswf-home | mackes: your question is to vague |
18:36.11 | bobkare | if you want a really cool utility for administering many *nix boxes have a look at cssh |
18:36.12 | tzafrir_home | Hence you need to provide there information with high redundancy and reduce the ammount of information passed there |
18:36.35 | snazm | Sorry if mine was a stupid question but I'm a little baffled :$ |
18:36.50 | tzafrir_home | With proper use of command-line you can minimize the ammount of data that needs to be passed |
18:36.55 | asdx | what i was wondering is: if i have register => username:password@somehost under [general], do i still need username/password entries in my [user] entry? |
18:37.21 | jameswf-home | missed snazm's wuestion due to venting.. Im back :) |
18:37.33 | tzafrir_home | Or better: get them to use an IM, and serve as a sort of remote terminal for you |
18:38.20 | tzafrir_home | asdx, yes, you do |
18:38.28 | snazm | jameswf-home: lol :) |
18:38.32 | snazm | In Asterisk, is it possible to have it redirect a call if the end point goes dead without clearing the call down properly? |
18:38.59 | jameswf-home | snazm: probably with a creative dialplan |
18:39.56 | snazm | lol @ creative |
18:40.02 | snazm | You mean much hackery? |
18:40.12 | mackes | Just so I am clear james, Can you configure Apache, Sendmail, Postfix, MySQL, et all from the command line, quickly, so much so that you think Webmin is a waste of time? |
18:40.18 | jameswf-home | mackes: the generic answer is I was born with it..... I popped out attached to a mainframe(my mom was pissed) |
18:40.23 | mvanbaak | mackes: I can |
18:40.29 | jameswf-home | yes |
18:40.46 | *** join/#asterisk alephcom_ (n=chatzill@h66-112-187-16.mcsnet.ca) |
18:40.54 | jameswf-home | if its a mundane task I use a thumbdrive and a bash script |
18:40.56 | *** join/#asterisk lemanal (n=lemanal@Paawc.gbis.com) |
18:41.03 | mvanbaak | I use vim |
18:41.05 | *** join/#asterisk irule (n=irule@200.53.61.4) |
18:41.15 | tzafrir_home | mackes, yes. You go and read the docs |
18:41.22 | jameswf-home | some people still write their own scripts |
18:41.38 | mackes | wait, wait.. I'm taking from scratch, not sitting down with Docs and scripting it first... |
18:41.47 | mvanbaak | from scratch |
18:41.53 | jameswf-home | or steal and rewrite tzafrir's |
18:42.03 | mvanbaak | just last week I did setup 2 new postfix boxen and 5 Mysql boxen |
18:42.07 | mvanbaak | I never use webmin |
18:42.21 | mackes | New Install, open vi, and edit the confs, and start the service.... |
18:42.26 | mvanbaak | yup |
18:42.44 | tzafrir_home | actually apache used to be a nightmare, but now distros have tamed it quite nicely |
18:42.46 | mackes | What ever.. I know what you guys are saying, and its not apples to apples |
18:43.10 | mackes | you spend a morning working on it, or you use a premade script for rollout. |
18:43.11 | mvanbaak | tzafrir_home: most distro's corrupt it |
18:43.13 | tzafrir_home | Postfix has always had saner defaults |
18:43.15 | mackes | That is your webmin |
18:43.26 | jameswf-home | lol mackes spend a few months purelly in the black window it wont seem so insane |
18:43.26 | mvanbaak | mackes: no, ssh != webmin |
18:44.48 | tzafrir_home | webmin badly lacks scriptability |
18:45.01 | mvanbaak | and security |
18:45.14 | jameswf-home | webmin is like a woman al nice and sweet but one wrong move you lose half your stuff |
18:45.14 | tzafrir_home | for instance: how do you automate the simple task of allowing a certain IP to control it? |
18:45.28 | tzafrir_home | Or allowing every IP? |
18:45.33 | asdx | [Nov 17 18:44:49] NOTICE[7886]: chan_iax2.c:7241 socket_process: Rejected connect attempt from 190.52.149.212, who was trying to reach 'test@' |
18:45.34 | mvanbaak | tzafrir_home: you open the conf in vi ;) |
18:45.36 | asdx | i get that now |
18:45.51 | asdx | but i get AUTHENTICATED |
18:45.55 | asdx | when i connect |
18:46.09 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
18:46.46 | tzafrir_home | mvanbaak, I did it once. But finding exactly the file to edit and what to put there is not trivial |
18:46.55 | tzafrir_home | not to mention not documented anywhere |
18:47.08 | mvanbaak | I hate webmin |
18:47.23 | mvanbaak | it's a memory hog, it's insecure, and it's a mess without documentation |
18:47.28 | jameswf-home | we have a consensus on webmin lol |
18:47.37 | tzafrir_home | If they designed it with a nice command-line interface to configure everything it might have helped |
18:47.59 | jameswf-home | tzafrir_home: at that rate use ssh |
18:48.04 | mvanbaak | tzafrir_home: no. I dont like some setuid perl giant to be reachable on port 80 |
18:49.50 | jameswf-home | I wonder if i can get googles mobile platform to load on my blackberry without it crapping out and voiding my warranty |
18:50.24 | jameswf-home | my boss would be pissed lol |
18:50.50 | jameswf-home | he would say this is why you cant give geeks nice things |
18:53.34 | *** join/#asterisk lemanal (n=lemanal@Paawc.gbis.com) |
18:53.39 | *** join/#asterisk mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
18:53.39 | *** mode/#asterisk [+o mog] by ChanServ |
18:56.07 | jameswf-home | http://code.google.com/android/ << neat |
18:56.25 | *** join/#asterisk gardo (n=gardo@121.97.110.119) |
18:58.26 | jameswf-home | time to learn java |
18:59.21 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
19:00.07 | unixdog | if you build it they will come |
19:03.19 | jameswf-home | unixdog: the apperant answer to your question about getting rich on asterisk is build a hybrid hosted solution |
19:03.37 | tzafrir_home | jameswf-home, it smells too much like "almost free software" |
19:03.49 | tzafrir_home | read their usage license carefully |
19:04.00 | asdx | this is my configuration now: http://pastebin.com/m27b9f8a3 seems like i can authenticate but i get this: [Nov 17 19:02:25] NOTICE[8034]: chan_iax2.c:7241 socket_process: Rejected connect attempt from 190.52.149.212, who was trying to reach 'test@' |
19:04.09 | tzafrir_home | the neo1973 has a nicer license and is closer to reality |
19:05.11 | jameswf-home | you should hear the prices to get your stuff added to trixbox pro |
19:05.14 | jameswf-home | eek |
19:05.15 | tzafrir_home | Just a small reality check: how can cellular providers really support an open handset device? |
19:05.35 | jameswf-home | they can verry easily... they wont |
19:06.13 | tzafrir_home | officially some of them are behind this Android |
19:06.26 | unixdog | lol |
19:06.32 | bobkare | why on earth should cell providers care what software a users handset is running? |
19:06.55 | jameswf-home | bobkare: because they make a fortune by locking it down |
19:07.44 | bobkare | what country do you live in? (just so I know where not to move) |
19:07.56 | jameswf-home | you know howmuch extra "blackberry services" are, it doesnt cost them a penny but they charge for everything |
19:08.30 | bobkare | no such thing here in norway |
19:08.43 | tzafrir_home | bobkare, http://gizmo5.com/ |
19:09.08 | tzafrir_home | Not that I recommend them or anything. |
