| 00:02.50 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 00:14.40 | TellarHK | This is annoying, we've got the Cisco 7941's registered with chan_sccp, but we're getting a fast busy tone trying to use them at all. |
| 00:18.32 | joobie | that's what happens when u go for overpriced equipment |
| 00:18.41 | TellarHK | Not overpriced when it's free. |
| 00:18.52 | joobie | hook a brotha up ;P |
| 00:19.23 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 00:20.13 | TellarHK | Ha. |
| 00:20.25 | TellarHK | Got 'em banged up from a roomie's workplace, cracked housings, a cracked LCD. |
| 00:27.56 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 00:29.13 | *** join/#asterisk siera08 (n=sosoriri@218.207.141.90) |
| 00:29.32 | BadHAL | well |
| 00:29.35 | BadHAL | this sucks |
| 00:29.39 | BadHAL | I can't figure out what is causing this |
| 00:31.05 | *** join/#asterisk siera08 (n=sosoriri@218.207.141.90) |
| 00:33.40 | *** join/#asterisk siera08 (n=siera08@218.207.141.90) |
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| 00:35.08 | *** part/#asterisk siera08 (n=siera08@218.207.141.90) |
| 00:37.10 | drucko | hi has anyone here experience with the ISDN-Cards from Hermstedt Leonardo X4? |
| 00:44.14 | *** part/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net) |
| 00:55.25 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 01:01.42 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-104-80-249.pskn.east.verizon.net) |
| 01:05.21 | *** join/#asterisk rootforce (n=chatzill@MT2-52.aircanopy.net) |
| 01:05.48 | rootforce | does asterisk 1.6.0.5 support t.38 endpoint |
| 01:09.59 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 01:12.01 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 01:15.18 | *** join/#asterisk bbryant (n=brett@adsl-243-239-183.chs.bellsouth.net) |
| 01:16.58 | *** join/#asterisk TellarHK (n=zillacha@75.141.193.106) |
| 01:24.10 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 01:24.31 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:27.03 | *** join/#asterisk animositisomina (n=chatzill@201-75-35-138-ma.cpe.vivax.com.br) |
| 01:29.27 | rootforce | 228 people in here and no one is talking |
| 01:29.54 | *** join/#asterisk Vec (n=vec@217.33.90.50) |
| 01:30.06 | jaytee | except for you |
| 01:30.29 | hads | and you |
| 01:30.46 | jaytee | y tu |
| 01:30.55 | Vec | Hi, does anyone know where asterisk stores its live queue information, I know it stores the status in the queuelog, but where does it keep track of where everyone is in the queue etc ? |
| 01:31.04 | rootforce | im sorry i missed the memo that we were playing the "quiet game" |
| 01:33.03 | Vec | I hope asterisk doesn't only store its queue information in memory :) ?? |
| 01:36.51 | jaytee | only one of those really astute developer types could answer that question |
| 01:37.09 | jaytee | maybe ask in asterisk-dev |
| 01:37.37 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 01:39.04 | Vec | thanks will give it a try |
| 01:43.29 | rootforce | has anyone in here had experience with the t.38 pass-through in 1.4 |
| 01:43.32 | *** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 01:54.30 | jaytee | I've used it |
| 01:56.46 | TellarHK | kerframil, you around by any chance? |
| 01:56.47 | rootforce | does it work well? |
| 01:58.16 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 01:58.20 | jaytee | it's ok. not great for faxing large multi-page faxes but the occassional 1 or 2 page fax works ok. |
| 01:59.30 | rootforce | what happens to the larger faxes |
| 01:59.39 | rootforce | packetloss or jitter |
| 02:00.06 | jaytee | usually it craps out and the fax is either several pages of black toner or the whole shebang freezes up |
| 02:00.45 | rootforce | hmm |
| 02:00.56 | rootforce | that is disconcerting |
| 02:01.22 | *** join/#asterisk genoobie (n=genoobie@pool-71-241-171-185.buff.east.verizon.net) |
| 02:01.34 | Damin | exit |
| 02:01.35 | Damin | quit |
| 02:01.43 | genoobie | anyone ever unlock a PAP box |
| 02:02.07 | rootforce | I buy em unlocked |
| 02:02.08 | jaytee | you can buy ones that are already unlocked |
| 02:02.16 | genoobie | yeah, i know |
| 02:02.40 | genoobie | but here's my ?, what's the point |
| 02:02.51 | rootforce | to buying them |
| 02:02.58 | rootforce | to use them |
| 02:03.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:03.18 | genoobie | well, right, what's the point seeing as you still need a service providers |
| 02:03.25 | genoobie | *provider |
| 02:03.29 | rootforce | i am a service provider |
| 02:03.53 | jaytee | because the locked PAPs only work with whatever provider had them provisioned that way, usually Vonage |
| 02:04.15 | carrar | hi service provider, I am a HUMAN |
| 02:04.40 | rootforce | human? |
| 02:04.46 | rootforce | all will fall to the MCP |
| 02:04.47 | carrar | HUE MON |
| 02:04.56 | TellarHK | End of Line. |
| 02:05.00 | genoobie | well i just signed up with vonage |
| 02:05.12 | genoobie | and I just got their ata box |
| 02:05.13 | jaytee | Bon Vonage! |
| 02:05.35 | carrar | Isn't everyone in here a service provider? |
| 02:05.44 | jaytee | I provide a service |
| 02:05.47 | genoobie | I was wondering if it was worth it to unlock it |
| 02:05.51 | TellarHK | See, if I wasn't the new guy around here I'd make a 'your mom' joke. |
| 02:05.51 | jaytee | and people pay me for it |
| 02:06.01 | carrar | haha |
| 02:06.08 | jaytee | worth what? money? no way! |
| 02:06.15 | rootforce | it isnt worth it |
| 02:06.39 | rootforce | if you want to use a byod provider just buy an unlocked device |
| 02:06.56 | genoobie | right, that's what I figured |
| 02:06.56 | rootforce | unless you are doing it for the challenge |
| 02:07.00 | genoobie | nah |
| 02:07.11 | jaytee | in fact, I'm willing to let you pay me five bucks for every unlocked PAP-2 that you give me. |
| 02:07.15 | rootforce | broadvoice has a BYOD service |
| 02:07.53 | genoobie | yah, I looked at other services and vonage sort of fit the bill |
| 02:08.08 | nix8n82 | dumb question what does PAP stand for? |
| 02:08.18 | genoobie | given the quantity of calls that I want to make, quality of service, etc. |
| 02:08.27 | TellarHK | I'm actually using Gizmo for my setup, because Broadvoice wanted too much setup. |
| 02:08.39 | TellarHK | Too much in setup fees, that is. |
| 02:09.04 | genoobie | yeah vonage charged me the $15 to ship and that was it |
| 02:09.10 | rootforce | quality of service? |
| 02:09.19 | genoobie | but I had to complain about the fees, etc. |
| 02:09.52 | genoobie | rootforce, to allow faxes to the PSTN and a few other features |
| 02:10.13 | rootforce | ah how is that faxing working for you? |
| 02:10.19 | genoobie | fine |
| 02:10.24 | rootforce | cool |
| 02:10.47 | rootforce | nix8n82: maybe phone adapter port? |
| 02:11.04 | genoobie | Just curious, if it was an exercise in skill or if there was really a reason for doing it |
| 02:11.29 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 02:11.49 | rootforce | genoobie: i can't see a reason. the interface is not that interesting |
| 02:13.15 | rootforce | genoobie: the dialplan might be fun to play with |
| 02:13.38 | genoobie | yeah, but I already get the idea, so otherwise there's not that much interesting config |
| 02:13.49 | rootforce | genoobie: some nat related stuff |
| 02:14.01 | nix8n82 | cool |
| 02:14.07 | rootforce | genoobie: but vonage has probably already set most of that |
| 02:14.08 | genoobie | isn't there also a little bit of a risk of configuring it to a point where I can't use it with vonage |
| 02:14.17 | genoobie | rootforce, right |
| 02:14.59 | rootforce | genoobie: about the only way I can think of to break into it would require that |
| 02:14.59 | *** join/#asterisk n3hxs (n=HAMming@72-58-106-11.pools.spcsdns.net) |
| 02:14.59 | *** part/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 02:14.59 | genoobie | I mean if I'm paying $30/mo for "phone service" then so be it |
| 02:15.00 | genoobie | stupid phone #'s |
| 02:15.09 | genoobie | oh well, there's not much way around it |
| 02:15.26 | rootforce | genoobie: so do you use asterisk? |
| 02:15.45 | genoobie | nah, I was thinking about it, but there's not really much need. |
| 02:15.55 | genoobie | my wife's biz was the only app. I had for it |
| 02:16.02 | rootforce | it is pretty rewarding what you can do |
| 02:16.18 | genoobie | I know, I've seen it, it's pretty amazing |
| 02:16.45 | genoobie | but with just her biz, there's not much need, i.e. setup time outweighs benefits |
| 02:16.45 | rootforce | at the very least you should try a setup to mess with telemarketers |
| 02:16.52 | genoobie | I should |
| 02:17.02 | rootforce | try asterisk@home |
| 02:17.10 | genoobie | but with 3 kids there's not much time to do it :) |
| 02:17.17 | genoobie | actually I gotta run now too. |
| 02:17.18 | rootforce | or some other live cd version |
| 02:17.27 | genoobie | tx for the info! |
| 02:17.27 | rootforce | have a good evening |
| 02:17.33 | rootforce | or morning |
| 02:17.35 | *** part/#asterisk genoobie (n=genoobie@pool-71-241-171-185.buff.east.verizon.net) |
| 02:31.23 | *** join/#asterisk sjobeck (n=sjobeck@206.173.193.34.ptr.us.xo.net) |
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| 02:40.36 | *** join/#asterisk siera08 (n=_hm@218.207.141.90) |
| 02:42.07 | siera08 | hi, i test asterisk realtime with mysql now. |
| 02:42.24 | siera08 | i successed to register sipuser with sip_buddies table. |
| 02:42.32 | siera08 | but i have problem. |
| 02:42.58 | manveru | why is it that everytime i call asterisk with twinkle, asterisk crashes with no backtrace? |
| 02:43.29 | siera08 | how can i sole something like "register=123:1234@192.168.15.25:5060/123" with mysql? |
| 02:43.44 | *** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 02:43.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:44.23 | siera08 | i want to use "register=123:1234@192.168.15.25:5060/123" with mysql. |
| 02:45.05 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 02:47.36 | TrentCreek | who has used route5060.com? |
| 02:48.08 | TellarHK | Not me... Escrow service? |
| 02:48.17 | TrentCreek | yeah |
| 02:48.36 | TrentCreek | trying to figure out how to configure. they have nothing on their site |
| 02:48.51 | TellarHK | Huh. |
| 02:49.11 | TrentCreek | you have to choose your routes by seller |
| 02:49.20 | TellarHK | Since I'm stuck on my asterisk stuff, playing with RouterOS now that my RB433AH arrived. |
| 02:49.23 | TellarHK | Damn, this is nice. |
| 02:49.56 | TellarHK | I think I can get a 600mw radio for it. Three of them, actually. |
| 02:54.15 | TrentCreek | and a linera amp to boost it to 5000Watts |
| 03:04.54 | rootforce | TellarHK: you should try an rb1000 |
| 03:10.41 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 03:10.41 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:11.06 | TellarHK | rootforce: No need for that much power, and no money to get it with. :) |
