| 00:00.57 | [TK]D-Fender | NovceGuru, sure you can. |
| 00:01.13 | [TK]D-Fender | NovceGuru, Go read up on Presence on the WIKI |
| 00:01.19 | *** join/#asterisk evilkiksass (n=do@75.35.230.5) |
| 00:01.45 | evilkiksass | Does anyone know of an avaya phone that supports LLDP, CDP and works with Asterisk? |
| 00:03.15 | NovceGuru | [TK]/win 12 |
| 00:03.22 | NovceGuru | erm, thanks [tk] |
| 00:04.01 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 00:04.50 | *** join/#asterisk moy (n=moyhu@189.169.83.74) |
| 00:05.14 | NovceGuru | I suppose I have found one issue with going with a hosted solution |
| 00:06.33 | *** join/#asterisk s0lid (n=s0lid@210.213.199.2) |
| 00:09.51 | *** join/#asterisk b1shop (n=b1shop@c-71-194-197-216.hsd1.il.comcast.net) |
| 00:14.06 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 00:14.07 | [TK]D-Fender | NovceGuru, Funny, the rest of us take this stuff for granted and see it on our PHONE's |
| 00:14.08 | gitguy | hi |
| 00:14.28 | gitguy | is asterisk 1.6 (latest one) good for production already? |
| 00:14.58 | [TK]D-Fender | gitguy, its in BETA. Do the math. And generally you do't want to take X.0 into "production" in anything. |
| 00:15.18 | drmessano | It's not even an RC |
| 00:15.44 | gitguy | hm okay |
| 00:16.32 | drmessano | wonders when "beta" went from "experimental, testing only" to "got my whole company running on it" |
| 00:17.43 | ManxPower | drmessano: sometime before Asterisk 1.0, but it really accelerated between 1.0 and 1.2 |
| 00:17.53 | NovceGuru | [TK]D-Fender: ? |
| 00:18.15 | drmessano | lol |
| 00:18.55 | NovceGuru | [TK]D-Fender: you mean the person that wants it |
| 00:19.00 | [TK]D-Fender | NovceGuru, like I said... GO READ THE WIKI |
| 00:19.02 | NovceGuru | thinks it "should be simple" |
| 00:19.11 | [TK]D-Fender | NovceGuru, What you're looking for is PRESENCE. |
| 00:19.27 | NovceGuru | Right I just didn't get your second comment is all |
| 00:20.47 | puzzled | evening all |
| 00:21.34 | puzzled | in 1.4.20-rc2 did speex get disabled too like ilbc? I have speex & speex-devel installed, ./configure shows all is ok yet in make menuselect I only see it disabled XXX and I can't enable it |
| 00:22.48 | jeffspeff | hey, what's the best way to stress test a system? I want to find out how many concurrent calls mine can handle. Do I have to setup a bunch of phones and have several people call me? |
| 00:23.08 | puzzled | jeffspeff: google for sipp and sipsak. voip-info.org has some info on this too |
| 00:23.28 | _ShrikE | jeffspeff: google sipp |
| 00:29.59 | gitguy | can asterisk transcode, eg: when one of your endpoints uses GSM and the other one ULAW/ALAW, will it work? |
| 00:30.09 | _ShrikE | gitguy: yes |
| 00:30.20 | gitguy | if A (gsm) dials b (ULAW) |
| 00:30.45 | _ShrikE | g.729a requires a license however |
| 00:31.13 | gitguy | i'm doing that right now and getting "Call failed: 503 Service Unavailable" in xlite |
| 00:31.23 | LinuxMafia | hi |
| 00:31.25 | LinuxMafia | me again |
| 00:31.34 | gitguy | guess i should enable sip debug |
| 00:32.07 | LinuxMafia | i want to buy a voip phone that support sip |
| 00:32.33 | Qwell | ~phones |
| 00:32.34 | 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 phones. Ever. places like such as |
| 00:32.46 | LinuxMafia | Qwell, in canada though |
| 00:32.52 | LinuxMafia | i did not find any of those |
| 00:35.27 | [TK]D-Fender | LinuxMafia, how many, and for what kind of use? |
| 00:35.49 | LinuxMafia | [TK]D-Fender, just 1 , for home use for now |
| 00:35.58 | [TK]D-Fender | LinuxMafia, And down the road? |
| 00:36.14 | LinuxMafia | hum not really |
| 00:36.19 | *** join/#asterisk mandd (n=dache@dsl-129-156.aei.ca) |
| 00:36.24 | mandd | hello |
| 00:36.29 | LinuxMafia | but i want to be able to try as company phone too |
| 00:36.33 | [TK]D-Fender | LinuxMafia, What do you expect out of buying a phone? |
| 00:36.34 | LinuxMafia | just want to try |
| 00:36.41 | mandd | how do I add another SIP account for an outgoing calls? |
| 00:37.09 | mandd | so that when users dial 9 before the nimber, and alternative SIP is used. |
| 00:37.16 | mandd | hi [TK]D-Fender! |
| 00:37.35 | LinuxMafia | [TK]D-Fender, i want it to work with asterisk and i can use it also as company/home phone (just want to check ) |
| 00:38.02 | LinuxMafia | [TK]D-Fender, but it gotta be in toronto |
| 00:38.05 | rob0 | haha, that was my project for the day ... an alternate SIP trunk |
| 00:38.13 | mandd | :) |
| 00:38.38 | mandd | any helpful urls ? |
| 00:38.40 | *** join/#asterisk Frogzoo (n=Frogzoo@124.184.18.213) |
| 00:38.45 | rob0 | actually mine was slightly different, yours is simpler to do |
| 00:38.53 | rob0 | (I did a failover) |
| 00:38.54 | [TK]D-Fender | LinuxMafia, waitasec.. I already answered all of this for you before... |
| 00:38.57 | mandd | i hope :) |
| 00:39.02 | mandd | yeah, not like that |
| 00:39.10 | mandd | just an alternative on demand |
| 00:39.12 | mandd | not on fail |
| 00:39.38 | rob0 | Yours is just a matter of setting different sip.conf peers and referencing those from the dialplan. |
| 00:39.40 | LinuxMafia | [TK]D-Fender, yeah but people told me that dlink is blocked |
| 00:40.03 | [TK]D-Fender | LinuxMafia, No I recall you menioning you can't order via CC, etc... |
| 00:40.11 | rob0 | What I did, keep poking around the wiki and trying things until they work :) |
| 00:40.33 | [TK]D-Fender | LinuxMafia, And I never suggested d-link for anything more than PoE switches. |
| 00:40.48 | LinuxMafia | [TK]D-Fender, yeah that is why i want it to be in toronto |
| 00:40.56 | rob0 | You can have as many SIP registers and peers as you need. |
| 00:41.03 | mandd | rob0 any samples, for sip.conf and extension.conf? |
| 00:41.15 | mandd | i sorta have an idea how it should work |
| 00:41.17 | rob0 | sure, at the wiki, lots. |
| 00:41.19 | LinuxMafia | [TK]D-Fender, i have to go to the store by person , other wise i can not buy |
| 00:41.52 | [TK]D-Fender | LinuxMafia, and I gave you a link you could take the friggen Metro too and you didn't write this shit down. |
| 00:42.14 | [TK]D-Fender | LinuxMafia, I spoon-fed you all of this and you can't even seemed to be bothered to write it down. |
| 00:42.25 | LinuxMafia | [TK]D-Fender, you gave me link for canadacomputers right? |
| 00:42.44 | [TK]D-Fender | LinuxMafia, Go read some logs. |
| 00:43.15 | LinuxMafia | [TK]D-Fender, and that was some hardware that changes the ordinary phone to voip phone |
| 00:43.19 | LinuxMafia | i remember that |
| 00:43.27 | LinuxMafia | and that device was dlink |
| 00:43.34 | [TK]D-Fender | LinuxMafia, No, it WASN'T. |
| 00:44.04 | LinuxMafia | i remember it was canadacomputers |
| 00:44.10 | LinuxMafia | i checked it out |
| 00:44.23 | LinuxMafia | there was a branch close to me |
| 00:44.31 | mandd | rob0 any urls to those wiki samples? |
| 00:44.35 | mandd | i cant seem to find any |
| 00:44.38 | LinuxMafia | and i even called them |
| 00:44.41 | mandd | for alternative sip trunks |
| 00:45.14 | vector | mandd? |
| 00:45.18 | vector | THE mandd? |
| 00:45.21 | mandd | hahahah |
| 00:45.24 | mandd | vec |
| 00:45.26 | mandd | no way |
| 00:45.30 | jeffspeff | hey, i'm trying to do sipp to test... when i run the "./sipp -sn uas -p 5061 -mp 6001" command i get this error > Unable to bind main socket, errno = 98 (Address already in use). |
| 00:45.33 | vector | what are the chances |
| 00:45.44 | mandd | i thought you said you didnt know # |
| 00:46.11 | jeffspeff | what am i doing wrong? |
| 00:46.47 | JT | LinuxMafia: do you have a phobia of online shopping? |
| 00:47.07 | LinuxMafia | JT, nothing for online shopping |
| 00:47.22 | JT | it's easy to buy ip telephones online |
| 00:47.31 | LinuxMafia | JT, and what is the point it gonna take a week until you get your stuff |
| 00:47.33 | jblack | I wouldn't think a distrust of online retailers as unreasonable |
| 00:47.33 | rob0 | mandd: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 00:47.42 | JT | LinuxMafia: err then at least you get it |
| 00:48.52 | mandd | awesome |
| 00:49.05 | mandd | thanks rob0 |
| 00:51.53 | JT | LinuxMafia: how long have you been shopping for ip phones? |
| 00:52.05 | jeffspeff | can anybody help me with sipp ? |
| 00:52.28 | [TK]D-Fender | JT : read above. |
| 00:52.59 | LinuxMafia | JT, first time |
| 00:53.22 | JT | [TK]D-Fender: i didn't see a timeframe |
| 00:53.27 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-2b9a3d1cff348c14) |
| 00:53.40 | JT | LinuxMafia: today is the first day ever? |
| 00:54.01 | [TK]D-Fender | JT : in this past week |
| 00:54.14 | JT | [TK]D-Fender: lol |
| 00:54.25 | LinuxMafia | no |
| 00:54.46 | [TK]D-Fender | JT : living prrof that you lead a horse to the water, but the SPCA won't let you hold its head under... |
| 00:55.08 | JT | hehe |
| 00:57.15 | LinuxMafia | brb |
| 00:58.45 | *** join/#asterisk mogorman (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:58.45 | *** mode/#asterisk [+o mogorman] by ChanServ |
| 01:01.59 | rob0 | was just RoIP'ed (rickrolled over IP) |
| 01:02.10 | mwalling | has a new way to... yeah, i rickrolled rob0 |
| 01:02.15 | rob0 | This is a tough neighborhood |
| 01:07.32 | drmessano | IRRX <-- InterRickRollXchange |
| 01:07.56 | mwalling | heh |
| 01:08.00 | drmessano | IRRX is an XML based format used for the open exchange of Rickrolls |
| 01:08.25 | drmessano | Any app that can process IRRX can properly process a rickroll |
| 01:08.40 | rob0 | It's pronounced "Irrix" like "irritating" |
| 01:08.46 | drmessano | Regardless of programming language and operating platfrom |
| 01:08.53 | drmessano | platform too |
| 01:08.54 | mogorman | heh |
| 01:09.03 | drmessano | No |
| 01:09.12 | mwalling | needs to add in answering machine detection |
| 01:09.14 | drmessano | IRRX like "Irks" |
| 01:09.19 | rob0 | ah yes |
| 01:09.30 | mwalling | i left my cellphone a voicemail |
| 01:09.40 | drmessano | mwalling: What a pathetic life you lead |
| 01:09.54 | drmessano | heh, sorry.... you were wide open |
| 01:09.59 | mwalling | drmessano: just from 1700 to 0800 |
| 01:10.09 | rob0 | Don't be sorry, it's true. |
| 01:10.26 | drmessano | Oh, so it's ok to call him a douchebag? |
| 01:10.30 | drmessano | Cool |
| 01:10.33 | drmessano | takes notes |
| 01:10.43 | rob0 | starts setting up a 7125 extension |
| 01:11.12 | mwalling | files.markwalling.org/rickroll.gsm |
| 01:11.29 | drmessano | I cant wait for the Cisco 9000 |
| 01:12.09 | drmessano | "Danny, you just missed a call.. it didn't seem important, so I hung up on them" |
| 01:12.11 | rob0 | Hey, I have a real albeit offtopic question; what do I need for PoE? What do I search for at newegg? Or is it cheaper on a small scale to just buy AC adapters? |
| 01:12.22 | drmessano | Power and Ethernet |
| 01:12.27 | drmessano | and the Power must run over Ethernet |
| 01:12.39 | rob0 | "power ethernet" turns up a lot |
| 01:12.40 | drmessano | PoE is too expensive for small scale |
| 01:12.48 | drmessano | 120V AC on pins 2 and 3 |
| 01:12.50 | drmessano | No |
| 01:12.53 | rob0 | ok that's pretty much what I figured |
| 01:12.56 | mwalling | *blink* |
| 01:13.14 | mwalling | drmessano: PoE done right is expensive. PoE done like a hick is cheap |
| 01:13.27 | drmessano | We use PoE injectors at work for 8 devices or less |
| 01:13.31 | rob0 | hmmm |
| 01:14.07 | drmessano | I've made a freakin art of wire management and mounting of Cisco PoE injectors |
| 01:15.02 | [TK]D-Fender | rob0, Since no-one asked.... how many PoE ports are you looking for? |
| 01:15.14 | rob0 | just a couple at nost |
| 01:15.16 | rob0 | most |
| 01:15.25 | [TK]D-Fender | rob0, a number please... |
| 01:15.47 | drmessano | at-nost, is Latin for "Not enough to buy a PoE swi.. holy crap that's expensive" |
| 01:16.08 | drmessano | Couple being 4 or 5? |
| 01:16.11 | drmessano | 3 or 4? |
| 01:16.16 | rob0 | well, not sure if this old ATA would support it, so probably only one for the foreseeable future. |
| 01:16.16 | drmessano | 5 or 6 is a handful |
| 01:16.37 | [TK]D-Fender | rob0, I've never heard of an ATA that runs off PoE |
| 01:16.59 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 01:16.59 | rob0 | I was looking at a polycom |
| 01:17.03 | rob0 | 330 |
| 01:17.19 | rob0 | I don't think it comes with AC adapter |
| 01:17.43 | [TK]D-Fender | rob0, Ok, you don't seem to be reading from the same program as the rest of us. how MANY are you looking at? please narrow down the range and provide actual NUMBERS. |
| 01:17.55 | drmessano | He said ONE |
| 01:18.03 | rob0 | yup |
| 01:18.19 | [TK]D-Fender | 1 is a divorcee, 2 is a couple :p |
| 01:18.21 | drmessano | Two if his ATA supports it |
| 01:18.29 | rob0 | which it does not |
| 01:18.39 | [TK]D-Fender | rob0, just buy the power brick if needed. |
| 01:18.41 | drmessano | What kind of ATA? |
| 01:18.57 | rob0 | old Sipura spa2000 |
| 01:19.03 | rob0 | pre linksys |
| 01:19.14 | [TK]D-Fender | rob0, no, they don't |
| 01:20.08 | [TK]D-Fender | Yay, China Roby is back and advertising on the WIKI. Here's hoping they don't go spamming it again... |
| 01:20.20 | [TK]D-Fender | *cough*cheapcrap*cough* |
| 01:20.51 | ManxPower | I thought 1 is happy, 2 is a couple |
| 01:21.13 | rob0 | That Polycom, US$138 shipped, but I'd need the power brick too: http://www.newegg.com/Product/Product.aspx?Item=N82E16876129004 |
| 01:21.14 | [TK]D-Fender | ManxPower, that too. |
| 01:21.43 | [TK]D-Fender | rob0, For your needs I might suggest an IP 501 in its place |
| 01:22.13 | [TK]D-Fender | rob0 : well... actually.. that is shipped... |
| 01:22.20 | ManxPower | All polycoms my customers have purchased over the past 2 years came with a power supply in the box. |
| 01:22.43 | ManxPower | Maybe they just didn't order the part number that does not include the AC adapter, I don't know. |
| 01:22.55 | jeffspeff | i'm trying to test my server with sipp... i can get the test to run, but i'm getting a few errors, does anybody have experience with sipp? or is there another suggestion for stress testing my server? |
| 01:23.02 | gitguy | what ip based phone should i buy? |
| 01:23.14 | ManxPower | ~phones |
| 01:23.14 | 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 phones. Ever. places like such as |
| 01:23.32 | ManxPower | or maybe, since he has seen the info a dozen times already... |
| 01:23.34 | ManxPower | ~troll |
| 01:23.34 | jbot | i guess troll is a race on some muds; a guy under a bridge: an annoying robot; a port scanner, or somebody faking being clueless to be shown the One Linux Way so he can argue against it for his psychology thesis on linux advocates, or an employee of trolltech, or at http://www.kuro5hin.org/story/2001/7/27/51233/2979 or http://www.catb.org/~esr/jargon/html/entry/troll.html |
| 01:24.25 | gitguy | i wasnt reading the channel |
| 01:24.56 | *** join/#asterisk dFence (n=chatzill@p549810F8.dip0.t-ipconnect.de) |
| 01:25.01 | jeffspeff | ManxPower, any suggestions or ideas? |
| 01:25.05 | dFence | grrr.. sorry guys |
| 01:25.23 | ManxPower | jeffspeff: My idea is that 99% of Asterisk people don't understand SIP well enough to use SIPP to do anything. |
| 01:26.09 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 01:26.13 | dFence | hey, is it possible to include a bash-command in the sip.conf? want to include the MD5Secret in the template for the sip-channels so i dont have to run echo "xyz" | md5sum 100 times |
| 01:26.20 | jeffspeff | ManxPower, I'm wanting to stress test my system to see how man concurrent calls it can handle... i asked in here earlier and sipp was suggested by 2 people that are no longer in the room... |
| 01:26.22 | florz | ManxPower: so few? you are including the developers? |
| 01:26.48 | LinuxMafia | ~phones |
| 01:26.49 | jbot | from memory, 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 phones. Ever. places like such as |
| 01:26.51 | jeffspeff | ManxPower, how would suggest testing # of concurrent calls? |
| 01:28.30 | ManxPower | jeffspeff: No idea. I never have to worry about it. |
| 01:28.49 | dFence | oook..... maybe i'll wish i never asked that question but... apart from being absolutely hideous - what's it with GrandStream phones? |
| 01:29.04 | [TK]D-Fender | dFence, Cheap crap as the comments allude to. |
| 01:29.10 | rob0 | Thanks for the suggestions ( drmessano & [TK]D-Fender ) |
| 01:29.37 | jeffspeff | does anybody else have any ideas of how to stress test? i want to find how many concurrent calls my system can take... |
| 01:29.55 | ManxPower | jeffspeff: I live in a corporate environment where, in the unlikely event of a system not being able to handle the load they just buy a much faster system -- cheaper than paying someone to spend days trying to figure out wher4 performance can be improved. In a service provider enviroment, of course, things are totally different. |
| 01:29.55 | dFence | wasn't there sth in the source-folder!? |
| 01:30.09 | rob0 | irrx://you.dontlike.us/ |
| 01:30.19 | [TK]D-Fender | jeffspeff, jfgi |
| 01:30.37 | mwalling | heh |
| 01:30.45 | jeffspeff | [TK]D-Fender, jfgi ? what is that? |
| 01:30.49 | [TK]D-Fender | ~jfgi |
| 01:30.50 | jbot | http://www.google.com/search?q=jfgi |
| 01:30.55 | jeffspeff | lol |
| 01:30.56 | rob0 | jfgi.us |
| 01:31.08 | ManxPower | dFence: Grandstream has the buggiest firmware this side of SIP/Wifi Phones. |
| 01:31.11 | rob0 | oops, nm |
| 01:31.36 | jeffspeff | rob0, the link you posted is dead |
| 01:31.51 | rob0 | yes I see that :) |
| 01:31.51 | dFence | ManxPower: aight, thx.. |
| 01:31.54 | ManxPower | dFence: They have had YEARS to fix the problems. To this day, you basically have to keep trying different versions of the firmware until you find one that works for you and does not crash in your enviroment. |
| 01:31.55 | mwalling | <PROTECTED> |
| 01:32.01 | mwalling | my site is dead? |
| 01:32.06 | rob0 | 01:30 < rob0> jfgi.us |
| 01:32.13 | ManxPower | They hardware is nothing special |
| 01:32.16 | rob0 | Used to belong to ARob. |
| 01:32.28 | rob0 | in fact it was hosted on this machine. |
| 01:32.28 | mwalling | ah |
| 01:32.30 | jeffspeff | [TK]D-Fender, i get it... nice one. :p |
| 01:32.47 | dFence | ManxPower: ok, think i got it... will stay away from them - promise :) |
| 01:32.54 | [TK]D-Fender | jeffspeff, now lift your skirt, grab your balls, AND MAN UP! |
| 01:33.08 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-edaa0dfe028d2ff4) |
| 01:45.47 | *** join/#asterisk hfb (n=hfb@cpe-76-87-167-79.socal.res.rr.com) |
| 01:51.10 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58-8-2-30.revip2.asianet.co.th) |
| 01:51.47 | HaMYaI | I got a "YELLOW/RED" alarm on my Tor2. What does it indicate? |
| 01:51.48 | *** join/#asterisk shido6 (n=shido6@74-130-51-159.dhcp.insightbb.com) |
| 01:52.48 | *** join/#asterisk freezey (n=freezey@c-68-36-242-87.hsd1.nj.comcast.net) |
| 01:52.58 | drmessano | Red/Yellow usually means "I guess i'll sleep tomorrow night" |
| 01:53.07 | freezey | if i was trying to build a medium size call center thats scalable... what specs on a asterisk machine should i go for? |
| 01:53.10 | [TK]D-Fender | HaMYaI, That you desperately should be considering modern hardware. |
| 01:53.44 | [TK]D-Fender | freezey, everything is scalable so log as you aren't using it to its potential. |
| 01:54.07 | [TK]D-Fender | freezey, please get SPECIFIC for your starting point, and ENDING points. |
| 01:55.44 | *** join/#asterisk BeeBuu (n=beebuu@218.13.81.208) |
| 01:56.06 | freezey | [TK]D-Fender: starting point is 1 t1 line coming from a trunk with an 800 number and trunk size is roughly 50 lines and ending point would be a voip solution that can start off with roughly 20 people and expand to possibly 500 |
| 01:56.55 | [TK]D-Fender | freezey, now try to put solid terms to those numbers an describe usage and maybe we'll be able to suggest something. |
| 01:57.21 | freezey | [TK]D-Fender: lets say its a 24hr service center and want to be able to support 200+ concurrent calls |
| 01:57.42 | [TK]D-Fender | freezey, All over T1? |
| 01:57.52 | freezey | [TK]D-Fender: yeah |
| 01:58.46 | [TK]D-Fender | freezey, ok, well if we're talking local, the calls themselves aren't too serious. HWEC required, transcoding should be avoided, then it comes down to call recording. |
| 01:59.18 | freezey | [TK]D-Fender: calls could be long distance as well... and call recording capabilities are a must.. |
| 01:59.59 | [TK]D-Fender | freezey, fast big RAID array, native codec recording recommended. |
| 02:01.56 | freezey | [TK]D-Fender: so say like a dell poweredge 2950 with about 5 drives with a raid 5 array |
| 02:02.22 | [TK]D-Fender | freezey, Aim for RAID 6. |
| 02:02.29 | freezey | [TK]D-Fender: slap asterisk on their and grab a t1 digium card with all polycom phones? |
| 02:02.54 | [TK]D-Fender | freezey, I'd use a Sangoma A108d in this case. |
| 02:03.45 | JT | i'd use 2 systems and a 4 port card in each |
| 02:03.49 | ManxPower | You would need 8.7 T-1s for 200 calls |
| 02:03.50 | dFence | *garrr* now that the test-server is set up, my isdn-card fucks around... am using a fritcard pci via chan_capi... modprobe capi throws a kernel-oops, and since prolly 2 hours it just won't work anymore... no specific errormessage, just kcapi: appl 2 ncci 0x10101 down in the syslog |
| 02:04.15 | freezey | and that supposed 4 t1's? what would i need all those for just for future scalability? |
| 02:04.53 | freezey | JT: why do you say 2 systems? for redundancy? |
| 02:05.00 | JT | 200 calls is 8.7 t1 PRIs as ManxPower said |
| 02:05.04 | JT | yeah |
| 02:05.05 | [TK]D-Fender | freezey, when you say "call center" and simultaneous calls, we're figuring thats all LINES coming in |
| 02:05.07 | ManxPower | freezey: A PRI supports up to 23 channels. YOU do the bath |
| 02:05.17 | [TK]D-Fender | ... I feel dirty... |
| 02:05.18 | Strom_C | the bath, eh? |
| 02:05.32 | freezey | haha |
| 02:06.02 | *** join/#asterisk HighOctane (n=HighOcta@68-185-143-114.dhcp.jcsn.tn.charter.com) |
| 02:06.08 | freezey | so a dell 2950 / raid 6 / signoma card / asterisk for application and i should be pretty good? |
| 02:06.15 | freezey | and if anything get another machine for redundancy |
| 02:06.17 | ManxPower | freezey: does your call volume go thru spikes? Is it OK for callers to get a busy? |
| 02:06.27 | jeffspeff | could somebody tell me if this phone is IP or not... it doesn't say... http://cgi.ebay.com/Polycom-Soundpro-Office-Speakerphone_W0QQitemZ160237773164QQihZ006QQcategoryZ61835QQssPageNameZWDVWQQrdZ1QQcmdZViewItem thanks. |
| 02:06.48 | freezey | ManxPower: busy signal would be no good... |
| 02:07.04 | ManxPower | jeffspeff: it is not. Soundpoint IP is what you are looking for. |
| 02:07.37 | jeffspeff | ManxPower, thanks... :) |
| 02:07.46 | ManxPower | freezey: if you do the research and manage expectations, you could get fewer PSTN channels, enough to handle the call volume 90% of the time, busyover/failover to using VoIP to handle the overflow |
| 02:08.51 | HighOctane | errr. You there? |
| 02:08.52 | JT | freezey: you definitely want a minimum of 2 machines |
| 02:09.00 | freezey | hmm |
| 02:09.12 | freezey | yeah def gonna need something for redundancy |
| 02:09.17 | ManxPower | Just don't expect VoIP to be as reliable as PSTN/PRI |
| 02:09.55 | *** join/#asterisk profounded (n=Bryan_Ru@pool-71-242-14-39.phil.east.verizon.net) |
| 02:10.09 | freezey | yeah see that what i figured... i wanted to go all pstn but they want voip |
| 02:10.50 | ManxPower | freezey: A mix of PSTN/VoIP always is better than only using one or the other |