19:09.37 | tzafrir_home | Just as an example of how this gives you a way to pay less to your mobile provider |
19:09.43 | bobkare | here they sell you a sim card, or optionally lease you a phone as well (which is a real ripoff) |
19:10.15 | tzafrir_home | But a phone they don't approve will have a hard time connecting to the network |
19:10.40 | unixdog | I wish I could find a softphone for the blackberry |
19:10.45 | tzafrir_home | The neo1973 devices will find all sorts of difficulties connecting to some providers |
19:10.46 | unixdog | 7520 |
19:10.57 | tzafrir_home | let alone getting support for connectivity problems |
19:11.51 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
19:11.55 | bobkare | luckily that kind of behaviour will get them lots of unwanted attention from the government here |
19:13.03 | jameswf-home | the us goverment is owned by big business..... |
19:13.14 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
19:14.07 | jameswf-home | microsoft temporarily stopped paying their "protection money" and bam anti trust.. they are back on trac |
19:14.59 | jameswf-home | Ma Dell has started catching back up on payments and you notice that they are reforming ma bell |
19:15.23 | jameswf-home | 2/Dell/Bell |
19:15.25 | bobkare | my condolances to anyone living with a government like that |
19:16.15 | *** part/#asterisk PaulAviles (n=salinas9@dsl-7-36.cofs.net) |
19:16.26 | jameswf-home | its all good cause we can carry guns and kill our kids without much hassle |
19:17.12 | jameswf-home | 7% of our bill of rights still applies in 4 states lol |
19:18.35 | jameswf-home | really if you goy rid of the people america would be perfect |
19:18.41 | jameswf-home | *got |
19:21.54 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
19:23.17 | dlynes | Anyone know how buffer overruns, framing errors, collisions, or carrier errors can affect voice quality? |
19:24.05 | mamep | someone can help me with ooh323 channel? |
19:27.08 | rfxr | how do you prevent "Auto fallthrough"? I tried setting Set(TIMEOUT(response)=7) but as soon as the Background(enter-ext-of-person), it falls through and hangs up :( |
19:27.19 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
19:27.50 | rfxr | as soon as it completes the playback that is |
19:28.55 | tzafrir_home | jbot, tell mamep about ask |
19:29.22 | tzafrir_home | rfxr, you provide a different dialplan |
19:29.25 | mamep | how can i add peer with username and password to ooh323 channel? |
19:29.38 | tzafrir_home | Please pastebin the relevant dialplan context |
19:31.05 | rfxr | http://pastebin.com/m14b1bae3 |
19:31.40 | rfxr | this is from the docs except the timout which I found elsewhere trying it out |
19:31.49 | dlynes | rfxr: in your [general] section, set autofallthrough=no |
19:31.55 | rfxr | thanks |
19:33.31 | rfxr | awesome, thanks |
19:33.35 | mamep | http://pastebin.ca/778608 |
19:33.39 | mamep | can someone help me with this.. |
19:33.44 | rfxr | hope that makes it to the version 2 docs ;) |
19:34.28 | dlynes | rfxr: version 2 docs? |
19:34.36 | dlynes | rfxr: version 2 of what? |
19:34.52 | asdx | damn i cant make my pstn phone ring |
19:34.58 | rfxr | the AsteriskTFOT.pdf |
19:35.16 | dlynes | tfot? |
19:35.27 | rfxr | the future of telephoney |
19:35.32 | dlynes | oh |
19:35.33 | rfxr | phony :P |
19:35.53 | dlynes | rfxr: look in your sample extensions.conf file that comes with asterisk 1.4 |
19:35.59 | dlynes | rfxr: it's documented in there |
19:36.09 | rfxr | I saw that there, but the docs say to start fresh |
19:36.18 | rfxr | and it was omitted |
19:37.09 | mamep | someone? |
19:38.25 | dlynes | mamep: well, for one, you don't have your ulaw sound files installed |
19:38.43 | dlynes | mamep: try installing them first, and then rerunning your issue |
19:38.49 | mamep | hmm |
19:38.52 | tzafrir_home | mamep, I don't really know ooh323c, but I figure you better pastebin also the relevant parts of extensions.conf and of the ooh323c conf file (with passwords and such obfuscated, of course) |
19:38.52 | mamep | yeah beside that |
19:38.56 | dlynes | mamep: there'll be less spam then, and maybe the error will be more apparent |
19:39.12 | tzafrir_home | dlynes, by why ulaw? |
19:39.15 | mamep | tzafrir_home : i don't know how to add username pass to peer in ooh323 |
19:39.18 | tzafrir_home | it can be any format, right? |
19:39.26 | mamep | dlynes : let me check |
19:39.40 | tzafrir_home | mamep, isn't there a sample ooh323c config file? |
19:39.48 | dlynes | tzafrir_home: perhaps he's only got the g729 sound files installed and no g729 codec installed, or perhaps he doesn't have any sound files installed |
19:39.55 | mamep | yeah but i can't find user pass |
19:41.14 | dlynes | mamep: you also have a sip peer defined with a 'mailbox=' entry, but no corresponding entry in your voicemail.conf file |
19:41.58 | dlynes | mamep: but anyways, it seems the number you're calling doesn't exist |
19:42.23 | mamep | i mean h323 gateway is working? |
19:42.45 | dlynes | mamep: seems to be, but I would clear up the other spam, and then up your verbosity level |
19:42.53 | mamep | k |
19:42.55 | mamep | trying |
19:42.55 | dlynes | mamep: you might find that there's a dialplan bug |
19:43.00 | mamep | ? |
19:43.32 | dlynes | mamep: i.e. you're trying to dial an extension from your phone, for which asterisk doesn't currently handle |
19:43.56 | dlynes | mamep: the reason I'm getting at that, is that you're executing the following lines from your dialplan: |
19:44.05 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
19:44.15 | mamep | well i actually want to forward my calls through cisco's callmanager |
19:44.21 | dlynes | mamep: exten => _X.,n,Playback(the-number-u-dialed) |
19:44.29 | dlynes | mamep: exten => _X.,n,Playback(is-currently) |
19:44.34 | dlynes | mamep: exten => _X.,n,Playback(unavailable) |
19:44.43 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
19:45.18 | dlynes | mamep: please note that the extension and priorities I've used do not necessarily match what is in your dialplan |
19:45.28 | mamep | yeah ok sure.. |
19:46.13 | rfxr | my playback doesn't quite play from the beginning and cuts off the first half second or so... is there a way to adjust that? |
19:46.27 | mamep | dlynes : http://pastebin.ca/778624 |
19:46.36 | mamep | check my outgoing using ooh323 |
19:48.56 | dlynes | rfxr: pastebin your entire extensions.conf file |
19:49.13 | *** join/#asterisk CVirus (n=GoD@196.205.192.118) |
19:50.51 | *** join/#asterisk Greek-Boy (n=Greek-Bo@41.221.58.2) |
19:51.58 | rfxr | dlynes, http://pastebin.com/d29bfb1b9 |
19:52.06 | dlynes | mamep: try the following, and dump the output you get to pastebin: http://pastebin.ca/778631 |
19:52.18 | mamep | which one d3wayne ? |
19:52.24 | mamep | ah |
19:52.40 | mamep | one sec |
19:52.57 | dlynes | rfxr: which line number is it where the playback is cutting off after the first half or so? |
19:53.13 | dlynes | rfxr: oh...nvm |
19:53.14 | rfxr | it cuts off the beginning |
19:53.16 | dlynes | rfxr: i see it |
19:53.38 | rfxr | it cuts off the beginning of line 13 background |
19:54.04 | dlynes | rfxr: try the following: http://pastebin.com/m3a12e5 |