| 03:11.40 | TellarHK | I wish these 7941's would work though. I got the sccp module installed, the phones are back on SCCP firmware, and the phones register, but they won't actually dial. |
| 03:14.53 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 03:21.50 | *** join/#asterisk vjr (n=nomail@c-76-121-138-221.hsd1.wa.comcast.net) |
| 03:22.09 | vjr | hello? |
| 03:28.14 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 03:29.35 | *** part/#asterisk vjr (n=nomail@c-76-121-138-221.hsd1.wa.comcast.net) |
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| 03:38.48 | Qwell | an IRC client for emacs.. that's just excessive |
| 03:39.05 | TellarHK | emacs is a great operating system, but a lousy text editor, I hear. |
| 03:40.53 | *** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 03:45.17 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
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| 03:51.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:56.52 | *** join/#asterisk keebler (n=keebler@h1.224.20.98.dynamic.ip.windstream.net) |
| 03:58.22 | *** join/#asterisk Gopaul (n=chatzill@59.97.121.53) |
| 04:05.28 | *** join/#asterisk lvlefisto (n=corleone@189.152.16.193) |
| 04:05.54 | lvlefisto | what are the minimum system requirements for the installation of asteriskNOW? |
| 04:06.34 | TellarHK | Probably depends a lot on your system scale. |
| 04:07.40 | keebler | lvlefisto: The smallest system I installed it on was an intel Atom, 512MB ram and a 4GB CF card. |
| 04:08.00 | lvlefisto | keebler: thank you very much |
| 04:08.41 | keebler | lvlefisto: I suggest installing from source with a bare Distro install and use the cli though. :) |
| 04:08.58 | keebler | lvlefisto: Get to know it the proper way. :) |
| 04:09.15 | lvlefisto | thanks |
| 04:09.36 | *** part/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 04:09.49 | lvlefisto | my wish is to install it in a Virtual Box |
| 04:09.57 | TellarHK | I'm running it in a VM with 256M right now. |
| 04:10.01 | TellarHK | Working fine, too. |
| 04:10.08 | lvlefisto | TellarHK: thanks |
| 04:12.34 | TellarHK | Technically I'm running trixbox, but I can't imagine AsteriskNOW having more hardware needs. And I did try that one first. |
| 04:15.16 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.74) |
| 04:19.07 | justdave | how do I change the domain on the To: address in a SIP trunk if it needs to be different than the host= value? |
| 04:21.39 | justdave | hmm, googling for this provider, it looks like people are putting in the IP that goes with the real host into their /etc/hosts file with the domain that's supposed to be in the header, then using the name from the header for host= |
| 04:21.46 | justdave | that seems like a hack :) |
| 04:22.48 | *** join/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 04:22.54 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-250-236.dsl.pltn13.sbcglobal.net) |
| 04:25.37 | PeterFA | My friend had an Asti box that compromised and it was making robo calls asking for debit numbers. |
| 04:25.42 | PeterFA | Anyone heard of this? |
| 04:28.19 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 04:28.40 | *** join/#asterisk Gopaul_ (n=chatzill@59.97.121.49) |
| 04:30.41 | justdave | PeterFA: yeah, that's been going around. |
| 04:30.58 | justdave | PeterFA: usually people have weak SIP passwords or somesuch |
| 04:31.59 | justdave | PeterFA: the bots that try to do that usually try to brute force the passwords on your SIP, and then once they succeed in registering just start placing calls |
| 04:32.14 | TellarHK | How do the bots get access, just infiltrate the LAN? |
| 04:32.43 | justdave | most of the cases I've heard of they don't actually get into your box and run anything on it, they're just relaying off your Asterisk to place calls |
| 04:32.55 | *** join/#asterisk imchandave-BB (n=imchanda@fw-e.isp.sunday.com) |
| 04:33.44 | justdave | they just repeately try to register to your system with random extension numbers and passwords |
| 04:33.58 | TellarHK | Hmm. |
| 04:34.11 | justdave | lots of folks have 2 to 4 digit extension numbers and do stupid things like using the extension number for the password |
| 04:35.59 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 04:39.14 | PeterFA | I use passwords like: |
| 04:39.29 | PeterFA | m@9z8pKDas!ZAgm,z>Qc |
| 04:40.02 | PeterFA | Them is secure. |
| 04:41.05 | justdave | yeah, me too |
| 04:41.32 | justdave | even for the softphone accounts, which pisses off my users ocassionally. :) |
| 04:42.02 | justdave | I'm like "hey, if your using a softphone that's any good, it should remember it for you and you'll only have to type it in once" |
| 04:50.25 | *** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 04:56.19 | *** part/#asterisk lvlefisto (n=corleone@189.152.16.193) |
| 05:00.20 | *** join/#asterisk Talkradio (n=talkradi@208.80.96.84) |
| 05:01.19 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
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| 05:20.17 | drmessano | Anyone heard anything lately on the Skype for Asterisk Beta |
| 05:20.24 | drmessano | or is it vaporware? |
| 05:21.58 | TellarHK | I haven't yet. |
| 05:24.17 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 05:31.01 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-250-236.dsl.pltn13.sbcglobal.net) |
| 05:35.51 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 05:40.40 | coppice | digium have talked about it in a bit of detail - e.g. its inclusion of a G.729 licence - and said the beta is in progress. that was a few weeks ago, though |
| 05:41.05 | Qwell | the beta is still in progress |
| 05:42.18 | coppice | is this a Googlish beta? :-) |
| 05:43.04 | Qwell | coppice: well, let's put it this way. Kevin is using Skype now. |
| 05:43.26 | Qwell | we *want* to release it, obviously. I don't know specifics though |
| 05:45.07 | coppice | sadly shipping is when the real testing starts :-) |
| 05:45.59 | Qwell | indeed |
| 05:46.12 | Qwell | that's when you get people trying all sorts of wacky stuff you'd never think of |
| 05:46.25 | Qwell | but, hell, that's half the fun, right? |
| 05:46.28 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 05:46.34 | Qwell | it's why we write software, isn't it? ;) |
| 05:46.50 | coppice | well, it may mean beta programs are a useless stretching of any timescale |
| 05:47.12 | Qwell | well, knowing that basic functionality actually works is a good thing. |
| 05:47.38 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 05:47.56 | coppice | I guess if you are offering something genuinely new, like this channel, people will be more eager to test. for V2.0 you're sunk :-\ |
| 05:48.42 | Qwell | I think the RCs of Asterisk that we've done have been pretty well tested. so, I don't know if that's entirely true. |
| 05:49.10 | Qwell | it seems like people do like testing certain things. sure, if it's new and different, they'll jump through hoops to do so |
| 05:49.42 | Qwell | maybe we've just been lucky |
| 05:51.02 | coppice | well * V1.2.0 and V1.4.0 were not stellar successes, and the more you get a reputation for such releases the less people are inclined to try even a .0 final release. its a difficult problem |
| 05:51.31 | Qwell | coppice: heh, well, you and I have both used 1.0 |
| 05:52.05 | Qwell | It seems people are really happy with 1.4 now |
| 05:52.17 | coppice | which is why all releases should be based on random numbers |
| 05:52.23 | Qwell | uuid? |
| 05:52.29 | Qwell | You may be on to something there. |
| 05:52.35 | coppice | reasonably happy with 1.4.something highish :-) |
| 05:52.54 | Qwell | how about we just MD5 the release numbers, and call it that? |
| 05:53.07 | coppice | or use the date |
| 05:53.20 | TellarHK | As a new guy to Asterisk, what I think is needed most is release numbers in hexadecimal. |
| 05:53.25 | TellarHK | That's what I really wanted it to have. |
| 05:53.44 | TellarHK | But it doesn't, so screw it, I'm going back to Intertel. ;) |
| 05:53.49 | Qwell | TellarHK: April is coming up soon.. |
| 05:53.58 | Qwell | That would be rather hilarious, I must admit. |
| 05:54.19 | *** join/#asterisk SwK_ (n=SwK@user-24-236-97-182.knology.net) |
| 05:54.23 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 05:54.43 | Qwell | "The Asterisk development team is pleased to announce the latest release of the Asterisk 1.4 series - 1.4.2F." |
| 05:55.01 | TellarHK | Ha. :) |
| 05:55.36 | BadHAL | Just out of curiosity, why is development on an older version going on? Is 1.6 not the newest? |
| 05:55.52 | Qwell | BadHAL: Same reason the 2.4 kernel is still being maintained. |
| 05:55.56 | BadHAL | :P |
| 05:56.08 | BadHAL | old habits die hard and embeded devices? |
| 05:56.11 | Qwell | You can't force people to upgrade like that over night. There are new features and things people don't want to use yet. |
| 05:56.42 | TellarHK | I just wish SCCP were handled a bit better, but I don't know if that's Asterisk or Trixbox or anything in between. |
| 05:56.54 | Qwell | TellarHK: It's trixbox. Use chan_skinny. |
| 05:56.55 | BadHAL | I blame cisco |
| 05:56.57 | TellarHK | I'm too new to have -good- complaints. |
| 05:56.59 | Qwell | and that |
| 05:57.09 | *** join/#asterisk op3r (n=op3r@124.104.89.57) |
| 05:57.11 | TellarHK | Yeah, back on SIP for now. |
| 05:57.17 | BadHAL | I wish cisco would stop developing proprietary protocols |
| 05:57.26 | Qwell | BadHAL: http://www.ntbox.com/cisco-openletter.html |
| 05:58.13 | Qwell | TellarHK: There are 2 (basically) Skinny channel drivers for Asterisk. chan_sccp and chan_skinny |
| 05:58.40 | Qwell | TellarHK: chan_sccp is an out-of-tree and unsupported module. Compared to older versions of chan_skinny, it was marginally better. |
| 05:58.48 | TellarHK | Qwell: Yeah, am trying a recent chan_sccp right now. Didn't know if skinny was supposed to be all that good, I read a lot of conflicting things. |
| 05:58.53 | Qwell | As of 1.4 and especially 1.6, it has been *vastly* improved. |
| 05:58.57 | TellarHK | Okay. |
| 05:59.16 | Qwell | about 4 of us (myself less lately) have given it a ton of love |
| 05:59.19 | TellarHK | Are there any good places with documentation on using skinny right now? |
| 05:59.26 | Qwell | TellarHK: not really |
| 05:59.31 | TellarHK | Ugh. |
| 05:59.47 | Qwell | BadHAL: check out that link.. you'll hate Cisco even more |
| 05:59.51 | BadHAL | ys |
| 05:59.57 | BadHAL | yes* I am reading it now |
| 05:59.59 | Qwell | that = me |
| 06:00.00 | TellarHK | Okay. Just got a pair of 7941's that don't like SIP all that much. |
| 06:00.13 | Qwell | TellarHK: yeah.. SIP on those is rather terrible |
| 06:00.22 | TellarHK | Isn't the license something insanely cheap though? |
| 06:00.23 | Qwell | 7941's should work pretty well with chan_skinny |
| 06:00.34 | Qwell | cheap? Cisco? no :) |
| 06:00.38 | Qwell | see the above URL, heh |
| 06:00.52 | TellarHK | I thought I saw someone that said $15 for a support contract on a used 7941 or something, okay. |
| 06:00.56 | Qwell | nope |
| 06:01.03 | Qwell | well, sort of... |