| 02:10.51 | errr | HighOctane: yes |
| 02:11.28 | ManxPower | freezey: you would have to work with your carrier to get calls to your PRIs to be forwarded to your VoIP service when the PRIs are full |
| 02:11.32 | jeffspeff | ManxPower, why is VOIP not as reliable? |
| 02:11.47 | JT | gee i wonder |
| 02:11.48 | jeffspeff | if you go full voip... |
| 02:11.49 | freezey | ManxPower: yeah thats what i am currently pitching out there... i just want to put together this asterisk solution and make it cheaper than nortel's, dell's, and avaya's |
| 02:11.58 | ManxPower | jeffspeff: because it uses the internet |
| 02:12.07 | JT | voip providers are not at the same level as telcos |
| 02:12.16 | HighOctane | errr: The other day, you tried initiating a blind transfer from your cell phone. I have not been able to get mine to work. Could you verify some settings for me? |
| 02:12.26 | errr | sure |
| 02:12.32 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 02:12.32 | JT | the only way to come close to TDM is to have dedicated data circuits going direct to your voip provider |
| 02:12.43 | HighOctane | errr: How many trunks you have? |
| 02:12.51 | errr | HighOctane: 3 |
| 02:13.07 | HighOctane | errr: What are your dtmf settings on those trunks? |
| 02:13.44 | HighOctane | errr: What is your dtmfmode set to on the extension which you were connected through when you got it to work? |
| 02:13.52 | jeffspeff | ~phones |
| 02:13.53 | jbot | well, 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 phones. Ever. places like such as |
| 02:13.55 | ManxPower | ~trunk |
| 02:13.56 | jbot | i guess trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 02:14.48 | freezey | ManxPower: i know the solution will work i just gotta set this up |
| 02:14.58 | jer | hrmm... trying to think of something clever to play with on my * tonight |
| 02:15.01 | jer | suggestions? |
| 02:15.37 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582051.dsl.bell.ca) |
| 02:16.13 | ManxPower | freezey: The system you are going to have to build will be incredibly complex. |
| 02:16.26 | errr | HighOctane: dtmfmode=rfc2833 on all of them |
| 02:17.05 | HighOctane | errr: What type of internet connection do you have? |
| 02:17.14 | HighOctane | errr: Are you behind a router? |
| 02:17.21 | ManxPower | jer: write a macro that acts like "saydigits", but supports chars in the string to indicate pauses, etc. i.e. "1w504w555w1234" |
| 02:18.13 | errr | HighOctane: I have a business class cable connection, and yes I have a router, but I have all the ports forwarded to the pbx |
| 02:18.38 | freezey | ManxPower: how complex do you think its going to get? |
| 02:18.50 | freezey | ManxPower: and when you say complex with which parts? |
| 02:18.56 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 02:20.06 | HighOctane | errr: I have "consumer-grade" cable connection. However, my machine is a vmware virtual machine with asterisk running on a quad core AMD phenom. You think this could be a latency issue? It worked once on a land line, but I can't get it to work since, especially on cell phones. My xLite sip phones work file though. |
| 02:20.47 | dFence | if hthere's something as justice in this world i WON'T have to compile a new kernel!!!!!!!!!! |
| 02:21.02 | errr | HighOctane: no idea. I have never had great experince with vmware and * |
| 02:21.48 | freezey | HighOctane: are you using the free version of vmware? |
| 02:21.51 | errr | once I got past a proof of concept with asterisk I took it out of vmware and gave it its own hardware |
| 02:24.20 | errr | I kind of went all out on my home system too |
| 02:24.22 | JT | does the free version of vmware do hardware virtualisation? |
| 02:24.34 | freezey | no |
| 02:24.39 | errr | pIII 500mhz with 512 RAM |
| 02:24.43 | freezey | and the free version limits you to 2 processors |
| 02:25.40 | HighOctane | freezey: VMWare 6 Professional |
| 02:26.30 | freezey | ahh |
| 02:26.32 | freezey | nm then |
| 02:26.56 | JT | errr: that's going all out? |
| 02:27.28 | freezey | i wouldnt run a production asterisk system on a vmware machines tho... its kind of a critical app if your running prod |
| 02:27.28 | HighOctane | Freezey: Correction VMWare 6 Workstation |
| 02:27.51 | drmessano | PIII 500MHZ is going "All out"? |
| 02:28.05 | errr | oh you know it |
| 02:28.08 | HighOctane | Freezey: I'm just about ready to go into production. I have a P4 2.4ghz with 512MB Ram. Guess I should go ahead and migrate, huh? |
| 02:28.21 | drmessano | I throw away anything below 1 GHZ |
| 02:28.34 | freezey | HighOctane: i would suggest that maybe throw the least amount a gig in that machine |
| 02:28.34 | HighOctane | errr: PIII 500 works well? |
| 02:28.35 | errr | drmessano: throw it my way |
| 02:28.39 | freezey | lol |
| 02:28.44 | errr | HighOctane: for my house it works fine |
| 02:29.26 | drmessano | lol |
| 02:29.38 | errr | I use it to filter any call that isnt my parents, my office or the wifes parents or her office straight to voicemail |
| 02:30.00 | drmessano | I can buy a used 1GHZ machine with 256MB Ram and a 20GB HD for $49 with no OS |
| 02:30.10 | jbeez | nice |
| 02:30.11 | errr | then it uses followme to send calls from her folks to her cell and from my parents to my cell |
| 02:30.15 | drmessano | Not worth it to save anything less |
| 02:30.17 | jbeez | sounds like a firewall winner |
| 02:31.15 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:31.21 | drmessano | You get to the point where you need to decide "Am I really going to use this 700MHZ AMD DURON box, or am I still in 2001? |
| 02:31.30 | errr | my firewall/router box here is a pII machine with 128M of RAM |
| 02:32.17 | JT | i beat that |
| 02:32.26 | HighOctane | errr: Sounds cool. I've only been doing this about two weeks tops. Got my system setup with a conference room, extensions with follow-me, a "callback" service, and a DISA for long distance. It's amazing how much you can do with asterisk. |
| 02:32.30 | Qwell | I've got that beat *cold* |
| 02:32.30 | JT | my firewall/router is a Pentium 166MHz |
| 02:32.33 | JT | 64MB ram |
| 02:32.38 | drmessano | lol |
| 02:32.41 | Qwell | My firewall is a SparcStation 5, 110mhz |
| 02:32.52 | errr | awesome |
| 02:32.52 | drmessano | My first linux firewall box was a P133MHZ with 32MB Ram |
| 02:33.06 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 02:33.10 | Qwell | the quad NIC is...sbus. |
| 02:33.20 | Qwell | 1995's finest |
| 02:33.23 | drmessano | ROFL |
| 02:33.24 | errr | =) |
| 02:33.24 | HighOctane | All: On another subject: Anyone here know the ins and outs of how conference bridging works. I can tie in 4 or 5 people without any increase in bandwidth, as if the calls were not "going through" my pbx. I can't figure that one out. |
| 02:34.13 | JT | my firewall still outperforms a lot of brand new cheap embedded units |
| 02:34.49 | Qwell | mine outperforms like... I don't know |
| 02:34.51 | jaytee | that's cuz embedded units never get out of bed so they don't get as much exercise |
| 02:34.54 | Qwell | maybe a windows 3.1 firewall |
| 02:34.57 | drmessano | I got rid of the full PC firewall boxes when I found a WRT54G with some other firmware would do the trick |
| 02:35.30 | HighOctane | drmessano: The WRT54G? Using DD-WRT? |
| 02:35.42 | Qwell | I actually don't use the SS5 anymore. My router/firewall is a Digium S800i :D |
| 02:35.44 | errr | drmessano: I will get one of those when my router box dies, but I cant bring my self to throw it away since it works |
| 02:35.45 | drmessano | OpenWRT right now.. switching to DD-WRT |
| 02:35.58 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 02:35.58 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:37.28 | drmessano | Theres just some things I WANT to admin, and some things I want to be a stupid end-user with.. I decided my firewall box was stupid to spend time on, but I wasnt going to use the stock firmwares either |
| 02:37.50 | drmessano | OpenWRT has so far done the trick, and I think dd-wrt will dumb it down even more and still rock |
| 02:37.57 | jbeez | some guy was just complaining about his poor transfer speeds through his wrt w/ ddwrt installed on it, he was getting like 25mbit |
| 02:38.13 | jbeez | I was helping him troubleshoot the problem, we came to the conclusion it was the limitation of his hardware |
| 02:38.24 | drmessano | heh |
| 02:38.29 | drmessano | I'd like to have that problem |
| 02:39.35 | jbeez | from his description, he had a cable modem account with multiple public ips, several boxes "outside" of the dd-wrt plugged into a switch using public ips, and his lan computers behind the dd-wrt and when he would transfer the files between his lan computers and the servers in his little dmz as he called it, he would get these janky transfer rates |
| 02:39.35 | drmessano | It's only been in the last year or so that Comcast was fast enough to bottleneck a Wireless B box |
| 02:40.01 | drmessano | Ah |
| 02:40.12 | drmessano | Sounds like he was using way too many boxes for the job anywway |
| 02:40.30 | drmessano | A VLAN using 5 routers is not a VLAN |
| 02:40.57 | jbeez | he didn't really have a vlan, he had an unmanaged switch with those servers plugged into it and the cable modem, and the wan interface of the ddwrt |
| 02:41.14 | jbeez | and home computers i guess behind the ddwrt box, and these boxes outside were hosting things, website, maybe email |
| 02:41.44 | drmessano | There's an x86 image of DD-WRT as I understand it |
| 02:41.50 | jbeez | the best part about it though, |
| 02:42.29 | jbeez | I was in his channel last year, and he ended up kicking me out because I was critizing some butcher job he did on some cat5, and i had a big argument on how to properly terminate it, and last week he strolls into #cisco on undernet and is asking me to help him with his network |
| 02:42.31 | *** join/#asterisk gmaruz1 (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 02:44.44 | drmessano | ROFL |
| 02:45.39 | HighOctane | Anyone know the difference between DTMF= and DTMFMODE= in the trunk settings PEER details. |
| 02:51.07 | [TK]D-Fender | HighOctane, yes. The latter is actually a legitimate options. |
| 02:52.12 | HighOctane | Are the two settings the same? |
| 02:53.35 | JT | HighOctane: read what he said carefully :P |
| 02:54.08 | HighOctane | lol |
| 02:56.27 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 02:56.50 | *** join/#asterisk ghostrdr (i=ghostrdr@c-68-80-149-157.hsd1.pa.comcast.net) |
| 03:01.31 | [TK]D-Fender | Once again well hidden in the BIG PRINT. |
| 03:02.12 | HighOctane | Good night all. |
| 03:05.35 | ftp3 | are there any voip providers that allow unlimited usa outgoing calls ? |
| 03:06.51 | *** join/#asterisk scubasteve (n=steve@cpe-066-026-095-197.nc.res.rr.com) |
| 03:07.28 | [TK]D-Fender | ftp3, depending on a certain point of view, yes. Go research : |
| 03:07.31 | [TK]D-Fender | ~itsplist-us |
| 03:07.32 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 03:07.47 | scubasteve | I'm trying to configure FOP with 1.4. I see all of the IP addresses for the extensions, but the flash isn't showing in use (everything is always green)... pulling hair out... anyone have any advice? |
| 03:08.05 | luke-jr | no les.net on jbot? |
| 03:08.14 | [TK]D-Fender | luke-jr, let.net = canadian |
| 03:08.17 | [TK]D-Fender | ~itsplist-ca |
| 03:08.18 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 03:08.20 | luke-jr | oh? |
| 03:08.21 | [TK]D-Fender | les* |
| 03:08.29 | scubasteve | How about Gafachi on the ITSP list? |
| 03:08.38 | luke-jr | [TK]D-Fender: they sure have US DIDs :þ |
| 03:08.42 | [TK]D-Fender | luke-jr, http://www.les.net/contact.php |
| 03:08.52 | ghostrdr | i have a usa DID from les.net |
| 03:08.57 | ghostrdr | works pretty good |
| 03:08.59 | *** join/#asterisk bintut (n=bintut@cm199.omega112.maxonline.com.sg) |
| 03:09.05 | ghostrdr | 3.99 a month flat for inbound 2 channels |
| 03:09.07 | luke-jr | wonders why les.net has a 4-channel cap on per-minute DIDs |
| 03:09.21 | ghostrdr | but term is perminute |
| 03:09.31 | ghostrdr | i dont use for term anyway |
| 03:09.35 | ftp3 | yeah, i want term |
| 03:09.37 | [TK]D-Fender | luke-jr, Yes, but when you want a good german sausage you clearly must be shopping in Dover, Delaware ;) |
| 03:09.40 | ghostrdr | but inbound works well ,just my 2 cents |
| 03:09.40 | Ritzerisk | or does anyone know of a type of asterisk system that can use the auto dialer |
| 03:09.43 | ftp3 | or at least a lot of min for a rate |
| 03:09.51 | [TK]D-Fender | ftp3, GO READ |
| 03:09.52 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 03:09.58 | teknoprep | does asterisk run well inside of qemu ? |
| 03:10.04 | ftp3 | i have been :-) |
| 03:10.07 | luke-jr | ftp3: Voipjet works great, if you qualify as wholesale |
| 03:10.12 | ftp3 | thought someone might has a suggestion |
| 03:10.14 | ftp3 | thank you |
| 03:10.17 | luke-jr | teknoprep: are you joking? |
| 03:10.25 | teknoprep | luke-jr, no why ? |
| 03:11.31 | luke-jr | teknoprep: Asterisk runs well with a dedicated box doing nothing else |
| 03:11.58 | teknoprep | luke-jr, so if i have a 32 CPU server running vmware-server or Xen |
| 03:12.09 | teknoprep | luke-jr, using asterisk on that would be crazy |
| 03:12.31 | luke-jr | teknoprep: would it? |
| 03:12.47 | luke-jr | teknoprep: if you don't have high volume for 32 CPUs, then I'd go OpenVZ on it |
| 03:12.50 | teknoprep | luke-jr, you just told me that it would be craz pretty much |
| 03:13.02 | luke-jr | emulators are just going to add too much latency |
| 03:13.54 | JT | luke-jr: not if you have hardware virtualisation |
| 03:14.22 | luke-jr | JT: last I checked, hardware virtualization was still slower than VMWare's software, and on top of that, the virtual hardware is still emulated |
| 03:14.27 | luke-jr | eg, the network card |
| 03:14.29 | JT | teknoprep: it should work fine inside kvm/qemu |
| 03:14.55 | JT | luke-jr: i doubt that |
| 03:15.02 | luke-jr | also, just having another link in the route is always going to add some latency |
| 03:15.10 | JT | what link? |
| 03:15.17 | luke-jr | VM->virtual NIC->host virtual NIC->real host NIC->router |
| 03:15.30 | luke-jr | instead of VM->real host NIC->router |
| 03:15.35 | teknoprep | luke-jr, paravirtualization |
| 03:15.46 | JT | luke-jr: negligible |
| 03:15.51 | luke-jr | teknoprep: there's still a virtual NIC in paravirt |
| 03:15.56 | JT | hardware virtualisation is very fast |
| 03:16.02 | teknoprep | yes it is |
| 03:16.09 | JT | much faster than uml/old vmware |
| 03:16.13 | luke-jr | sticks to OpenVZ since it has no overhead. |
| 03:16.31 | JT | and not that much flexibility either |
| 03:16.39 | JT | it's great if you have a hetrogenous environment |
| 03:16.46 | Ritzerisk | i was going to just do that similar setup but use a real nic instead in a vmware envirorment |
| 03:17.02 | JT | the nic is not the big issue anyway |
| 03:17.07 | luke-jr | Ritzerisk: if VMWare supports passing a real NIC to a VM, it's news to me |
| 03:17.19 | luke-jr | JT: makes a big difference in Xen vs OpenVZ in my experience |
| 03:17.24 | Ritzerisk | i just got a quad nic with the power edge |
| 03:17.32 | JT | xen in what mode? |
| 03:17.42 | luke-jr | JT: no idea, I didn't know there were multiple modes |
| 03:17.46 | JT | sure |
| 03:17.51 | JT | there's full virtualisation |
| 03:17.58 | JT | that is slow, but not near as slow as uml |
| 03:18.03 | JT | there's paravirtualisation |
| 03:18.06 | JT | that is fast |
| 03:18.07 | luke-jr | UML != OpenVZ |
| 03:18.20 | Ritzerisk | i hear virtualization uses resources better then hardware because its like maxing but i could be wrong.... |
| 03:18.26 | JT | xen, kvm and vmware esx can do paravirtualisation |
| 03:18.29 | teknoprep | full virtualization in Xen with Intel-V chips is NOT slow |
| 03:18.36 | JT | luke-jr: didn't say thery were identical |
| 03:18.39 | JT | teknoprep: that's paravirt |
| 03:18.43 | JT | teknoprep: not full |
| 03:18.49 | JT | full means no processor support |
| 03:19.20 | JT | i've been using kvm with amd-v cpus and it's fast |
| 03:19.29 | teknoprep | so virt-manager is using the wrong terminology ? |
| 03:19.37 | danp | JT: you have it backwards |
| 03:19.58 | luke-jr | teknoprep: OpenVZ is best ☺ |
| 03:19.58 | JT | danp: how so? |
| 03:20.06 | JT | luke-jr: only in some situations |
| 03:20.31 | luke-jr | JT: in situations where you need performance |
| 03:20.35 | JT | ... |
| 03:20.43 | luke-jr | sure, kernel hackers need qemu |
| 03:20.44 | JT | stop with this blind praise |
| 03:20.53 | Ritzerisk | i got an account to where this customer wants the phone to dial about 5 calls every second and give the callers a recording and he has a list of like 100 thousand numbers to start with.... |
| 03:20.53 | Ritzerisk | <drmessano> Hmmm |
| 03:21.10 | JT | hardware virtualisation is quite fast, and is way more flexible if not all guests can be running the same os/distro |
| 03:21.15 | danp | when you say 'full means no processor support', do you mean it doesn't require VT-x or AMD-V? |
| 03:21.32 | luke-jr | JT: OpenVZ works fine with multiple OS, as long as they're all Linux |
| 03:21.44 | JT | luke-jr: that's the one OS. |
| 03:22.00 | luke-jr | JT: Linux isn't an OS, it's a kernel |
| 03:22.04 | danp | you also said full virtualization is slower than paravirtualization which i don't think is accurate |
| 03:22.09 | JT | don't get all RMS on me |
| 03:22.19 | JT | i will call it linux |
| 03:22.21 | JT | it's an OS |
| 03:22.46 | luke-jr | nope |
| 03:22.56 | luke-jr | an OS is Fedora or Debian or Gentoo |
| 03:22.59 | JT | no |
| 03:23.02 | JT | that's a distribution |
| 03:23.08 | JT | you seem to be confused |
| 03:23.12 | JT | linux is definitely the OS |
| 03:23.13 | luke-jr | then Windows XP is a distribution? |
| 03:23.19 | luke-jr | Linux is a mere kernel |
| 03:23.20 | danp | full virtualization uses VT-x/AMD-V to run on the processor; paravirtualization requires a modified OS |
| 03:23.24 | JT | GNU/Linux if you follow RMS |
| 03:23.32 | Ritzerisk | but i heard of vicidial but i dont know if it does the auto dialing |
| 03:23.37 | luke-jr | GNU could be an OS, if FSF actually finished it |
| 03:23.42 | JT | danp: ah right my mistake |
| 03:24.01 | JT | luke-jr: i will just go off what 99% of people call an OS, in this case, it's linux |
| 03:24.09 | JT | you're evading the issue anyway |
| 03:24.18 | JT | openvz allows only linux |
| 03:24.25 | JT | whatever you want to call linux. |
| 03:24.25 | luke-jr | JT: 99% of people who have no clue how computers work |
| 03:24.39 | JT | luke-jr: i'm going to pretend you just didn't say that |
| 03:24.42 | luke-jr | probably 90% of modern OS support the Linux kernel |
| 03:25.53 | [TK]D-Fender | .... |
| 03:25.56 | [TK]D-Fender | um... |
| 03:26.06 | [TK]D-Fender | luke-jr, might want to rephrase that last one. |
| 03:26.20 | luke-jr | s/support/require ? |
| 03:29.40 | drmessano | Wow |
| 03:29.59 | drmessano | I haven't heard the GNU/Linux argument since... 1997 or so |
| 03:30.11 | drmessano | Yay, my time machine is fixed! |
| 03:30.17 | JT | drmessano: his argument seems to go further than RMS's :P |
| 03:30.27 | [TK]D-Fender | argv[-1] |
| 03:30.37 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 03:30.56 | drmessano | Screw you guys, i'm headed back to 2050.. Have fun with your TELEPHONEs.. neanderthals! |
| 03:31.25 | *** join/#asterisk hd2 (n=blah22@adsl-76-195-128-35.dsl.crchtx.sbcglobal.net) |
| 03:31.26 | drmessano | JT: Further than RMS? Thats.. extreme |
| 03:31.52 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 03:31.57 | drmessano | JT: I guess the GNUish Inquistion was just in hiding |
| 03:32.16 | drmessano | NOBODY expect the GNUish Inquisition |
| 03:32.21 | JT | haha |
| 03:32.25 | hd2 | Someone help me with a switchvox issue? On IVR an IVR menu when for example 2 is pressed it is not recognizing it any clue on what might be going on? |
| 03:32.45 | drmessano | hd2? wrong channel? |
| 03:32.56 | hd2 | Which channel should I be in :) ? |
| 03:33.01 | drmessano | #switchvox |
| 03:33.06 | drmessano | or ##nothere |
| 03:33.08 | hd2 | sounds logical :) |
| 03:34.30 | hd2 | Well no one is around in there apparently :P |
| 03:35.00 | hd2 | everying is working except for the darn key presses on the IVR menus lol |
| 03:35.04 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 03:35.10 | drmessano | hd2: That stinks.. but #asterisk isn't failover for "I went to ____ and no one was there to help" :) |
| 03:35.24 | drmessano | Maybe there's some forums you can try |
| 03:35.35 | hd2 | I've been googling |
| 03:35.38 | hd2 | no much info |
| 03:35.42 | drmessano | :( |
| 03:35.45 | hd2 | I was using asterisknow before |
| 03:35.49 | hd2 | which was pretty stable |
| 03:35.56 | hd2 | tried trixbox |
| 03:36.00 | hd2 | and it kept crashing lol |
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| 05:03.28 | adeel | is it possible to dump the sip channel variables (e.g. from sip show channel foo) to a log file after each call? |
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| 05:26.57 | Bhaal | Hey guys, got a quick question.. How do I stop asterisk from sending the extension number to my ATA along with the callerID for incoming calls? My handset will only display the extension number, not the actual calling number... |
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| 06:19.18 | Maxo22 | hi, i tried to call my asterisk server from extern but i could not come through |
| 06:19.40 | Maxo22 | outgoing calls work fine, the server has a static ip |
| 06:19.52 | Maxo22 | some ideas why this happens? |
| 06:29.59 | *** join/#asterisk drbrown (n=drbrown@rrcs-24-123-237-219.central.biz.rr.com) |
| 06:30.14 | drbrown | what could cause distortion with gsm files |
| 06:38.55 | lsodi | hi, can anyone recommend call center front end for asterisk (operator panel, call statistics, recording)? |
| 06:40.13 | lsodi | maxo22: ports are opened in firewall if you have any? |
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| 06:44.03 | *** join/#asterisk vortex` (n=vortex@202-136-108-213.static.adam.com.au) |
| 06:45.52 | vortex` | So i've gone to the voxbone.com site looking into getting a VOIP number, but 1) wont tell you prices until you sign up and 2) says there's a EUR500 min charge per month! Can anyone reccomend VOIP providers with Australian numbers that doesnt suck? |
| 06:55.30 | yang | vortex`: check voip.ms they got international DIDs |
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| 07:05.02 | kannan | hello, i have the problem rported at http://bugs.digium.com/view.php?id=11141. I am building on slack 12 , kernel 2.6.25.6 latest stable. |
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| 07:06.41 | HighOctane | Anyone here using bandwidth.com? |
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| 07:20.01 | HighOctane | Anyone here? |
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| 07:46.18 | Dr-Linux | any voicemail guru? |
| 07:46.23 | Dr-Linux | aournd |
| 07:47.14 | Dr-Linux | i'm sorry i mean queues guru :) |
| 07:47.41 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
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| 08:00.13 | fcois | hello aa50 channel !!! |
| 08:00.59 | JT | this is not aa50 channel |
| 08:04.57 | fcois | Yes I know but yesterday, there was a large fun around aa50 ... |
| 08:05.09 | fcois | I know thats asterisk channel, JT |