19:54.14 | mamep | dlynes : http://pastebin.ca/778635 |
19:54.43 | mamep | btw i have the file in gsm format |
19:54.48 | dlynes | mamep: now, do a dialplan reload |
19:54.55 | dlynes | mamep: and then redo it, and repastebin it |
19:55.06 | dlynes | mamep: you forgot to reload your dialplan before trying the new dialplan |
19:55.32 | dlynes | mamep: either that, or you never updated your dialplan, or the dialplan code you gave me, is not where it's failing |
19:55.49 | dlynes | mamep: which file? |
19:56.16 | mamep | http://pastebin.ca/778639 |
19:56.21 | mamep | the-number- |
19:57.12 | rfxr | dlynes, it doesn't cut of the end of the playback. For example, "Please enter the extension of the person you are trying to reach" plays back as "...ter the extension..." |
19:57.26 | dlynes | rfxr: ah |
19:57.28 | rfxr | cutting off the beginning |
19:57.35 | rfxr | the rest is fine |
19:57.46 | dlynes | rfxr: Right after the 'Answer()', add in exten => _XXXXXX.,n,Wait(1) |
19:58.06 | dlynes | rfxr: replace the _XXXXX. with whatever your extension pattern was |
19:58.15 | rfxr | ok, thank you |
19:58.18 | rfxr | again ;) |
19:58.24 | rfxr | this stuff is great! |
19:58.27 | linagee | does anyone know why a voip phone won't work? :( |
19:58.36 | dlynes | linagee: it's not plugged in? |
19:58.41 | linagee | i can ping it, i haven't changed the config. they've rebooted it... i get the gui... wtf |
19:58.41 | rfxr | :) |
19:58.44 | linagee | polycom 320 |
19:58.52 | linagee | s/gui/web gui/ |
19:58.58 | linagee | it worked before |
19:59.16 | dlynes | linagee: I would start with a sip debug, if nothing is showing up in verbose mode |
19:59.17 | linagee | and i'm using the same config scripts. i'm like, wtf? (i tried rebooting mine too. works just fine) |
19:59.24 | dlynes | linagee: also check your syslog for indications of dhcp issues |
19:59.38 | linagee | dlynes: exactly. and nothing shows up in sip debug! hah. i looked at the polycom log file and it looks normal. |
19:59.39 | dlynes | linagee: it might not be grabbing the latest tftp scripts |
19:59.54 | alephcom_ | lol, I like the "not plugged in one". We had a customer upset because their phone didn't work recently. Umm, well, it was unplugged from the network. |
19:59.54 | linagee | dlynes: i'm using ftp and it puts the logs up just fine |
19:59.58 | dlynes | linagee: i'm guessing htat's how polycom does autoconfig, anyways |
20:00.01 | rfxr | dlynes, works like a charm now, thanks ;) |
20:00.09 | linagee | dlynes: do you know if there's a way to do SIP to SIP calls with polycom? |
20:00.29 | dlynes | linagee: no idea...I use aastra, not polycom |
20:00.54 | dlynes | linagee: polycom's become quite anal lately...they'll only deal with you, if you're certified by one of their channel partners |
20:00.59 | linagee | users at other remote sites work just fine |
20:01.05 | linagee | s/users/family/ |
20:01.12 | mamep | dlynes : did u check it? |
20:01.13 | linagee | jbot: stop being annoying |
20:01.13 | jbot | ACTION leaps to his feet and stops being annoying |
20:01.49 | dlynes | mamep: yeah...you'll need to pastebin the entire extensions.conf file |
20:01.57 | mamep | k |
20:02.00 | linagee | dlynes: nothing at a sip debug is pretty weird when nothing has changed (that i know of) |
20:02.07 | dlynes | mamep: you're not showing me the correct area of your dialplan, or you're not adding my changes to your dialplan |
20:02.11 | dlynes | mamep: i'm not sure which it is |
20:02.59 | dlynes | linagee: have you monitored your /var/log/xferlog to make sure the phone is grabbing the latest configs? |
20:03.12 | linagee | dlynes: yes |
20:03.18 | mamep | here you go http://pastebin.ca/778647 |
20:03.27 | linagee | dlynes: is there a way to see what sip phones are currently registered? (in asterisk) |
20:04.02 | dlynes | linagee: have you double checked the contents of those config files to see if someone hasn't inadvertently changed them on you, without your knowledge? |
20:04.05 | dlynes | linagee: sip show users |
20:04.30 | dlynes | linagee: also sip show peers will show you the qualifies, and whether they're dynamic or not |
20:06.20 | linagee | dlynes: just checked them. did a diff on the regular <MAC HERE>.cfg and overrides/<MAC HERE>.cfg |
20:06.36 | linagee | dlynes: both are the same on the first one, the overrides has only the extension as the difference. |
20:06.48 | linagee | (as it should be) |
20:09.11 | linagee | dlynes: weird. all three are registered using sip show users.... |
20:09.18 | linagee | (and i still go straight to voicemail) |
20:10.09 | linagee | dlynes: here is the actual error: "Everyone is busy/congested at this time (1:0/0/1)" (in logs) |
20:10.13 | mamep | dlynes : any chance? |
20:11.33 | linagee | dlynes: wtf? sip show peers lists it differently. |
20:11.38 | linagee | hostname = unknown |
20:12.44 | linagee | wtf this is strange |
20:13.14 | linagee | (yes i have tried reloading asterisk. i have also tried restarting and stopping/starting asterisk) |
20:16.45 | dlynes | mamep: http://pastebin.ca/778655 |
20:17.07 | *** join/#asterisk CrashHD (n=crashhd@67-107-9-130.starstream.net) |
20:17.12 | dlynes | mamep: you'll find it's completely reworked...I eliminated all the guesswork by using a macro...it also compressed your dialplan down to half the size in the process |
20:17.50 | dlynes | linagee: hostname=unknown means it hasn't registered, or it had an error registering |
20:18.02 | dlynes | linagee: set core verbose=100 |
20:18.36 | dlynes | linagee: then reboot the phone in question, and monitor the registration in asterisk |
20:18.36 | dlynes | linagee: you'll probably find the username and/or password doesn't match for the registration |
20:18.55 | dlynes | linagee: double check by comparing your ftp config files to your sip.conf file for the phone |
20:19.06 | dlynes | linagee: also do a sip reload to be on the safe side |
20:19.37 | dlynes | linagee: perhaps some changes are made to the sip.conf file, that are not reflected in the running asterisk |
20:19.59 | linagee | dlynes: set core verbose? |
20:20.06 | mamep | declined |
20:20.06 | dlynes | linagee: yeah |
20:20.10 | dlynes | mamep: ? |
20:20.24 | mamep | http://pastebin.ca/778660 |
20:20.26 | mamep | call declined |
20:20.43 | dlynes | mamep: oops...my mistake |
20:20.49 | mamep | what? |
20:20.50 | linagee | dlynes: i've restarted and reloaded asterisk to heck already. and diffed the phone config files. :( |
20:21.06 | linagee | dlynes: maybe i can packet dump just from that phone ID and see if it's even attempting to register |
20:21.21 | linagee | s / ID / IP |
20:21.30 | linagee | hah! i outsmarted jbot |
20:21.37 | dlynes | mamep: replace the macro code with this: http://pastebin.ca/778661 |
20:22.26 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-89-254.hag.east.verizon.net) |
20:22.47 | dlynes | mamep: actually you can get rid of the EXTEN = ${EXTEN} part of the noop in the macro |
20:22.51 | dlynes | mamep: it's not useful anymore |
20:23.01 | mamep | k |
20:23.04 | mamep | http://pastebin.ca/778664 |