| 06:01.04 | TellarHK | Glad my phones came from someplace with monster support and my roommate still has access. |
| 06:01.13 | Qwell | that letter actually explains the whole licensing thing with those |
| 06:01.30 | TellarHK | Wow. |
| 06:01.40 | TellarHK | Yeah, that's a lot worse than I imagined. |
| 06:01.41 | BadHAL | yes |
| 06:01.48 | BadHAL | quite annoyign |
| 06:02.05 | TellarHK | Of course, you don't see Cisco killing eBay sales. |
| 06:02.13 | TellarHK | Which according to this letter, they should be doing. |
| 06:02.16 | Qwell | it literally took me years to find that information. It took me going as far as emailing their CEO and general counsel. |
| 06:02.29 | Qwell | They took *2 weeks* to draft that response |
| 06:02.34 | BadHAL | im suprised they answered |
| 06:03.11 | TellarHK | Pity that First Sale doesn't apply to -anything- anymore. |
| 06:03.33 | Qwell | 2 days for initial response. 3 more days for response saying that they're getting more info. 8 more days for the real response. |
| 06:03.39 | BadHAL | I am a firm believer that software licenses SHOULD be transferable |
| 06:04.27 | BadHAL | especially something that is burned onto an embedded device that is only used as an appliance |
| 06:04.53 | BadHAL | it's not like I am going to make my own phone around a cisco rom image |
| 06:05.38 | BadHAL | to me that would be like, for instance any other device that has firmware on it |
| 06:05.46 | BadHAL | your hard drive, your car stereo |
| 06:05.47 | BadHAL | your car |
| 06:05.52 | BadHAL | anything and everything these days |
| 06:06.03 | TellarHK | Yeah. |
| 06:06.08 | BadHAL | if I sold my car to someone and told them they could not have the software that came with it... |
| 06:06.11 | BadHAL | and flashed the rom |
| 06:06.12 | BadHAL | haha |
| 06:06.17 | BadHAL | how would that go over? |
| 06:06.26 | TellarHK | Here we go. |
| 06:06.30 | *** join/#asterisk prg3 (n=prg3@playground.cein.ualberta.ca) |
| 06:06.32 | BadHAL | I'll stop there :P |
| 06:06.43 | TellarHK | License fees for Cisco phones are $8.99/year, per phone. |
| 06:06.50 | BadHAL | which is insane |
| 06:06.52 | TellarHK | Not anything I'd pay for personal use, I'll admit. |
| 06:06.53 | Qwell | TellarHK: nope |
| 06:06.59 | Qwell | it's more like $100 per phone per year |
| 06:07.05 | BadHAL | Whoa, hold on |
| 06:07.08 | TellarHK | Oh? |
| 06:07.10 | BadHAL | who is your vendor? |
| 06:07.17 | TellarHK | http://www.cdw.com/shop/products/default.aspx?edc=993513&enkwrd=ALLPROD%3aCON-SNT-CP7941 |
| 06:07.21 | coppice | BadHAL: MS claims just that when you sell your BMW 7 |
| 06:07.25 | Qwell | BadHAL: smartnet is not a use license |
| 06:07.30 | Qwell | it is merely support. |
| 06:07.33 | TellarHK | Hmm. |
| 06:07.36 | BadHAL | this is true TellarHK |
| 06:07.40 | BadHAL | 8.99 is a bit low |
| 06:07.47 | BadHAL | I work for the goverment so perhaps we get a lower rate |
| 06:08.05 | BadHAL | but I know we pay just over $30 per phone per year |
| 06:08.05 | Qwell | "Answer: Yes, Cisco requires customers to obtain a Cisco license to use a Cisco Unified IP phone (SCCP or SIP protocol) with either the Cisco Unified Communications Manager product or with any other Cisco-branded or third party call control device." |
| 06:08.05 | Qwell | :) |
| 06:08.28 | BadHAL | of course when we deploy a system we cover an entire base with them |
| 06:08.39 | BadHAL | 3-10k phones |
| 06:08.39 | TellarHK | Might be worth asking Cisco to clarify the cost on that, really. What contract applies to some example phones. |
| 06:08.43 | BadHAL | depending on the base |
| 06:08.47 | Qwell | remember - this was a lawyer that drafted this response, and it was reviewed by a team. they were *VERY* careful about every word they used. |
| 06:09.07 | Qwell | When they said "license", they meant license. |
| 06:09.24 | Qwell | smartnet, however, is merely an "agreement" or "contract" |
| 06:09.50 | BadHAL | smartnet is just a hardware/software service agreement |
| 06:10.08 | BadHAL | if you have the product (and the needed licenses) |
| 06:10.09 | Qwell | actually, the license may not be yearly.. |
| 06:10.30 | BadHAL | it basicly provides a place to download updated firmware for devices in most cases :P |
| 06:10.36 | prg3 | Is festival worth getting working? |
| 06:10.39 | Qwell | I think it's just a one-time fee, that is not included with the price of the phone (and is not transferable) |
| 06:10.44 | BadHAL | prg3, in my experience no |
| 06:10.46 | coppice | basically the policy is to make things so vague you never know if you are in compliance or not. while this works well for governments making laws, it can work badly for corporations. you might as well try to pay nothing at all, and just wait to see what happens |
| 06:10.52 | TellarHK | I see Cisco CallManager licenses, but no phone licenses on CDW. |
| 06:11.02 | prg3 | BadHAL: Flite a better option, or just record voices? |
| 06:11.03 | BadHAL | the quality was terrible |
| 06:11.06 | Qwell | BadHAL: yep, and worse, it only gives you access to download the firmware - it doesn't mean you're allowed to use it :) |
| 06:11.11 | BadHAL | I ended up recording voices |
| 06:11.14 | Qwell | TellarHK: correct |
| 06:11.19 | BadHAL | Qwell, correct |
| 06:11.31 | BadHAL | prg3, if you can find a better option let me know |
| 06:11.39 | BadHAL | I did not care for festival |
| 06:11.45 | BadHAL | I never tried flite |
| 06:11.54 | Qwell | coppice: probably ends up cheaper too, depending on who you are. heh |
| 06:12.25 | prg3 | BadHAL: so far, festival is a non-option for me, it accepts the connection from asterisk and drops it right away. I'll go after Flite. I just need some voice for testing, and I don't really want to listen to mine :) |
| 06:12.27 | [TK]D-Fender | Don't know what the fuss is about when Cisco is just trouble with * and SIP to begin with and issues with licensing and cost right off the start... |
| 06:12.54 | TellarHK | I think that response really begs for a followup letter. |
| 06:13.08 | Qwell | TellarHK: They haven't responded to any of my followups |
| 06:13.11 | TellarHK | Going into the details of asking how people can acquire licenses, since that is far from readily apparent. |
| 06:13.13 | TellarHK | Hmm. |
| 06:13.27 | BadHAL | you will need to contact a cisco vendor |
| 06:13.29 | TellarHK | That sucks. |
| 06:13.49 | TellarHK | Well, if they're going to make it that tough, I'm just not going to pay a license. 'nuff said. ;) |
| 06:14.01 | BadHAL | for personal use do whatever you want |
| 06:14.01 | TellarHK | At least not for home use. Business, sure, I'd jump through hoops. |
| 06:14.10 | BadHAL | it's not like cisco is going to come knocking on your door |
| 06:14.27 | BadHAL | yea, stay clear from stuff like that when doing it for a real buisness |
| 06:14.38 | BadHAL | go buy some polycom devices |
| 06:14.42 | BadHAL | I have had good luck with them |
| 06:14.50 | TellarHK | I hear Polycom's really good. |
| 06:14.54 | BadHAL | I like them |
| 06:15.02 | TellarHK | Pricy, but good. |
| 06:15.05 | BadHAL | I have a soundpoint 330 sitting on my desk at home |
| 06:15.10 | BadHAL | pricy? Compared to what? Cisco? |
| 06:15.11 | BadHAL | ha |
| 06:15.50 | TellarHK | Pricy compared to free Ciscos. ;) |
| 06:15.50 | coppice | polycoms have nice hardware, but their software QA seems a bit iffy |
| 06:15.50 | BadHAL | the webinterface? |
| 06:15.50 | BadHAL | yes |
| 06:15.50 | TellarHK | Cracked cases and LCD's make Ciscos pretty cheap. |
| 06:15.50 | BadHAL | garbage |
| 06:15.50 | BadHAL | the remote configuration options are outstanding for polycom |
| 06:15.53 | BadHAL | I only have 4 phones at home |
| 06:16.02 | BadHAL | but it is quite nice to just plug them in and BAM |
| 06:16.02 | BadHAL | done |
| 06:16.05 | BadHAL | all configured |
| 06:16.24 | BadHAL | vs most people log into the web interface |
| 06:16.28 | BadHAL | and dick around for an hour |
| 06:16.33 | Micc | Is there anyway to evaluate a variable and pass multiple variables declared as a single variable in the dialplan? I want to call a macro like this Macro(xfer,${args}) |
| 06:16.48 | Micc | and ${args} could be like 6007|SIP/6007 |
| 06:17.11 | Micc | but it is treating it all as one parameter as ARG1. and it cuts it off at the pipe |
| 06:17.25 | Micc | commas get passed through but they don't turn it into another argument either. |
| 06:19.46 | *** join/#asterisk UQlev (n=kvirc@91.184.220.73) |
| 06:27.57 | [TK]D-Fender | Micc: PB your dialplan and CLI output |
| 06:28.50 | prg3 | Is there a way to check where asterisk thinks the full path to sound files are for Playback? |
| 06:29.53 | BadHAL | asterisk.conf |
| 06:31.03 | prg3 | astsounds? it's specified to my sound directory.. |
| 06:31.15 | [TK]D-Fender | prg3: varlib <- |
| 06:37.22 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 06:41.19 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 06:41.26 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-49f40ffb5cd804d0) |
| 06:44.22 | prg3 | something is messed up then, astsound and astsounds are defined in asterisk.conf as /var/lib/asterisk/sounds, test.wav exists, and is readable by the asterisk user in that directory, yet asterisk errors with "Unable to open test.wav" |
| 06:44.51 | BadHAL | strange |
| 06:45.43 | [TK]D-Fender | prg3: "astsounds" Where did you come up with that name from? |
| 06:46.15 | prg3 | guess.. the asterisk.conf page on voip-info doesn't have a directory for sounds.. |
| 06:46.30 | [TK]D-Fender | prg3: "astvarlibdir => " <- this is what you set, and it looks for "sounds" under this |
| 06:46.44 | [TK]D-Fender | prg3: WIKI = "random crap" |
| 06:46.55 | prg3 | astvarlibdir => /var/lib/asterisk |
| 06:46.56 | [TK]D-Fender | prg3: Read the samples in your source tarball |
| 06:46.58 | prg3 | True |
| 06:49.08 | prg3 | Any ideas what might be going on here? |
| 06:49.31 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 06:49.45 | [TK]D-Fender | prg3: You could start by showing use the actual failed call. |
| 06:50.30 | prg3 | exten => s,n,Playback(test.wav) |
| 06:50.42 | [TK]D-Fender | prg3: Not legal and you should know better. |
| 06:50.46 | BadHAL | Playback(test) |
| 06:50.48 | BadHAL | first |
| 06:50.51 | [TK]D-Fender | prg3: you NEVER specify the EXTENSION <- |
| 06:50.52 | *** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 06:51.00 | BadHAL | the reason it cant find it |
| 06:51.09 | BadHAL | is because it's really looking for test.wav.wav |
| 06:51.10 | BadHAL | when you call that |
| 06:51.13 | [TK]D-Fender | prg3: * will pick it based on the best format available for transcoding. |
| 06:51.18 | BadHAL | ^^ |
| 06:51.31 | prg3 | [Jan 30 23:51:17] WARNING[19393]: file.c:568 ast_openstream_full: File test does not exist in any format |
| 06:52.03 | [TK]D-Fender | prg3: its looking for "test.wav.moreshitherebecausewetoldyoudon'tputthedamnextensionon!" |
| 06:52.19 | prg3 | there's no longer an extension |
| 06:52.38 | [TK]D-Fender | prg3: please pastebin the OCMPLETE output and your configs AND the folder dump. |