| 08:05.19 | Dr-Linux | JT: are you aware of queues functioanlities? |
| 08:06.37 | Dr-Linux | i want to it say after every 15 minites "press 1 for continue" if the caller doesn't press 1 in 15 seconds call should be hangup? |
| 08:06.44 | Dr-Linux | how can i do that? |
| 08:12.25 | JT | lol every 15minutes, patient callers |
| 08:15.07 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 08:17.06 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 08:20.14 | Dr-Linux | JT: yes |
| 08:20.30 | Dr-Linux | JT: is there an guidance to do that? |
| 08:25.02 | arbuser | Dr Linux, do you by any chance work for a government institution? |
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| 08:32.55 | Dr-Linux | arbuser: why? |
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| 08:33.20 | arbuser | The only time I ever wait 15 minutes for anything is when dealing with the government. |
| 08:33.41 | gr0mit | hehe! |
| 08:34.31 | Dr-Linux | lol |
| 08:35.05 | fcois | in france, it is to contact the support of our internets providers |
| 08:35.26 | fcois | upto 1h |
| 08:35.36 | Dr-Linux | i want periodic announcment with input |
| 08:36.59 | arbuser | Dr Linux, I think you should code a quiz game |
| 08:37.15 | arbuser | and then the people with the highest scores get pushed to the front of the queue. |
| 08:37.29 | arbuser | Another Awesome Idea (tm) |
| 08:37.54 | Dr-Linux | :( |
| 08:39.53 | gr0mit | fastest finger first? |
| 08:39.54 | arbuser | Dr-Linux: Don't cry. |
| 08:40.20 | arbuser | gr0mit: or general knowledge |
| 08:45.42 | Dr-Linux | arbuser: just want to know , as i want is possible or not? |
| 08:45.52 | arbuser | Does anyone know of a working implementation of a "Press # to change the hold music genre"? |
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| 09:04.29 | whymarkwh | hi there i found a link elastic complete pbx is this worth having a look at? |
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| 09:12.52 | tzafrir_home | whymarkwh, it's Elastix . http://elastix.org/ |
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| 09:22.19 | SuD | hi, i'm trying to dial from console but it doesnt work (no such extension). |
| 09:22.19 | SuD | The command i tried: dial Zap/1/55512345 |
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| 09:29.00 | SuD | ok, i must create a context first then call hangup |
| 09:29.03 | whymarkwh | must be in your default contect |
| 09:29.21 | SuD | then call 55512345@mycontext |
| 09:29.48 | whymarkwh | past your extensions.conf and iwill try and help u |
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| 09:32.19 | SuD | it works now, thank you anyway |
| 09:32.44 | SuD | i installed that asterisk remotely and i didn't know the number :) |
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| 09:39.16 | Vec | Hi, is there a way I can check how often, and when last a SIP phone registers with Asterisk ? |
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| 09:59.53 | xiaoxuanzi78 | Have any one tell us the diffenrence of Tor2 and tor3 card ? |
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| 10:13.53 | Rico29 | does anybody knows how tu make auto provisioning for a cisco phone with a dhcp server ? |
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| 10:17.13 | gr0mit | Rico29, you need to specify a tftp server in the dhcp options |
| 10:21.24 | Rico29 | gr0mit> yes of course |
| 10:21.39 | Rico29 | but what the conf file looks like ? |
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| 10:24.10 | zxd | hi |
| 10:24.18 | gr0mit | it is somewhat complex! |
| 10:24.27 | zxd | how are passwords sent in SIP channel? |
| 10:24.40 | gr0mit | are you running the SCCP image or the SIP image? |
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| 10:25.04 | zxd | talking to me? |
| 10:25.27 | gr0mit | no, Rico29 |
| 10:25.41 | gr0mit | Rico29, are you running SCCP or SIP? |
| 10:26.11 | mort_gib | Anyone know a good VOIP provider in Switzerland?? |
| 10:26.11 | Rico29 | SIP |
| 10:26.12 | Rico29 | sorry |
| 10:26.26 | Rico29 | gr0mit> SIP image |
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| 10:26.51 | gr0mit | mort_gib, i know of one - can find out |
| 10:27.09 | mort_gib | gr0mit: -Yeah??? |
| 10:27.44 | gr0mit | need to find out |
| 10:27.44 | mort_gib | I have found a few using Google, but if anyone know a good one... |
| 10:30.34 | gr0mit | mort_gib, see pm |
| 10:31.02 | mort_gib | ?? |
| 10:31.17 | gr0mit | i sent you details in a private message tab |
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| 10:33.10 | gr0mit | there are good howto's on voip-info.org, Rico29 |
| 10:34.14 | Rico29 | i'll see |
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| 10:49.46 | doke | Hey everybody, does anyone here has been able to successfully register a Cisco 7975 SIP with Asterisk? |
| 10:51.34 | doke | Rico29: did you manage to provision your phone? |
| 10:51.50 | Rico29 | no |
| 10:52.15 | doke | What you're looking for is the right option for the DHCPd to provision your phone ? |
| 10:53.31 | Rico29 | doke> i'm looking for the way to do it |
| 10:53.47 | Rico29 | to give sip conf to my phone via shcp+tftp |
| 10:53.50 | Rico29 | dhcp |
| 10:53.51 | Rico29 | sorry |
| 10:54.13 | doke | Rico29: I wrote a config to you in a private window |
| 10:59.34 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 10:59.57 | mort_gib | how do I find a corresponding SIP user from the transmission ID |
| 11:00.16 | mort_gib | chan_sip.c: Maximum retries exceeded on transmission 3c363d083d7f-qgfmvnz46ter for seqno 1 (Critical Response) |
| 11:00.52 | awk | anyone seen this or kow what could cause it? |
| 11:00.54 | awk | [May 13 12:26:54] WARNING[32606]: pbx.c:1832 pbx_extension_helper: No application 'rxfax' for extension (macro-all-faxreceive, s, 6)? |
| 11:01.04 | awk | trying to use spandps / fax to email |
| 11:01.21 | awk | I get this on 2 boxes its working on many others |
| 11:02.19 | RoyK | http://www.freebeer.org/blog/ |
| 11:04.16 | jduggan | hey guys, im using a digium 4port fxo, seems there's an issue where the card is probably not telling asterisk that the line calling in has hung up, so asterisk continues to hold the line open, can someone point me in the right direction to debug/fix this? |
| 11:04.31 | jduggan | excuse any ignorance on my behalf, i've never done telephony stuff before |
| 11:04.44 | JT | awk: is rxfax/spandsp installed? |
| 11:05.10 | JT | jduggan: this is normal for an analogue line unless you have polarity reverse on far end disconnect enabled on the line |
| 11:05.53 | awk | JT: spandsp is, not sure about rxfax |
| 11:06.05 | JT | i'm guessing no |
| 11:06.23 | awk | but never used a package called rxfax |
| 11:06.26 | awk | only spandsp |
| 11:06.33 | JT | that's not the name of the package |
| 11:06.45 | awk | spandsp-0.0.4-1.el4 |
| 11:06.56 | JT | that's not the name of the package that contains rxfax |
| 11:07.56 | *** join/#asterisk igascream (n=igascrea@80.179.192.178.cable.012.net.il) |
| 11:08.04 | JT | http://sourceforge.net/projects/agx-ast-addons |
| 11:08.07 | JT | you need that |
| 11:08.08 | jduggan | JT: ok, its completely normal?, its just a case of someone answering the line and hanging up i guess? |
| 11:08.26 | JT | jduggan: right, analogue has pretty crappy call progress handling |
| 11:08.37 | JT | if you want really good call process handling, get digital lines |
| 11:08.47 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:08.54 | JT | if you want slightly better than really crappy, get polarity reverse enabled by the telco |
| 11:09.26 | JT | s/process/progress/ |
| 11:10.21 | jduggan | JT: unfortunately we're based on a big Science Park.. we cant get our own provider into the building as the Park has their own trunking around the estate, i believe our connection is fed via their own big PBX, maybe they can enable it? |
| 11:10.37 | JT | maybe but i don't like your chances |
| 11:10.45 | JT | doesn't sound very scientific to me |
| 11:10.48 | igascream | hi all have a problem with picking up the zap channels |
| 11:11.37 | jduggan | JT: shrug, there's a few big 'science' type co's on the estate ;), pharmaceuticals ISP etc |
| 11:11.50 | mort_gib | igascream: Do you see the incoming calls?? |
| 11:11.56 | *** join/#asterisk Dovid (n=Dovid@bzq-79-181-121-232.red.bezeqint.net) |
| 11:12.03 | JT | the phone system doesn't sound hi-tech :P |
| 11:12.08 | jduggan | but you're right, phone is a bit crap, maybe we can get our own ISDN/E1 in |
| 11:12.27 | jduggan | we already have 3x fiber feeds into the building, i'll query |
| 11:12.33 | jduggan | thanks for your help |
| 11:12.39 | Dovid | does anyone know if I can pass multiple values to externnotify= in voicemail.conf / |
| 11:12.42 | JT | no probs |
| 11:12.46 | JT | voip is another option |
| 11:12.50 | Dovid | ?* |
| 11:12.50 | JT | but it has its own issues |
| 11:13.24 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
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| 11:13.30 | igascream | I have this exten=_6.,1,Pickup(${EXTEN:1}) |
| 11:22.19 | igascream | in console I receave this No target channel found for 124. |
| 11:23.03 | igascream | whan can I try to do? |
| 11:26.05 | Dovid | JT: Do you use externnotify in voicemail.conf ? |
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| 11:40.08 | mort_gib | I have an issue with WaitExten |
| 11:40.38 | LinuxMafia | what adaptors you guys suggest? |
| 11:45.01 | Rico29 | doke> are you there? |
| 11:46.05 | LinuxMafia | ANY one |
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| 12:01.33 | mort_gib | LinuxMafia: ??Adapters?? |
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| 12:13.19 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 12:13.51 | puzzled | hi |
| 12:14.03 | *** join/#asterisk Skarmeth (n=Skarmeth@201009042244.user.veloxzone.com.br) |
| 12:14.12 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:16.07 | puzzled | morning [TK]D-Fender |
| 12:16.17 | [TK]D-Fender | puzzled: mornin' |
| 12:17.05 | puzzled | [TK]D-Fender: any idea what the best practice is with irqbalance. Iirc it should be turned off. Do you agree? |
| 12:17.42 | [TK]D-Fender | puzzled: Never heard of actually. |
| 12:18.08 | puzzled | ah ok. it's a service on RHEL/CentOS boxes. It dynamically spreads irqs over cpu's |
| 12:21.30 | tzafrir_laptop | But if you have just 2 CPUs, isn't it best to let 1 CPU handle all the hits and keep its cache warmer? |
| 12:24.28 | awk | does anyone have agx-ast-addons for centos 4 (rpm) ? |
| 12:25.09 | puzzled | tzafrir_home: yes I guess so unless it gets pretty overloaded. in the past I would fiddle with smp_affinity settings and assign the nic to one cpu and raid/e1 card to the other cpu |
| 12:25.39 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:25.39 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:28.14 | JT | awk: just compile it |
| 12:40.04 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 12:40.26 | *** join/#asterisk jicksta_ (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 12:42.51 | *** join/#asterisk jicksta (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 12:43.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 12:51.25 | tzafrir | puzzled, so why mess with it? |
| 12:52.01 | tzafrir | Distributing the interrupts evenly means a colder cache, right? |
| 12:52.15 | puzzled | tzafrir: just read that irqbalance switches irq beteen cpu's and that may result in dropped irq's which you obviously don't want |
| 12:52.23 | tzafrir | Let them hog specific CPUs, if reasonable |
| 12:52.27 | awk | JT cmake doesn't compile. so its 1 story after the next.. |
| 12:52.39 | awk | cmake requires ceratin vars that are not defined and its turning into such a hack |
| 12:53.05 | awk | and ive tried the agx-ast rpm package but it doesn't come with rxfax.. or app_fax.so |
| 12:53.55 | tzafrir | puzzeled: on a different matter: |
| 12:54.31 | tzafrir | would you consider moving the default extensions.conf to /usr/share/asterisk/configs/extensions.conf |
| 12:54.43 | tzafrir | and have in /etc/asterisk/extensions.conf: |
| 12:54.59 | tzafrir | #include /usr/share/configs/extensions.conf |
| 12:55.15 | tzafrir | (with perhaps an extra line or two of notes) |
| 12:55.55 | tzafrir | The idea is to let the sample config file update on version updates, and thus keep serving as a useful reference |
| 12:56.29 | puzzled | tzafrir: I always stick them in /etc/asterisk-samples_<version> so they never interfere with the real config |
| 12:56.31 | tzafrir | Without running over the user's configuration |
| 12:57.33 | tzafrir | /etc/asterisk-samples_<version> ? Why the <version>? Do you expect two versions installed ocncurrently? |
| 12:58.15 | puzzled | no just in case configs changed between versions so it's just a reminder |
| 12:58.49 | tzafrir | If it's something the user shouldn't edit, why place it under /etc ? |
| 12:59.56 | puzzled | well I have them in /usr/share/doc/asterisk-<version>/ too :) |
| 13:00.11 | puzzled | but that way it's easier to copy stuff between samples and the real config dir |
| 13:00.41 | JT | awk: what about cmake from a package? |
| 13:00.46 | tzafrir | points puzzled to the option -s in ln(1) |
| 13:01.01 | JT | awk: i found agx-ast to be a breeze to get going with an existing 2.4 install |
| 13:01.03 | puzzled | tzafrir: thanks :) |
| 13:01.05 | JT | err 1.4 |
| 13:04.26 | Katty | mew. |
| 13:05.13 | tzafrir | wonders if that's Katty's preffered MUA |
| 13:05.33 | lsodi | skyy consulting asterisk call center solution, anyone having any experience with it? |
| 13:08.35 | *** join/#asterisk djs (n=djs@unaffiliated/djs26) |
| 13:08.59 | Katty | tzafrir: mua? |
| 13:10.45 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 13:11.32 | tzafrir | http://mew.org/ |
| 13:12.41 | *** join/#asterisk intralanman (n=lanman@209.85.58.2) |
| 13:14.25 | x86 | I want to randomly start a mixmonitor before dialing, how would you guys recommend I do this? |
| 13:15.28 | x86 | aha..... Random() |
| 13:16.15 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:16.15 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:16.22 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.46 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:17.05 | LinuxMafia | any suggestion of --> http://voipstorecanada.ca/catalog/index.php?cPath=59 |
| 13:18.17 | *** join/#asterisk JimTheBeef (n=jdavies@79.121.201.192) |
| 13:18.20 | JimTheBeef | Hi |
| 13:18.30 | JimTheBeef | can anyone give me a hand with asterisk now ? |
| 13:18.31 | JimTheBeef | now :) |
| 13:19.17 | jaytee | try in #asterisknow, this is for standard Asterisk |
| 13:19.31 | JimTheBeef | No ones talking in there :) |
| 13:19.54 | jaytee | is it that no one is talking? or no one is listening? |
| 13:19.58 | rob0 | Now, with asterisk, but never with asteriskNow. :) |
| 13:20.02 | JimTheBeef | Would you say asteriskNow is a lot more limited that the standard asterisk ? |
| 13:20.24 | JimTheBeef | lol |
| 13:20.26 | rob0 | Don't know, but I imagine it has to be. |
| 13:20.27 | LinuxMafia | any one |
| 13:20.28 | jaytee | the GUI is just glued onto asterisk but it limits you |
| 13:20.46 | JimTheBeef | can you not pass all the standard commands though the console then ? |
| 13:21.13 | rob0 | If you do things in the console / config files that the GUI doesn't understand, the GUI breaks. |
| 13:21.33 | JimTheBeef | oh dear :( |
| 13:21.44 | JimTheBeef | maybe i need to install the proper version then |
| 13:22.06 | jaytee | and if there are things that are not simple setups that requires some system tweaking then you can't do it in the GUI in the first place. |
| 13:22.10 | *** join/#asterisk Defraz (n=T0tal@69.92.19.83) |
| 13:22.10 | JimTheBeef | I'm looking at trying to set it up as a SIP GW as opposed to and endpoint like a phone. |
| 13:22.18 | JimTheBeef | ahh ok i see |
| 13:22.35 | JimTheBeef | what distro do most people install it on ? |
| 13:23.18 | jaytee | whatever they tend to feel comfortable with |
| 13:23.19 | JimTheBeef | linux disto that is |
| 13:23.46 | JimTheBeef | ok well thanks for the heads up, guess i'm going to have to install it manually |
| 13:23.54 | jaytee | it runs great on Debian, CentOS, Ubuntu server, RHEL 5, Fedora. Take your pick. |
| 13:24.12 | JimTheBeef | i'm trying to run trunks to it from a Genband M6 |
| 13:24.23 | jaytee | I imagine it runs well on a bunch of others but never tried it on any but the above mentioned. |
| 13:24.34 | *** join/#asterisk shinao1 (n=shinao1@41.219.235.80) |
| 13:24.35 | *** join/#asterisk znoG (n=gs@host46.190-31-3.telecom.net.ar) |
| 13:24.35 | JimTheBeef | ok thanks jaytee |
| 13:24.43 | jaytee | yw |
| 13:24.48 | JimTheBeef | and rob0 |
| 13:24.54 | JimTheBeef | :) |
| 13:25.25 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-a74478956108437d) |
| 13:27.37 | JimTheBeef | Think i'm going to go for Debian |
| 13:27.50 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 13:28.12 | x86 | not a bad choice if you had a time machine and could go back 5 years to when a brand new Debian install would have current software ;) |
| 13:28.58 | doke | hey people I'm still trying to register my Cisco 7975G phone with Asterisk... Any success here? |
| 13:29.06 | x86 | get an iso, burn CD, hop in the time machine and go back 5 years (at least), install the CD, now you've got a current OS! |
| 13:29.06 | doke | does anyone has a config file that he would like to share? |
| 13:29.26 | x86 | doke: are you running the SIP firmware or skinny? |
| 13:29.32 | jaytee | will you give it back when you are done? |
| 13:29.32 | doke | SIP firmware |
| 13:29.41 | *** join/#asterisk Dr-Linux (n=somethin@202.125.139.198) |
| 13:29.51 | Dr-Linux | hi guys |
| 13:30.24 | Dr-Linux | which Cisco ip phone model support video calls with asterisk? |
| 13:30.25 | *** join/#asterisk Zyna (n=IceChat7@p54BCEFD4.dip.t-dialin.net) |
| 13:31.03 | doke | x86: any idea? I only have the SIP image here... and I'm fighting with it since a week now |
| 13:31.36 | doke | before I didn't get anything... now at least I have a SIP dialog working but Asterisk replies with a 401 unauthorized for an unknown reason |
| 13:31.38 | *** join/#asterisk shinao1 (n=shinao1@41.219.235.80) |
| 13:32.15 | Zyna | I am sooo screwed... I have 9 days left to write a konceptional work to implement an asterisk for my company plus a project documantation on that and I don't even know how to connect asterisk to a point-to-point isdn line |
| 13:32.39 | *** join/#asterisk ManxPower (n=manxpowe@79.sub-75-201-0.myvzw.com) |
| 13:33.14 | gr0mit | Zyna, that was a cry from the heart! |
| 13:33.27 | doke | :) |
| 13:33.39 | JimTheBeef | x86 lol |
| 13:33.41 | Zyna | I'm about to put my head between my legs and kiss my but good bye |
| 13:33.47 | JimTheBeef | what would you recomment then ? |
| 13:33.50 | gr0mit | nah - don't go there! |
| 13:33.56 | doke | Zyna: are you connecting a BRI card to the pstn or are you connecting a point to point isdn line to interface your asterisk with another pbx |
| 13:34.00 | Zyna | Considering this is my final test for apprentice ship |
| 13:34.13 | gr0mit | how many users? |
| 13:34.23 | JimTheBeef | its only a etst machine |
| 13:34.29 | Zyna | doke, lol... I don't even have a machine... this is a completly conzeptional work... everything happens in my head and nowhere eelse |
| 13:34.32 | JimTheBeef | test even |
| 13:34.42 | gr0mit | aaah ok Zyna |
| 13:34.58 | Zyna | but it has to work if it is tested |
| 13:35.04 | gr0mit | well in that case no-one will ever know if it really works then!1 |
| 13:35.05 | doke | Bristuff would be your friend I suppose |
| 13:35.10 | doke | junghanns.net |
| 13:35.19 | Zyna | And I have never ever done anything with asterisk... not even anything with isdn to be honest... lol |
| 13:35.21 | JimTheBeef | x86 got any ideas on which one to go for then ? |
| 13:35.31 | doke | x86: any update for my Cisco? |
| 13:35.32 | JimTheBeef | solaris 10? |
| 13:36.01 | Zyna | SO here I am... a complete noob to all main subjects of the test with 9 days left to do the impossible for the ungreatful |
| 13:36.05 | doke | what I'll suggest is to get yourself a BRI card and have a bit of experimentaiton |
| 13:36.10 | doke | experimentation |
| 13:36.18 | doke | even if it's only a conceptional work |
| 13:36.29 | gr0mit | or pay someone here to do it all for you ;-) |
| 13:36.31 | doke | openvox have unexpensive cards |
| 13:36.32 | Zyna | doke... I have like 15 bucks eft on my account for this month... ;P |
| 13:36.53 | doke | Zyna: you're in the us? |
| 13:36.57 | doke | where are you located |
| 13:36.57 | Zyna | nope... DE |
| 13:37.00 | doke | ok |
| 13:37.07 | jaytee | Zyna, do you have the PDF of the book Asterisk, The Future of Telephony? |
| 13:37.12 | gr0mit | uses the hfc card for his bri line with bristuff |
| 13:37.13 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:37.16 | *** join/#asterisk Asterisk-nob (n=somethin@117.20.21.66) |
| 13:37.17 | doke | pm me your email |
| 13:37.21 | doke | I have to go |
| 13:37.27 | Asterisk-nob | which Cisco ip phone model support video calls with asterisk? |
| 13:37.29 | doke | I have a couple of production machines in Germany |
| 13:37.31 | Zyna | Yeah, I read through the entire book, but it was too much in too little time to compensate everything |
| 13:37.34 | doke | 2 dev machines |
| 13:37.50 | gr0mit | has a machine in .de with a quad-bri |
| 13:38.34 | jaytee | Zyna, www.voip-info.org is a good resource also. There are lots of examples and tutorials there. If you don't have to build an actual system with hardware you can at least grab info and simulate it. |
| 13:39.17 | gr0mit | Zyna, how many channels, how many extensions? |
| 13:39.22 | *** join/#asterisk axisys (n=axisys@202.79.19.72) |
| 13:39.29 | Asterisk-nob | anybody clue on my question? |
| 13:39.46 | Zyna | We have 2 isdn lines in point2point and there will be approx. 15-20 extensions |
| 13:39.50 | powerkill | hi |
| 13:39.56 | ManxPower | Asterisk-nob: none that I'm aware of. |
| 13:40.11 | powerkill | did the behaviour of forkcdr change between asterisk 1.2 and 1.4 ? |
| 13:40.12 | Katty | tzafrir: ahhh. |
| 13:40.28 | Katty | tzafrir: different type of mew. |
| 13:40.43 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 13:40.49 | Zyna | I have plenty of ressource sites, but the entire topic is just too freakin big to overview in 9 days plus writing the portfolio, plus writing the documentatzion |
| 13:40.52 | x86 | doke: i was just wondering, I don't know anything about cisco phones |
| 13:40.57 | gr0mit | ok well bristuff is what i wolud use |
| 13:40.58 | x86 | doke: I don't use that crap ;) |
| 13:41.06 | x86 | doke: I use real phones... aka Polycom ;) |
| 13:41.07 | ManxPower | Zyna: Expect to spend at least a month |
| 13:41.23 | Zyna | ManxPower: due day is teh 22nd |
| 13:41.31 | Zyna | ;P |
| 13:41.31 | ManxPower | "due day" |
| 13:41.36 | gr0mit | well you need to start 2 weeks ago then |
| 13:41.48 | ManxPower | Is this for a class? |
| 13:41.59 | Zyna | yeah... I'll spend the next 9 days writing a timemachine... xD |
| 13:42.18 | gr0mit | hands Zyna a tardis |
| 13:42.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:42.57 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:43.33 | ManxPower | Nothing quite deciding to work on one of the most complex systems on the internet -- VoIP PBX for a class. |
| 13:44.25 | Asterisk-nob | ManxPower: i'd like to ask some more quick important questions |