20:25.16 | dlynes | mamep: did you do a dialplan reload before trying the call again? |
20:25.55 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
20:26.20 | mamep | yeah |
20:26.58 | mamep | first of all how can i get rid of message about sound files? |
20:28.13 | dlynes | mamep: Are you in Europe? |
20:29.07 | dlynes | mamep: do a make menuselect in your asterisk source directory, go to the section for sound files, and make sure you install all the sound files, all the extra sound files, and all the music on hold files |
20:29.29 | mamep | i have only in gsm format |
20:29.31 | mamep | [Nov 17 22:29:05] NOTICE[4682]: chan_sip.c:5335 process_sdp: No compatible codecs, not accepting this offer! |
20:29.33 | mamep | i get this one |
20:29.36 | mamep | if i use only gsm |
20:30.13 | dlynes | mamep: nod...just install all the sound files |
20:30.44 | dlynes | mamep: I suspect your issue is that you're not using autoload=yes in your modules.conf file, and that there's certain modules you're not loading, or loading in the wrong order |
20:31.05 | mamep | hmm |
20:31.30 | mamep | installing sounds |
20:31.50 | dlynes | mamep: replace your modules.conf file with the following lines: |
20:31.52 | dlynes | mamep: [modules] |
20:31.56 | dlynes | mamep: autoload=yes |
20:31.58 | dlynes | and nothing else |
20:32.07 | linagee | dlynes: wow that's strange. i had the remote person try to call and saw nothing on a packet dump from that IP. hah |
20:32.18 | linagee | dlynes: but yet it fetches it's config. weird |
20:32.37 | mamep | dlynes : http://pastebin.ca/778670 |
20:32.40 | mamep | my modules.conf |
20:32.42 | dlynes | linagee: perhaps the config is telling it that the sip proxy and sip registrar are something other than your asterisk server |
20:33.11 | dlynes | mamep: yeah, your modules.conf file is fine |
20:33.22 | dlynes | mamep: I suspect you don't have certain modules built |
20:33.32 | mamep | which ones? |
20:33.34 | linagee | dlynes: it could be something weird like that, but all the other phones would be acting up. (i did a diff and rebooted the other phones. hrm) |
20:33.50 | dlynes | mamep: some of the format_... and codec_... modules |
20:34.06 | mamep | just a sec let me install first sounds |
20:34.06 | [TK]D-Fender | go prove is. "show modules like codec" <---- |
20:34.08 | [TK]D-Fender | it* |
20:34.12 | dlynes | linagee: I don't know about the polycoms, but on the aastras |
20:34.19 | [TK]D-Fender | Geez... stop guessing and start SHOWING. |
20:34.40 | linagee | dlynes: i was about to say maybe it's the cable provider, but i remembered this is over an encrypted VPN. :) |
20:35.01 | mamep | http://pastebin.ca/778673 |
20:35.05 | dlynes | linagee: you can override the web settings with dialpad settings |
20:35.19 | dlynes | linagee: get them to factory default the phone |
20:35.32 | linagee | dlynes: on polycoms its pretty much the same way. when you do a dialpad setting, it gets uploaded to the overrides dir |
20:35.35 | *** join/#asterisk Squeeb (n=squirt@87-194-8-66.bethere.co.uk) |
20:35.39 | dlynes | linagee: also, does the polycom allow you to get a dump of the current config used by the phone? |
20:35.41 | Squeeb | Hello. |
20:35.51 | Squeeb | Does anybody know when the asterisk docs page will be back up? |
20:35.53 | linagee | dlynes: i haven't heard of that if it does exist |
20:35.57 | [TK]D-Fender | mamep, Good, you have GSM, now pastebin your phone's entries, the CLI output of the failed call at verbose 10 and with channel debug enabled |
20:36.01 | dlynes | linagee: you might want to do that, and compare it with your ftp config files |
20:36.07 | [TK]D-Fender | dlynes, No |
20:36.31 | dlynes | [TK]D-Fender: ah..pretty crappy |
20:36.53 | dlynes | [TK]D-Fender: it really helps with debugging issues on the aastra phones...i'm sure it would help on the polycoms, too |
20:37.02 | [TK]D-Fender | dlynes, only if you royally screwed stuff up. Then again, anyone capable of that kind of damage won't be saved by an export either |
20:37.24 | [TK]D-Fender | dlynes, Wouldn't need debugging if you didn't spend so much time BUGGING them :p |
20:37.47 | dlynes | [TK]D-Fender: well, if their phones worked properly, i wouldn't spend so much time bugging them |
20:37.59 | [TK]D-Fender | PEBKAC <- |
20:38.04 | dlynes | pebkac? |
20:38.12 | [TK]D-Fender | dlynes, feel free to point fingers at charis :) |
20:38.14 | mamep | http://pastebin.ca/778678 |
20:38.18 | dlynes | charis? |
20:38.23 | [TK]D-Fender | chairs* |
20:38.43 | [TK]D-Fender | dlynes, http://en.wikipedia.org/wiki/PEBKAC |
20:39.23 | dlynes | [TK]D-Fender: nothing to do with me, or my users |
20:39.32 | dlynes | [TK]D-Fender: their phones aren't terribly well tested |
20:39.44 | [TK]D-Fender | mamep, ooh323_request - data ucnet format 0x8 (alaw) <--- why is this only asking for **ALAW**? |
20:39.45 | dlynes | [TK]D-Fender: i have one phone locking up like clockwork every 1/2 hour |
20:39.54 | [TK]D-Fender | mamep, fix your endpoint. |
20:40.11 | mamep | you mean i need to change it to gsm? |
20:40.22 | [TK]D-Fender | mamep, means it'd better bloody well match |
20:40.51 | [TK]D-Fender | mamep, don't have it asking for apples when all you have is ORANGES |
20:41.47 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
20:43.08 | mamep | ? |
20:46.07 | *** part/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com) |
20:47.04 | mamep | cdr.c:434 ast_cdr_free: CDR on channel 'OOH323/ucnet-73f9' not posted |
20:47.24 | snazm | After a bit of investigation in to the subject further and considering all the options, I've decided to go for a USB VoIP desk phone (with headset capabilities) and use Linux on the desktop to manage the VoIP connections and the phone hardware |
20:48.01 | snazm | Do people have preferences for the best type of USB desk phone to use and the best Linux software to enable it all to happen? |
20:48.23 | [TK]D-Fender | snazm, I highly recommend against that... |
20:49.26 | snazm | I have considered all of the options though and it seems to be the most practical solution |
20:49.28 | [TK]D-Fender | snazm, soft-phones are kludgy, low performing & usually low on features. This is not something you'd use for a business and I couldn't even tell you a USB handset that would work for Linux for that purpose. |
20:49.46 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-181-11.hsd1.or.comcast.net) |
20:49.51 | [TK]D-Fender | snazm, That'd mean you'd have to KNOW all the options. What have you thought about so far? |
20:50.23 | snazm | Well all it would be is a phone-shaped microphone and speaker essentially, just like a normal headset, but with a keypad and hopefully a screen also |
20:50.40 | snazm | Why is that so wrong or different if softphones through headsets have been used so successfully for so long? |
20:51.31 | [TK]D-Fender | snazm, For one its a question of making the dialpad of any use. What will TELL the soft-phone to use it? And then the display... that too. |
20:51.39 | obnauticus | How do I have multiple phones ring on 1 extension? |
20:51.52 | [TK]D-Fender | Softphones should only be used by people like those on laptops |
20:52.05 | [TK]D-Fender | obnauticus, "core show application dial" |