| 06:52.50 | [TK]D-Fender | prg3: You've alos been modding crap all around with invented values |
| 06:52.59 | prg3 | They're gone now.. |
| 06:53.08 | [TK]D-Fender | prg3: your dialplan line was clearly wrong, but trust factor for the remainer is equally nil |
| 06:54.54 | prg3 | http://pastebin.com/m5fbbcd0c |
| 06:55.04 | prg3 | Hopefully that's enough.. I can add more if you'd like. |
| 06:56.30 | baliktad | lines 23 and 33 are telling you the answer |
| 06:57.18 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 06:57.46 | prg3 | baliktad: because root owns it? |
| 06:57.51 | drmessano | apparently my wakeup call script has a problem |
| 06:57.58 | drmessano | Damnit |
| 06:58.32 | baliktad | "Unable to open test (format 0x44 (ulaw|slin))" <-- it's looking for ulaw or signed linear (.sln) |
| 06:59.55 | [TK]D-Fender | prg3: Go fix the owner and rights |
| 07:00.15 | baliktad | http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 07:00.55 | *** join/#asterisk fiddur (n=fiddur@92-32-123-186.tn.glocalnet.net) |
| 07:01.38 | prg3 | [Jan 31 00:00:03] WARNING[19107]: file.c:897 ast_readfile: Unable to open /usr/share/asterisk/sounds/test.wav |
| 07:01.47 | prg3 | Where is it getting the /usr/share/asterisk from? |
| 07:02.01 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 07:02.43 | baliktad | if you pasted all your configs we might be able to tell you |
| 07:02.49 | [TK]D-Fender | prg3: Did you restart * following these config changes? |
| 07:03.00 | prg3 | Yup |
| 07:04.30 | fiddur | Hello. Can anyone explain the autopause function in app_queue? Why does it auto-pause queue-members who's interfaces return "Busy" directly, and not only interfaces which ring until timeout is reached? If an interface is busy, why would I want to auto-pause it?! |
| 07:07.34 | prg3 | baliktad: there's a lot of config.. what do you want to see? grep "usr/share" * in /etc/asterisk doesn't show it redefining varlibdir.. could it be a hard coded ubuntu thing? |
| 07:08.16 | prg3 | I just tried using one of the stock sounds, and it doesn't error any longer, but I don't get sound on the phone. The line is a analog line on a sangoma zaptel card |
| 07:09.10 | prg3 | Specifically a A200 with wanpipe 1.1 |
| 07:16.55 | prg3 | I'm giving up for the day.. thanks for the help tonight |
| 07:30.04 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.86) |
| 07:47.34 | riddlebox | have you guys seen asterikast? |
| 08:08.02 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 08:29.23 | *** join/#asterisk myselfhimself (n=jonathan@ip-33.net-82-216-240.rev.numericable.fr) |
| 08:29.26 | myselfhimself | hi |
| 08:34.00 | *** join/#asterisk vi390 (n=fc@unaffiliated/vi390) |
| 08:38.55 | *** join/#asterisk xkr47 (i=xkr47@a88-112-18-173.elisa-laajakaista.fi) |
| 08:43.09 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 08:44.16 | *** join/#asterisk Gary_ (n=Gary@freenode/staff/colchester-lug.gary) |
| 08:45.40 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.199) |
| 09:05.31 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:24.42 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 09:30.10 | TellarHK | Are Trixbox and AsteriskNOW pretty much the only two half-decent free implementations with a GUI? |
| 09:31.24 | drmessano | PBX In a Flash |
| 09:31.26 | drmessano | Trixbox sucks |
| 09:31.45 | drmessano | You can also roll your own FreePBX box |
| 09:31.52 | drmessano | not that hard |
| 09:32.18 | TellarHK | Hmm. |
| 09:32.45 | TellarHK | Is there any good compelling reason to do anything other than a Trixbox install, though? |
| 09:33.00 | TellarHK | From a "Not going to be deploying it commercially anytime soon" standpoint? |
| 09:33.23 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 09:33.57 | *** join/#asterisk makkksimal (n=makkksim@91-66-249-217-dynip.superkabel.de) |
| 09:40.08 | drmessano | Yes |
| 09:40.27 | drmessano | Trixbox sucks.. Its a horrible product, backed by idiots.. |
| 09:40.40 | drmessano | Its unstable, horribly supported, and bloated |
| 09:40.59 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 09:41.10 | drmessano | Do yourself a favor and get PIAF or Asterisknow if you MUST use an ISP |
| 09:41.12 | drmessano | Do yourself a favor and get PIAF or Asterisknow if you MUST use an ISO |
| 09:41.14 | tzafrir_laptop | pbxinaflash does not include the source. I can't help them. I can't provide them patches |
| 09:41.27 | TellarHK | None of which has -really- impacted me yet, but it's only been up a couple days with two lines and Gizmo integration. |
| 09:41.36 | drmessano | Trixbox doesnt accept patches |
| 09:41.38 | TellarHK | I hadn't heard of PIAF until you mentioned it just now. |
| 09:42.15 | TellarHK | And we tried AsteriskNOW, but it collapsed in on itself somehow. Pissed off the roommate. |
| 09:42.32 | drmessano | Well, PIAF would be best |
| 09:42.43 | drmessano | I've used it a little, its not too bad |
| 09:43.43 | drmessano | trixbox is far more likely to suffer from "PBX collapse disorder" as well |
| 09:44.00 | drmessano | Avoid YUM and updating at all costs |
| 09:44.11 | TellarHK | Too late for that. Amazingly it worked pretty well though. |
| 09:44.17 | drmessano | No its not |
| 09:44.21 | TellarHK | No collapses, and up to date. |
| 09:44.23 | drmessano | Just install something else |
| 09:44.30 | drmessano | Ok, TellarHK, whatever |
| 09:44.32 | TellarHK | Well, too late for YUM. I upgraded the whole thing last night. |
| 09:44.39 | TellarHK | No, I'm not saying too late to try something else. |
| 09:44.46 | TellarHK | Not at all, I'm actually kinda looking for other options to try. |
| 09:45.05 | TellarHK | Just saying that Trixbox hasn't been -that- catastrophic yet. |
| 09:45.12 | drmessano | Probably not |
| 09:45.16 | drmessano | YOu've been at this a WEEK? |
| 09:45.23 | TellarHK | Two days. ;) |
| 09:45.27 | drmessano | Yeah |
| 09:45.28 | TellarHK | So yeah, I know. |
| 09:45.40 | drmessano | trixbox has a long horrid history |
| 09:45.40 | TellarHK | Will probably play with another VM this weekend, that's why I asked. |
| 09:45.45 | drmessano | Which continues to this day |
| 09:45.50 | TellarHK | That I've started to hear a bit of. |
| 09:46.22 | drmessano | They snoop on users, have horrible RPMs, have been known to push updates that irreversibly destroy installs |
| 09:46.25 | drmessano | "oops" |
| 09:46.35 | TellarHK | There are things about it that just rub me the wrong way, it's like it's trying to hold my hand -sometimes-, and then leaves me in the fucking dark other times. |
| 09:46.39 | drmessano | and their line is "Why did you update if it was working" |
| 09:46.45 | TellarHK | Eww. |
| 09:46.51 | drmessano | and then when you get haxored by some lame ass security hole |
| 09:46.59 | drmessano | "WHY DIDNT YOU UPDATE: |
| 09:47.06 | drmessano | "Um, you told me not to?" |
| 09:47.07 | TellarHK | Hah. |
| 09:48.01 | TellarHK | I'm an independent contract guy, I have some customers that just let me remote in and update things whenever I want. |
| 09:48.22 | TellarHK | Because they just can't hack sorting that sort of thing out. |
| 09:48.40 | drmessano | Right now, the guy maintaining the repos is the original developer.. After they bought him out and renamed it to trixbox however long ago its been, 2 years or so, whenever they have fired a dev and hes taken back over, they've nuked installed, rolled out shit updates, etc |
| 09:48.47 | drmessano | Hes not an idiot, but hes careless |
| 09:49.08 | drmessano | installs* |
| 09:49.12 | TellarHK | Ouch. Okay. |
| 09:49.26 | TellarHK | How tough is it to slap a GUI on top of a roll your own? |
| 09:49.50 | drmessano | http://wiki.2l2o.com/index.php/Asterisk_without_Tricks |
| 09:49.53 | drmessano | Not hard |
| 09:49.58 | drmessano | Follow that guide, maybe 2 hours |
| 09:50.21 | *** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 09:50.25 | drmessano | I wrote that, but dont blame me if you skip a step and end up with poo jello |
| 09:51.25 | TellarHK | Heh. |
| 09:51.38 | drmessano | I get that shit all the time |
| 09:51.44 | drmessano | Permissions issues, etc |
| 09:51.50 | drmessano | "Oh, didnt think I needed to.." |
| 09:51.56 | TellarHK | Now how much of a problem am I going to run into with DNS on another box? |
| 09:51.57 | drmessano | Yes, I LOVE writing EXTRA steps |
| 09:52.11 | TellarHK | Hopefully none if said DNS is right? |
| 09:52.29 | drmessano | Theres no reason to not just follow the steps |
| 09:53.15 | drmessano | If you choose not to install BIND, and end up with DNS problems, good luck with that.. Its only being used as a local DNS resolver |
| 09:53.18 | TellarHK | Okay. Just trying to make sure the Windows box serving DNS/DHCP isn't going to make things -too- bad. |
| 09:53.21 | TellarHK | Okay. |
| 09:53.39 | drmessano | I've built dozens of boxes using that same set of steps |
| 09:53.42 | drmessano | Never had a problem |
| 09:53.56 | TellarHK | Hey, just for the record? I'm not a total noob, I just haven't done much serious stuff in Linux for a few years. And I'm at the point where I just want crap to work. |
| 09:54.05 | drmessano | Thats fine |
| 09:54.11 | drmessano | Thats what that guide is for |
| 09:54.16 | drmessano | Dont think, just type |
| 09:54.31 | drmessano | If you start thinking, you will end up with poo jello |
| 09:54.43 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:56.27 | TellarHK | Fortunately VM's full of that are easy to nuke. |
| 09:56.46 | drmessano | So is any box |
| 09:57.12 | TellarHK | Yep. |
| 09:57.32 | TellarHK | May want to do that, and start trying to play around with skinny instead of chan_sccp. |
| 09:58.37 | drmessano | I just avoid Cisco phones |
| 09:58.51 | drmessano | Overpriced and pointless on non-Cisco systems |
| 09:59.03 | TellarHK | Well, they were free. So I have them. |
| 09:59.16 | drmessano | What models? |
| 09:59.21 | TellarHK | 7941's. |
| 09:59.53 | TellarHK | Cracked housing on one, cracked LCD on another but it's still fully usable. Nobody would take them on desk though, so they got put in a trash pile. |
| 09:59.56 | drmessano | Should be easy to eBay them and buy real phones :) |
| 10:00.06 | TellarHK | Bah. :) |
| 10:00.20 | TellarHK | These look like they've been beaten up by a drunk husband. |
| 10:00.30 | drmessano | ebay'ers are suckers |
| 10:00.41 | *** join/#asterisk Miccster (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 10:00.49 | drmessano | Put that in the desciption, they'll still sell |
| 10:00.55 | TellarHK | Hah. |
| 10:01.10 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 10:01.13 | TellarHK | To be honest, I really like 'em. The speakers are great, and the microphone's amazing on the speakerphone. |
| 10:01.30 | drmessano | Polycoms are 10x better |
| 10:01.37 | drmessano | For speakerphone anyway |
| 10:02.21 | Miccster | does func_odbc have a small timeout in case it can't get a response it doesn't lock up the dialplan? |
| 10:02.34 | TellarHK | I'm not surprised to hear that about Polycom at all. |