| 13:44.36 | *** join/#asterisk thepacmanfan (n=pacmanfa@12-218-140-89.client.mchsi.com) |
| 13:44.43 | ManxPower | ~ask |
| 13:44.44 | jbot | extra, extra, read all about it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:45.21 | Asterisk-nob | ManxPower: i don't think asterisk does it, but still wannna ask: Can we route calls to multiple phones at the same time (e.g. call comes in, it rings on my cell phone, home phone, and work desk phone)? |
| 13:45.21 | thepacmanfan | so i'm having a pesky problem... zaptel can't find my kernel source during compile. |
| 13:45.35 | Zyna | gr0mit, you located in DE? |
| 13:46.04 | ManxPower | Asterisk-nob: Yes, you can do that -- as long as you do not use FXO ports and only have to let one phone answer per call (not answer the same call on multiple phones) |
| 13:47.10 | thepacmanfan | i'm running debian. i've got linux-source in /usr/src. |
| 13:47.18 | Asterisk-nob | ManxPower: you mean same call can not ring on multiple phones? |
| 13:47.24 | thepacmanfan | maybe zaptel is looking for kernel-source, not linux-source? |
| 13:47.40 | gr0mit | Zyna, no, .uk |
| 13:50.01 | ManxPower | Asterisk-nob: the same call can RING on multiple phones, you just can't ANSWER the call on more than one phone. |
| 13:52.25 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 13:52.47 | Zyna | What's a nother good company to buy voip phones besides grandstream? |
| 13:53.01 | JT | another? |
| 13:53.06 | Zyna | yes |
| 13:53.07 | JT | you mean "a" |
| 13:53.07 | Zyna | sry |
| 13:53.22 | JT | grandstream is not an example of good :P |
| 13:53.35 | Zyna | what would be then? |
| 13:53.44 | *** join/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 13:53.48 | *** join/#asterisk profounded (n=Bryan_Ru@cpe-74-66-233-176.nyc.res.rr.com) |
| 13:53.54 | JT | polycom |
| 13:54.37 | coppice | be specific. the polycom 320 and 330. the other polycoms are not such good value |
| 13:55.18 | gr0mit | avoids grandstreams like the plague |
| 13:55.20 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 13:55.41 | coppice | I don't avoid them. I often pass their building :-) |
| 13:55.49 | JT | the other polycoms are still good quality |
| 13:56.03 | ManxPower | ~phones |
| 13:56.03 | jbot | it has been said that 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 phones. Ever. places like such as |
| 13:56.24 | glaz | I love my 7940 from cisco and also my 480i CT from AAstra |
| 13:57.03 | Zyna | What's so bad about grandstream phones? |
| 13:57.20 | gr0mit | have you tried them? |
| 13:57.24 | Zyna | nope |
| 13:57.27 | Zyna | none ever |
| 13:57.31 | gr0mit | well save your money |
| 13:57.48 | gr0mit | and spend the same amount on almost any other phone |
| 13:57.56 | *** join/#asterisk adr3nalin3_ (n=afink@66.172.245.81) |
| 13:58.02 | Zyna | could you possibly provide some kind of senseful arguement I can put in my text? |
| 13:58.04 | [TK]D-Fender | Zyna: flakey firmware, crappy use of LCD, cheap build quality, second rate audio, history of echo issues, etc |
| 13:58.11 | [TK]D-Fender | Zyna: How's taht sum it up? |
| 13:58.19 | Zyna | that helps thx! |
| 13:58.24 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:58.35 | gr0mit | ah, if you want consultancty, I will PM you my BIC/IBAN |
| 13:58.42 | coppice | grandstream needs to hire a good plastics designer. |
| 13:59.19 | [TK]D-Fender | coppice: Aastra too.... 5i = BLEH |
| 13:59.21 | Zyna | gr0mit, I am a student with about 15 bucks left on my bank account... I don't think I could buy you a burger for your help, even if I would liek to |
| 13:59.26 | gr0mit | hehehehe ! |
| 13:59.34 | gr0mit | was only kidding Zyna |
| 13:59.41 | Zyna | I know ;P |
| 13:59.42 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:59.57 | Zyna | I just wish I had more time for this... |
| 14:00.00 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.224) |
| 14:00.16 | gr0mit | where are you in .de, Zyna? |
| 14:00.18 | Zyna | Either I'm not gonna get finished, which is not an option... or it's just gonna be a crappy piece of work |
| 14:00.21 | Zyna | Berlin |
| 14:00.25 | coppice | [TK]D-Fender I've never seen an aastra in the flesh. they look ugly in pictures, but I thought people liked them |
| 14:00.51 | gr0mit | aaah go see Klaus then |
| 14:01.03 | gr0mit | www.junghanns.net |
| 14:01.04 | Zyna | Who's Klaus? |
| 14:01.15 | coppice | santa klaus? |
| 14:01.17 | gr0mit | hehe |
| 14:01.37 | [TK]D-Fender | coppice: Have some good points, but bad materials usage. |
| 14:01.42 | gr0mit | he is also in Berlin |
| 14:02.02 | Zyna | gr0mit, do you know him? |
| 14:02.06 | gr0mit | yup |
| 14:02.06 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:02.21 | Zyna | omg, that's lke 10 minutes from here ;P |
| 14:02.23 | Zyna | awesome!!! |
| 14:02.29 | coppice | gr0mit: have you heard from him recently? |
| 14:02.41 | gr0mit | spoke to him a couple months ago |
| 14:02.50 | JT | it's klaus the recluse |
| 14:04.31 | Zyna | gr0mit, you know what... I might as well do that... that should give me a little boost... I can imagine, that if I ask nicely... they'll answer me some questions andf give me alittle push here and there... |
| 14:04.32 | gr0mit | he is busy doing voip call centres in Slovenia and other former east-block countries |
| 14:04.44 | [TK]D-Fender | Zyna: For your money, Linksys is probably the best choice. |
| 14:04.45 | gr0mit | tell him i sent you |
| 14:04.52 | gr0mit | shudders |
| 14:05.01 | Zyna | gr0mit: I'll do that |
| 14:05.08 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:05.18 | gr0mit | linksys phones do not make me happy |
| 14:05.26 | gr0mit | the audio is not so good |
| 14:05.38 | gr0mit | Snom (also aus Berlin ) are good |
| 14:05.51 | [TK]D-Fender | gr0mit: Lower than cisco/Polycom, but not "bad" |
| 14:06.08 | gr0mit | have had very good results (i.e. happy customers!) with ST2030 |
| 14:06.37 | gr0mit | customers like the BLF which Snom and Thomson do |
| 14:07.08 | gr0mit | constantly curses C****o |
| 14:07.22 | gr0mit | with their orrible config |
| 14:07.49 | gr0mit | but they are about the best in terms of audio quality etc |
| 14:09.02 | [TK]D-Fender | gr0mit: I've heard Thomson is decent, though they aren't popular on this side of the ocean. Snom has a history of flakeyness and is overpriced |
| 14:09.20 | gr0mit | nor here |
| 14:10.12 | coppice | I think thomson might sell mostly in volume accounts for service providers |
| 14:14.04 | gr0mit | but i have been plesantly surprised |
| 14:18.20 | *** join/#asterisk af_ (n=getsmart@88-149-241-145.dynamic.ngi.it) |
| 14:25.26 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 14:27.16 | thepacmanfan | now zaptel is throwing a zompile error. |
| 14:28.04 | thepacmanfan | *compile |
| 14:28.04 | thepacmanfan | it's in /kernel/pciradio |
| 14:28.27 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 14:28.32 | thepacmanfan | i get a *lot* of "struct pciradio" has no member named "membername" errors |
| 14:34.40 | JT | then don't compile pciradio, i very much doubt you need it |
| 14:36.03 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 14:36.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:36.54 | thepacmanfan | JT, how do i do a "make" and ommit pciradio? |
| 14:37.08 | thepacmanfan | i'm mostly a newbie at compiling things. :o |
| 14:39.38 | powerkill | hi coppice |
| 14:39.51 | coppice | hi |
| 14:39.58 | powerkill | did you find something on my division by 0 problem ?: :D |
| 14:41.05 | coppice | I posted a solution the same day. didn't you see it? |
| 14:41.15 | powerkill | No sorry I didn't :( |
| 14:41.59 | coppice | then you'll never know |
| 14:42.03 | powerkill | lol |
| 14:42.24 | adr3nalin3_ | Hey guys I am having trouble with voicemail quality on asterisk. Is there a way to define the compression of the audio file? whether it be gsm or wav gsm(wav). Or if there is anything else anybody can suggest please let me know. |
| 14:43.33 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 14:43.34 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
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| 14:45.18 | ManxPower | adr3nalin3_: the format= specifies the format, including compression. |
| 14:45.20 | powerkill | coppice it was a patch ? |
| 14:45.35 | ManxPower | what SPECIFIC problem are you having with voicemail audio? listening on phone? listening via e-mail? |
| 14:45.37 | coppice | it might have been. |
| 14:45.48 | *** join/#asterisk CVirus (n=GoD@196.205.193.11) |
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| 14:47.37 | adr3nalin3_ | ManxPower, it is both in email and on the phone. It is choppy and almost impossible to understand. |
| 14:47.54 | powerkill | 16:11.01coppicepowerkill: as a quick fudge try changing line 661 in v17rx.c from |
| 14:47.54 | powerkill | 16:11.03coppice<PROTECTED> |
| 14:47.54 | powerkill | 16:11.04coppiceto |
| 14:47.54 | powerkill | 16:11.06coppice<PROTECTED> |
| 14:48.19 | powerkill | I got it protected can you tell me what to change there ? |
| 14:48.30 | coppice | huh? |
| 14:48.52 | ManxPower | adr3nalin3_: that is not what is supposed to happen. What format are you using? |
| 14:49.12 | powerkill | I got what you write to me on 30 April 2008 on http://purl.rikers.org/%23asterisk/20080430.html.gz |
| 14:49.21 | adr3nalin3_ | ManxPower, I have tried all three formats....I think I was setting via AsteriskGUI and I do not always trust guis. |
| 14:50.09 | rob0 | Well damn, my Polycom hopes are dashed for now. |
| 14:50.22 | coppice | powerkill: there are two occurances of |
| 14:50.24 | coppice | <PROTECTED> |
| 14:50.26 | coppice | <PROTECTED> |
| 14:50.27 | coppice | find the second one, and change it to |
| 14:50.29 | coppice | <PROTECTED> |
| 14:50.30 | coppice | <PROTECTED> |
| 14:50.36 | ManxPower | adr3nalin3_: WAV49 normally gives the best audio for the low bandwidth it uses, perfect for E-mail. |
| 14:51.05 | ManxPower | adr3nalin3_: set it in voicemail.conf and issue a reload in Asterisk |
| 14:51.39 | powerkill | perfect got it :) |
| 14:52.23 | adr3nalin3_ | The attachfmt is the one I should be changing correct? |
| 14:53.43 | ManxPower | adr3nalin3_: I just set the format, I don't differ it for e-mail |
| 14:53.56 | adr3nalin3_ | ok thanks. |
| 14:55.55 | thepacmanfan | ok, my bad on misdiagnosing the compile error. |
| 14:56.33 | thepacmanfan | it's going back to a "Symbol version dump .... Module.symvers is missing" error |
| 14:56.35 | adr3nalin3_ | ManxPower, strange there is still a lot of static on the message. |
| 14:56.53 | ManxPower | adr3nalin3_: you have some OTHER problem. |
| 14:57.37 | *** join/#asterisk eth01 (i=foo@gentoo/user/eth01) |
| 14:57.54 | adr3nalin3_ | Its strange b/c call quality is very good. |
| 14:57.55 | iCEBrkr | So, anyone know why the 'a' flag (mark as administrator) in MeetMe() nagates the join/leave sound? |
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| 15:05.28 | adr3nalin3_ | ManxPower, I think this may be my problem...http://forums.digium.com/viewtopic.php?p=62527&sid=6020a3b05dc5d0a3a489139e1a213165 |
| 15:06.50 | *** join/#asterisk gitguy (n=diego@adsl-134-171.click.com.py) |
| 15:07.25 | *** part/#asterisk lsodi (n=root@213.168.26.50) |
| 15:07.58 | Nugget | http://www.debian.org/security/2008/dsa-1571 <-- oops |
| 15:08.01 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 15:08.09 | Nugget | "It is strongly recommended that all cryptographic key material which has been generated by OpenSSL versions starting with 0.9.8c-1 on Debian systems is recreated from scratch. Furthermore, all DSA keys ever used on affected Debian systems for signing or authentication purposes should be considered compromised; the Digital Signature Algorithm relies on a secret random value used during signature generation." |
| 15:08.29 | b11d` | lol |
| 15:08.31 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 15:10.30 | matnel | just upgrading all systems :P |
| 15:11.23 | Asterisk-nob | ManxPower: another question, I know OpenFire has a plugin with Asterisk, can it do streaming video in the chat then divert to a video phone call? |
| 15:11.50 | *** join/#asterisk mknerd (i=3f951603@gateway/web/ajax/mibbit.com/x-96242e96042f1c4c) |
| 15:11.50 | [TK]D-Fender | adr3nalin3_: there have been very specific cases of GCC 4.1 causing real distortion of GSM codec for compile. a recompile to lowe version usually solved taht instantly. |
| 15:12.25 | thepacmanfan | ooo.. i had no idea i had to make symlinks before compiling zaptel! |
| 15:13.12 | ManxPower | thepacmanfan: you don't |
| 15:13.44 | ManxPower | unless, of course, your distro's kernel modules are in a different directory than your kernel source points to. |
| 15:13.50 | ManxPower | Mandriva is one of the distros that does that. |
| 15:13.56 | ManxPower | most distros do not. |
| 15:15.36 | LinuxMafia | hi |
| 15:15.55 | Maliuta | people using distro kernels beyond installation should be shot |
| 15:16.19 | Maliuta | roll your own, know what your systems are and need. |
| 15:16.25 | LinuxMafia | [TK]D-Fender, i was looking every where to buy the ip-phone , they got only panasonic and philips |
| 15:16.26 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 15:16.34 | adr3nalin3_ | [TK]D-Fender: Thanks, I was missing the gsm libraries but I am also using gcc 4.2. Right now I am recompiling with gsm libs installed but if that doesn't work I will use a lower version of gcc. Thanks! |
| 15:16.40 | Maliuta | less code, less chance of compromise from bad code |
| 15:16.49 | LinuxMafia | [TK]D-Fender, can i use panasonic |
| 15:18.16 | [TK]D-Fender | LinuxMafia: You should just give up now. I've handed you the answer and you are still looking for trouble. I've never even HEARD of either of those 2 companies making a SIP phone compatible with *. |
| 15:19.37 | LinuxMafia | [TK]D-Fender, http://www.futureshop.ca/catalog/proddetail.asp?sku_id=0665000FS10097498&catid=23014&logon=&langid=EN |
| 15:19.39 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 15:19.48 | gitguy | how much bugs are being fixed for 1.6? where can i see this? |
| 15:19.55 | gitguy | changelog i guess? uhmm |
| 15:20.07 | [TK]D-Fender | LinuxMafia: Dear God you seriously don't have a clue at all. |
| 15:20.10 | jayrod422 | does any know how to look up the correct OCN or number to see if it was ported from something different than lerg |
| 15:20.22 | LinuxMafia | [TK]D-Fender, no i dont |
| 15:20.25 | [TK]D-Fender | gitguy: CHANGELOG <-- and 1.6 doesn't fix 1.4 bugs |
| 15:20.44 | LinuxMafia | i looked over almost 20 computer stores today |
| 15:20.47 | gitguy | [TK]D-Fender: it doesn't? i though it was a improvement of 1.6? |
| 15:20.48 | LinuxMafia | neither one had |
| 15:20.51 | LinuxMafia | any of |
| 15:20.54 | LinuxMafia | ~phones |
| 15:20.54 | jbot | rumour has 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 phones. Ever. places like such as |
| 15:21.26 | [TK]D-Fender | LinuxMafia: You aren't going to FIND this stuff is chitty consumer stores! What don't you get? consumers don't BUY SIP PHONES. |
| 15:21.57 | *** join/#asterisk mratliff (n=mratliff@cust-baileys-90-146.mounet.com) |
| 15:22.04 | mratliff | hello |
| 15:22.04 | [TK]D-Fender | LinuxMafia: And what you linked me is a shitty USB phone that relies on being plugged into your PC and using a soft-phone. Its the equivalent of a stupid headset |
| 15:22.08 | *** join/#asterisk jarrod (n=jarrod@theos.org) |
| 15:22.34 | mratliff | guys...I have a question about a new asterisk deployment |
| 15:22.34 | LinuxMafia | [TK]D-Fender, oh i wish i could buy of the internet |
| 15:22.51 | jarrod | anyone know why a polycom, with forwarding enabled, would forward calls received from PSTN, but not calls that were forwarded to it from other local polycoms? |
| 15:22.51 | mratliff | I need a design that will support roughly 800 phones |
| 15:23.08 | [TK]D-Fender | LinuxMafia: I told you a company you can take the friggen metro to get to and buy from. Whats the problem? |
| 15:23.48 | LinuxMafia | [TK]D-Fender, you said factorydirect.ca i went there too |
| 15:23.58 | LinuxMafia | [TK]D-Fender, i looked at the logs |
| 15:24.14 | LinuxMafia | http://www.futureshop.ca/catalog/proddetail.asp?sku_id=0665000FS10061150&catid=23014&logon=&langid=EN |
| 15:24.18 | mratliff | what server specs do you recommend for this? |
| 15:24.22 | ManxPower | jarrod: Many carriers reject calls with invalid callerid. Do your internal extensions have valid PSTN callerid? |
| 15:24.24 | LinuxMafia | it was look like that ^^ |
| 15:24.48 | [TK]D-Fender | LinuxMafia: NO I DIDN'T. For the LAST TIME, I never send ANYONE to Facotrydirect.com for ANYTHING |
| 15:24.59 | jarrod | manx: the softswitch sets the callerid on any outbound calls, and yes, each phone is configured with a valid callerid |
| 15:25.05 | LinuxMafia | [TK]D-Fender, just a sec |
| 15:25.21 | jayrod422 | manx i have the correct caller id |
| 15:25.31 | jayrod422 | what i am trying to do is find the valid OCN for routing |
| 15:25.45 | jayrod422 | i know you get it from SS7 but I dont even know where to get SS7 |
| 15:25.47 | jarrod | the user has set the forward using the softkey on their polycom... it works GREAT when the call comes in from pstn, but not when a call is transfered locally |
| 15:25.48 | [TK]D-Fender | LinuxMafia: and "Linksys Vonage VOIP Adapter (E41580)" <---- NO. |
| 15:26.00 | jarrod | i thought it might be able to distinguish a local call from an 'external' call |
| 15:26.36 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 15:26.39 | Asterisk-nob | [TK]D-Fender: question, we setup an environment where someone is in a call on their cell then they arrive to work (or home), can that call be transferred to their desk phone without disruption of the call? |
| 15:27.06 | Asterisk-nob | i guess this is not possible but wanna confirm it |
| 15:27.13 | gitguy | [TK]D-Fender: what kind of improvement is 1.6 over 1.4 then? |
| 15:27.21 | [TK]D-Fender | Asterisk-nob: Only if your cell carrier has a transfer feature. |
| 15:27.25 | [TK]D-Fender | gitguy: read the changelog. |
| 15:27.29 | LinuxMafia | [TK]D-Fender, http://www.canadacomputers.com/index.php?do=ShowProduct&cmd=pd&pid=014833&cid=828.480 |
| 15:27.38 | LinuxMafia | right this one you gave me |
| 15:27.48 | gitguy | [TK]D-Fender: i was hoping for them to fix bugs instead of adding more stuff |
| 15:28.00 | [TK]D-Fender | LinuxMafia: http://www.canadacomputers.com/index.php?do=ShowProduct&cmd=pd&pid=013131&cid=828.480 |
| 15:28.11 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 15:28.12 | [TK]D-Fender | LinuxMafia: There, in stock all over. |
| 15:28.19 | *** join/#asterisk n00m (i=n00m@c-24-12-177-254.hsd1.il.comcast.net) |
| 15:28.37 | *** join/#asterisk frieze (n=frieze@cpe-69-203-15-230.nyc.res.rr.com) |
| 15:28.45 | Zyna | back |
| 15:29.03 | LinuxMafia | [TK]D-Fender, i was in that store |
| 15:29.04 | *** join/#asterisk axisys (n=axisys@202.79.19.72) |
| 15:29.13 | rickross | when recording from Asterisk, is there any easy way to keep the discrete IN and OUT wave files, instead of having them combined at the end? |
| 15:29.15 | LinuxMafia | [TK]D-Fender, today few minutes ago |
| 15:29.33 | [TK]D-Fender | rickross: Don't use the "m" option or MixMonitor |
| 15:29.58 | rickross | Fender - thx |
| 15:30.06 | Zyna | gr0mit, got an appointment tomorrow ;P didn't see him yet, but I'll make sure to say hi from you |
| 15:30.10 | LinuxMafia | [TK]D-Fender, thanks alot |
| 15:30.16 | LinuxMafia | i will buy that one then |
| 15:30.18 | rickross | we're just initiating the recording in the call with *1 |
| 15:30.29 | rickross | I'll have to find what options that is invoking |
| 15:30.34 | [TK]D-Fender | rickross: Go check your features.conf |
| 15:31.46 | rickross | Fender - is this discrete channel method the best option to record for podcasting? (so we can adjust the in/out volumes independently if needed) |
| 15:32.29 | [TK]D-Fender | rickross: Why would you broadcast each half of the call separately? |
| 15:32.37 | rickross | we wouldn't |
| 15:32.41 | [TK]D-Fender | rickross: and I've never broadcast anything like that. |
| 15:32.48 | [TK]D-Fender | rickross: then why do you want them separate? |
| 15:32.58 | rickross | but often there is a significant variation in volume between in and out |
| 15:33.20 | rickross | so, for post-production, it would be helpful to have discrete channels that can be independently processed |
| 15:33.26 | rickross | and ultimately combined |
| 15:33.32 | *** join/#asterisk mort___ (n=mort@user-3e8886cc.tcl115.dsl.pol.co.uk) |
| 15:33.35 | [TK]D-Fender | rickross: Shouldn't be. This is a sign that you haven't balanced one of your connections. |
| 15:33.55 | rickross | it's just telephone |
| 15:34.39 | rickross | and seems to vary depending on who we are talking to |
| 15:34.55 | rickross | phone interviews of people in all different places |
| 15:36.08 | [TK]D-Fender | rickross: well ifs its different PSTN points over the same link then yeah I guess you might want to split. |
| 15:36.24 | rickross | Fender - that's right |
| 15:39.37 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:41.07 | *** join/#asterisk Dr-Linux (n=somethin@202.125.139.198) |
| 15:41.10 | Dr-Linux | anybody tried Video with Asterisk? |
| 15:41.58 | *** join/#asterisk patrick-- (n=patrick@sam.openroot.de) |
| 15:42.17 | patrick-- | does anyone know a good asterisk frontend to manage voicemail/call-redirections, etc? |
| 15:42.40 | Maliuta | patrick--: vim |
| 15:42.51 | patrick-- | web frontend |
| 15:42.55 | patrick-- | * |
| 15:43.20 | Maliuta | leaves stupid ideas well alone |
| 15:44.12 | keith4 | patrick--: not sure, but the FOP might provide some of that |
| 15:44.23 | *** join/#asterisk Dr-Linux|home (n=somethin@117.20.21.66) |
| 15:44.33 | keith4 | or... try #asterisk-now, #asterisk-gui, etc. |
| 15:44.36 | patrick-- | FOP? |
| 15:46.14 | *** join/#asterisk b1shop (n=b1shop@c-76-16-229-8.hsd1.il.comcast.net) |
| 15:47.10 | *** join/#asterisk b1shop (n=b1shop@c-76-16-229-8.hsd1.il.comcast.net) |
| 15:47.11 | [TK]D-Fender | ~fop |
| 15:47.12 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 15:47.12 | rob0 | http://www.google.com/search?q=fop+asterisk |
| 15:47.55 | jarrod | man could a polycom distinguish between a call transfered by another local phone and a call from the pstn when both source from the same asterisk switch? |
| 15:48.03 | jarrod | its forwarding calls properly from one, but not the other |
| 15:50.08 | ManxPower | jarrod: My idea did not help? |
| 15:50.13 | jarrod | no |
| 15:50.18 | jarrod | not has a proper callerid configured |
| 15:50.22 | jarrod | and im using a digium appliance |
| 15:50.32 | jarrod | er.. no it has a proper callerid configured |
| 15:50.41 | ManxPower | so you are totally sure you are sending valid PSTN callerid, no quotes, not dashes, no extra 1 or 9? |
| 15:50.54 | ManxPower | paste me the "proper" callerid, just the one line |
| 15:51.05 | jarrod | have you ever used a digium appliance? |
| 15:51.21 | ManxPower | You should also put a Noop before the Dial, Noop(CALLERID(num)=${CALLERID(num)}) |
| 15:51.28 | ManxPower | jarrod: The Digium appliance is not supported here. |
| 15:51.33 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:51.40 | ManxPower | If you want our help you are going to edit your config files like the rest of us. |
| 15:52.01 | jarrod | i have no issue 'editing the config files' |
| 15:52.10 | jarrod | but the caller-id is properly configured |