20:52.26 | obnauticus | I know that silly |
20:52.29 | snazm | I am hopefully going to find a softphone which will allow the keypad to be read just as if it were another input device (I've seen that mentioned a few times) |
20:52.40 | obnauticus | Do i put an other line with the same priority? |
20:52.50 | snazm | [TK]D-Fender: I don't think it would be too hard to tweak the softphone driver to pick this up and act on it |
20:53.12 | [TK]D-Fender | snazm, Ok well if your set on your path, best of luck with that.... |
20:53.48 | [TK]D-Fender | obnauticus, No, its 1 line. Read the instructions. |
20:53.50 | snazm | [TK]D-Fender: Well it seems to be the best option with regards on price sensitivity, features provided and upgradability |
20:53.58 | obnauticus | k |
20:54.27 | [TK]D-Fender | snazm, Which soft-phone do you have in mind? |
20:54.49 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
20:54.50 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
20:55.16 | snazm | [TK]D-Fender: I don't yet, I am now looking in to this. All I've decided so far is that according to all the criteria, it seems to be the best option to go for, that's why I was asking what the recommendations were |
20:56.16 | [TK]D-Fender | snazm, You're expecting normal users to go through this for business purposes. there will be a very noticable performance hit, lack of quality, redundancy, etc. |
20:56.53 | [TK]D-Fender | snazm, And puts more things into the picture that can go wrong. |
20:57.22 | snazm | [TK]D-Fender: Performance hit where? How can it be lack of quality also if the computer and softphone is doing exactly what a dedicated phone does anyway? |
20:58.11 | *** join/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com) |
20:58.13 | snazm | [TK]D-Fender: I also think it simplifies things, a dedicated VoIP system (with expensive phones :S) would double the ethernet connections, more cabling, more switches and loads of other stuff |
20:58.19 | [TK]D-Fender | snazm, soundcard quality and control sucks. if they touch the mixer, etc. USB handsets are usually junk quality speaker & mic themselves |
20:58.42 | [TK]D-Fender | snazm, Or you just buy phones with a pass-through port |
20:59.09 | snazm | [TK]D-Fender: Well yes I do know this, I have a softphone handset which I've been playing around with, but it's a cheapo. I'm sure that a professional desk phone would have a much superior audio quality, some have had good reviews |
21:00.02 | obnauticus | <PROTECTED> |
21:00.05 | obnauticus | Err, what does that mean |
21:00.10 | [TK]D-Fender | snazm, well you've heard my recommendation. Hope you find a way to make it work and be worth the difference |
21:00.17 | [TK]D-Fender | obnauticus, means a phone is forwarded |
21:00.26 | obnauticus | Huh? |
21:00.33 | obnauticus | Oh I see |
21:00.34 | obnauticus | nevermind |
21:00.38 | snazm | [TK]D-Fender: Yes I know what you're recommending is the dogs bollocks of VoIP. But I have to justify everything I do including cost |
21:01.11 | obnauticus | lol |
21:01.14 | [TK]D-Fender | snazm, I don't know a manager out there who'd sacrifice so much for so little savings. |
21:01.24 | obnauticus | I got a lot of stuff done thanks to you [TK]D-Fender |
21:01.28 | snazm | [TK]D-Fender: You don't know many managers then :D |
21:01.40 | snazm | [TK]D-Fender: Don't get me wrong, I would love it |
21:01.58 | snazm | [TK]D-Fender: But unfortunately for 80 seats, the added costs soon add up :( |
21:02.21 | [TK]D-Fender | snazm, I know plenty and have sold several projects. People want a phone they can hold in their hands and feels natural. Something of quality that is intuitive. |
21:02.40 | [TK]D-Fender | snazm, Whats your budget for this? |
21:02.54 | snazm | That's what I mean by a USB VoIP desk phone, it looks like a normal office phone with a USB plug |
21:03.04 | [TK]D-Fender | snazm, And what USB phone & spftphone are you looking at? |
21:03.17 | snazm | rob0: Always :D |
21:03.27 | snazm | I've been looking at loads |
21:03.40 | [TK]D-Fender | snazm, price? and which soft-phone? |
21:03.52 | snazm | There isn't a budget as such but some quotes have been in the region of £20k/$40k |
21:04.09 | snazm | That was rejected quite spectacularly |
21:04.24 | [TK]D-Fender | snazm, 40$k!? lol |
21:04.32 | snazm | :-o? |
21:04.41 | [TK]D-Fender | snazm, Dunno where you're getting pricing.. |
21:04.49 | snazm | The suppliers :-| |
21:04.59 | snazm | That's why I'm of the DIY mentatlity now |
21:05.36 | Nukemizer | is there a core command list, I am trying to fing like {core:userlogon} but for acd login ? |
21:05.46 | snazm | I tell you what then, give me your recommendation of a good dedicated desk phone and I will look in to it again |
21:06.00 | [TK]D-Fender | snazm, 80 x $100USD (Linksys SPA-941) = $8000 for phones. How much is a kludgy USB setup going to cost you? |
21:06.19 | snazm | [TK]D-Fender: I am of course still very much open to ideas, but as I said, as far as I know, that would be the most cost effective and practical option |
21:06.26 | [TK]D-Fender | Nukemizer, "show applications" |
21:06.38 | Nukemizer | thank you |
21:07.21 | [TK]D-Fender | snazm, So at $8000 for phones, 1000$ for a PRI card, and $2000 for a decent RAID server, you're looking at $11,000, not $40k |
21:08.23 | snazm | OK you're making it sound quite attractive, but the only thing is a softphone approach would still be cheaper than $11l |
21:08.49 | snazm | And that still seems like quite a bit for what it is (there is already a working POTS PABX in place, so replacing it needs to be as justifiable as possible) |
21:09.11 | [TK]D-Fender | snazm, if you have a pots PBX, then just get FXS gateways! |
21:09.58 | [TK]D-Fender | snazm, and reuse your phones |
21:10.08 | snazm | It's a crap PABX, it's even stoped working - some users can't even transfer calls! |
21:10.11 | [TK]D-Fender | snazm, Still much better than soft--phones. |
21:10.30 | snazm | It's also not very flexible at all, has no auto-login or follow-me features |
21:10.35 | [TK]D-Fender | snazm, if its POTS (for the stations), then its the PBX's fault, not the phones. |
21:10.42 | snazm | I know |
21:10.51 | snazm | The whole system is flaky |
21:10.59 | [TK]D-Fender | snazm, So keep the phones & wiring, and jsut dropin in * in place o fthe PBX itself |
21:11.02 | snazm | Looks very old and worn |
21:11.10 | snazm | All the wiring is being ripped out too lol |
21:11.28 | snazm | To explain, the building was build in the 1800's and the current wiring was done about 20 years ago |
21:11.35 | snazm | No Cat5e also |
21:12.53 | snazm | This whole project is very big, and even a slight saving per seat on VoIP would reduce the total cost of it by a lot. Sorry for being more sensitive to cost than technical excellence (although I wouldn't like to be of course), but I have to present options and reasons with the prices before anything is agreed |
21:13.39 | snazm | The SPA941 looks nice though |
21:13.42 | [TK]D-Fender | snazm, Well it sounds like you won't be PRESENTING anything but what you find as the lowest costs. |