| 10:02.53 | TellarHK | If I ever buy business phones, Polycom's high on the "If we can afford it" list. |
| 10:03.07 | joobie | polycom isnt that much more expensive |
| 10:03.08 | drmessano | Um |
| 10:03.18 | drmessano | Polycoms arent expensive |
| 10:03.22 | joobie | heck, for a once off investment, for a business.. an extra 50-100$ per phone is not much |
| 10:03.27 | drmessano | IP330 is $110 |
| 10:03.32 | coppice | the low end polycoms aren't too expensive. the snag is their prices rise rapidly with small extra features |
| 10:03.38 | drmessano | They're not extra |
| 10:03.45 | drmessano | coppice: True |
| 10:03.47 | joobie | they are in AU |
| 10:03.55 | TellarHK | I've been working for a non-profit primarily for the last 2 years. I couldn't get them to spend $800 on a server rack, and they've had four power supplies die in the last week because of it. |
| 10:03.57 | drmessano | 50-100$ more than what? |
| 10:03.59 | joobie | anyway, back to CSS.. |
| 10:04.01 | TellarHK | I'm not used to thinking with budgets. ;) |
| 10:04.02 | joobie | linksys |
| 10:04.09 | drmessano | Not true |
| 10:04.13 | joobie | in AU it is |
| 10:04.16 | drmessano | SPA-941 is $89 USD |
| 10:04.23 | drmessano | IP 330 is $100 USD |
| 10:04.28 | drmessano | err |
| 10:04.30 | drmessano | $110 |
| 10:04.32 | joobie | ur lucky |
| 10:04.38 | drmessano | $20 a phone is not a big deal |
| 10:04.48 | joobie | nod |
| 10:04.59 | TellarHK | Wow, that's a lot better price than I expected. |
| 10:05.01 | *** join/#asterisk af_ (n=getsmart@88-149-230-97.dynamic.ngi.it) |
| 10:05.05 | TellarHK | Every Polycom I see at Fry's is like $300. |
| 10:05.10 | drmessano | HA |
| 10:05.16 | drmessano | Two days? |
| 10:05.19 | drmessano | Thats about right |
| 10:05.21 | TellarHK | And I don't expect Fry's to be THAT much higher. |
| 10:05.27 | drmessano | http://www.telephonydepot.com |
| 10:05.44 | drmessano | Dont buy VoIP devices retail |
| 10:05.45 | drmessano | Ever |
| 10:06.00 | *** join/#asterisk af_ (n=getsmart@88-149-230-97.dynamic.ngi.it) |
| 10:06.20 | coppice | isn't not that easy to VoIP stuff retail in most places |
| 10:06.38 | drmessano | Yeah, margins suck.. low volume |
| 10:06.58 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 10:06.58 | drmessano | Buy from a real retailer |
| 10:07.04 | drmessano | online retailer |
| 10:07.18 | drmessano | Not brick and mortar best buy/frys/etc |
| 10:07.20 | TellarHK | I've seen Polycom stuff ever since college. And the ubiquitous conferencing rigs they have everywhere. Always -big- money. didn't know they even made regular desk-type phones. |
| 10:07.41 | drmessano | Yep |
| 10:07.49 | drmessano | Go to that link |
| 10:07.53 | drmessano | If you havent |
| 10:08.08 | drmessano | They're decent.. decent prices, good to work with |
| 10:08.26 | *** join/#asterisk af_ (n=getsmart@88-149-230-97.dynamic.ngi.it) |
| 10:08.29 | drmessano | TK loves them, so that must be worth something.. he loves no one/nothing |
| 10:08.56 | drmessano | He types in ALL CAPS to his MOM |
| 10:09.04 | drmessano | But loves telephonydepot |
| 10:09.10 | drmessano | *shrug* |
| 10:09.18 | TellarHK | Yeah, I'm looking there now. |
| 10:11.01 | TellarHK | Some really nice Polycom phones there. |
| 10:14.10 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:14.11 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:15.30 | TellarHK | Wow, there's a brand name that should have spent some time in focus groups. |
| 10:16.02 | TellarHK | Snom. |
| 10:19.30 | mort_gib | Snoms are nice |
| 10:20.31 | TellarHK | Sounds like it needs an umlaut and a tag at Ikea. |
| 10:21.49 | mort_gib | Yeah, but the actual phones are great |
| 10:23.45 | mort_gib | One thing that makes a huge difference for my users is the website, Snom bundles a nice website that allows each use to "play" around with the phone. I lock the essential settings, and they love that |
| 10:23.51 | mort_gib | Less resistance to change |
| 10:24.05 | TellarHK | Okay, that's pretty nice. |
| 10:24.12 | TellarHK | I like that they have a DECT set. |
| 10:24.36 | coppice | the DECT set sucks |
| 10:26.29 | TellarHK | I need something with a decent speaker, that's the big thing. |
| 10:26.34 | TellarHK | So probably Polycom. |
| 10:26.42 | TellarHK | If/when I replace what's on my desk now. |
| 10:27.44 | coppice | polycoms seem the best for speakerphone. it shouldn't be hard to get reasonable results these days, but people just don't seem to apply even minimal effort |
| 10:30.43 | TellarHK | Unless you're a Cisco or a Polycom, you probably don't have a reputation enough to worry about losing by using crap. |
| 10:30.59 | TellarHK | And Cisco's been squandering that for a while, particularly post-Linksys. |
| 10:31.24 | coppice | you don't think Avaya or Nortel have a reputation to loose? |
| 10:31.44 | TellarHK | I wasn't going to list -all- the big names. :) |
| 10:32.05 | TellarHK | I'd put those two up there in the Cisco/Polycom range though. Ones that do have something to defend, though I know less about Avaya than any of the others. |
| 10:32.32 | coppice | I hate the lead weighted handsets cisco use to try to make a cheap bit of flimsy plastic appear more solid |
| 10:33.44 | TellarHK | I'm really surprised the sound quality of these Ciscos is as good as it is. |
| 10:34.08 | TellarHK | Especially on speaker. Using Gizmo's probably pretty weak service, I was able to have a pretty good conversation with my folks. |
| 10:34.58 | coppice | but it isn't all that good as something that price ought to be. they only just started introducing wideband voice on their newest stuff. its pathetic |
| 10:38.37 | TellarHK | Yeah. |
| 10:39.09 | TellarHK | I don't know much about the codecs involved yet, only started digging into those a little yesterday. |
| 10:39.20 | TellarHK | But I'm surprised at how low the bitrates are. |
| 10:40.14 | coppice | the farce is the amount RTP overheads add to swamp the low bit rate of the codec :-) |
| 10:41.26 | TellarHK | Pretty nasty percentage? |
| 10:42.15 | *** join/#asterisk af_ (n=getsmart@88-149-230-97.dynamic.ngi.it) |
| 10:42.28 | coppice | the 8k G.729 bulks up to nearly 30k :-) |
| 10:42.36 | TellarHK | Wow. |
| 10:42.51 | TellarHK | I think we're using 711 |
| 10:47.48 | *** join/#asterisk af_ (n=getsmart@88-149-230-108.dynamic.ngi.it) |
| 11:09.42 | TellarHK | Yeah, I think I'm going to see about shifting over to a self-rolled Asterisk+FreePBX install tomorrow. |
| 11:10.04 | mort_gib | Have any of you played around with the CTI software that "comes" with the Snom's |
| 11:40.31 | *** join/#asterisk mafrac (n=chatzill@84.121.145.40.dyn.user.ono.com) |
| 11:41.26 | mafrac | Hello all |
| 11:41.43 | mafrac | Anybody could answer a question? |
| 11:42.40 | mafrac | Is it possible to use a movile as a gateway to asterisk? |
| 11:43.22 | mort_gib | mafrac: Use a GSM gateway |
| 11:43.52 | mafrac | Yes, I know it |
| 11:44.06 | mort_gib | Forget mobile |
| 11:44.16 | mafrac | ah ok |
| 11:44.41 | mafrac | wich one do you recomend me? |
| 11:45.15 | mort_gib | I use portech |
| 11:45.28 | mafrac | thanks |
| 11:45.47 | mort_gib | http://www.voipon.co.uk/portech-voip-gsm-gateway-c-3_192_193.html |
| 11:46.13 | mort_gib | Don't ask these guys questions, they are pretty clueless, but they are fast and reliable |
| 11:46.23 | mort_gib | This is supposing your in Europe |
| 11:46.35 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 11:46.59 | mafrac | Yes, in spain |
| 11:47.13 | mort_gib | Ok, like me |
| 11:47.22 | mafrac | jajaja |
| 11:47.34 | mafrac | so, you speak spanish, not? |
| 11:47.36 | mort_gib | Pero, soy Danes |
| 11:47.51 | Greek-Boy | Which open source billing package for asterisk is highly recommended? |
| 11:48.02 | mafrac | Bueno, yo soy español, jeje |
| 11:48.03 | mort_gib | Si, hablo Espanol |
| 11:48.11 | mort_gib | Vale |
| 11:48.13 | mafrac | Pues encantado |
| 11:48.44 | mort_gib | Egualmente, pero recuerdo que esto channel es en Ingles |
| 11:48.47 | mafrac | La verdad es que estoy interesado en crear una centralita con asterisk y tengo algunas dudas que no encuentro en la documentación |
| 11:49.04 | mafrac | hablamos en privado? |
| 11:56.02 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-23-158.w86-215.abo.wanadoo.fr) |
| 11:58.23 | frogonwheels | ~x es en La verdad es que estoy interesado en crear una centralita con asterisk y tengo algunas dudas que no encuentro en la documentación |
| 12:05.18 | *** join/#asterisk ITguru (n=ITGuru@5ac10610.bb.sky.com) |
| 12:10.54 | *** join/#asterisk neurosys (n=vinix@c-71-196-8-216.hsd1.fl.comcast.net) |
| 12:56.39 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 12:59.07 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 13:08.55 | *** join/#asterisk xacatecas (n=jkroon@41.240.145.87) |
| 13:09.58 | xacatecas | hi, quickly need some confirmation, if a call comes into the system, it gets answered by an "IVR" type system, then another channel is created and sent into a macro before being bridged, can I get it set up such that the incoming and outgoing channel is logged in the CDRs seperately? |
| 13:19.49 | *** join/#asterisk ScriptFanix (i=vincent@2a01:e35:2f43:ae90:21a:70ff:fea3:44ab) |
| 13:19.56 | ScriptFanix | Hi |
| 13:20.40 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 13:20.42 | ScriptFanix | I'm running asterisk from debian etch (1.2), and use cdr_odbc to store CDRs in a mysql database. |
| 13:21.04 | ScriptFanix | sometimes I get > cdr_odbc: Error in Query -1 |
| 13:21.46 | ScriptFanix | and calls aren't logged until I restart asterisk. |
| 13:22.13 | ScriptFanix | any idea why this happens, and how i could avoid this ? |
| 13:31.05 | ScriptFanix | looks strangely like this old closed bug report: http://bugs.digium.com/view.php?id=3618 |
| 13:31.24 | *** join/#asterisk MrNaz (n=mrnaz@203-217-86-29.dyn.iinet.net.au) |
| 13:39.40 | *** join/#asterisk ITguru (n=ITGuru@5ac10610.bb.sky.com) |
| 13:49.03 | *** join/#asterisk D-m0nj0 (i=d44621d2@gateway/web/ajax/mibbit.com/x-8d1ec992a3665c72) |
| 13:49.14 | D-m0nj0 | hello |
| 13:49.26 | xacatecas | hi |
| 13:49.38 | D-m0nj0 | is there a way to connect a traditional PBX to asterisk using a TDM11B port? |
| 13:49.54 | D-m0nj0 | or the TDM11B is only used to connect traditional telephones |
| 13:49.54 | D-m0nj0 | ? |
| 13:50.16 | xacatecas | as long as FXS connects to FXO and vice versa you're fine. |
| 13:50.34 | D-m0nj0 | but what if i have extensions associated with the PBX? |
| 13:50.39 | D-m0nj0 | how can i reference them |
| 13:51.02 | xacatecas | how long does the pbx take to answer? |
| 13:51.26 | xacatecas | Dial(DAHDI/${PBXCHAN}/,,,,,,${EXT}) perhaps ? |
| 13:51.29 | D-m0nj0 | what do you mean sorry i didnt get it |
| 13:51.45 | D-m0nj0 | the scenario is as follow: |
| 13:52.01 | xacatecas | well, if you simply do Dial(DAHDI/${PBXCHAN}) how long would it take to answer the ringing line? |
| 13:52.03 | D-m0nj0 | i have a traditional PBX running mytelephony system and i have like 100 extensions associated with it |