| 15:52.10 | ManxPower | If not, then use whatever support methods the Digium appliance has. |
| 15:52.16 | jarrod | this appears to be more of an issue on a polycom |
| 15:52.36 | ManxPower | jarrod: Best of luck. I cannot help you further if you do not follow my suggestions. |
| 15:52.47 | jarrod | thats because your suggestions are of no help |
| 15:52.49 | ManxPower | jarrod: I mange over 300 polycoms and have NEVER EVER had your issue. |
| 15:52.49 | jarrod | you arrogant prick |
| 15:53.25 | ManxPower | jarrod: best of luck |
| 15:53.44 | jarrod | thanks |
| 15:54.15 | mratliff | What server specs are recommended for 800 phones? |
| 15:54.37 | [TK]D-Fender | jarrod: if you want help, I highly suggest you show us 2 pastebins. With the phone forwarded, #1 = "normal call", and #2 of one that didn't react as expected. This would be at verbose 10, SIP debug enabled. |
| 15:54.51 | jarrod | im ok, i just need to troubleshoot it |
| 15:54.54 | [TK]D-Fender | mratliff: Go lookup "asterisk dimensioning" on the WIKI |
| 15:54.55 | jarrod | thanks |
| 15:54.56 | ManxPower | mratliff: that varies depending on if you have PSTN cards in the server, what transcoding, recording, reinvites, etc |
| 15:55.25 | mratliff | 400 are analog |
| 15:55.42 | ManxPower | mratliff: I suggest you write up a requirements list before asking. |
| 15:55.43 | mratliff | plus would like to have a distributed design |
| 15:55.47 | mratliff | ok |
| 15:56.03 | ManxPower | ask on the mailing list, putting your requirements in your message. |
| 15:56.13 | ManxPower | Obviously, also follow [TK]D-Fender's recommendation |
| 15:56.29 | mratliff | awesome! Thanks! I'll do just that |
| 15:57.02 | ManxPower | mratliff: your question is like "what specs do you recommend for 800 web sites", we can't recommend anything until you know more about your requirements (SSI, CGI, etc) |
| 15:57.16 | Qwell | SSI? |
| 15:57.24 | ManxPower | Qwell: Server Side Includes |
| 15:57.27 | Qwell | oh |
| 15:57.47 | [TK]D-Fender | mratliff: At while point you can show me the machine spec that you feel will support said 800 sites... and then I'll design a SINGLE SITE that would completely KILL your "setup". |
| 16:00.16 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 16:01.00 | mratliff | what is the mailing list address? |
| 16:01.18 | ManxPower | lists.digium.com is where you can signup |
| 16:01.28 | mratliff | i c |
| 16:01.33 | mratliff | thx |
| 16:06.32 | keith4 | holy crap. 400 analog phones? |
| 16:07.56 | [TK]D-Fender | keith4 : nothing wrong with that. |
| 16:08.09 | ManxPower | I cry for people using analog |
| 16:08.09 | [TK]D-Fender | keith4 : Easy enough to setup for. |
| 16:08.16 | keith4 | channel banks? |
| 16:08.24 | [TK]D-Fender | keith4 : SIP gateways. |
| 16:08.30 | keith4 | ooh |
| 16:08.33 | ManxPower | A feeling like a million people suddenly using the switchook |
| 16:08.35 | keith4 | tell me more |
| 16:08.50 | keith4 | oh, like... 400 ATAs? |
| 16:08.55 | [TK]D-Fender | keith4 : 17 x AudioCodes MP-124 |
| 16:09.07 | [TK]D-Fender | keith4 : you = crazy |
| 16:09.13 | keith4 | grins |
| 16:09.15 | keith4 | it's been said |
| 16:09.58 | [TK]D-Fender | keith4 : Single relay-rack, amphenol cross-over to existing demarc. |
| 16:09.59 | keith4 | holy crap, those are expensive |
| 16:10.31 | keith4 | sometimes i wish telephonydepot had bigger pictures. i like pictures |
| 16:10.55 | keith4 | ahahaha, ManxPower someone actually called your an arrogant prick, instead of just implying it, as they usually do ;-) |
| 16:11.09 | keith4 | s/your/you |
| 16:11.19 | *** join/#asterisk SQLDarkly (n=nospam@199-117-163-66.dia.static.qwest.net) |
| 16:11.40 | keith4 | the topic should read "There are really only 2 people in here who know what they are talking about. Don't insult either of them. It's up to you to figure out which 2." |
| 16:12.29 | rob0 | You can insult me. |
| 16:12.40 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 16:13.00 | ManxPower | keith4: Don't worry, he'll insult the other one soon enough. |
| 16:13.22 | keith4 | oh, I'm not worried |
| 16:13.36 | ManxPower | keith4: me neither |
| 16:13.40 | keith4 | some people just can't take constructive criticism! |
| 16:14.23 | ManxPower | keith4: not that, some people just think they know more than the experts and refuse to provide any supporting documentation. I refer to those as "people I don't help" |
| 16:14.45 | keith4 | well, in his defense, there is a fine line between "confident" and "arrogant" ;-) |
| 16:15.01 | keith4 | but you're correct |
| 16:15.06 | ManxPower | keith4: I don't see him providing [TK]D-Fender with the requested info either. |
| 16:15.23 | fenlander | more of a dashed line than a fine line ;) makes it easier to cross |
| 16:15.24 | keith4 | so, maybe the topic should include "we will assume you are an idiot until you give us reason to suspect otherwise." |
| 16:15.47 | iCEBrkr | grrr |
| 16:15.51 | ManxPower | Nobody reads the /topic, as evidenced by all the GUI people that try to use this channel. |
| 16:15.52 | keith4 | [TK]D-Fender: how much bandwidth usage would you expect between one of these MP124s and asterisk? |
| 16:15.56 | iCEBrkr | I missed any replies.. but.. |
| 16:16.01 | iCEBrkr | So, anyone know why the 'a' flag (mark as administrator) in MeetMe() nagates the join/leave sound? |
| 16:16.27 | ManxPower | keith4: you can assume around 80kbps for an ulaw/alaw call. i.e. .08Mbps |
| 16:16.27 | keith4 | uh... because administrators are supposed to be able to sneak around in conference rooms? ;-) |
| 16:17.55 | iCEBrkr | haha |
| 16:17.59 | keith4 | and these are what.. 24 analog lines each? |
| 16:18.03 | *** join/#asterisk codefreeze-lap (n=murf@71-36-6-234.chyn.qwest.net) |
| 16:18.22 | keith4 | so, <2 mbit for 24 ulaw calls? |
| 16:18.24 | keith4 | that's not bad at all |
| 16:18.25 | iCEBrkr | Well, Admins aren't notified if join/leaves |
| 16:18.58 | *** join/#asterisk thepacmanfan (n=thepacma@12-218-140-89.client.mchsi.com) |
| 16:19.03 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 16:19.15 | ManxPower | keith4: with GSM it gets much better. |
| 16:20.28 | *** join/#asterisk acxty (n=acxty@201.220.132.141) |
| 16:20.32 | keith4 | audicode's website is making me sad |
| 16:20.44 | *** part/#asterisk jarrod (n=jarrod@theos.org) |
| 16:21.28 | murdock_ut | Ok, i'll bite... Why? |
| 16:21.34 | ManxPower | Personally, I'd never use a SIP/analog gateway |
| 16:21.44 | ManxPower | I'll stick to T-1 cards and channel banks, thankyouverymuch |
| 16:21.55 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:21.59 | keith4 | I'm looking for more information about their MediaPack stuff |
| 16:22.08 | keith4 | but... their search functionality sucks |
| 16:22.24 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 16:22.29 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582051.dsl.bell.ca) |
| 16:22.31 | keith4 | ManxPower: yah... but that gets very expensive, very quicklyt |
| 16:22.42 | keith4 | what's a quad T1 card go for these days? |
| 16:22.44 | cpm | ManxPower, wanna buy my T1 card and channel bank? |
| 16:23.25 | cpm | wonders why folks think that telephone is supposed to be cheap? |
| 16:24.02 | jbeez | because comcast offers triple play and phone is just $33/month zomg |
| 16:24.10 | keith4 | hehe |
| 16:24.17 | keith4 | it doesn't have to be "cheap" |
| 16:24.31 | keith4 | but T1s and channel banks for 400 analog phones gets a bit ridiculous |
| 16:24.39 | jbeez | im really cheap, I have a 500 minute vonage plan at home because I rarely use my home phone, little linksys pap2, it comes to like $20/month w/ taxes and all |
| 16:24.54 | jbeez | 400 analog phones is ridiculous in and of itself |
| 16:25.52 | cpm | handling 400 analog phones with adapters, and expecting to not have endless headaches, failures, outages, irritated users, et al, is also ridiculous. carrier grade channel banks (most are) and T1s is the /right/ way to handle that. |
| 16:26.10 | cpm | and what jbeez said. |
| 16:26.22 | *** join/#asterisk dgzdd (n=chatzill@bdy93-5-82-235-80-162.fbx.proxad.net) |
| 16:26.25 | dgzdd | hi |
| 16:26.25 | cpm | in a roll out like that, I'd seriously consider going with sip phones |
| 16:27.01 | keith4 | yah, i would too |
| 16:27.12 | dgzdd | hi |
| 16:27.14 | keith4 | i forget who was asking about it... but he was talking about 400 sip and 400 analog, I think |
| 16:27.19 | dgzdd | Bandwidth information is it importante to calculate to have number of simultaneous calls ? |
| 16:27.27 | *** part/#asterisk SuD (n=Ask@89.140.32.2.static.user.ono.com) |
| 16:27.29 | ManxPower | cpm: I have like 4 spare Adtran TS750s |
| 16:27.50 | keith4 | dgzdd: for SIP? |
| 16:27.52 | dgzdd | having 8MB bandwith how many simulaneous calls |
| 16:27.54 | ManxPower | But thanks for the offer |
| 16:27.57 | dgzdd | can supply |
| 16:28.01 | dgzdd | yes sip of course |
| 16:28.05 | ManxPower | dgzdd: you must calculate that |
| 16:28.12 | keith4 | depends on the codec |
| 16:28.19 | ManxPower | voip-info.org has a link to a calculator for BW |
| 16:28.33 | dgzdd | how ? |
| 16:28.52 | *** join/#asterisk cli4me (n=root@cpe-071-070-229-009.nc.res.rr.com) |
| 16:29.26 | cli4me | anyone have a solution to bad DTMF when call is destined for * 1.2? |
| 16:29.34 | jbeez | 8Megabytes? |
| 16:29.42 | dgzdd | yes |
| 16:30.09 | jbeez | very nice |
| 16:30.37 | keith4 | uhhh |
| 16:30.54 | keith4 | dgzdd: you have 64Mbps connection? |
| 16:31.15 | jbeez | I think the question is, who doesn't? It's 2008, come on people |
| 16:31.19 | dgzdd | not for the moment |
| 16:31.37 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:31.51 | dgzdd | isp is working to have fiber optical connection |
| 16:33.10 | cli4me | ive checked online and the only solutions I find are for calls sourcing from the asterisk box, does anyone have a suggestion? |
| 16:33.44 | dgzdd | is it possible to reunify sevreal pc to have on one power pc runing with asterisk ??? |
| 16:33.57 | keith4 | i have a suggestion. describe your problem more thoroughly |
| 16:34.01 | dgzdd | is it possible ? |
| 16:34.12 | dgzdd | to make it possible ? |
| 16:34.19 | cli4me | ok. Inbound calling to asterisk box shows up with multiple DTMF sometimes but not all the time |
| 16:34.27 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 16:34.39 | cli4me | I would assume its an inband out of band thing, but from what I know that cannot be controlled on inbound calls |
| 16:35.04 | keith4 | what's the source? |
| 16:35.34 | dgzdd | hello |
| 16:35.44 | cli4me | different all the time. Land line, cell phone, etc. The carrier the call comes in on is different to. |
| 16:36.16 | keith4 | and what's your interface to the PSTN? |
| 16:36.33 | cli4me | my number goes through Level3 |
| 16:37.43 | keith4 | so you're using IAX or SIP upstream? |
| 16:37.59 | cli4me | yes I am using SIP upstream |
| 16:38.07 | keith4 | i don't know if there's anything you can do about that, then |
| 16:38.21 | keith4 | isn't that a problem with your upstream provider? |
| 16:38.50 | cli4me | would you agree that the duplicate digits (sometimes) are related to inband and out-of-band being sent? |
| 16:39.09 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:39.19 | cli4me | and if so, is there a way to 'ignore' one or the otheron the * box? |
| 16:39.48 | keith4 | dunno. i haven't seen that problem |
| 16:39.59 | keith4 | you should ask TK, or pray that manxpower comes back in a good mood |
| 16:40.44 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 16:41.02 | b11d` | can anyone tell me the name of a vitelity server they connect to? I'd just like to test latency from my location to there.. their support people dont seem to be getting back to me. |
| 16:41.17 | b11d` | i dont want to sign up for service and find 200ms ping responses :| |
| 16:42.01 | cli4me | TK? |
| 16:43.01 | *** join/#asterisk mltlnx (n=mltlnx@209.10.153.194) |
| 16:43.26 | *** join/#asterisk jets (n=brian@pdpc/supporter/active/jets) |
| 16:44.18 | b11d` | nevermind.. speak of the devil.. |
| 16:47.04 | [TK]D-Fender | here |
| 16:47.05 | *** join/#asterisk nezza-_- (i=troth@unixforge.de) |
| 16:47.39 | cli4me | TK, dont know if you can halp but, I was directed to you by keith4 |
| 16:47.41 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:47.41 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:48.06 | keith4 | whoa whoa, let's not get carried away here |
| 16:48.10 | cli4me | ha ha |
| 16:48.27 | cli4me | he said its either YOU, or it cant be done! |
| 16:48.36 | nezza-_- | Hi there! I've got the following question: I'm using a mediatrix 1204 VoIP to PSTN Gateway.. can anybody tell me howto do a call via a VoIP phone connected to the network to the PSTN line without an external sip server? |
| 16:48.40 | cli4me | j/k |
| 16:48.53 | [TK]D-Fender | cli4me: pick ONE mode and use it. |
| 16:49.19 | [TK]D-Fender | nezza-_-: Set aup a SIP peer in * and dial out that. |
| 16:49.40 | cli4me | the call is not sourced from the * box, its coming in to the asterisk box |
| 16:49.51 | cli4me | so I cant choose inband/out-of/band |
| 16:50.18 | cli4me | *out-of-band |
| 16:50.35 | [TK]D-Fender | keith4 : I'd use SIP gateways over channel banks any day. CB makes your T1 card the poitn of failure, has clocking concerns, adds the cost of the card, etc. SIP gateways can allow you to have a HA setup w/ redundant servers, removes the need for stupid DMTF features, zaptel, etc. |
| 16:50.36 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 16:50.41 | cli4me | i have sip.conf set for rfc2833 |
| 16:51.30 | keith4 | [TK]D-Fender: yah, I agree. It seems to be a polarizing question, though. |
| 16:52.38 | *** join/#asterisk ManxPower (n=manxpowe@79.sub-75-201-0.myvzw.com) |
| 16:53.07 | cli4me | hmm, so no real known solution? |
| 16:53.09 | [TK]D-Fender | keith4 : that'd add the cost of 5 x 4port T1 cards to the equation, meaning what, a minimum of 3 servers? (2 cards max each), and WORSE. |
| 16:53.58 | keith4 | yah, I don't know why the hell you would do that, when a single server can easily handle thousands of sip calls |
| 16:55.00 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:57.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:57.25 | adr3nalin3_ | Is there a driver I need to load for the TE122P? Shouldn't the zaptel driver load the driver for it? |
| 16:57.26 | [TK]D-Fender | keith4 : if you have a spare port or two fine, but to make it for a large scale deployment? No thanks. |
| 16:57.42 | [TK]D-Fender | adr3nalin3_: Zaptel, and Libpri for * |
| 16:57.57 | keith4 | adr3nalin3_: no, but udev should load it |
| 16:58.22 | keith4 | er, no the zaptel driver won't load it, but udev should load it |
| 16:58.35 | adr3nalin3_ | If I had compiled Zap and libpri before the card was installed that should be a problem should it? |
| 16:59.02 | adr3nalin3_ | I just put it in and it shows as 07:04.0 Ethernet controller: Digium, Inc. Unknown device 8001 (rev 11) when I lspci |
| 16:59.06 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 16:59.08 | adr3nalin3_ | Also no lights on the card show |
| 16:59.15 | [TK]D-Fender | adr3nalin3_: What ver of * / zaptel? |
| 16:59.34 | [TK]D-Fender | adr3nalin3_: And have you initialized the driver? modprobed? |
| 16:59.41 | [TK]D-Fender | adr3nalin3_: ztcfg -vvvv ? |
| 16:59.50 | cli4me | manxpower can you offer any insight on why duplicate digits come through on inbound to * 1.2? |
| 16:59.56 | [TK]D-Fender | adr3nalin3_: Got something comprehensive to show us? How about your configs? |
| 17:00.15 | BCS-Satori | What in asterisk makes up an encrypted password response to a SIP 407 Proxy Authentication Required. |
| 17:01.44 | keith4 | cli4me: what did I say about describing your problem better?? |
| 17:01.46 | keith4 | :-P |
| 17:02.09 | cli4me | you'd think I'd learn |
| 17:02.39 | nezza-_- | [TK]D-Fender: i've set the SIP Server source to static, the port to 3336 and the sip domain blank... but where can i add this SIP peer? |
| 17:02.55 | adr3nalin3_ | [TK]D-Fender: modprobed and I think it took care of it |
| 17:02.58 | [TK]D-Fender | nezza-_-: sip.conf like you would most any other itsp |
| 17:03.08 | coppice | anyone get echo problems with the linksys ATAs? |
| 17:03.38 | [TK]D-Fender | coppice: I had what I think is termed side-tone once a long time ago, but that was on the Sipura models. |
| 17:03.44 | nezza-_- | [TK]D-Fender: i want to do this ON the mediatrix box.. without any external asterisk server |
| 17:03.58 | cli4me | I have inbound calls to an * 1.2 box. the call when answered asks for digits. THe digits sometimes are duplicated (I.e. 1234 = 122344 or 1122344) Im thinking its because im receiving both inband and out-of-band from the carrier, but is there a way to ignore one or the other from the * box? |
| 17:04.03 | [TK]D-Fender | nezza-_-: You need a PEER ion * to tall * to USE IT. |
| 17:04.08 | [TK]D-Fender | tell* |
| 17:04.23 | coppice | [TK]D-Fender: http://www.rowetel.com/blog/?p=64 |
| 17:04.49 | nezza-_- | [TK]D-Fender: and how do i do this? |
| 17:05.14 | [TK]D-Fender | coppice: Yeah, it was on my SPA-3000 |
| 17:05.31 | [TK]D-Fender | nezza-_-: As I said... just like you would another ITSP. |
| 17:05.42 | igascream | Hi all need help can I invite someone to conference while talking there without holding the call? |
| 17:07.05 | [TK]D-Fender | igascream: something has to trigger a "call-out" in *. Either your phone, or a PC telling * to take this action. |
| 17:07.20 | keith4 | nezza-_-: are you asking if you can place a call from a SIP phone to an analog phone, through a media gateway, without an asterisk server involed at all? |
| 17:07.32 | [TK]D-Fender | coppice: They made a DSP to do OSLEC on the IP04? |
| 17:07.48 | nezza-_- | keith4: yeah |
| 17:07.59 | coppice | the IP04 is based on a DSP |
| 17:08.27 | keith4 | nezza-_-: let me ask you a question then. would you expect to be able to place a call from one analog phone to another, because they are both plugged into the same phone splitter? |
| 17:08.39 | BCS-Satori | Could someone tell me what in asterisk makes up an encrypted password response to a SIP 407 Proxy Authentication Required. Like username:password:relamn:port? Also does asterisk user its own tool to gernerate the md5 or does it use the linux md5sum tool? |
| 17:09.00 | nezza-_- | keith4: someone told me that the mediatrix 1204 is able to do such things.. |
| 17:09.30 | [TK]D-Fender | nezza-_-: Go read your admin manual. |
| 17:09.36 | keith4 | nezza-_-: well then maybe you should go ask that someone how to do it. because you're asking for help in #asterisk, but you're not using asterisk |
| 17:09.43 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:09.58 | nezza-_- | keith4: okay, thank you for your help. |
| 17:11.18 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-83-43.vif.net) |
| 17:12.55 | *** part/#asterisk nezza-_- (i=troth@unixforge.de) |
| 17:14.00 | igascream | <[TK]D-Fender>, and for example if I have two persons on two lines talking with can I put three of us to the conference without transfering eachone by itself. |
| 17:14.02 | [TK]D-Fender | coppice: Do you happen to know what kind of processing load OSLEC places per channel to get an idea of scalability for small (embedded) systems other than blackfin? |
| 17:14.43 | [TK]D-Fender | igascream: I jsut told you. You're either doing it YOURSELF on your phone, or via some external interface using AMI, etc. |
| 17:15.16 | coppice | it should be comparable to other good cancellers, though considerably higher than crude ones. on a PC I think it now uses MMX, so it should be fairly quick |
| 17:15.36 | tzafrir | [TK]D-Fender, try searching for OSLEC and MIPS |
| 17:17.00 | tzafrir | Can you safely use MMX in the kernel? |
| 17:17.02 | [TK]D-Fender | tzafrir : looking up now. See your name plastered all over it :) |
| 17:18.00 | tzafrir | We had that funny thread in the oslec list of someone complaining he suddenly can't properly login via ssh to the machine |
| 17:18.20 | [TK]D-Fender | tzafrir : .... and in other unrelated news .... |
| 17:18.21 | tzafrir | Which eventualy turned out to be MMX-related issue |
| 17:18.22 | [TK]D-Fender | lol |
| 17:19.38 | coppice | MMX or floats in the kernel can do some real funky obscure stuff, if you get the saves and restore wrong |
| 17:19.53 | *** join/#asterisk xenonex (n=xenonex@89.218.236.221) |
| 17:20.04 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:23.41 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:28.52 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 17:30.45 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:32.38 | keith4 | heh... is it bad to consider it a victory when you tell someone to (essentially) F off, and he thanks you before leaving the channel? |
| 17:32.51 | *** join/#asterisk smash- (n=smash@66.236.19.230) |
| 17:33.15 | smash- | hey, anyone point me in a direction to find a fair priced sip trunk provider? |
| 17:33.27 | rob0 | ~itsp |
| 17:33.28 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 17:33.36 | ManxPower | ~trunk |
| 17:33.36 | jbot | it has been said that trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 17:34.20 | smash- | damn i feel like i got reject at prom manx |
| 17:34.31 | smash- | stop bringing up repressed memories |
| 17:35.01 | *** join/#asterisk DarylVoip (n=daryl@c-71-224-53-6.hsd1.pa.comcast.net) |
| 17:35.02 | smash- | ok |
| 17:35.04 | rob0 | Elephant's trunk, storage trunk, swim trunks, automobile trunk |
| 17:35.05 | smash- | so whats a url for itsplist-us |
| 17:35.06 | smash- | itsplist-us |
| 17:35.24 | rob0 | The bot likes the ~ |
| 17:35.41 | rob0 | ~botsnack |
| 17:35.41 | jbot | rob0: :) |
| 17:35.42 | jets | LOL a SIP Provider I suppose |
| 17:35.43 | jets | LOLOL |
| 17:36.46 | [TK]D-Fender | ~itsplist-us |
| 17:36.46 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 17:39.01 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 17:41.50 | destructure | what qualifies as "more respected"? |
| 17:42.39 | ManxPower | destructure: ones where many people have good experiences. |
| 17:42.48 | ManxPower | The less respected ones are where many people had bad experiences |
| 17:43.05 | [TK]D-Fender | destructure: "not Vonage" and "has some decent feedback from people in here" |
| 17:45.16 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 17:46.04 | *** join/#asterisk mltlnx (n=mltlnx@static-64-115-158-106.isp.broadviewnet.net) |
| 17:47.26 | gitguy | is snom good? |
| 17:47.57 | ManxPower | gitguy: you already know the answer to that |
| 17:48.25 | gitguy | i heard they aren't good, but they use linux, how can't they be no good? :p |
| 17:48.26 | [TK]D-Fender | gitguy: How many more times would you like to ask the same questions? |
| 17:49.07 | gitguy | sorry, i didn't wanted to be annoying |
| 17:49.15 | cpm | can haz asterisk? |
| 17:49.27 | gitguy | but that's why my nick starts with "git" |
| 17:50.28 | gitguy | polycom is the best then? |
| 17:50.51 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 17:54.59 | gitguy | i'm getting a polycom, yeah xD |
| 17:56.12 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:59.45 | [TK]D-Fender | gitguy: Yes, Polycom is probably the best product to use, but not always the best value. |
| 17:59.56 | [TK]D-Fender | gitguy: that depends on where you are buying from/for. |
| 18:00.31 | mratliff | okay guys...another question...this time regarding raid |
| 18:00.40 | mratliff | would 1+0, 5, or 6 be best |
| 18:01.08 | mratliff | this server would have a heavy load potentially |