21:14.09 | [TK]D-Fender | snazm, and that will cost you for a usage & support performace hit. |
21:14.29 | snazm | I will offer a few options, but judging by the response to the previous quotes, I don't want it thrown back in my face |
21:14.37 | [TK]D-Fender | snazm, You will spend a asignificant amount of time trying to tweak your plan into something only second-rate |
21:15.14 | [TK]D-Fender | snazm, I jsut showed you $11K ($8K of phones). That already undercuts the other buy 2/3 |
21:15.31 | snazm | Well the price did include setup and consultancy |
21:15.37 | snazm | Which still needs to happen |
21:16.18 | snazm | I'm still not an expert by any means, but I feel as though I could manage a lot of what they wanted, and maybe with one expert on board for a week we could get it done which would be cheaper |
21:16.30 | [TK]D-Fender | snazm, Keep your old analog phones and you can drop that amout from $8000 to $2100 |
21:16.48 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-9-152.bchsia.telus.net) |
21:16.59 | [TK]D-Fender | snazm, using 10x SPA-8000 |
21:17.09 | snazm | None of the POTS wiring will remain though, it's a mega mess (there are even exposed live mains wires in places lol) |
21:17.23 | snazm | The whole thing needs to be ripped out and done properly and safely |
21:17.53 | snazm | SPA 8000 |
21:18.12 | Qwell | 8000... 8 port? |
21:19.04 | snazm | How easy would it be to implement follow-me and auto-logon on analogue POTS phones with this device? |
21:20.34 | snazm | I don't even know how that would work TBH |
21:21.41 | [TK]D-Fender | Qwell, yup, 8 ports at $200. almost impossible to beat |
21:22.08 | snazm | It really effs me off when you order diet coke and they bring you normal coke 8-) |
21:22.17 | snazm | How hard is it for the freaks to understand it's not the same thing |
21:23.45 | De_Mon | snazm the diet stuff is actually worse for you dontcha know |
21:23.56 | snazm | Aspartame ftw :D |
21:24.06 | snazm | I know but I'm low-carbing at the moment |
21:24.13 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
21:24.17 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
21:24.19 | Qwell | water > diet anything |
21:24.42 | snazm | Not in the UK lol, in my home county they add fluoride to everything :S |
21:24.52 | Qwell | and? |
21:24.58 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
21:25.05 | snazm | Do you know how nasty fluoride is? :-O |
21:25.12 | Qwell | fluoride is...tasteless |
21:25.26 | snazm | It's also a neurotoxin that doesn't get flushed out of your system, you accumulate it for life |
21:25.38 | snazm | And it's more poisonous than a lot of other chemicals including lead and arsenic |
21:25.53 | Qwell | besides - you think they don't use the same water in coke that you drink? |
21:26.10 | snazm | I doubt it, most of this coke comes from abroad lol |
21:26.18 | lirakis | my asterisk server "crashed" this morning. I rebooted it and now i can not get it to start. 'asterisk -c' shows 'ERROR[13931]: chan_zap.c:7053 mkintf: Unable to open channel 1: No such device or address' |
21:27.18 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
21:27.31 | lirakis | here is a full pastebin of the asterisk -c http://pastebin.ca/778728 |
21:27.51 | lirakis | <10 lines FYI |
21:29.11 | lirakis | i have a sangoma A101 card |
21:33.41 | asdx | i registered two users and it has too much noise/echo when i call the other user |
21:33.57 | asdx | 300 ms of latency |
21:34.07 | asdx | gsm codec |
21:35.15 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
21:41.07 | [TK]D-Fender | lirakis, if it bombs on your first channnel odd are zaptel didn't load. Either for not being called, or for wanpipe not having been ready |
21:42.39 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
21:43.16 | lirakis | [TK]D-Fender: ive been working with the modules... i have manually modprobed zaptel.. and still got nothing. I backed up zapata.conf and wiped it out to a empty file, then mod probed ztdummy and asterisk started. |
21:43.54 | [TK]D-Fender | lirakis, you aren't supposed to be using ztdummy. |
21:44.08 | [TK]D-Fender | lirakis, and go verify wanpipe. Then do "ztcfg -vvvv" |
21:44.09 | lirakis | [TK]D-Fender: honestly.. im unsure of what wanpipe is... i didnt setup the card on this box. It seems odd.. that when i do lspci .. it shows up as a "Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card" |
21:44.25 | [TK]D-Fender | lirakis, then you have NO slue what the hell you're doing ! :p |
21:44.37 | lirakis | [TK]D-Fender:yeah i know.. (about ztdummy) i just did it to make sure asterisk would start. |
21:44.40 | [TK]D-Fender | lirakis, Wanpipe is Sangoma's drive which needs to be loaded first |
21:45.21 | *** join/#asterisk dlynes_ (n=dlynes@d154-20-9-152.bchsia.telus.net) |
21:45.25 | lirakis | [TK]D-Fender: okay.. i wasnt sure if it enabled some "special features" or some thing.. or if it was needed.. i tried to do a quick look on their website.. but it wasnt immediately apparent. |
21:45.33 | lirakis | [TK]D-Fender: i will check out wanpipe |
21:45.53 | [TK]D-Fender | lirakis, "wanrouter status" <- PB it |
21:49.01 | lirakis | <PROTECTED> |
21:49.16 | [TK]D-Fender | lirakis, yup, that'd kill it |
21:49.55 | [TK]D-Fender | lirakis, when that happens on my server I lose internet as I use an S518 for ADSL :) |
21:51.00 | lirakis | <PROTECTED> |
21:51.18 | *** part/#asterisk PepOSX (n=pepOSX@190.72.149.231) |
21:51.19 | [TK]D-Fender | lirakis, saves me a lot of wiring though... I love it |
21:51.51 | lirakis | [TK]D-Fender: im sure.. and one less appliance in the network |
21:53.43 | [TK]D-Fender | lirakis, 1 ls appliance, 1 less power brick. |
21:56.29 | lirakis | [TK]D-Fender: okay .. its up now with a recompile and reloading of the driver |
21:56.38 | lirakis | [TK]D-Fender: thanks.. i |
21:56.41 | lirakis | * -i |
21:59.24 | [TK]D-Fender | lirakis, np |
22:00.14 | *** join/#asterisk anthm (n=anthm@adsl-70-226-55-121.dsl.milwwi.ameritech.net) |
22:00.14 | *** mode/#asterisk [+o anthm] by ChanServ |
22:00.25 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
22:07.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
22:24.35 | *** join/#asterisk cyberpass2 (n=mataz@ppp-64-219-79-16.dsl.hstntx.swbell.net) |
22:24.49 | cyberpass2 | which modems can you reprogram to make them into phone adapters? |
22:24.57 | *** join/#asterisk jozu (n=torrent@84.120.184.91.dyn.user.ono.com) |
22:25.02 | jozu | hello to all |
22:25.08 | BBHoss | cyberpass2: none of them |
22:25.37 | cyberpass2 | <BBHoss> hrmm..r u sure? |
22:26.04 | cyberpass2 | <BBHoss> can you suggest a cheap phone adapter that works well with asterisk? |
22:26.08 | BBHoss | cyberpass2: there may be some out there but you don't want to use them |
22:26.27 | BBHoss | cyberpass2: what kind of ports do you want |
22:26.35 | BBHoss | fxo, fxs or both? |
22:26.56 | cyberpass2 | regular US phone jack...whats the diff between fxs and fxo? |
22:27.10 | BBHoss | heh |
22:27.31 | BBHoss | http://www.patton.com/technotes/fxs_fxo.pdf read that and come back |
22:27.50 | BBHoss | or ~book |
22:27.56 | BBHoss | ~book |
22:27.57 | jbot | 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 |