| 13:52.25 | D-m0nj0 | i would like to integrate now asterisk into my network so i benefit from all its features |
| 13:52.39 | D-m0nj0 | so my concern is which PCI card i need to buy to do that |
| 13:52.48 | D-m0nj0 | i only have 1 external line going to PSTN |
| 13:53.00 | xacatecas | depends on the type of connections and where exactly in your network you want to put it. |
| 13:53.03 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 13:53.07 | D-m0nj0 | and like 50 wires going from PBX to each telephone client |
| 13:53.47 | xacatecas | TDM11B should be ok then. how come you have line a hundred extensions with only one line? |
| 13:54.03 | D-m0nj0 | its actually 2 PBX |
| 13:54.32 | D-m0nj0 | we use it for internal communication more than outside comm |
| 13:54.37 | BBHoss | just get an X100-P |
| 13:54.44 | BBHoss | :] |
| 13:55.04 | xacatecas | ok, well, you could also use the m option to Dial() to get the dtmf codes for the extension sent to the pbx... |
| 13:55.06 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 13:55.09 | D-m0nj0 | ok but please can you explain how does asterix know each extension associated with the PBX? |
| 13:55.16 | xacatecas | M sorry. |
| 13:55.28 | BBHoss | D-m0nj0: you dial ports using the dialplan |
| 13:55.31 | xacatecas | it doesn't, you need to inform it as part of your dialplan. |
| 13:56.02 | xacatecas | so if you want to reach extension 1234 you would probably do something like Dial(DAHDI/1/,,,,,,,1234). |
| 13:56.07 | BBHoss | are all of your extensions SIP, or are they POTS (FXS Devices)? |
| 13:56.25 | D-m0nj0 | POTS i think |
| 13:56.27 | D-m0nj0 | no ip used |
| 13:56.29 | xacatecas | he has a single line going into an analog pbx and he's trying to reach the individual extensions behind it. |
| 13:56.39 | D-m0nj0 | just regular traditional most bsed PBX |
| 13:56.47 | D-m0nj0 | yea exactly xacatecas |
| 13:56.58 | D-m0nj0 | i would like to know if asterix can prove all these extensions somehow |
| 13:57.02 | D-m0nj0 | probe* |
| 13:57.03 | xacatecas | D-m0nj0, the answer really depends on how that pbx handles calls coming in from the PSTN. |
| 13:57.25 | xacatecas | that's a different question. and again depends on the PBX. |
| 13:57.48 | xacatecas | BBHoss, ever used ForkCDR()? |
| 13:57.59 | BBHoss | ideally, you would want a PRI card with a channel-bank to hook all of your extensions to |
| 13:58.01 | BBHoss | xacatecas: nope |
| 13:58.15 | D-m0nj0 | in case i would like to connect all my phones directly into the asterik box i need to buy a huge card no? |
| 13:58.23 | xacatecas | *sigh* ... i'm looking to split a cdr, the T option doesn't quite do what I want. |
| 13:58.36 | xacatecas | so instead I'm wondering whether there's a way to only lock the variables instead of the end time too. |
| 13:58.41 | BBHoss | well you wouldn't buy a big card, just a t1 card with a channel bank |
| 13:59.03 | D-m0nj0 | whats that cost? do you have any idea? |
| 13:59.15 | xacatecas | depends on what country you're in :) |
| 13:59.25 | D-m0nj0 | hehe |
| 13:59.25 | BBHoss | plus, if you have hundreds of extensions, you're going to want more than 1 FXO (Telephone Line) coming in |
| 13:59.34 | BBHoss | ideally you'd have a t1 coming IN as well |
| 13:59.42 | xacatecas | he already answered that one... |
| 13:59.46 | BBHoss | so a te210p |
| 13:59.51 | BBHoss | xacatecas: wasnt around |
| 13:59.57 | D-m0nj0 | but there should be a way to connect my traditional pbx... |
| 14:00.14 | BBHoss | oh you want to link it to another pbx? |
| 14:00.19 | xacatecas | BBHoss, how exactly does a channel bank work ... the T1 has like 30 lines, so it somehow tells the channel bank which channel it wants to speak with? |
| 14:00.42 | xacatecas | would personally throw out the old pbx ... just much simpler. |
| 14:00.48 | xacatecas | and less support issues. |
| 14:01.03 | D-m0nj0 | actually i want to connect my traditional pbx to my asterix but i want to keep all the extensions on my traditional pbx running |
| 14:01.05 | BBHoss | well a t1 has 24 lines, an E1 has 30. The channel bank lets you split the PRI (usually a RJ-45 connector, sometimes others), into the individual lines |
| 14:01.25 | xacatecas | but still the exact same number of them? |
| 14:01.33 | BBHoss | the channel bank generates dialtone and ring voltage |
| 14:01.38 | BBHoss | yeah |
| 14:01.39 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 14:01.39 | riddlebox | hrmm I cannot get a sip client to connect from outside my network to asterisk, stupid linksys firmware |
| 14:01.44 | xacatecas | ah ok. |
| 14:01.57 | xacatecas | riddlebox, NAT issues? |
| 14:02.01 | BBHoss | basically a channel bank transforms digital to analog |
| 14:02.14 | xacatecas | ah ok. makes sense now. thanks. |
| 14:02.18 | xacatecas | needs to look into those. |
| 14:02.35 | xacatecas | will come in handy with some upcoming installs. |
| 14:02.36 | riddlebox | xacatecas, it has to be, I have the ports forwarded but I dont even see the client trying to connect to the server so I know its not asterisk |
| 14:02.40 | BBHoss | http://www.rhinoequipment.com/cb.html |
| 14:02.52 | xacatecas | riddlebox, then get rid of the NAT :) |
| 14:02.54 | D-m0nj0 | why do we need to convert from digital to analog? |
| 14:02.57 | BBHoss | D-m0nj0: you want the FXS ones if you want to connect extensions to them |
| 14:03.05 | riddlebox | I will be in a second |
| 14:03.13 | BBHoss | D-m0nj0: you have to, unless the other pbx supports digital (PRI) |
| 14:03.14 | *** join/#asterisk oh207 (n=oh207@nylug/member/oh207) |
| 14:03.15 | riddlebox | the thing is it worked fine when I was using dd-wrt |
| 14:03.24 | xacatecas | D-m0nj0, you don't. but most phones are analog. |
| 14:03.28 | BBHoss | if it does support PRI, this will be much cheaper and easier |
| 14:03.34 | riddlebox | but dd-wrt's wireless crashed when I downloaded something |
| 14:03.36 | D-m0nj0 | i see |
| 14:03.39 | BBHoss | because you only need the card |
| 14:03.40 | xacatecas | BBHoss, other way round, he wants to put asterisk between PSTN and PBX. |
| 14:03.48 | D-m0nj0 | asterix talks digital right? |
| 14:03.59 | D-m0nj0 | so we need that card to convert from digital to analog |
| 14:04.09 | xacatecas | yes. |
| 14:04.15 | BBHoss | let me get this straight |
| 14:04.21 | BBHoss | can you re-state what you want to do? |
| 14:04.26 | D-m0nj0 | yes sure |
| 14:04.37 | D-m0nj0 | i have a running old PBX with extensions associated with it |
| 14:04.55 | D-m0nj0 | i would like to integrate asteriks in the network without doing again wiring |
| 14:04.59 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:05.00 | D-m0nj0 | so i want to keep the old pbx |
| 14:05.04 | D-m0nj0 | and just connect it to asterix |
| 14:05.16 | D-m0nj0 | same time i would like to keep the extensions associated with the old pbx |
| 14:05.24 | D-m0nj0 | is it possible? |
| 14:05.25 | BBHoss | why would you have to re-do wiring to swap PBXs? |
| 14:05.37 | D-m0nj0 | i dont mind if i readd the extensions on asterix asa long as it works |
| 14:06.00 | D-m0nj0 | its just that the guys payed a lot for the pbx and want to keep it for some dumb reasons |
| 14:06.01 | BBHoss | so you're ok with switching them out? |
| 14:06.08 | D-m0nj0 | no... |
| 14:06.11 | D-m0nj0 | sadly not |
| 14:06.14 | BBHoss | well thats kinda stupid :), but i understand |
| 14:06.18 | BBHoss | how much did they pay for it? |
| 14:06.25 | D-m0nj0 | i dont know |
| 14:06.26 | D-m0nj0 | but a lot |
| 14:06.28 | D-m0nj0 | ... |
| 14:06.53 | BBHoss | well its going to cost a good deal to do it like they want |
| 14:07.00 | D-m0nj0 | even if i want to connect the clients to asterix i will need to introduce new cards which the decision maker doesnt want |
| 14:07.01 | D-m0nj0 | :) |
| 14:07.03 | BBHoss | much easier to just swap them out |
| 14:07.23 | D-m0nj0 | coz i have like 50 wires |
| 14:07.24 | xacatecas | *sigh* ForkCDR doesn't quite do what I want ... I want T (which sets DONT_TOUCH) except I still want the end time to be updated when the channel gets hung up. |
| 14:07.26 | BBHoss | you will have to do cards either way, just depends on how much |
| 14:07.44 | D-m0nj0 | i see |
| 14:07.59 | BBHoss | D-m0nj0: the wires aren't a problem, you could easily plug them into a channel bank instead of the old PBX, which would then run into asterisk |
| 14:08.20 | D-m0nj0 | whats the cost of that bank do you have any idea? |
| 14:08.20 | BBHoss | then you could get a regular analog card for your PSTN phone service |
| 14:08.26 | BBHoss | look at that link |
| 14:08.33 | BBHoss | http://www.rhinoequipment.com/cb.html |
| 14:08.42 | D-m0nj0 | 2k |
| 14:08.45 | BBHoss | there are many manufacturers, thats just one i remembered |
| 14:09.05 | D-m0nj0 | its 1500USD |
| 14:09.10 | D-m0nj0 | they wont accept |
| 14:09.11 | D-m0nj0 | ... |
| 14:09.11 | BBHoss | do you know the exact specs of the old pbx? If it supports PRI, you can link them relatively easily |
| 14:09.28 | D-m0nj0 | i dont think it supports PRI |
| 14:09.43 | D-m0nj0 | if they support PRI i wont use a channel bank right? |
| 14:10.08 | BBHoss | right, you can just link them directly |
| 14:10.14 | BBHoss | over an RJ-45 cable |
| 14:10.41 | D-m0nj0 | yea but which chip i would use instead use? |
| 14:10.48 | D-m0nj0 | instead of the channel bankd |
| 14:10.51 | D-m0nj0 | bank* |
| 14:10.52 | BBHoss | heh |
| 14:11.02 | BBHoss | well its the same card, you just don't buy the channel bank |
| 14:11.14 | BBHoss | all the bank does it splits up the PRI into analog |
| 14:11.15 | D-m0nj0 | ah its like a feature ? |
| 14:11.26 | D-m0nj0 | ok i get it |
| 14:11.35 | BBHoss | a channel bank just plugs in via rj-45 or similar |
| 14:11.36 | D-m0nj0 | so i would be buying a bank but not using it |
| 14:11.44 | D-m0nj0 | :) |
| 14:11.59 | BBHoss | no, if your current pbx supports PRI, you don't NEED a bank? |
| 14:12.09 | BBHoss | whoops, that wasnt a question :) |
| 14:12.17 | *** join/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 14:12.30 | BBHoss | hang on i'll get you a link to what you need |
| 14:13.23 | [TK]D-Fender | If you're not using a T1 card on * for something else then channel banks ramp up your total cost for probably nothing |
| 14:14.09 | BBHoss | D-m0nj0: http://www.digium.com/en/products/digital/te120p.php or similar is what you would need if your pbx supports PRI |
| 14:14.46 | BBHoss | which costs just below $500USD |
| 14:14.54 | D-m0nj0 | 1 port/ |
| 14:14.55 | D-m0nj0 | ? |
| 14:15.23 | BBHoss | yeah, thats a PRI |
| 14:15.30 | BBHoss | it connects directly to the other PBX |
| 14:15.43 | D-m0nj0 | ok i see |
| 14:16.28 | [TK]D-Fender | BBHoss: Don't forget it isn't just if he want's PRI... its if he wants to USE that channelbank at all |
| 14:16.28 | D-m0nj0 | and if not :) i need somethin with 20 ports to connect my clients to * (swapping old pbx) |
| 14:16.28 | D-m0nj0 | what do i need in the latter case? |
| 14:16.36 | BBHoss | now for phone service, you would also need a POTS card, which is like a TDM410P |