| 18:01.08 | [TK]D-Fender | mratliff: 6 clearly. |
| 18:01.13 | [TK]D-Fender | mratliff: 6 > 5 |
| 18:01.19 | mratliff | lol |
| 18:01.29 | [TK]D-Fender | mratliff: 1+0 < 5 < 6 |
| 18:01.33 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.4.19.2 (2008/05/13), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 18:01.35 | gitguy | [TK]D-Fender: ok |
| 18:03.12 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 18:03.26 | whymarkwh | hi there anyone know what openfire is? |
| 18:03.46 | *** join/#asterisk DaSkreech (n=skreech@katapult/ninja/daskreech) |
| 18:03.49 | DaSkreech | Hello |
| 18:03.58 | DaSkreech | is there a jingle plugin for asterisk? |
| 18:04.12 | whymarkwh | jingle? |
| 18:04.20 | whymarkwh | like in jingle bells? |
| 18:04.22 | russellb | chan_gtalk / chan_jingle, yes |
| 18:04.32 | russellb | whymarkwh: jabber/xmpp + voice |
| 18:04.54 | whymarkwh | what do you use it for? |
| 18:05.09 | beek | whymarkwh: openfile is a jabber server. |
| 18:05.13 | *** join/#asterisk mltlnx (n=mltlnx@static-64-115-158-106.isp.broadviewnet.net) |
| 18:05.21 | DaSkreech | whymarkwh: speechifying ? |
| 18:05.24 | whymarkwh | define jabber please |
| 18:05.25 | russellb | interconnection with googletalk for thing .. |
| 18:05.31 | russellb | sighs |
| 18:05.32 | DaSkreech | XMPP |
| 18:05.33 | whymarkwh | k insead of text its voice |
| 18:05.50 | coppice | i thought openfire was something you did to telemarketers |
| 18:05.54 | DaSkreech | Hmm I wonder if google caches that on it's server as well :) |
| 18:06.03 | beek | whymarkwh: http://www.igniterealtime.org/projects/openfire/index.jsp |
| 18:06.38 | DaSkreech | russellb: It's supposed to ship with the next stable release of asterisk ? |
| 18:06.50 | whymarkwh | if i google it i find:"police openfire on sivilians" and "us soldiers openfire on innocent iraques" |
| 18:06.52 | whymarkwh | lol |
| 18:07.10 | russellb | it's in 1.4 ... |
| 18:07.33 | DaSkreech | russellb: any caveats with it? |
| 18:07.36 | russellb | which has been out for 1.5 years |
| 18:07.37 | russellb | um ... |
| 18:07.45 | whymarkwh | does anyone know where i can find a working demo site to check out what it does? |
| 18:08.03 | whymarkwh | thx beek |
| 18:08.07 | russellb | DaSkreech: probably, heh, but i don't have any off of the top of my head |
| 18:08.07 | DaSkreech | Oh.. |
| 18:08.25 | DaSkreech | For somereason I though it was introduced into svn recently |
| 18:09.53 | *** part/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 18:09.57 | DaSkreech | beek: I'll assume the openfire is not in response to my jingle question? |
| 18:10.21 | beek | DaSkreech: You assume correctly. |
| 18:12.26 | DaSkreech | Is there a comparison between 1.6 and 1.4 ? |
| 18:16.07 | *** join/#asterisk mltlnx (n=mltlnx@static-64-115-158-106.isp.broadviewnet.net) |
| 18:19.59 | [TK]D-Fender | DaSkreech: yes, in upgrade.txt |
| 18:20.24 | DaSkreech | thanks |
| 18:20.25 | russellb | also see CHANGES |
| 18:20.30 | russellb | UPGRADE.txt, stuff you need to know when upgrading |
| 18:20.33 | russellb | CHANGES, list of new features |
| 18:20.44 | russellb | (mostly not including architecture and performance improvements ...) |
| 18:22.36 | *** join/#asterisk shido6 (n=shido6@74-130-51-159.dhcp.insightbb.com) |
| 18:23.21 | gitguy | russellb: do you also do changes in architecture? |
| 18:23.31 | russellb | yes |
| 18:23.37 | gitguy | nice |
| 18:25.27 | DaSkreech | Anyone running on 1.6 ? |
| 18:27.05 | *** join/#asterisk tobias (n=tobias@adsl-068-213-147-159.sip.rdu.bellsouth.net) |
| 18:31.12 | thepacmanfan | can anyone recommend a good 16 port PoE switch? |
| 18:31.28 | thepacmanfan | cheap would be nice, but i'm sure that won't happen. |
| 18:32.04 | [TK]D-Fender | thepacmanfan: 24 is about the same price actually... D-Link DES-1228P |
| 18:32.50 | [TK]D-Fender | http://www.newegg.com/Product/Product.aspx?Item=N82E16833127228 |
| 18:34.27 | DaSkreech | Hmm |
| 18:34.37 | DaSkreech | Asterisk didn't get a GSoC ? |
| 18:36.32 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 18:40.52 | bkruse | DaSkreech: nah |
| 18:41.06 | DaSkreech | Hmm |
| 18:41.09 | DaSkreech | Wesnoth did |
| 18:41.17 | bkruse | Digium could have, just not this year |
| 18:41.20 | bkruse | maybe next year :] |
| 18:51.51 | thepacmanfan | D-Fender: cool... i don't think we need one right now. |
| 18:52.31 | thepacmanfan | i've never used a Cisco router before... is it hard to set one up as a simple gateway, with DHCP? |
| 18:52.55 | ManxPower | thepacmanfan: yes, if you've never done it before. |
| 18:53.02 | *** join/#asterisk Hawk36 (n=me@modemcable202.30-70-69.static.videotron.ca) |
| 18:53.08 | Hawk36 | Hi all |
| 18:53.31 | Hawk36 | Is there a basic extensions.conf using les.net for incomming calls? |
| 18:53.47 | Hawk36 | I can dial out but can't receive calls :( |
| 18:53.52 | ManxPower | Hawk36: If there is, it would come from les.net |
| 18:54.05 | Hawk36 | I tried but they don't have it |
| 18:54.12 | thepacmanfan | well, i need something a step above the average Linksys junk, but i'm not a CCNA... |
| 18:54.18 | Hawk36 | They only give me the dial plan but nothing for dialing in |
| 18:54.29 | thepacmanfan | i've had too many WRT54Gs and BEFSR41s die on me |
| 18:54.49 | ManxPower | Dial(SIP/${EXTEN}@sipconfriendorpeerentryforlessnet |
| 18:54.56 | ManxPower | remember the closing ) |
| 18:55.12 | ManxPower | thepacmanfan: I use almost all Cisco routers and switches |
| 18:55.41 | [TK]D-Fender | thepacmanfan: Um. Its just a SWITCH... |
| 18:56.02 | ManxPower | For incoming calls, exten => yourdidnumber,1,Whatever |
| 18:56.11 | Hawk36 | Manx that is four outbound no? |
| 18:56.28 | Hawk36 | exten => _X.,n,Dial(SIP/lesnet_peer/${EXTEN}) |
| 18:56.29 | ManxPower | in the context incoming calls come into. You of course have eliminated and registration problems first, right? |
| 18:56.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:56.36 | thepacmanfan | D-Fender: yeah... i have a number of non-PoE switches around, and i'll make do with them, but i do need a new router. |
| 18:56.40 | ManxPower | Hawk36: you don't want that. |
| 18:56.49 | ManxPower | You want something more specific of what you are dialing. |
| 18:56.57 | ManxPower | Maybe _NXXNXXXXXX,1,Whatever |
| 18:57.20 | ManxPower | this is no different than the setup for any of the gadzillion service providers out there. |
| 18:57.23 | Hawk36 | So what I have is wrong? |
| 18:57.44 | mratliff | <[TK]D-Fender>: how hard is it to build Asterisk on a distributed server design? ...example would be to have redundant servers for call processing, voicemail, and maybe a pstn gatway (may not be necessary though)....any thoughts |
| 18:58.01 | ManxPower | Hawk36: no idea, you've not provided us with ANY pastes of failed incoming, outgoing calls, nor the relevant parts of your dialplan. Use pastebin.ca for that |
| 18:58.07 | [TK]D-Fender | mratliff: I'm getting the impression you've never worked with * before. |
| 18:58.40 | mratliff | actually I have ,but it's been 3 years back...built one to integrate into an avaya sys |
| 18:58.42 | Hawk36 | Manx, basically when I dial my DID, my asterisk answers the call |
| 18:58.56 | Hawk36 | Then just hangs up |
| 18:59.02 | mratliff | was just one server though...pretty easy |
| 18:59.12 | ManxPower | Hawk36: until you provide some pastebins I cannot help you further. |
| 18:59.16 | Hawk36 | [lesnet-incoming] |
| 18:59.16 | Hawk36 | exten => _X.,1,Answer |
| 18:59.16 | Hawk36 | exten => _X.,n,Goto(incoming,s,1) |
| 18:59.43 | ManxPower | Hawk36: CLI. OUTPUT. ON. PASTEBIN.CA |
| 18:59.50 | ManxPower | I will not ask again |
| 19:00.05 | Hawk36 | Sorry |
| 19:00.40 | Hawk36 | do you have info on that |
| 19:00.45 | ManxPower | ~pb |
| 19:00.45 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:00.59 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:01.03 | ManxPower | the site is pretty self explainatory |
| 19:02.20 | ManxPower | I'm leaving shortly to lay out and work on my tan, so we don't have a lot of time to get this working for you. |
| 19:03.30 | Hawk36 | I did paste it at the web site |
| 19:03.36 | Hawk36 | How do I know if you got it |
| 19:03.45 | ManxPower | you give me the link it gives you |
| 19:04.26 | mratliff | <[TK]D-Fender>: so no thoughts...? |
| 19:04.44 | Hawk36 | I feel stupid, but I see no links |
| 19:04.54 | [TK]D-Fender | mratliff: I think you'd better do some real research of your own or just hire a consultant. |
| 19:05.03 | ManxPower | Your paste has been accepted an added to the database. You will be redirected to it momentarily. The URL for it is:http://pastebin.ca/1017168You may use that URL for referencing your submission from now on. |
| 19:05.07 | ManxPower | Ya know, that message |
| 19:05.20 | Hawk36 | http://pastebin.com/d5b9be99a |
| 19:05.27 | Hawk36 | is that it |
| 19:05.41 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
| 19:05.47 | ManxPower | that is extensions.conf I need the CLI output, the stuff from "asterisk -rvvv" |
| 19:05.58 | ManxPower | Hawk36: perhaps you should read The Book before you do anything else. |
| 19:06.00 | Hawk36 | ok |
| 19:06.55 | ManxPower | For one thing EVERY extension MUST start with priority 1, I see at least one extension in your extensions.conf paste that does not have a priority 1 |
| 19:07.11 | Hawk36 | http://pastebin.com/d2317ccb2 |
| 19:07.34 | ManxPower | You even have a Goto pointing to the non existent priority 1: Goto(incoming,s,1) |
| 19:08.11 | thepacmanfan | manxpower, is a 3640 worth $60 more than a 2610? |
| 19:08.13 | Hawk36 | I changed it |
| 19:08.14 | ManxPower | Hawk36: You do not have the sound files installed that you arre trying to play. |
| 19:08.27 | jer | thepacmanfan, yes |
| 19:08.30 | Hawk36 | That is true |
| 19:08.37 | ManxPower | thepacmanfan: I think so, you can compare specs at cisco.com |
| 19:08.53 | Hawk36 | But it just sits there and then hangs up after a few seconds |
| 19:09.01 | ManxPower | How is this message unclear? |
| 19:09.02 | ManxPower | [May 13 15:06:32] WARNING[8593]: file.c:607 ast_openstream_full: File enter-extension-or-2 does not exist in any format |
| 19:09.14 | Hawk36 | It is very clear |
| 19:09.17 | ManxPower | correct, it can't play the sound file so it sits in the waitexten until timeout. |
| 19:09.19 | Hawk36 | I tried many things |
| 19:09.44 | jbeez | I've seen some useless error messages.... that is def not one of them |
| 19:09.48 | ManxPower | Where is the file enter-extension-or-2 located? |
| 19:10.17 | ManxPower | and what format is it in? |
| 19:10.24 | Hawk36 | Ok even if I remove that file, used as a test it just does nothing |
| 19:10.32 | Hawk36 | Line I mean |
| 19:10.43 | ManxPower | Hawk36: of course it does something, it is waiting for you to dial an extension |
| 19:10.59 | Hawk36 | When I dial the extension, nothing happens |
| 19:11.06 | Hawk36 | No message |
| 19:11.09 | ManxPower | You don't have any other extensions defined. |
| 19:11.18 | Hawk36 | And it does not transfer to that extension |
| 19:11.21 | ManxPower | Hawk36: Asterisk will almost never play any message unless you tell it to. |
| 19:11.22 | [TK]D-Fender | ManxPower: prepare fora glowing example of not knowing what an "extension" is. |
| 19:11.36 | ManxPower | Hawk36: You HAVE no other extensions! |
| 19:11.49 | ManxPower | [TK]D-Fender: I'll beat them with a nerf bat until they do |
| 19:12.13 | *** part/#asterisk DaSkreech (n=skreech@katapult/ninja/daskreech) |
| 19:12.15 | Hawk36 | Keep beating, someday I will understand and be gratefull to those who were patient |
| 19:12.21 | ManxPower | unless you did not provide complete information in the pastebin |
| 19:12.34 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 19:12.39 | ManxPower | Well, actually I KNOW you did not provide accurate info in your first pastebin |
| 19:12.47 | ManxPower | As pasted that dialplan could NEVER work. |
| 19:12.52 | Hawk36 | Ok, I wish to have the call transfered to extension 100 when they dial 100 at the wait |
| 19:12.52 | mratliff | touché |
| 19:13.09 | ManxPower | Hawk36: OK, now start providing COMPLETE extensions.conf |
| 19:13.39 | Hawk36 | Hold on |
| 19:13.53 | ManxPower | Hawk36: There is no extension 100 in that dialplan, for example. |
| 19:13.58 | Hawk36 | http://pastebin.com/d689f5847 |
| 19:14.11 | Hawk36 | This is basic |
| 19:14.16 | [TK]D-Fender | Hawk36: Stop now and go read the book till your eyes bleed. You clearly do not understand the dialplan at all. |
| 19:14.21 | Hawk36 | Learning and playing with it as I go |
| 19:14.33 | [TK]D-Fender | Hawk36: Chapter 5 <----- |
| 19:14.48 | ManxPower | Add "include => internal" at the end of [incoming] |
| 19:14.51 | Hawk36 | Fender, I did that, and I don't understand how it transfers to the extensions |
| 19:14.58 | Hawk36 | Sorry but I don't |
| 19:15.05 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.80.140) |
| 19:15.14 | ManxPower | your incoming context did not know anything about extension 100 because extension 100 is in a different context and not include =>'d anyhwere. |
| 19:15.28 | Hawk36 | ahhhh |
| 19:15.30 | anonymouz666 | Does chan_sip use dnsmgr to refresh the hosts in sip.conf? |
| 19:15.44 | [TK]D-Fender | Hawk36: You don't "transfer" ANYTHING. What you are allowed to dail are grouped into CONTEXTS. You do not have an EXTENSION in there that you can dial to do anything whatsoever |
| 19:15.59 | ManxPower | I'm outta here |
| 19:16.47 | anonymouz666 | jpeeler: weren't you working on dnsmgr with sip? |
| 19:17.25 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 19:18.07 | *** join/#asterisk Mikeonline (i=Mike@p57A7F6F5.dip.t-dialin.net) |
| 19:18.11 | Mikeonline | hi |
| 19:18.11 | Hawk36 | Manx thanks |
| 19:18.34 | Hawk36 | Fender, not clear |
| 19:19.06 | Hawk36 | Is there a difference in having the include before or after? |
| 19:19.55 | anonymouz666 | I am just trying to find a way to perform DNS lookups (interval refresh) in my SIP peers hosts. |
| 19:20.25 | Mikeonline | hm can i query AstDB in voicemail.conf to use the mailbox password from the database? |
| 19:20.29 | jpeeler | anonymouz666: yes |
| 19:20.31 | anonymouz666 | dnsmgr does not seem to work, so I assume that chan_sip does not use it. |
| 19:20.38 | jpeeler | you mean the hosts that sip show registry reports? |
| 19:20.48 | *** join/#asterisk CrashHD (n=CrashHD@65.74.161.225) |
| 19:21.03 | anonymouz666 | [peer1] host=test.domain.com |
| 19:21.06 | anonymouz666 | and the you sip show peers |
| 19:21.19 | anonymouz666 | and it is listes as 127.0.0.1 - but the IP already changed to 127.0.0.2 |
| 19:21.25 | anonymouz666 | so I need to manually do a "sip reload" |
| 19:21.33 | jpeeler | dnsmgr refresh |
| 19:21.46 | anonymouz666 | the verbose is bigger than 3 |
| 19:21.50 | jpeeler | it periodically does that of course for you |
| 19:21.54 | anonymouz666 | i can't see any report on dnsmgr |
| 19:22.02 | anonymouz666 | strange |
| 19:22.11 | anonymouz666 | so it should work then... |
| 19:22.20 | jpeeler | no report? hmm |
| 19:22.32 | jpeeler | it should at least say refreshing dns lookups |
| 19:22.33 | Hawk36 | Ok, it answers |
| 19:22.58 | Hawk36 | but I can't reach my extension |
| 19:23.06 | Hawk36 | Even after adding the include as indicated |
| 19:23.48 | anonymouz666 | jpeeler: yeap, thats why I asked... something like this should be printed ast_verbose(VERBOSE_PREFIX_2 "refreshing '%s'\n", entry->name); |
| 19:24.08 | anonymouz666 | btw this is version 1.2 |
| 19:24.25 | *** join/#asterisk mltlnx (n=mltlnx@207-237-36-133.c3-0.nyw-ubr3.nyr-nyw.ny.static.cable.rcn.com) |
| 19:24.28 | jpeeler | oh, well... i don't think those changes made it back to that version |
| 19:25.31 | jpeeler | anonymouz666: in fact it isn't in 1.4 either |
| 19:25.35 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:27.45 | *** join/#asterisk mltlnx (n=mltlnx@207-237-36-133.c3-0.nyw-ubr3.nyr-nyw.ny.static.cable.rcn.com) |
| 19:27.50 | keith4 | uh oh. i see Hawk36 is back |
| 19:28.38 | anonymouz666 | jpeeler: so the dnsmgr should work with chan_sip in host= (peer)? |
| 19:28.59 | Hawk36 | yeah |
| 19:29.29 | Hawk36 | Slowly learning the basics |
| 19:29.31 | anonymouz666 | even if I am not using DNS SRV? |
| 19:30.37 | jpeeler | anonymouz666: that's correct, if no srv record is found it will just do a normal lookup. but this behavior is only present in 1.6 |
| 19:31.07 | b11d` | IAX is supposed to work well behind NAT correct? |
| 19:31.18 | jpeeler | b11d`: yep |
| 19:31.21 | [TK]D-Fender | b11d`: easier than SIP+RTP anyways |
| 19:31.23 | defsdoor | anyone here use sangoma a500 ? |
| 19:31.40 | [TK]D-Fender | b11d`: Forward UDP 4569 and thats it. |
| 19:31.57 | anonymouz666 | jpeeler: how are the behavior in 1.2/1.4 version? |
| 19:31.58 | b11d` | sweet.. |
| 19:32.00 | b11d` | thanks TK |
| 19:33.25 | anonymouz666 | I just need a normal lookup :) |
| 19:35.51 | jpeeler | anonymouz666: since dnsmgr isn't present, the IP change isn't going to be detected |
| 19:36.17 | jpeeler | i'm not sure if there is another way to force the lookup |
| 19:36.33 | jpeeler | that's why dnsmgr support was added :) |
| 19:36.53 | anonymouz666 | "sip reload" do it, but thats not a good fix |
| 19:37.32 | anonymouz666 | damn, if I was using IAX2 in this version, chan_iax2 supports it |
| 19:38.25 | Hawk36 | Finally got it |
| 19:38.51 | Hawk36 | Thanks to the ones who helped or guided me |
| 19:38.53 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:39.20 | Hawk36 | exten => s,n,Dial(SIP/100,30) |
| 19:39.45 | Hawk36 | Is there a way to dial the extension instead of forcing it too 100 |
| 19:40.39 | anonymouz666 | jpeeler: what about ast_dnsmgr_lookup()? |
| 19:41.27 | *** join/#asterisk killab33z (i=40166060@gateway/web/ajax/mibbit.com/x-6ca3855b18543ede) |
| 19:42.42 | killab33z | anyone got faxs working on their server? |
| 19:42.49 | [TK]D-Fender | Hawk36: put an actual extension they can dial. |
| 19:43.31 | Hawk36 | So I would have to do that line for each extension correct? |
| 19:43.39 | *** join/#asterisk grandpapadot (n=anonymou@mail.heavylogic.com) |
| 19:44.03 | Hawk36 | If I had 3 extensions let's say 100, 010 and 102 I would write the same line tree times |
| 19:44.10 | Hawk36 | for each extension correct? |
| 19:44.40 | [TK]D-Fender | Hawk36: not sure what you mean but I think its a "yes". |
| 19:45.00 | killab33z | i heard the best way to recieve faxes in asterisk is with iaxmodem? |
| 19:45.19 | [TK]D-Fender | killab33z: Most seem to agree with that. |
| 19:45.51 | killab33z | would you happen to know a decent article on using the two? |
| 19:46.04 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 19:46.57 | [TK]D-Fender | ~wikis |
| 19:46.57 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 19:47.08 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:47.52 | *** join/#asterisk eklof (i=jonas@trimix.eklof.eu) |
| 19:48.49 | eklof | Hi guys. I have a agi-script that won't execute due to a "permission denied" error, have tried setting the script to 777 and it still says the same. Anyone have a cvlue as to why ? |
| 19:49.43 | killab33z | chown? |
| 19:50.01 | waKKu | mount options on partition ? |
| 19:50.04 | eklof | I'v chown it to asterisk and tested aswell, it was root initially. |
| 19:50.26 | eklof | UUID=cdf2246a-bc26-4466-bca8-e757591f1155 /home ext3 defaults,usrquota,grpquota 0 2 |
| 19:50.34 | eklof | sorry wrong one |
| 19:50.43 | eklof | UUID=f9bfeb9b-1673-409d-858b-dd9122208d9f /usr ext3 defaults 0 2 |
| 19:50.58 | *** join/#asterisk Skarmeth (n=Skarmeth@201009042244.user.veloxzone.com.br) |
| 19:51.06 | jpeeler | anonymouz666: yes, that is the function. but it's not in chan_sip for 1.4 |
| 19:51.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:51.29 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 19:51.44 | jpeeler | although the changes are not extensive, it wasn't trivial to add |
| 19:52.27 | anonymouz666 | jpeeler: I am looking at your code in sip_registry() |
| 19:53.43 | jpeeler | and sip_peer |
| 19:54.29 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 19:55.26 | *** join/#asterisk clive- (n=pirch@dsl-242-156-73.telkomadsl.co.za) |
| 19:55.39 | anonymouz666 | jpeeler: do you think that could be a way to look at this changes and try to port to chan_sip running 1.2? |
| 19:56.02 | anonymouz666 | I really need the dnsmgr refresh working with chan_sip |
| 19:56.09 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:56.55 | jpeeler | it's possible i guess, i'm really very unfamiliar though with 1.2 |
| 19:57.09 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 19:59.24 | mratliff | Is SER considered a better alternative to using Asterisk as just a pstn gateway? |
| 20:00.31 | b11d` | how good is audio quality when using g729? is it worth it? or will there be a noticable degradation? |
| 20:00.43 | b11d` | or should I stfu, and try it to see? :) |
| 20:00.47 | [TK]D-Fender | mratliff: SER isn't a PSTN gateway, it is a SIP proxy and last I checked had no way to interface with PSTN hardware |
| 20:01.02 | [TK]D-Fender | b11d`: the latter goes without saying ;) |
| 20:01.07 | b11d` | :) |
| 20:01.10 | b11d` | figures lol |
| 20:01.17 | [TK]D-Fender | b11d`: but the question is what do you want it for? |
| 20:01.44 | mratliff | sorry for all of the novice questions...I'm slowly re-building my knowledge |
| 20:01.55 | mratliff | thx for your help so far |
| 20:02.03 | mratliff | i do appreciate it |
| 20:02.15 | b11d` | im just looking for the best quality VoIP service.. and ona 7mb down, 1mb up DSL connection.. it sounds OK now, but I get a little lag.. |
| 20:02.28 | b11d` | latency is latency..wasnt sure if g729 would actually help with that or not |
| 20:03.25 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 20:04.49 | [TK]D-Fender | b11d`: how many calls, what protocols? |
| 20:08.22 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 20:08.57 | b11d` | maybe three or four calls at a time.. I am using IAX right now.. |
| 20:09.37 | b11d` | im looking at about 80-90ms ping replies from the asterisk box to the far end.. |
| 20:10.12 | [TK]D-Fender | b11d`: codec? |
| 20:10.17 | b11d` | right now, g711.. |
| 20:10.47 | b11d` | far end supports g711 and g729a |
| 20:12.38 | *** join/#asterisk adr3nalin3 (n=afink@66.172.245.81) |
| 20:13.42 | [TK]D-Fender | b11d`: how is a single channel with little extra traffic? |
| 20:14.56 | b11d` | sounds fine.. like i said, i am only picking up a little bit of lag.. not much.. |
| 20:15.08 | b11d` | just trying to get the best out of it is all.. |
| 20:17.13 | adr3nalin3 | Hey guys I am having trouble hooking up my digium t1 card with the telco PRI. I got the digium driver loaded and everything. When I change the switch type to Nortel DMS all alarms clear but the telco says I am throwing errors on their end. Also in the zaptel.conf span=1,1,0,esf,b8zs line it should auto detect which one is being used correct? |
| 20:19.42 | [TK]D-Fender | b11d`: if you get that regardless on an single channel with little traffic, codec won't matter |
| 20:20.03 | b11d` | i figured as much.. the lag is not that noticable anyways.. i appreciate it TK. Thanks. |
| 20:20.31 | [TK]D-Fender | b11d`: note I HAVE seen IAX lag all by itself whereas SIP would not. I'd advise testing. |
| 20:20.53 | [TK]D-Fender | adr3nalin3: signallng is in zapata.conf. |
| 20:21.17 | [TK]D-Fender | adr3nalin3: And you'd be advised to pastebin both in their entirety |
| 20:22.25 | b11d` | will do TK.. |