22:30.28 | cyberpass2 | oh ok |
22:31.05 | cyberpass2 | well..what I want is to have multiple FXS's so I can have a number of phone lines to one box |
22:31.19 | cyberpass2 | maybe one FXO just to test |
22:31.36 | cyberpass2 | or do i have it backwards? |
22:32.23 | BBHoss | you need an FXO to connect to the phone company |
22:32.35 | BBHoss | and you need an fxs to connect to one of your phones |
22:32.37 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
22:32.49 | BBHoss | or you can ditch FXS and go for IP Phones |
22:33.08 | BBHoss | or SIP/IAX2 SoftPhones, which you install on your computer |
22:33.33 | cyberpass2 | well i want to set up my voip service |
22:33.50 | BBHoss | what kind of voip service? |
22:34.25 | BBHoss | are you paying for a voip service from an ITSP |
22:34.25 | cyberpass2 | so have multiple internet users connect to the box using their own voip gateways... |
22:34.34 | cyberpass2 | no im not...i dont intend to |
22:35.05 | cyberpass2 | then have multiple phone lines(POTS) connected to my asterisk box |
22:35.13 | BBHoss | so you want other people to be able to connect their asterisk boxes to you, or do you just want them to be able to link their phones to you |
22:35.33 | cyberpass2 | i want them to connect to me... |
22:35.49 | BBHoss | 1st or second? |
22:35.50 | cyberpass2 | i was going to give them those cheap linksys PAP2s |
22:36.26 | cyberpass2 | i want them to connect to my askterisk box which is connected to my telephone company using regular phone lines |
22:36.54 | cyberpass2 | so, they could make calls |
22:37.07 | cyberpass2 | maybe not recieve them...but make outgoing calls for sure |
22:37.12 | BBHoss | yeah you can use PAP2 |
22:37.18 | BBHoss | they could make and recieve |
22:37.27 | BBHoss | i would use the digium IAXy though |
22:37.31 | BBHoss | it works better with NAT |
22:37.35 | [TK]D-Fender | iaxy yuck.... |
22:38.01 | BBHoss | but a pap2 will work too |
22:38.22 | cyberpass2 | can i set asterisk so that any incoming calls from the POTS phone lines get automatically disconnected? |
22:38.33 | BBHoss | yeah |
22:38.42 | BBHoss | or you could just never answer them |
22:38.51 | rob0 | :) Not plug in the FXO. |
22:39.09 | cyberpass2 | <BBHoss> but if i never answer them, the phone lines wont be availble for outgoing calls |
22:39.20 | BBHoss | cyberpass2: yeah they will |
22:39.38 | cyberpass2 | damn...i really gotta read up more on this... |
22:39.40 | BBHoss | cyberpass2: only the incoming calls wouldnt be answered |
22:39.41 | BBHoss | yeah |
22:39.43 | BBHoss | ~book |
22:39.44 | jbot | 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 |
22:39.50 | rob0 | I use a menu for that. |
22:39.51 | BBHoss | FREE!!!^^^^^^^^^^^^^^^^^^^^ |
22:40.35 | cyberpass2 | so..i could still set up a call, even if there is an incoming call on that line? |
22:40.46 | BBHoss | no |
22:40.52 | BBHoss | i see what youre saying now |
22:41.06 | BBHoss | if someone called it and never hung up |
22:41.24 | cyberpass2 | anyways...i can call up POTS provider and tell them to make it outgoing phone line only |
22:41.31 | cyberpass2 | hopefully they will agree |
22:41.33 | rob0 | there's a thought |
22:41.34 | BBHoss | yeah |
22:41.46 | BBHoss | its called a "something" terminate |
22:42.18 | cyberpass2 | now...when all this is set up and running, would asterisk also allow for internal phone to phone routing? |
22:42.37 | rob0 | if you set it up that way :) |
22:42.37 | cyberpass2 | ie i give each one an internal number so they can call each other |
22:42.41 | BBHoss | YES READ THE BOOK |
22:42.56 | BBHoss | ITS A FREE DOWNLOAD |
22:42.59 | cyberpass2 | i wil i wil |
22:43.12 | BBHoss | you can do anything with asterisk |
22:43.21 | De_Mon | cyberpass2 all your wet dreams will come tru, just read the book and come back if you still have questions |
22:43.41 | De_Mon | my * box orders pizza automatically I just dial 2, its really cool... |
22:44.04 | cyberpass2 | so for what ive describe...how much am i looking for interms of hardware for the FXO/FXS? |
22:44.09 | BBHoss | what do you call the company and give them a voip menu? |
22:44.33 | BBHoss | well how many phone lines from the telco to you want? |
22:44.40 | cyberpass2 | <De_Mon> 3 |
22:45.00 | cyberpass2 | <De_Mon> do you automatically play a sound file when the pizza store answers? |
22:45.12 | BBHoss | so you need 3 lines? |
22:45.12 | cyberpass2 | <BBHoss> 3 |
22:45.14 | De_Mon | no, it waits 5 seconds and places the order, its always the same so :) |
22:45.15 | cyberpass2 | yea |
22:45.25 | cyberpass2 | oh ok |
22:45.39 | De_Mon | then it adds a if you have any questions press 0 for operator and will call me :) |
22:46.02 | BBHoss | cyberpass2: i would get a TDM400p with 3 FXO modules |
22:46.06 | De_Mon | I like smiley faces :) |
22:46.33 | *** join/#asterisk weazahl (n=revwease@adsl-68-93-176-137.dsl.ksc2mo.swbell.net) |
22:47.08 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
22:47.24 | jozu | i have and error installing 1.4.14, chan_iax2.o error |
22:47.26 | jozu | ??? |
22:47.40 | De_Mon | the actuall error will help |
22:47.47 | cyberpass2 | <De_Mon> thats sick!...so you pick up ur phone, dial 2, shut ur phone...then asterisk calls the pizza store, dials in the DTMF codes for your pizza and play a sound file after? |
22:48.47 | De_Mon | no.. it plays a recording of me saying "Hi id like 2 large peparonie pizzas <dramatic pause> that will be all. <long pause> If you have any questions press 0 |
22:49.22 | De_Mon | oiy I butchered pepperoni that time |
22:49.32 | De_Mon | crap I forgot what I was doing! |
22:51.19 | weazahl | ok, i have 2 boxes that use vitelity for in/out trunks. one has been running for almost a year without reboot and no problems. the other, on a regualar basis will be able to accept calls but not be able to make outbounds until i restart asterisk. it restarts at 3am daily. any ideas on why? |
22:51.46 | De_Mon | weazahl are you restarting asterisk at 3am daily? |
22:52.02 | De_Mon | weazahl lets see the actual errors when outbound calls fail instead of us guessing |
22:52.15 | weazahl | well, for a while it was just asterisk. now i reboot since it dint help |
22:52.30 | De_Mon | sounds like your hardware is hosed |
22:53.05 | weazahl | infortunatly i dont have any right now. laptop got wiped clean and havent had it happen in a few days |
22:53.54 | De_Mon | maybe you should come back when you actually have some useful info |
22:54.00 | weazahl | its a new poweredge core duo, 2 gigs of ram (overkill i know) and any memtest or other test i throw at it works fine |
22:54.08 | jozu | someone get an error installing 1.4.14? |
22:54.24 | De_Mon | jozu you did, duno what it was though |
22:54.36 | De_Mon | weazahl telephony hardware |
22:54.58 | weazahl | De_Mon: no analouge all VOIP |
22:55.05 | De_Mon | you know.. the things you are trying to call out on and cant? |
22:55.36 | De_Mon | weazahl okay... there is more than one VOIP protocol supported by asterisk do they both stop working? |
22:55.36 | weazahl | like i stated, vitelity.net no local hardware |