| 14:16.57 | BBHoss | if it doesnt support PRI, then you would need a channel bank |
| 14:16.58 | D-m0nj0 | this one connects to the PSTN right? |
| 14:17.06 | BBHoss | yeah, the TDM series does |
| 14:17.16 | D-m0nj0 | this is crazy |
| 14:17.24 | [TK]D-Fender | You do NOT require a channel bank |
| 14:17.24 | BBHoss | well, actually, they both do, but in your case, you only have one POTS line, correct? |
| 14:17.34 | D-m0nj0 | yes |
| 14:17.38 | [TK]D-Fender | this IS crazy |
| 14:17.46 | D-m0nj0 | and another going to the intenet for VOIP |
| 14:17.52 | D-m0nj0 | lan cable |
| 14:17.52 | BBHoss | [TK]D-Fender: he is not for sure if the PBX supports PRI, if it doesn't, he will need one to link the two |
| 14:18.04 | BBHoss | so this is all about saving money with VOIP? |
| 14:18.09 | D-m0nj0 | yea |
| 14:18.12 | BBHoss | heh |
| 14:18.19 | BBHoss | and the current system doesnt do IP? |
| 14:18.22 | [TK]D-Fender | BBHoss: If he's not sure that isn't a reason to go buy something. You get off your ass and LOOK |
| 14:18.28 | D-m0nj0 | i think i will stick to skype :D |
| 14:18.29 | D-m0nj0 | lol |
| 14:18.40 | D-m0nj0 | no the current systems doesnt do ip |
| 14:18.50 | BBHoss | well hell, there is a much easier solution to this |
| 14:19.08 | BBHoss | how many trunk lines does the old PBX have? |
| 14:19.21 | D-m0nj0 | in the end of the day i want my clients who are using their phones to dial an internaltional number and get redirected to the VOIP network |
| 14:19.31 | BBHoss | correct |
| 14:19.32 | riddlebox | xacatecas, hrmm I can ping the ip address -p5060 and I get reply's |
| 14:19.49 | BBHoss | riddlebox: how are you pinging UDP? |
| 14:19.54 | D-m0nj0 | trunk lines? |
| 14:19.55 | xacatecas | hping. |
| 14:19.59 | BBHoss | ahh ok |
| 14:20.00 | xacatecas | hopefully. |
| 14:20.03 | BBHoss | just checking :) |
| 14:20.06 | xacatecas | if he gets replies the port's closed. |
| 14:20.06 | D-m0nj0 | i am not familiar with PBXs to be honest |
| 14:20.07 | riddlebox | BBHoss, crap thats true |
| 14:20.09 | D-m0nj0 | i am an IP guy |
| 14:20.14 | BBHoss | D-m0nj0: the lines that you connect to the phone company |
| 14:20.16 | xacatecas | probably receiving RSTs. |
| 14:20.22 | D-m0nj0 | 2 |
| 14:20.25 | BBHoss | hmm |
| 14:20.26 | D-m0nj0 | i ahve 2 |
| 14:20.32 | xacatecas | open ports won't generate replies unless the application actually responds to the packets. |
| 14:21.02 | BBHoss | so the current PBX only has TWO lines that connect to the phone company (two conversations at once) |
| 14:21.06 | riddlebox | xacatecas, on the router I have it set to forward both TCP and UDP as well |
| 14:21.14 | D-m0nj0 | yes |
| 14:21.27 | xacatecas | what about rtp? |
| 14:21.38 | xacatecas | SIP + NAT == a huge mess |
| 14:21.41 | BBHoss | D-m0nj0: and thats all the capacity it has, there are no empty ports that could be used later for phone service? |
| 14:21.46 | riddlebox | 10,000 to 20,000 is forwarded as well |
| 14:22.02 | xacatecas | and you've set up asterisk to only use those ports as rtp ports? |
| 14:22.07 | riddlebox | yup |
| 14:22.08 | D-m0nj0 | i have some free ports that can be used for internal extensions |
| 14:22.08 | [TK]D-Fender | xacatecas: hardly |
| 14:22.09 | BBHoss | riddlebox: make sure you set externip and localnet in sip.conf |
| 14:22.16 | [TK]D-Fender | ~sipnat |
| 14:22.32 | 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 |
| 14:22.33 | BBHoss | D-m0nj0: yeah, those are FXS ports, we need FXO ports, which from your information, i have gathered that you have two |
| 14:22.37 | [TK]D-Fender | ^^^^^^^^^^^ |
| 14:22.42 | riddlebox | BBHoss, it is, I used this same setup before, on dd-wrt and it worked but had to move back to linksys firmware and now its not working |
| 14:23.03 | xacatecas | [TK]D-Fender, he's trying to do SIP in front of NAT ... ie, his phone is on the "public" side and his asterisk on the private. |
| 14:23.18 | xacatecas | the other way round works usually, but I've had cases where I simply could not make it work. |
| 14:23.19 | [TK]D-Fender | xacatecas: Changes little |
| 14:23.20 | BBHoss | D-m0nj0: so your biz guys are ok with only having two concurrent conversations? |
| 14:23.36 | xacatecas | ok, my experiences with SIP + NAT has been extremely bad. |
| 14:23.39 | xacatecas | anyway, gotta run. |
| 14:24.03 | [TK]D-Fender | xacatecas: Doesn't matter if the phone is public or private, nor asterisk |
| 14:24.27 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:24.33 | riddlebox | [TK]D-Fender, I am thinking it has more to do with the router firmware than anything |
| 14:24.54 | [TK]D-Fender | riddlebox: A lot of talk, but I'm not seeing debug PB'd anywhere... |
| 14:25.08 | BBHoss | D-m0nj0: if they are, if I were you, I'd just buy a two port ATA and be done with it, then plug the trunk lines from the current pbx into the FXS lines on the ATA |
| 14:25.35 | BBHoss | and not even use asterisk, you really don't need it for what you want anyways |
| 14:25.37 | riddlebox | [TK]D-Fender, if you saw earlier I said I am not even seeing it hit the asterisk box to try to register |
| 14:25.47 | BBHoss | brb |
| 14:26.17 | [TK]D-Fender | riddlebox: check your forwarding, confirm your port (No router on this planet should fuck up PORT FORWARDING) and then check your server firewall. |
| 14:26.41 | [TK]D-Fender | riddlebox: Check your IP's (WAN and inside private to *) |
| 14:26.49 | BBHoss | back |
| 14:27.09 | riddlebox | [TK]D-Fender, the problem is f**king ekiga |
| 14:27.27 | riddlebox | tried with gizmo works right away |
| 14:27.29 | BBHoss | riddlebox: wow, i was thinking you were running asterisk ON dd-wrt :) |
| 14:27.47 | BBHoss | riddlebox: is ekiga on the same network as the * box |
| 14:27.52 | [TK]D-Fender | ridddYou're saying Ekiga... out ther on a PUBLIC IP has a problem sending packets to "just any IP also on the internet"? |
| 14:28.34 | riddlebox | BBHoss, nope it is off of the networks |
| 14:28.38 | riddlebox | network |
| 14:28.51 | D-m0nj0 | i have a serial console onthat PBX |
| 14:28.54 | D-m0nj0 | jsut figured it |
| 14:29.12 | D-m0nj0 | i dont know if it can be of any help |
| 14:29.12 | riddlebox | [TK]D-Fender, I have checked my config on it and it matches gizmo, and gizmo works |
| 14:29.17 | BBHoss | riddlebox: then make sure that your config in ekiga is right |
| 14:29.27 | riddlebox | BBHoss, see above :) |
| 14:29.29 | BBHoss | gizmo might be tunneling through http or something fun |
| 14:29.36 | [TK]D-Fender | riddlebox: This is usually where I chalk it up to PEBCAK |
| 14:29.44 | [TK]D-Fender | riddlebox: This is usually where I chalk it up to PEBKAC |
| 14:30.08 | riddlebox | [TK]D-Fender, thanks now I will be going to google yet again after you tell me something :) |
| 14:30.23 | riddlebox | [TK]D-Fender, easy! |
| 14:30.29 | BBHoss | riddlebox: make sure the port is forwarded to the EKIGA client, that you have in the settings (usually 5060), also make sure that ekiga is sending the correct public IP |
| 14:30.37 | BBHoss | D-m0nj0: what? |
| 14:30.49 | BBHoss | what kind of pbx is it D-m0nj0? avaya? |
| 14:30.54 | [TK]D-Fender | riddlebox: I'd trust things better if you showed TCPDUMPS from the Ekiga side and imagebin's of the config screens and prrof that * is where you think it is |
| 14:31.15 | D-m0nj0 | i think i will buy a channel bank, connect it to my * , buy a network card mount it on my * (for internet access0 and connect via FXO port to the PSTN |
| 14:31.19 | D-m0nj0 | is that correct ? |
| 14:31.23 | riddlebox | [TK]D-Fender, or I can just use gizmo |
| 14:31.52 | [TK]D-Fender | riddlebox: Feel free to write it off... |
| 14:32.08 | BBHoss | D-m0nj0: well hang on here |
| 14:32.15 | riddlebox | [TK]D-Fender, its not that big of a deal to me |
| 14:32.25 | [TK]D-Fender | D-m0nj0: Umm... your description seems kinda spotty and I'd seriously wnat to double-check your plan |
| 14:32.33 | BBHoss | D-m0nj0: if you only have two trunk lines, then you don't need a channel bank |
| 14:32.44 | [TK]D-Fender | D-m0nj0: To use a chanel-bank you'd need to buy a T1 card like the TE120 you were shown earlier as well |
| 14:32.58 | BBHoss | you only need a TDM410P, with 2 FXS modules and one FXO module |
| 14:33.31 | BBHoss | D-m0nj0: this is crazy though because you're going to have two full-blown PBXs, and it will cost you MORE money, and you will get LESS value out of it |
| 14:34.16 | BBHoss | if i were you i'd either swap the old pbx out with asterisk, or buy a cheap little 2-port FXS ATA, and link the ATA to your SIP provider |
| 14:34.31 | D-m0nj0 | its a panasonic |
| 14:34.31 | D-m0nj0 | BBHoss: i want to swap my old pbx |
| 14:34.32 | D-m0nj0 | with * |
| 14:34.32 | D-m0nj0 | i will convince them |
| 14:34.32 | D-m0nj0 | its a headache otherwise |
| 14:34.40 | BBHoss | indeed |
| 14:34.43 | BBHoss | good luck |
| 14:34.55 | BBHoss | how many extensions do you have internally now? |
| 14:35.15 | D-m0nj0 | 20 |
| 14:35.53 | D-m0nj0 | my concern is how to connect these 20 extensions to asterix |
| 14:36.01 | D-m0nj0 | other than using a channel bank |
| 14:36.25 | BBHoss | ok, then you'll need this: 1 Computer, with a network card of course, 1 PRI card (TE120P), 1 Channel Bank(Whatever brand you choose), and ONE FXO card (TDM410P) |
| 14:36.37 | D-m0nj0 | yep |
| 14:36.38 | BBHoss | the bank is the cheapest way to go |
| 14:36.39 | [TK]D-Fender | D-m0nj0: What model of Panasonic phones? They might be unusable with * |
| 14:36.44 | [TK]D-Fender | ^^^ |
| 14:36.45 | D-m0nj0 | the channel bank should have 25 ports |
| 14:36.46 | D-m0nj0 | righy? |
| 14:36.52 | BBHoss | oh god i forgot about that [TK]D-Fender |
| 14:37.01 | [TK]D-Fender | Holy crap. |
| 14:37.13 | D-m0nj0 | any compatibility matrix? |
| 14:37.39 | [TK]D-Fender | D-m0nj0: STOP. NOW. I think we should look at your situation from the beginning here because I'm seeing more holes than a block of Swiss cheese. |
| 14:37.50 | D-m0nj0 | lol |
| 14:38.06 | BBHoss | let me ask you this, is there ethernet at each phone, if the phones are incompatible you might as well do 100% IP |
| 14:38.15 | [TK]D-Fender | D-m0nj0: Campatibility is : if thsoe are 2 wire digital sets = NONE |
| 14:38.40 | riddlebox | D-m0nj0, what Panasonic system is it, and what model phones? |
| 14:38.41 | D-m0nj0 | i will check |
| 14:38.48 | D-m0nj0 | i will come back |
| 14:38.53 | D-m0nj0 | with an answer |
| 14:38.55 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 14:38.55 | BBHoss | heh, ok |
| 14:39.06 | D-m0nj0 | but really thank you guys |
| 14:39.11 | D-m0nj0 | you enlightened me |
| 14:39.40 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-549871cf1735ba8e) |
| 14:40.04 | BBHoss | poor soul |
| 14:40.31 | riddlebox | 20 phones, I wonder if he wanted to run them on the same network as the data traffic too |
| 14:40.37 | [TK]D-Fender | BBHoss: and CB's = costly for nothing. single SIP gateway, no T1 card req'd |
| 14:40.50 | BBHoss | if the only reason he wants * is to save money with VoIP, i can't see why he needs asterisk |