| 20:25.43 | [TK]D-Fender | Ok, checkout time here. Later all |
| 20:28.33 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:33.12 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 20:33.36 | *** join/#asterisk comprookie2000 (n=comprook@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 20:34.01 | *** join/#asterisk MACscr (n=Mark@c-98-214-107-162.hsd1.il.comcast.net) |
| 20:34.44 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 20:34.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:34.45 | MACscr | I want to turn on monitoring for a sip trunk. Can't seem to remember the text I need to add to the conf. Anyone? |
| 20:36.49 | LinuxMafia | hi guys |
| 20:37.28 | LinuxMafia | i bought this one -->http://www.canadacomputers.com/index.php?do=ShowProduct&cmd=pd&pid=013131&cid=828.480 |
| 20:37.35 | LinuxMafia | and i have a router |
| 20:38.09 | LinuxMafia | is that device is like a PC? |
| 20:38.21 | LinuxMafia | or it needs a ethernet card? |
| 20:38.35 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:39.12 | MACscr | it doesn't need ethernet card, it has its own. You just need to plug it into your network |
| 20:39.43 | LinuxMafia | MACscr, yeah but what will happen to my PC? |
| 20:40.07 | LinuxMafia | MACscr, then how do i connect it to internet? |
| 20:40.15 | MACscr | why would it affect your pc? if your really asking these questions, you shouldn't be messing with voip |
| 20:40.21 | MACscr | er, asterisk |
| 20:40.55 | LinuxMafia | MACscr, look the internet from ISP , first should go to the router or the adaptor? |
| 20:41.01 | LinuxMafia | which one first? |
| 20:41.17 | MACscr | router, you can't do it any other way |
| 20:41.29 | LinuxMafia | ok |
| 20:41.41 | *** part/#asterisk MACscr (n=Mark@c-98-214-107-162.hsd1.il.comcast.net) |
| 20:41.51 | b11d` | anyone here use vitelity? |
| 20:42.05 | b11d` | i cant seem to figure out why im getting inbound busy signals.. everything looks good.. obviously isnt :) |
| 20:42.21 | Strom_C | busy signals, or reorder tone ("fast busy")? |
| 20:42.29 | b11d` | nah normal busy.. not reorder. |
| 20:42.29 | rob0 | sip debug is your friend |
| 20:42.34 | b11d` | im using IAX though |
| 20:42.36 | *** join/#asterisk razu (n=razu@195.222.7.33) |
| 20:42.44 | rob0 | s/sip/iax/ |
| 20:42.46 | b11d` | doh |
| 20:42.48 | b11d` | :) |
| 20:42.48 | LinuxMafia | guys |
| 20:42.48 | b11d` | lol |
| 20:42.54 | LinuxMafia | some one please direct me |
| 20:43.00 | Strom_C | points |
| 20:43.02 | Strom_C | THAT WAY |
| 20:43.02 | razu | hey ... has anyone used sangoma A500 isdn card with asterisk 1.6 ? |
| 20:43.04 | b11d` | i dont seem to have iax2 debugging :/ |
| 20:43.22 | LinuxMafia | i have a router and this adaptor --> http://www.canadacomputers.com/index.php?do=ShowProduct&cmd=pd&pid=013131&cid=828.480 |
| 20:43.24 | LinuxMafia | so |
| 20:43.34 | LinuxMafia | how i connect this |
| 20:43.34 | b11d` | doh |
| 20:43.37 | b11d` | iax2 set debug :) |
| 20:44.24 | b11d` | nothing comes across my console.. |
| 20:44.32 | b11d` | core, iax2, and debug all set to 100 |
| 20:44.43 | b11d` | outbound works, for what its worth |
| 20:44.59 | b11d` | i'll pastebin my confs.. |
| 20:46.13 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 20:48.16 | *** join/#asterisk RoyK (n=roy@72.84-48-10.nextgentel.com) |
| 20:50.01 | b11d` | http://www.pastebin.ca/1017267 |
| 20:50.07 | b11d` | iax.conf and extensions.conf |
| 20:50.30 | jayrod422 | anyone here use lsms |
| 20:50.54 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 20:51.17 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 20:52.33 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 20:53.16 | *** join/#asterisk whymarkwhy (n=dsfsdfsd@196.211.34.2) |
| 20:53.20 | keith4 | ~ask |
| 20:53.20 | jbot | somebody said ask was Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:53.56 | *** part/#asterisk eklof (i=jonas@trimix.eklof.eu) |
| 20:54.26 | b11d` | ARRRRRRRRRRRRRRRRRRRRRRRRRRR |
| 20:54.32 | b11d` | damn inbound busy !!@! |
| 20:54.51 | b11d` | sigh.. |
| 20:56.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:59.12 | rob0 | firewall? |
| 20:59.32 | b11d` | not that I can tell.. cant guarantee that it isnt either.. |
| 20:59.53 | b11d` | i dont BELIEVE my provider is blocking it.. but I cant be certain either. |
| 21:00.01 | rob0 | router or direct connection? |
| 21:00.06 | b11d` | direct.. no nat.. |
| 21:00.20 | rob0 | I could believe they'd block SIP, but IAX2 less likely. |
| 21:00.47 | b11d` | I do pass through a few of my providers firewalls.. no doubt.. it is very possible they are blocking it.. outbound works though.. oddly. |
| 21:01.15 | rob0 | Not odd at all, if your firewall is blocking inbound. |
| 21:01.19 | b11d` | aye |
| 21:01.34 | b11d` | i will be trying it from a different location tonight.. will have to see then.. |
| 21:01.47 | rob0 | just disable the firewall to test |
| 21:01.53 | b11d` | cant.. they dont belong to me. |
| 21:01.58 | b11d` | they belong to the state of MN.. |
| 21:02.18 | Hawk36 | What can cause for the voice to not travel during incomming calls? |
| 21:02.21 | b11d` | although reviewing the rules, i see nothign on it which appears to be blocking that port.. |
| 21:02.31 | b11d` | doesnt mean there isnt a firewall further upstream blocking it though |
| 21:02.43 | [TK]D-Fender | Hawk36, read up : |
| 21:02.44 | [TK]D-Fender | ~sipnat |
| 21:02.45 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:02.46 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 21:03.09 | rob0 | does the state of MN want you to get IAX2 phone calls on this IP? |
| 21:03.24 | b11d` | no they dont really know about it.. |
| 21:03.27 | b11d` | dont see why they wouldnt |
| 21:04.27 | Hawk36 | Fender thanks, exactly what I was looking for |
| 21:04.57 | *** join/#asterisk anil-ast (n=a@59.93.74.58) |
| 21:05.00 | rob0 | so figure out what hosts will be sending those inbound calls, and accept all from them. |
| 21:05.04 | *** join/#asterisk LinuxMafia_ (n=awatt@CPE001346a4c4cb-CM00159a642d7e.cpe.net.cable.rogers.com) |
| 21:05.07 | LinuxMafia_ | hi |
| 21:05.21 | anil-ast | hello - does anyone experience with speech recognition on asterisk? Not lumenvox - ? |
| 21:05.29 | LinuxMafia_ | any one can help me set my ip phone |
| 21:06.09 | LinuxMafia_ | i have a modem , a router , and phone adaptor |
| 21:06.16 | LinuxMafia_ | so what i have to do |
| 21:06.17 | LinuxMafia_ | ? |
| 21:06.37 | [TK]D-Fender | LinuxMafia_, www.voxilla.com <- go read to forums. |
| 21:06.52 | [TK]D-Fender | the* |
| 21:06.57 | LinuxMafia_ | [TK]D-Fender, hi i bought that thing |
| 21:07.17 | anil-ast | Is there a command which plays a file and stops while the caller is speaking something. Right now, we have solution integrated with nuance but the caller has to wait till the prompt is over as anything he speaks during does not stop the recording. |
| 21:07.44 | [TK]D-Fender | anil-ast, No, there is no "stop on audio" option. |
| 21:08.04 | LinuxMafia_ | [TK]D-Fender, it does not load |
| 21:08.08 | anil-ast | how does lumevox do it? |
| 21:08.44 | anil-ast | there is also a speech api which is not documented properly. I am willing to write a connector but the documentation on that is poor |
| 21:08.47 | [TK]D-Fender | anil-ast, must be part of their playback option, not part of * |
| 21:10.10 | LinuxMafia_ | [TK]D-Fender, that page wont load |
| 21:10.11 | [TK]D-Fender | LinuxMafia_, http://www.google.ca/search?hl=en&q=SPA-2102+asterisk+setup+guide&btnG=Search&meta= |
| 21:10.38 | LinuxMafia_ | [TK]D-Fender, i can not set up the hardware |
| 21:10.48 | anil-ast | there is something similar but not exactly I want.. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BackGroundDetect |
| 21:11.26 | [TK]D-Fender | LinuxMafia_, Whats that supposed to mean? |
| 21:11.55 | LinuxMafia_ | [TK]D-Fender, i went by manual , i have the manual |
| 21:12.08 | anil-ast | we plan to use in agi and I am not sure how to do it there. also, the original spoken speech should be retained before it jumps. speech engines require it |
| 21:12.21 | [TK]D-Fender | LinuxMafia_, JFGI |
| 21:12.47 | LinuxMafia_ | i can not get connected to internet |
| 21:12.59 | LinuxMafia_ | when i go by manual |
| 21:13.00 | [TK]D-Fender | LinuxMafia_, then how the #%^$ are we chatting now? |
| 21:13.17 | LinuxMafia_ | [TK]D-Fender, i disconnect the device |
| 21:13.33 | LinuxMafia_ | it says disconnect you pc from modem |
| 21:13.59 | LinuxMafia_ | then connect ethernet cable to the port |
| 21:14.00 | [TK]D-Fender | anil-ast, Actually I seem to be mistaken in that backgroundDetect is a valid * native app. Have you tried it? |
| 21:14.17 | LinuxMafia_ | connect the other end to pc |
| 21:14.17 | [TK]D-Fender | LinuxMafia_, You don't have to disconnect from the internet for that device. JFGI |
| 21:14.22 | anil-ast | we plan to use in agi |
| 21:14.33 | anil-ast | also, off topic.. |
| 21:14.46 | anil-ast | any good termination provider in US for less than 1.3c |
| 21:14.50 | LinuxMafia_ | [TK]D-Fender, it says on the manual |
| 21:15.19 | anil-ast | we terminate over 1M calls.. |
| 21:15.35 | [TK]D-Fender | anil-ast, at that point I think I'd call Level3 directly... |
| 21:16.04 | anil-ast | level3 - whats the minimum - i think they have 50k min every month. |
| 21:16.20 | jduggan | hey guys, i understand hang up detection on analogue is generally fubarred, but when someone calls in via my analogue line and a sip extension answers, if the sip extension hangs up the call, the phone on the other end stays open, it doesnt know it disconnected - is there something i can tweak to sort this? |
| 21:16.20 | LinuxMafia_ | [TK]D-Fender, i am so confused , there is an ethernet jack and internet jack on device |
| 21:16.46 | [TK]D-Fender | LinuxMafia_, JFGI <------------------ |
| 21:17.00 | LinuxMafia_ | [TK]D-Fender, what is JFGI? |
| 21:17.06 | [TK]D-Fender | ~jfgi |
| 21:17.06 | jbot | http://www.google.com/search?q=jfgi |
| 21:18.07 | mocker | So when adding users to the sip.conf, do most people do extensions for the username, or an actual username for the username? |
| 21:18.23 | mocker | I'm trying to decide if I should switch away from extensions as my standard. |
| 21:18.34 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 21:18.37 | [TK]D-Fender | mocker, hard to say doesn't really matter so much. |
| 21:19.26 | mocker | [TK]D-Fender: What do you do? |
| 21:20.44 | [TK]D-Fender | mocker, for most installs I have it match the "logical extension" it'd be associated with |
| 21:21.04 | adeel | anyone upgrade their polycom phones to 3.0 firmware yet? |
| 21:21.47 | mocker | [TK]D-Fender: So [5555] for extension 5555? |
| 21:21.55 | mocker | That's the way that I generally do it. |
| 21:21.56 | [TK]D-Fender | mocker, yup |
| 21:22.16 | mocker | Ok, just a general practice question that I thought I'd throw out therer. |
| 21:22.18 | mocker | :) |
| 21:24.59 | [TK]D-Fender | mocker, some prefer to use a "friendly name". Others (like Manxpower) prefer something a little lower level like using the MAC of a hardware device thats associated with it |
| 21:25.04 | jackson__ | I have a multihomed Asterisk box; eth0-Internet, eth1-LAN. When disconnecting the eth0 ethernet cable (simulate Internet problems), I find that my LAN based sip phones are unable to maintain registered status. Any idea why? I can still ping the sip phones (on the LAN) from the Asterisk box. |
| 21:25.17 | mocker | [TK]D-Fender: Ohh, that's pretty hot. |
| 21:25.31 | [TK]D-Fender | mocker, in the end, its really jsut a name. |
| 21:25.42 | mocker | Except that everyone here uses softphones so I'd be a pain to find them all. |
| 21:25.50 | mocker | Yeah. |
| 21:26.05 | [TK]D-Fender | mocker, at which point we have plans A & B |
| 21:26.23 | jduggan | anyone able to andwer my question regardin hangup problelsm? |
| 21:27.06 | [TK]D-Fender | jduggan, this is likely a zaptel zone issue where the telco is expecting a "wink" or "flash" |
| 21:27.17 | [TK]D-Fender | jduggan, I'd search the WIKI on this. |
| 21:27.33 | TJNII | Which feature records calls? Is it monitor? |
| 21:28.15 | [TK]D-Fender | TJNII, thats one of them |
| 21:31.39 | jackson__ | bah, it was a dns resolv issue... |
| 21:33.02 | *** join/#asterisk adr3nalin3 (n=afink@66.172.245.81) |
| 21:34.15 | defsdoor | jduggan: what card ? |
| 21:34.24 | jduggan | defsdoor: digium 410p |
| 21:34.35 | defsdoor | jduggan: how many lines ? |
| 21:34.57 | jduggan | defsdoor: 4 |
| 21:35.06 | defsdoor | jduggan: BT Multiline ? (all on same number) |
| 21:35.13 | jduggan | only 1 is plugged in during testing |
| 21:35.29 | adr3nalin3 | Has anyone seen a problem where Asterisk GUI doesn't write to config files properly? I am not getting any warning but I am seeing what looks like a parsing error on my zaptel.conf |
| 21:35.35 | jduggan | defsdoor: well, actually we're on a science park that have their own PBX, so its a multiline from them |
| 21:35.38 | defsdoor | ask BT to ensure Disconnect Supervision is on - they sometimes call it clear-disconnect |
| 21:35.48 | defsdoor | oh - same thing though |
| 21:35.51 | jduggan | defsdoor: right now im testing on a BT line though |
| 21:35.56 | defsdoor | at home ? |
| 21:35.59 | jduggan | can i just call 151? |
| 21:36.01 | jduggan | nah, @ work |
| 21:36.03 | jduggan | on a late shift |
| 21:36.09 | jduggan | out of hours maintenance etc |
| 21:36.15 | defsdoor | jduggan: a single analog wont have disconnect supervision on |
| 21:36.18 | defsdoor | (by default) |
| 21:36.27 | defsdoor | multiline usually does |
| 21:36.27 | jduggan | ok thats the issue then, i guess |
| 21:36.29 | adr3nalin3 | I am getting this for lines 1-12 of zaptel.conf in asterisk/messages Unknown directive '' at line 1 of /etc/asterisk/../zaptel.conf |
| 21:36.35 | jduggan | defsdoor: are you from UK ? |
| 21:36.36 | defsdoor | I had major probs |
| 21:36.39 | defsdoor | yes |
| 21:36.43 | jduggan | ok |
| 21:37.00 | jduggan | well if i put it live on the 4 lines we have coming in it should be ok, technically |
| 21:37.10 | defsdoor | I had one line out of 6 not hanging up - just happened to be the primary |
| 21:37.12 | jduggan | since our previous (avaya) pbx didnt suffer this issue |
| 21:37.28 | defsdoor | you tried it on the 4 line ? |
| 21:37.32 | jduggan | not yet |
| 21:37.44 | jduggan | i have to wait untill the morning for the science park to re-route the numbers to this system |
| 21:38.05 | defsdoor | I used a sangoma card and set it up at home - single analog - didnt hang up |
| 21:38.21 | defsdoor | ended up trusting that my home line wasn't setup correctly by BT for this |
| 21:38.34 | defsdoor | and was true apart from the one duff line - which was eventually fixed |
| 21:38.41 | jduggan | basically we had a pbx outage so had them move our number mapping to their own internal extensions and they gave us some nortel handsets, so we can only test on a BT line which was used for a redcare system, i believe |
| 21:38.46 | defsdoor | (coincided with CLI being added incidientally) |
| 21:38.51 | jduggan | ah |
| 21:39.21 | defsdoor | I used to log in routinely and clear the line :) |
| 21:39.38 | defsdoor | whereabouts are you btw ? |
| 21:39.48 | jduggan | ok, so this disconnect supervsion is two way, right? |
| 21:39.53 | jduggan | im in northants |
| 21:40.16 | defsdoor | jduggan: yes - it seemed to affect incoming calls hang ing up not being detected and out going |
| 21:40.25 | jduggan | ok great |
| 21:40.31 | defsdoor | I'm just up the road in Warwickshire |
| 21:40.34 | jduggan | you've filled me with confidence |
| 21:40.42 | jduggan | oh great, i travel through it on teh bypass |
| 21:40.46 | jduggan | to get home to wales |
| 21:40.51 | jduggan | whenever i get the chance to go home |
| 21:40.52 | jduggan | ;P |
| 21:40.53 | defsdoor | nice commute :o |
| 21:40.59 | jduggan | uhg, its so not |
| 21:41.02 | jduggan | :) |
| 21:41.07 | defsdoor | I'm at j3 M6 |
| 21:41.13 | jduggan | ah |
| 21:41.22 | jduggan | j15 M1 here :D |
| 21:41.49 | jduggan | oh well, 3days to get a working asterisk system, including waiting for the digium card to arrive |
| 21:41.56 | defsdoor | where from ? |
| 21:41.59 | jduggan | ..first time for me |
| 21:42.05 | defsdoor | I get all my stuff from voipon.co.uk |
| 21:42.09 | jduggan | same |
| 21:42.16 | jduggan | we bought grandstream handsets from them also |
| 21:42.19 | defsdoor | got reseller discount ? |
| 21:42.22 | jduggan | cheap things |
| 21:42.25 | jduggan | but work great |
| 21:42.32 | defsdoor | I've used aastra exclusively so far |
| 21:42.40 | defsdoor | and some samsung dects |
| 21:42.47 | jduggan | ah |
| 21:42.56 | jduggan | ive never touched telephony stuff before |
| 21:43.00 | jduggan | kinda got thrown into it |
| 21:43.04 | defsdoor | nor had I till the first time :) |
| 21:43.11 | jduggan | hehe |
| 21:44.13 | jduggan | no reseller discount btw.. never bought from them before |
| 21:45.07 | jduggan | right well, gotta shoot, planned outtage this evening for one customer - have to move 2 racks worth of kit into a nother suite :o.. thanks for your advice defsdoor |
| 21:45.23 | defsdoor | give me a shout if you get stuck |
| 21:45.26 | defsdoor | andy@defsdoor.org |
| 21:45.43 | jduggan | i'll make a note of it, thanks |
| 21:50.00 | *** join/#asterisk LinuxMafia (n=awatt@CPE001346a4c4cb-CM00159a642d7e.cpe.net.cable.rogers.com) |
| 21:50.08 | LinuxMafia | hummmmm |
| 21:50.14 | LinuxMafia | i can not figure it out |
| 21:51.48 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 21:52.35 | *** join/#asterisk tobias (n=tobias@user-0c998nt.cable.mindspring.com) |
| 21:53.14 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:53.51 | LinuxMafia | hey guys |
| 21:55.03 | LinuxMafia | i have DI-604 and LinkSys SPA2102 phone adaptor , i can not make them work , any idea? |
| 21:55.22 | mwalling | duct tape. |
| 21:55.27 | adeel | LinuxMafia, what exactly is your problem? |
| 21:55.29 | mwalling | preferably red or green. |
| 21:55.35 | rob0 | tin cans and string |
| 21:55.50 | Maliuta | mwalling: really? I find the black works best |
| 21:55.53 | LinuxMafia | adeel, i dont know how to set it up |
| 21:56.07 | rob0 | yikes. |
| 21:56.14 | adeel | LinuxMafia, i'm assuming the DI-604 is a router right? |
| 21:56.15 | LinuxMafia | adeel, i connect the adaptor as client into router |
| 21:56.22 | LinuxMafia | adeel, right |
| 21:56.30 | LinuxMafia | i spent 100$ |
| 21:56.33 | mwalling | Maliuta: http://www.redgreen.com/ |
| 21:56.35 | LinuxMafia | for these |
| 21:57.03 | adeel | LinuxMafia, pretty much you just need to forward ports UDP/5060 & UDP/10000-20000 to the ATA |
| 21:57.17 | adeel | LinuxMafia, but then you need the ATA to register with a Voice Service Provider/ ITSP |
| 21:58.13 | LinuxMafia | adeel, oh but phone light is not on on the adaptor |
| 21:58.32 | adeel | LinuxMafia, the phone light will only come on once the ATA has registered with the VSP/ITSP |
| 21:58.51 | adeel | LinuxMafia, you still need someone to terminate your calls to the PSTN (typically) |
| 21:59.19 | LinuxMafia | adeel, oh how do i find out what ip address ata has? |
| 21:59.39 | adeel | LinuxMafia, there should be a status page on the DI-604 that should give you the ip |
| 21:59.56 | LinuxMafia | right |
| 22:01.03 | mwalling | Maliuta: duct tape is a handymans best friend |
| 22:01.15 | [TK]D-Fender | mwalling, that and WD-40 |
| 22:01.21 | *** join/#asterisk Siya (n=djerk@194.60.207.239) |
| 22:01.24 | mwalling | 17:56 < mwalling> Maliuta: http://www.redgreen.com/ |
| 22:01.31 | mwalling | [TK]D-Fender: duct tape. |
| 22:02.05 | [TK]D-Fender | mwalling, Nope, you need BOTH. If it moves, and shouldn't : duct tape. If it doesn't, and should : WD-40 |
| 22:02.12 | Maliuta | duct tape can remove skin, I prefer to play with a low tak cellotape |
| 22:02.28 | mwalling | [TK]D-Fender: Red never, ever, used wd40. ever. |
| 22:02.50 | [TK]D-Fender | mwalling, never said it applied to HIM |
| 22:02.57 | Maliuta | [TK]D-Fender: wd40 is not so good inside servers though |
| 22:03.24 | mwalling | [TK]D-Fender: you were out of context :P |
| 22:03.46 | [TK]D-Fender | mwalling> Maliuta: duct tape is a handymans best friend <- sub context, entirely valid :) |
| 22:04.09 | mwalling | 17:55 < mwalling> preferably red or green. |
| 22:04.15 | mwalling | context hint. |
| 22:04.40 | [TK]D-Fender | mwalling, yes I fully got your joke, but my point still flows. |
| 22:04.40 | LinuxMafia | adeel, DHCP lease IP 192.168.0.101 to SipuraSPA |
| 22:04.45 | mwalling | NEVER! |
| 22:04.46 | LinuxMafia | is that the one? |
| 22:04.58 | Maliuta | takes mwalling and puts him in the [silly] context |
| 22:05.26 | mwalling | :) |
| 22:08.03 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET) |
| 22:08.49 | [TK]D-Fender | ok, steping out for a bit |
| 22:10.33 | *** join/#asterisk RoyK (n=roy@ip-99-58-149-91.dialup.ice.no) |
| 22:10.37 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 22:16.16 | *** join/#asterisk RoyK (n=roy@ip-117-23-149-91.dialup.ice.no) |
| 22:16.42 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 22:23.09 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:24.32 | whymarkwhy | what is chanspy |
| 22:25.01 | ManxPower | whymarkwhy: if we told you, we'd have to kill you. |
| 22:25.16 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:25.17 | whymarkwhy | no no don't kill me |
| 22:25.26 | whymarkwhy | just please tell me |
| 22:25.45 | *** part/#asterisk RoyK (n=roy@ip-117-23-149-91.dialup.ice.no) |
| 22:27.01 | whymarkwhy | got it |
| 22:27.36 | adeel | LinuxMafia, yes |
| 22:28.34 | LinuxMafia | adeel, but they dont support me |
| 22:28.34 | adeel | LinuxMafia, who doesn't? |
| 22:28.34 | LinuxMafia | if i can not get to adaptor page |
| 22:28.34 | LinuxMafia | les.net |
| 22:28.56 | LinuxMafia | they are asking me to get to webpage for that linksys (which has ip of 192.168.0.1) |
| 22:29.05 | LinuxMafia | but my router also has same address |
| 22:29.15 | LinuxMafia | even if i dont connect my router |
| 22:29.28 | adeel | LinuxMafia, your SPA has an ip address of 192.168.0.101 when connected behind the router |
| 22:29.36 | adeel | so just point your web browser to that address |
| 22:29.43 | LinuxMafia | let me check it out |
| 22:30.08 | adeel | anyone have any experience installing * behind a MS Small Business Server? |
| 22:31.57 | LinuxMafia | adeel, Firefox can't establish a connection to the server at 192.168.0.101. |
| 22:32.51 | LinuxMafia | adeel, 102,103,same thing |
| 22:33.16 | adeel | LinuxMafia, try 100? |
| 22:33.31 | LinuxMafia | 100 is my computer |
| 22:33.50 | ManxPower | LinuxMafia: Of course it can't. Asterisk doesn't come with a web server |
| 22:34.45 | LinuxMafia | ManxPower, it is about my syslink adapter |
| 22:35.08 | adeel | LinuxMafia, try plugging your machine directly into the linksys adapter |
| 22:35.14 | adeel | LinuxMafia, or try reading the book on the SPA |
| 22:35.14 | ManxPower | Ah. The vendor was no help? |
| 22:35.23 | ManxPower | This isn't really an Asterisk issue. |
| 22:36.18 | LinuxMafia | adeel, i did that too , then 192.168.0.1 shoot me to router |
| 22:36.32 | LinuxMafia | ManxPower, it is the begining of * |
| 22:37.26 | adeel | LinuxMafia, then you need to connect your desktop/laptop DIRECTLY into the SPA without the router or anything |
| 22:37.38 | adeel | a lot of these devices restrict which port you can access it from |
| 22:37.49 | LinuxMafia | adeel, funny thing is even if my router is not connected , 192.168.0.1 takes me to router |