22:56.16 | weazahl | De_Mon: thank you, i will switch it to IAX in my window tonight |
22:56.25 | weazahl | i will see what happens then |
22:56.40 | De_Mon | if you can actually get an ERROR we could tell you why its happening (the error probably does that already, but hey) |
22:58.09 | weazahl | the sip debug is hard to use with 32 stations chattering constanly. and it is infreuqent. just a pain. |
22:58.36 | De_Mon | so, it stops some outgoing calls not all of them? |
22:59.04 | weazahl | only calls going out on that trunk. |
22:59.16 | De_Mon | but all calls going out on that trunk, not "some" |
22:59.24 | weazahl | hang on a minute. i think the logs havent rotated out yet. |
22:59.31 | weazahl | correct all |
23:00.29 | De_Mon | Its hard to believe restarting asterisk doesn't solve problems with a SIP/IAX provider |
23:00.40 | De_Mon | and that rebooting the box does |
23:04.00 | weazahl | restarting does solve it. the problem is, i restart at 3am. and sometimes the problem apears at 4am, sometimes at 22pm |
23:04.24 | weazahl | 11pm rather |
23:05.05 | weazahl | real problem when i get woke up at 5 |
23:05.16 | De_Mon | if restartign asterisk fixes the problem, why are you rebooting the box? |
23:05.37 | De_Mon | SVM support vector machines? Never heard of one of those before |
23:06.30 | *** part/#asterisk deeperror (n=deeperro@d14-69-9-250.try.wideopenwest.com) |
23:06.34 | weazahl | accually no. and the reboot is really just something i was trying since a 3am restart didnt really help anything |
23:07.10 | weazahl | it is a RANDOM and INTERMITTENT problem |
23:07.45 | weazahl | log is taking a while to load. 600+mb |
23:17.09 | weazahl | De_Mon: Nov 14 09:38:21 NOTICE[5667] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
23:17.13 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
23:17.40 | weazahl | it is pingable at that time though, and is still registered and accepting inbound |
23:21.43 | [TK]D-Fender | weazahl, that measn nothing. |
23:22.17 | [TK]D-Fender | weazahl, next time enable SIP DEBUG, pand pastebin the full output. Also describe in detail the networking path between * and your provider |
23:27.37 | weazahl | [TK]D-Fender: ok. i'll do so. network path is quite simple, Static DSL to KC MO, Dallas TX, Denver CO terminating at coloc on XO.NET |
23:27.59 | weazahl | same as another machine 3 blocks away that NEVER has a problem |
23:29.42 | weazahl | i have replaced the DSL router because the forst one cooked in the summer while there was no AC during renovations |
23:29.45 | [TK]D-Fender | weazahl, So you * server has a public fixed IP? |
23:30.10 | weazahl | sure does. is there any other way? :D |
23:30.22 | *** join/#asterisk yannj_fr (n=yannj_fr@tvn95-3-82-237-158-147.fbx.proxad.net) |
23:30.31 | [TK]D-Fender | weazahl, Sure... behind behind a NAT router with a dynamic ip, duh..... |
23:31.10 | weazahl | i know that. but i WAY perfer the static route. |
23:31.23 | weazahl | makes maintnence much easier |
23:32.39 | *** join/#asterisk _matt (i=matt@2001:770:168:1:20b:cdff:fe04:843a) |
23:33.06 | *** join/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
23:33.26 | katsuodo | everyone hello |
23:34.28 | katsuodo | Does anyone know how many calls does dsl support using sip and asterisk |
23:34.48 | [TK]D-Fender | katsuodo, Depends on what codec, and how much bandwidth |
23:35.21 | katsuodo | Halo [Tk]D-Fender |
23:37.40 | rob0 | 42 |
23:38.01 | [TK]D-Fender | 6 * 7? |
23:38.27 | weazahl | with 6016/768 and ulaw, at 5 you will have signifigant jitter. with GSM, dont know cant get that many channels going. i figure about 20 though |
23:38.38 | rob0 | The answer! |
23:39.10 | weazahl | rob0: took me a second but, what is the question to the answer |
23:40.17 | Corydon76-dig | How many roads must a man walk down before they call him a man? |
23:40.35 | weazahl | NULL SET |
23:40.44 | [TK]D-Fender | Corydon76-dig, Just one.... the one with the red light... she'll make a man outta you ;) |
23:41.10 | Corydon76-dig | [TK]D-Fender: no, it's rumored that is the question, to which the answer is 42 |
23:44.42 | katsuodo | [TK]D-Fender G.729 not sure of bandwidth? Is dsl idea for office? |
23:45.40 | [TK]D-Fender | katsuodo, you don't know what your bandwidth is, and you didn't describe your needs including any suitabe description of your "office". Just how psychic do you think we are? |
23:46.14 | katsuodo | pardon |
23:48.56 | katsuodo | two company, one asterisk pbx, (4) polycom 301 ip phone, (4) RCA digital phone, one user needs the extension to connect to other extension in other company, (8) pots only use (5), of (5) one for fax, dsl (is all I am told), no have bandwidth number yet |
23:49.45 | katsuodo | Six extension in both company total 12 extension |
23:52.05 | [TK]D-Fender | katsuodo, what are you planning on doing over the internet? |
23:53.28 | katsuodo | there was discussion of iax to other office |
23:53.58 | katsuodo | so really three office. other office (3) ip phones connect to asterisk |
23:54.15 | [TK]D-Fender | katsuodo, you made it sound like there were 2 companies sharing 1 PBX localluy. Perhaps you should split your description up and try again |
23:55.01 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
23:56.09 | katsuodo | yes, (2) company in one office sharing asterisk; other company different location with asterisk pbx connect iax (maybe!) no not yet decide |
23:56.32 | *** join/#asterisk jero (n=jerome@modemcable169.212-70-69.mc.videotron.ca) |
23:56.36 | jero | hi all |
23:56.42 | [TK]D-Fender | katsuodo, how many channels do you want to have between the 2 systems? |
23:56.44 | katsuodo | one office two company internet no important for phone |
23:57.04 | katsuodo | unless connect to other asterisk pbx in other office |
23:57.36 | *** join/#asterisk asdx (n=diego@adsl-149-212.click.com.py) |
23:57.40 | asdx | hi |
23:57.41 | katsuodo | in one office two company two channels, no? |
23:57.44 | jero | I'm experiencing problems with asterisk 1.4 and polycom 601 expansion module: can't get the module to switch its light to ON for outgoing calls (asterisk does not notifies an event when a phone makes an outgoing call) |
23:57.44 | [TK]D-Fender | katsuodo, thats what I'm talking about. how many simultaneous calls do you figure you'll pass between these 2 boxes if your expansion happens? |
23:57.55 | asdx | i have a little echo in my pure voip calls |
23:58.05 | asdx | how can i remove echo |
23:58.08 | asdx | jitterbuffer? |
23:58.11 | [TK]D-Fender | asdx, what hardware? |
23:58.29 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
23:58.29 | asdx | [TK]D-Fender: asterisk is running on a vps |
23:58.37 | [TK]D-Fender | asdx, I mean the ENDPOINTS |
23:58.47 | katsuodo | guess maybe 20 in whole day? |
23:58.52 | katsuodo | if expansion |
23:59.01 | [TK]D-Fender | katsuodo, I'm talking simultaneous. |
23:59.27 | asdx | [TK]D-Fender: this is pure voip, no special hardware |
23:59.31 | katsuodo | inbound / outbound you speak of, yes? |
23:59.46 | [TK]D-Fender | asdx, what kind of PHONES........ what is on each end of the conversation?! |
23:59.58 | asdx | [TK]D-Fender: softphones (zoiper) |