| 22:38.04 | adeel | it's called caching |
| 22:38.08 | adeel | firefox is notorious for it |
| 22:38.10 | mwalling | clear your cache |
| 22:38.16 | LinuxMafia | oh got it |
| 22:38.18 | LinuxMafia | so brb |
| 22:38.20 | LinuxMafia | guys |
| 22:41.01 | *** join/#asterisk alancio (n=Alancio@190.75.3.207) |
| 22:41.26 | alancio | hi people, have a question |
| 22:41.52 | alancio | I am trying to place a call across the internet, using SIP, and the other party answers but he can't hear me, although I can |
| 22:41.57 | alancio | what can be wrong? |
| 22:42.01 | adeel | NAT? |
| 22:42.28 | adeel | alancio, you're probably behind a NAT/Firewall and you need to open ports |
| 22:42.34 | adeel | ~NAT |
| 22:42.35 | jbot | i guess nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 22:43.00 | adeel | ~firewall |
| 22:43.01 | jbot | somebody said firewall was This is a form of Internet security that stands between a private network and the Internet. It is like a wall in that it can prevent unwanted traffic from passing either way. Some firewalls have proxy functions built in. In fact, the distinction between a firewall and a proxy is often blurry. Add in the differences and similarities between a firewall and packet-filtering router and you've got one big ball of confusion. True ... |
| 22:43.05 | alancio | ok, I am not behind a NAT, but my friend is (although his asterisk box is the one doing nat) |
| 22:43.09 | *** join/#asterisk CaRb0n^ (n=playa@203.81.221.240) |
| 22:43.38 | adeel | ~1-way audio |
| 22:44.06 | adeel | alancio, he needs to setup his firewall to allow RTP packets in from ports 10000-20000 udp |
| 22:44.37 | alancio | but is this necesary even though I am calling to his asterisk, to its not natted interface |
| 22:44.46 | adeel | yep |
| 22:45.03 | alancio | what about if I use canreinvite=no, or something like that? so that asterisk is always in the middle |
| 22:45.15 | adeel | the reason why he can't hear you is because none of the RTP packets are getting to asterisk |
| 22:45.44 | adeel | alancio, look up 1-way audio/ nat & asterisk on www.voip-info.org |
| 22:45.50 | alancio | ok thanks |
| 22:46.22 | alancio | is there any way to solve it using a netfilter module for connection tracking of sip? |
| 22:46.35 | adeel | alancio, yes...it's possible |
| 22:46.54 | adeel | alancio, but you need to realize that SIP and RTP are related, but different things |
| 22:47.13 | adeel | SIP is used for control/signalling while RTP is the actual media...they operate on 2 different ports |
| 22:47.27 | alancio | mmm ok, I see, I think sip is actually working well, but RTP is not |
| 22:47.29 | rob0 | ip_conntrack_sip IIUC won't substitute for having the SIP UDP port open to the host that wants to initiate the call to you. |
| 22:48.12 | rob0 | (if you have registered to that host, I might be wrong) |
| 22:48.36 | alancio | we are both registering to the asterisk box |
| 22:48.43 | alancio | and the asterisk box does the nat |
| 22:48.56 | alancio | the thing is that one phone is inside and the other outside the nat |
| 22:49.25 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 22:50.14 | alancio | the asterisk box does sip, but it doesn't firewall anything, all gets into its public interface |
| 22:50.27 | alancio | I meant, the asterisk box does nat, but it doesn't firewall anything |
| 22:50.48 | alancio | I'm reading voip-info.org |
| 22:50.50 | rob0 | I'm not talking about -t nat, I'm talking about -t filter |
| 22:50.53 | Nugget | in my experience, asterisk doesn't handle multi-homed network topolgies (as you describe) very well. |
| 22:51.05 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 22:51.39 | *** join/#asterisk RoyK (n=roy@ip-117-23-149-91.dialup.ice.no) |
| 22:52.04 | ManxPower | Nugget: Actually it does, but everything gets so complex, unless you REALLY understand IP, SIP, RTP, DSP, NAT, packet filtering, iptables you'll have a lot of trouble getting it to work |
| 22:52.08 | *** part/#asterisk RoyK (n=roy@ip-117-23-149-91.dialup.ice.no) |
| 22:52.14 | ManxPower | SDP, not DSP |
| 22:53.23 | Nugget | I have nothing but respect for your asterisk skills, but I don't believe you are correct here. I found asterisk itself to be fundamentall lacking if you needed it to serve on multiple IPs at the same time |
| 22:53.38 | adeel | Nugget, i haven't had that problem |
| 22:53.42 | Nugget | I gave up on it and just put asterisk completely behind NAT. |
| 22:54.08 | ManxPower | the two biggest issues I've seen is the Asterisk box NATing packets from the internal network to the external IP of the box |
| 22:54.08 | Nugget | perhaps it has improved since 1.2 |
| 22:54.33 | ManxPower | the second is users try to use bindaddr to override the normal socket binding |
| 22:54.39 | *** join/#asterisk tobias (n=tobias@user-0c998nt.cable.mindspring.com) |
| 22:54.41 | adeel | Nugget, i've found (in my experience) that the majoirty of * connectivity problems are related to the admin's misunderstanding/mis-implementation of routing/ip related stuff...and not actually * |
| 22:55.07 | Nugget | well, I can't really argue with that since you'll just tell me that I'm clearly stupid and can't handle routing and ip related stuff. |
| 22:55.14 | Nugget | but I strenuously disagree with that. :) |
| 22:55.20 | alancio | ManxPower: what do you suggest in bindaddr for multiple interfaces? |
| 22:55.28 | ManxPower | alancio: don't use it. |
| 22:55.34 | ManxPower | you almost never need bindaddr |
| 22:55.36 | adeel | Nugget, no, it's not that people are stupid...it's just that we all make assumptions |
| 22:55.45 | alancio | I have it set to 0.0.0.0, I think it binds to all interfaces |
| 22:55.48 | ManxPower | Nugget: NAT and IP hare HARD |
| 22:55.50 | drmessano | What about Asterisk in a DMZ |
| 22:55.58 | drmessano | Where it's NAT'ed, but not, no wait |
| 22:56.04 | ManxPower | alancio: bindaddr has been buggy in the past. just remove it |
| 22:56.06 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:56.12 | drmessano | Thats a surefire clownkill of a scenario |
| 22:56.18 | `Sauron | "bindaddr has been buggy" ? |
| 22:56.24 | `Sauron | It's not rocket science. |
| 22:56.48 | Nugget | in my old topology, NAT wasn't a factor. The asterisk box sat on both the public and the private networks and no phones had NAT in between them and the asterisk box. |
| 22:56.56 | Nugget | and asterisk was flaky |
| 22:57.18 | alancio | if I remove bindaddr, what will be used? |
| 22:57.22 | adeel | Nugget, what IP did the phones register to? public or external? |
| 22:57.23 | alancio | it has to bind to something |
| 22:57.24 | ManxPower | People are lazy and set their nat rules to be "nat anything not destined for the local internal network", not realizing that they should not nat packets going to the OTHER local network, the external IP of the box |
| 22:57.46 | Nugget | adeel: the inside phones to the inside address, outside phones to the outside address (obviously) |
| 22:57.53 | ManxPower | once you fix that, register your phones to the external IP, allow the SIP and RTP packets and you're done |
| 22:58.06 | ManxPower | Nugget: no, not obviously |
| 22:58.15 | adeel | Nugget, why not just have ALL phones just register to the external ip? |
| 22:58.22 | ManxPower | you would still need canreinvite=no, of course. |
| 22:58.46 | `Sauron | adeel: Give me a good reason WHY one would have internal phones talking to the external interface? |
| 22:59.01 | `Sauron | Other than working around some bug in *. |
| 22:59.04 | ManxPower | `Sauron: to make them work? 8-) |
| 22:59.06 | adeel | `Sauron, reduce the complexity of your IPtables...make it easier to debug....whatever |
| 22:59.22 | rob0 | I think what he was talking about was an Asterisk connecting to an Asterisk with no NAT in the middle, not really a "multi-homed" thing as far as * is concerned, is it? |
| 22:59.25 | Nugget | adeel: to avoid nat traversal and the need for my nat device to do hairpin routing. |
| 22:59.26 | ManxPower | Your phones could also roam between internal and external networks with no config change |
| 22:59.38 | ManxPower | (you can do that using SRV records for endpoints that support it) |
| 22:59.51 | `Sauron | Umm. There's no iptable rules involved in having internal phones register to an internal address, and external phones register to an external attress. |
| 22:59.54 | adeel | `Sauron, the first thing is to make sure things work, then you can work on optimization and funky setups |
| 23:00.00 | `Sauron | Turn off IP forwarding, and it gets even simpler. |
| 23:00.07 | adeel | `Sauron, [15:56] <ManxPower> People are lazy and set their nat rules to be "nat anything not destined for the local internal network", not realizing that they should not nat packets going to the OTHER local network, the external IP of the box |
| 23:00.17 | ManxPower | `Sauron: Asterisk is pretty bad about picking the correct interface for RTP. |
| 23:00.17 | adeel | `Sauron, iptables is always involved with NAT |
| 23:00.47 | `Sauron | adeel: Huh. |
| 23:00.51 | ManxPower | I'm sure it's gotten better in recent releases, but for a long time bindaddr only bound SIP, not RTP. |
| 23:00.57 | `Sauron | Last I checked, it was PF under *bsd. |
| 23:01.20 | `Sauron | ManxPower: so all this work to fix a lousy bug in * that people are too lazy to fix? |
| 23:01.23 | adeel | `Sauron, PF for *bsd, iptables/netfilter *nix |
| 23:01.25 | *** join/#asterisk djs (n=djs@unaffiliated/djs26) |
| 23:01.34 | `Sauron | adeel: I am quite familiar with what it takes to do NAT. |
| 23:01.44 | `Sauron | However, if you paid attention, you would've realised 2 things. |
| 23:01.49 | `Sauron | 1) What if there is no NAT |
| 23:01.55 | ManxPower | `Sauron: you can fight Asterisk's bugs and live a miserable and pointless life, or you can accept Asterisk's oddities and live a happy life. |
| 23:01.57 | `Sauron | 2) What if there is no IP forwarding |
| 23:02.05 | adeel | `Sauron, if there's no nat, what's the need for 2 interfaces? |
| 23:02.18 | `Sauron | Oh boy. |
| 23:02.32 | ManxPower | `Sauron: Obviously you can do this in many ways -- hence everyone being confused about it. |
| 23:02.33 | adeel | if they're both on the same network, why not just bond the interfaces? |
| 23:02.44 | `Sauron | adeel: A bit narrow-minded are we? |
| 23:03.02 | adeel | no, just that everyone wants a 'simple' solution to a complicated problem |
| 23:03.02 | drmessano | Bondage? I'll come back... |
| 23:03.06 | `Sauron | Imagine the * box NOT being part of your general traffic routing. |
| 23:03.32 | ManxPower | `Sauron: now you are getting out of the realistic setup for a home user. |
| 23:03.42 | *** join/#asterisk Nasra (n=Nasra@CPE001839494bc9-CM00111ade9528.cpe.net.cable.rogers.com) |
| 23:04.02 | `Sauron | ManxPower: And yet, a perfectly valid scenario for the non-home user, no? |
| 23:04.19 | ManxPower | `Sauron: Sure! But I'm not talking about non-home users. |
| 23:04.21 | `Sauron | I am merely creating you a (perfectly valid) scenario in which * will misbehave. |
| 23:04.29 | ManxPower | If you are a corporation get off your cheap ass and buy a router |
| 23:04.39 | adeel | hehe |
| 23:04.41 | `Sauron | You don't get it either. |
| 23:04.50 | `Sauron | But that's okay. I'm done. |
| 23:05.18 | drmessano | Why should I buy a cisco router when I can just add a NIC to my asterisk box and make it suck at two things equally? |
| 23:05.26 | *** join/#asterisk jeffspeff2 (n=jeff@c-68-53-81-73.hsd1.ky.comcast.net) |
| 23:05.36 | ManxPower | drmessano: it's not SO bad for a simple home setup |
| 23:05.37 | adeel | `Sauron, i'd actually like to know what your point/scenario is |
| 23:06.06 | rob0 | I've been running Linux based routers for a long time, don't see anything wrong with it. |
| 23:06.23 | `Sauron | adeel: Have the * box BOTH service SIP/whatever from the outside, while also servicing SIP/whatever from the inside. |
| 23:06.27 | Nugget | there are legitimate reasons to want to have an asterisk box which is multi-homed. bob knows I tried to run that way for a year. |
| 23:06.32 | Nugget | but it doesn't really work right |
| 23:06.32 | `Sauron | WITHOUT the * box being your NAT gateway |
| 23:06.56 | Nugget | exactly |
| 23:06.56 | adeel | `Sauron, you can achieve that with some firewall rules on your router |
| 23:07.17 | adeel | `Sauron, i've done that without the need for 2 nics.... |
| 23:07.22 | adeel | hell, i do that right now |
| 23:07.49 | Nugget | I found that asterisk sometimes got confused and emitted the wrong address in SIP traffic to the "other" interfacce |
| 23:08.03 | adeel | to me, personally, i only see the need for 2 NICS if i want my * box to be a router/gateway |
| 23:08.22 | Nugget | that's fine. be aware that other people aren't so limited in their vision. |
| 23:08.48 | *** join/#asterisk nDuff (n=cduffy@rrcs-71-41-149-67.sw.biz.rr.com) |
| 23:09.41 | adeel | Nugget, can you enlighten me on to a 'broader' vision? |
| 23:09.53 | Nugget | I've described it twice now in the past 10 minutes. |
| 23:10.04 | adeel | 3rd time's the charm =cp |
| 23:10.05 | Nugget | a multi-homed asterisk box where there are SIP devices on both networks. |
| 23:10.47 | Nugget | where either I do not want routing between the two networks, or I want to avoid the NAT that exists between them |
| 23:10.47 | alancio | Nugget: I just made it work |
| 23:11.07 | alancio | I have another question |
| 23:11.11 | Nugget | I ran that way for about a year and faced a pretty stready stream of small problems and erratic behavior |
| 23:11.31 | alancio | what happens if a user registers from two different phones |
| 23:11.32 | Nugget | (all on asterisk 1.2.mumble or earlier, never on 1.4+) |
| 23:11.35 | alancio | and he receives a call |
| 23:11.51 | nDuff | alancio, whichever one registered most recently gets it. |
| 23:12.09 | adeel | unless you setup the new shared line stuff |
| 23:12.45 | alancio | and if the one that registered most recently deregisters, while the other one was always on, the call will be correctly routed to the first one |
| 23:13.19 | adeel | i don't think that'll happen |
| 23:13.20 | nDuff | alancio, the traditional answer if you want connect SIP behavior in interesting circumstances is to put a dedicated SIP proxy in front of asterisk. |
| 23:13.31 | nDuff | s/connect/correct/ |
| 23:14.03 | alancio | oh that would be overkill for me, I think I'll just have fun with these circumstances |
| 23:14.21 | nDuff | alancio, ...and no, there's not a queue like that -- deregistering one won't make any difference until the other reregisters. |
| 23:14.39 | adeel | alancio, one way to do what you want is to give the user multiple extensions and then make a ring group for it |
| 23:14.45 | alancio | oh ok, good to know that |
| 23:14.56 | alancio | thanks |
| 23:15.06 | adeel | and direct all calls to the ring group, so which ever device is registered will get the call...you can setup how the indvidual extensions will be rung |
| 23:16.13 | alancio | ok, in my case each person has a phone in his work place, but maybe he leaves the office and takes the laptop |
| 23:16.24 | alancio | and he runs a software sip phone on the laptop |
| 23:16.40 | Nugget | the easiest solution is to have the softphones use independent SIP credentials |
| 23:17.06 | alancio | ok, and then I would use a single extension that calls both of them at the same time? |
| 23:17.12 | adeel | alancio, yeah, so each device has it's own extension/pass, and then you have a ring group defined for them |
| 23:17.16 | Nugget | Dial(SIP/user&SIP/usersoftphone) |
| 23:17.36 | alancio | what is a ring group? |
| 23:17.46 | adeel | alancio, what Nugget just pasted |
| 23:18.00 | alancio | oh ok :) |
| 23:18.04 | rob0 | Oh, FWIW my very small * box is multihomed. SIP ATA's on the internal interface (canreinvite=no) and SIP inbound/outbound peers on the external one. |
| 23:18.58 | drmessano | Oh, AIFWIWIYC, my * box is multihorned. |
| 23:19.02 | adeel | alancio, you just define an extension to it....e.g if bob has extension 888 and his sip phone is 889, you can create a 3rd one 890 and just publish 890 to everyone, and 890 will call 888/889 until bob picks up |
| 23:19.39 | Nugget | or just put both SIP clients on the same extension |
| 23:19.57 | Nugget | there is absolutely no direct correlation between extensions and SIP clients. |
| 23:21.18 | rob0 | hey, if you do that (2 SIP clients same exten), if one picks up, what happens when the other one does? Dial tone? |
| 23:21.29 | adeel | yes |
| 23:21.31 | rob0 | (assuming the call is active) |
| 23:21.38 | Nugget | whichever grabs the channel first gets it |
| 23:22.20 | alancio | ok, I'll use that one, calling 2 sip clients on one extension, its easier to remember |
| 23:30.22 | *** join/#asterisk s0lid (n=s0lid@210.213.199.2) |
| 23:30.30 | `Sauron | adeel: do you really need us to give you the explanation again? :p |
| 23:30.54 | adeel | ? |
| 23:31.06 | `Sauron | oh, good, you forgot.. nvm. |
| 23:31.07 | `Sauron | :) |
| 23:31.16 | *** part/#asterisk clive- (n=pirch@dsl-242-156-73.telkomadsl.co.za) |
| 23:31.40 | adeel | i haven't forgotten, i just can't think of a real life situation where i'd do a deployment similar to what you and nugget had described |
| 23:32.08 | `Sauron | So, we all agree that * and NAT pretty much sucks. |
| 23:32.29 | `Sauron | So you have your NAT/fw gateway that deals with all your traffic to/from the internet |
| 23:32.33 | adeel | i'd go so far to say any application and NAT sucks |
| 23:32.46 | jbeez | some work perfectly fine with it |
| 23:32.52 | `Sauron | On the outside of said gateway is a router between "you" and your internet circuit (T1/whatever) |
| 23:32.55 | `Sauron | So |
| 23:33.07 | `Sauron | in paralell to your nat/fw machine, you put a *nix box running asterisk |
| 23:33.19 | `Sauron | with one interface on the "outside" with a public IP |
| 23:33.38 | `Sauron | and one interface on the "inside" with a RFC1819 address (or whatever internal numbering scheme you have) |
| 23:33.58 | adeel | `Sauron, what in gods name is the benefit of that setup? your just adding complexity for the hell of it |
| 23:34.03 | `Sauron | Now, in effect, asterisk becomes a sip/whatever proxy |
| 23:34.15 | `Sauron | adeel: No you are not. You're removing having to deal with SIP and NAT. |
| 23:34.30 | `Sauron | So in effect, you are REDUCING complexity. |
| 23:34.48 | `Sauron | You're also removing having to deal with * and NAT |
| 23:35.01 | adeel | `Sauron, not really...it only takes 3 lines of iptables rules to get * to have that same functionality |
| 23:35.09 | adeel | behind a NAT |
| 23:35.31 | `Sauron | And a line of proxy-arp |
| 23:35.38 | adeel | and i no longer have to worry about any dual homed problems that maybe lurking inside of * |
| 23:36.14 | adeel | i don't need to add proxy-arp unles i have a NEED to give * it's own public ip |
| 23:36.24 | jblack | Oh man. That ssh key vulnerability is a disaster |
| 23:36.33 | `Sauron | adeel: You're not listening. |
| 23:36.46 | `Sauron | 18:33 <`Sauron> with one interface on the "outside" with a public IP |
| 23:36.49 | adeel | if no other services on my network is using 5060 udp then why should i bother with anyone else? |
| 23:36.49 | `Sauron | So yes |
| 23:36.53 | `Sauron | * needs a public IP |
| 23:36.57 | jbeez | jblack: disaster for who? :D |
| 23:37.21 | jblack | jbeez: Anyone running a debian derived system that uses ssh keys |
| 23:37.58 | adeel | `Sauron, yeah, i can come up with complicated * setups where * will not work properly...but why bother? if you can do it a simpler way, why wouldn't you? there's no justifiable reason (to me anyway) where you'd go through the hassle of this exercise, especially if you KNOW * doesn't handle the situation well....next thing you'll suggest is running SIP & RTP purely over TCP |
| 23:38.51 | jbeez | thats not me :D |
| 23:39.04 | `Sauron | adeel: The problem is the following: |
| 23:39.05 | adeel | jblack, what ssh key vulnerability? |
| 23:39.11 | `Sauron | 1) It's not a complicated setup |
| 23:39.16 | `Sauron | 2) asterisk doesn't handle it |
| 23:39.20 | `Sauron | ergo, asterisk is broken |
| 23:39.32 | `Sauron | Which is what Nugget was trying to point out earlier |
| 23:40.12 | `Sauron | And having written numerous network-centric applications in the past, it is NOT rocket science to map your outbound connections (udp or otherwise) to the same address that the incoming packets arrived at. |
| 23:40.12 | adeel | `Sauron, and what i had tried to point earlier is that i've had a similar setup working before (hadn't tried before 1.4) and then decided against it |
| 23:40.27 | `Sauron | The * developers are lazy, and can't be arsed to fix it. |
| 23:40.29 | jblack | adeel: The one where debian broke the random number generator for key generator, resulting in a many, many easily reproduceable keys. http://article.gmane.org/gmane.linux.debian.security.announce/1614 |
| 23:40.39 | *** mode/#asterisk [+b %`Sauron!*@*] by russellb |
| 23:40.41 | jblack | Any key generated after 2006 needs to be checked. |
| 23:40.57 | russellb | I don't appreciate you calling Asterisk developers lazy .. we bust our asses fixing things every day |
| 23:40.59 | adeel | `Sauron, so why don't you help everyone out and fix it yourself? |
| 23:41.18 | adeel | jblack, ouch, that sucks....good thing i don't use ubuntu |
| 23:41.25 | adeel | or debian for that matter |
| 23:41.44 | jblack | Yeah. I have something like 30 systems to check |
| 23:42.25 | adeel | jblack, ouch...too bad you can't just force a global upgrade/key-regen on them all...but then key maintenance would be a problem |
| 23:42.29 | jblack | Thankfully, _my_ key is older than 2006, so that's not a couple hundred easy to root system out there. |
| 23:43.41 | *** part/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 23:43.41 | rob0 | <flame> Thankfully *my* distributor didn't break good code with a patch! ;) </flame> |
| 23:43.47 | adeel | hahah |
| 23:43.53 | adeel | rob0, what do you run? |
| 23:44.00 | rob0 | sorry, I like Debian, but that was pretty stupid. |
| 23:44.02 | shasta | ain't source routing what Sauron is asking for? |
| 23:44.21 | rob0 | Slackware and slamd64 (64-bit port of Slack) |
| 23:44.26 | adeel | shasta, not really....he wants * to be running on a separate network leg |
| 23:44.49 | shasta | ok, I didn't follow the conversation closely |
| 23:45.11 | jblack | rob0: Yeah. It's a true egg-on-the-face moment. |
| 23:45.12 | lmadsen | russellb: <3 |
| 23:45.18 | rob0 | me neither, but I tried to participate in the flames |
| 23:45.31 | rob0 | made up a new flame when the old ones had died down :) |
| 23:45.49 | jblack | To drag it bck on topic... Make sure you call anyone you know that may be at risk for the vulnerability |
| 23:45.54 | jblack | With *, of course. |
| 23:46.04 | jbeez | lol |
| 23:46.11 | tzafrir_home | rob0, the Debian maintainer asked upstream (openssl-dev mailing list) if the patch is OK, and was answered that it is OK |
| 23:47.12 | rob0 | Okay, I'll officially admit no knowledge of the topic, but I'm sure glad I don't have to scramble to fix mine. |
| 23:47.12 | adeel | it's not like the guy broke it on purpose...he was trying to improve something but messed up...it happens |
| 23:48.16 | *** join/#asterisk impl (n=impl@atheme/member/impl) |
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| 23:48.28 | freakmod | hello |
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