| 14:40.58 | BBHoss | [TK]D-Fender: i agree |
| 14:41.19 | BBHoss | thats what i was trying to tell him is that he just needed a simple ata/gateway to IP for his current system |
| 14:41.39 | [TK]D-Fender | BBHoss: So he wanted to replace his PBX's FXO's to go out VoIP? |
| 14:41.47 | BBHoss | yep |
| 14:41.52 | BBHoss | thats what i gathered |
| 14:42.00 | [TK]D-Fender | BBHoss: 25 analog lines? |
| 14:42.05 | BBHoss | yeah |
| 14:42.11 | *** join/#asterisk ITguru (n=ITGuru@5ac10610.bb.sky.com) |
| 14:42.17 | riddlebox | better have a good connection |
| 14:42.27 | BBHoss | riddlebox: he can only push 2 calls at once |
| 14:42.30 | [TK]D-Fender | BBHoss: AudioCodes MP-124 + Asterisk. Done. $1300 |
| 14:42.34 | BBHoss | due to the current PBX |
| 14:42.46 | [TK]D-Fender | BBHoss: What do you mean... 2 calls a t once? |
| 14:42.53 | [TK]D-Fender | BBHoss: You said 25 lines |
| 14:43.12 | [TK]D-Fender | BBHoss: Oh.. you're tlaking survivable BW? |
| 14:43.15 | BBHoss | he only has two trunk lines |
| 14:43.25 | BBHoss | he has 20 current FXS extensions to the phones |
| 14:43.44 | riddlebox | 2 trunks and 20 extensions hrmmm |
| 14:43.47 | [TK]D-Fender | BBHoss: ... OH. So he has 2 FXO and THAT has to go out VOIP? |
| 14:43.53 | BBHoss | yeah |
| 14:44.01 | BBHoss | the current pbx only has two fxo ports on it |
| 14:44.04 | [TK]D-Fender | BBHoss: Well WTF! Thats 1 stupid $50 ATA! |
| 14:44.07 | BBHoss | yeah |
| 14:44.10 | BBHoss | thats what i said |
| 14:44.19 | [TK]D-Fender | BBHoss: Why were you going on about channel banks and crap!? |
| 14:44.52 | BBHoss | [TK]D-Fender: well i was just giving him all the options, in case he didnt want asterisk JUST for voip savings |
| 14:44.57 | BBHoss | ie voicemail etc |
| 14:45.02 | [TK]D-Fender | BBHoss: EW! |
| 14:45.13 | BBHoss | but you cant do shit with 2 fxo ports anyways |
| 14:45.24 | [TK]D-Fender | BBHoss: the cost of T1 + channel bank is the cost of trashing his ENTIRE PBX for nice SIP phones |
| 14:45.28 | BBHoss | yep |
| 14:45.42 | BBHoss | thats what i'm gonna recommend if he comes back when i'm around |
| 14:45.47 | [TK]D-Fender | BBHoss: CB should never have been factored in unless he has no ethernet runs |
| 14:46.03 | BBHoss | [TK]D-Fender: he never replied to my question about that, so who knows |
| 14:46.24 | [TK]D-Fender | BBHoss: Should have pushed for the answer |
| 14:46.26 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 14:46.36 | BBHoss | shit someone fucked up google |
| 14:46.58 | BBHoss | google anything |
| 14:47.36 | dlewis | BBHoss: google works... |
| 14:47.48 | denon | he's talking about the "this site may harm your computer" stuff |
| 14:47.51 | denon | old news |
| 14:48.00 | BBHoss | yeah |
| 14:48.16 | *** part/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 14:48.51 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 14:49.35 | BBHoss | denon: old news is like 15 minutes ago? |
| 14:49.49 | dlewis | BBHoss: in the internet age, yes |
| 14:49.50 | dlewis | lol |
| 14:50.10 | denon | BBHoss: well, every irc channel I'm in has been talking about it .. |
| 14:50.21 | denon | so .. relatively old :) |
| 14:50.22 | dlewis | it seems to be a regional thing |
| 14:50.26 | BBHoss | denon: yeah i just picked it up in my client |
| 14:50.27 | dlewis | it doesn't say that for my friend |
| 14:51.01 | BBHoss | google is taking over |
| 14:51.13 | dlewis | already took over |
| 14:51.29 | denon | BBHoss: it's more fun to tell people "Dude, google 'Yahoo' .. they're giving other search engines a bad name!" |
| 14:51.39 | BBHoss | haha |
| 14:51.49 | BBHoss | i wonder if they could get a defamation lawsuit for this |
| 14:51.57 | denon | doubt it, since it covers everything |
| 14:52.06 | BBHoss | doesn't matter |
| 14:52.23 | dlewis | some illiterate computer user is going haywire right now |
| 14:52.24 | dlewis | lol |
| 14:52.26 | denon | well, sort of it does .. "Everything sucks" .. so can anyone sue me now? |
| 14:52.28 | BBHoss | just think about some 80 yo grandma that is looking for cookie recipies, and google tells her your site is bad |
| 14:52.44 | dlewis | exactly |
| 14:52.48 | dlewis | lawsuit? |
| 14:52.52 | riddlebox | hehehe |
| 14:52.52 | BBHoss | yep |
| 14:52.53 | dlewis | a stretch |
| 14:53.02 | BBHoss | aren't they all? |
| 14:53.14 | BBHoss | but this is AMERICA damnit! |
| 14:53.20 | denon | *rolls eyes* |
| 14:53.25 | denon | nobody's gettin sued |
| 14:53.50 | BBHoss | denon: you would really put it past today's america? |
| 14:53.50 | denon | goog doesn't charge for their search engine |
| 14:53.57 | BBHoss | doesn't matter |
| 14:54.05 | denon | and their paid ads are working |
| 14:54.35 | BBHoss | if i go out and start posting stuff thats false about digium, and it is REALLY false, then i'm being libelous |
| 14:54.45 | BBHoss | also defaming their name |
| 14:55.07 | denon | yes, but if you go out and say "ALL VOIP sucks" |
| 14:55.11 | denon | digium can't sue you |
| 14:55.12 | BBHoss | especially with the number of users google has |
| 14:55.17 | denon | and that's effectively what goog has done |
| 14:55.22 | BBHoss | denon: no, but thats not whats happening here |
| 14:55.32 | BBHoss | voip is a general area |
| 14:55.39 | denon | so is every single result on goog |
| 14:55.49 | [TK]D-Fender | VoIP doesn't suck... just the equipment you choose and the media you run it over :) |
| 14:55.51 | denon | so let me rephrase, "each and every voip solution in existance sucks" |
| 14:56.11 | BBHoss | when you click on a link to someone's page, and it says its bad, then obviously they are talking about that business's page |
| 14:56.29 | denon | BBHoss: actually, they say "may" harm your computer |
| 14:56.32 | denon | and their right, it may .. |
| 14:56.34 | BBHoss | remeber, lawsuits aren't supposed to be logical |
| 14:56.40 | denon | they don't need proof that it does to say it may |
| 14:56.52 | denon | any site may fill up the last 10k available on your hard drive |
| 14:56.54 | denon | and cause harm |
| 14:57.04 | keebler | Please tell me this Google thing isn't perm. |
| 14:57.13 | denon | haha |
| 14:57.25 | denon | Yahoo++ |
| 14:57.51 | [TK]D-Fender | denon: in that case : Life is a sexually transmitted disease which is in all cases fatal. If you're reading this then you're already fucked. |
| 14:58.01 | BBHoss | you can probably use google.co.uk |
| 14:58.04 | [TK]D-Fender | :) |
| 14:58.05 | denon | *rolls eyes* |
| 14:58.27 | BBHoss | nope |
| 14:58.29 | denon | actually, google.com.au is broken too |
| 14:58.30 | keebler | lol |
| 14:58.39 | BBHoss | must be based on geolocation |
| 14:58.56 | denon | prolly |
| 14:59.02 | BBHoss | i use cuil anyways, newbs |
| 14:59.18 | denon | away |
| 15:01.17 | BBHoss | just think of all of the tech support calls that people must be getting right now |
| 15:03.14 | *** join/#asterisk JoseBravo (i=JoseBrav@190.144.124.194) |
| 15:04.52 | JoseBravo | I have configured my asterisk with iaxmodem and hylafax. I can get faxes perfectly from one extension to another. But if the call comes from my TDM400 I get only the first part of the page or a full page but with very bad quality. Any idea? |
| 15:07.30 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 15:07.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:13.50 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 15:16.24 | BBHoss | here's a handly link to workaround the google problem, just make this a bookmark named google fix, and click it whenever you get the malware warning page "javascript:void((function(){location.href=location.href.replace(/^http\:\/\/www.google.com\/interstitial\?url=(.+)/,%22$1%22.replace(%22\:%22,%22.%22));})())"\ |
| 15:16.41 | BBHoss | without the quotes of course |
| 15:17.53 | JoseBravo | BBHoss, good |
| 15:18.25 | JoseBravo | BBHoss but I think google solved the problem. |
| 15:18.38 | BBHoss | JoseBravo: still intermittent for me |
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| 15:19.15 | JoseBravo | BBHoss ok |
| 15:24.40 | coppice | JoseBravo: depending on your motherboard and BIOS settings you may have serious data loss problems with the TDM400 |
| 15:30.46 | coppice | JoseBravo: some people have their gains set to crazy levels, so the received signal is in almost permanent overload. that distorts voice, and stops modems completely |
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| 15:56.11 | JoseBravo | coppice I need to share my line for voice and fax, what you recomend me to do? |
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| 16:00.55 | BBHoss | JoseBravo: you can detect faxes |
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| 16:30.46 | jaytee | morning ManxPower |
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| 16:36.23 | kerframil | TellarHK: yep |
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| 16:38.04 | ManxPower | 'morning |
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| 16:39.48 | ZenMasta | Good morning from the west coast. I am looking for some advice. I work for a small business that is considering abandoning its rented facility and allowing people to work from home |
| 16:40.21 | ZenMasta | they want to know how if possible, we can have a phone system where all the phone lines we have could ring at 2 locations at the same time, and we would still be allowed to transfer between the locations |
| 16:41.02 | Winkie | ZenMasta: do you mean two locations as in two individual phones? |
| 16:41.05 | ZenMasta | example, the owners home and another employees home, and if one person answers they could transfer or intercom to the other person etc |
| 16:41.15 | ZenMasta | no as in 2 literally different street addresses |
| 16:41.30 | ZenMasta | but each address may have more than one 4 line phone |
| 16:41.54 | Winkie | ZenMasta: that should be fine, there are some caveats as to how it is implemented but i can imagine what you want is possible |
| 16:42.04 | ZenMasta | currently our office uses regular phone lines and we have 6 offices each with a 4 line phone |
| 16:42.21 | Winkie | ZenMasta: why not set up asterisk as a test? the software is free, and there are free SIP phones for all operating systems |
| 16:43.02 | ZenMasta | I'd be interested in trying something like that but I'm not familiar with voip much. Can it do multiple locations like that? |
| 16:43.31 | Winkie | sure, but it depends on exactly what sort of rules you would apply, when calls come into a queue typically the strategies extend to ringing individuals in a certain order, or ringing everyone at once |
| 16:43.51 | Winkie | i haven't kept up with latest asterisk developments, but depending on what exact ringing rules you require it should be relatively easy |
| 16:43.56 | ManxPower | ZenMasta: You should read the Book |
| 16:43.59 | ManxPower | ~book |
| 16:44.09 | jbot | well, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 16:44.09 | ZenMasta | Winkie: and besides the hardware or internet connection, what kind of costs are associated with running asterisk. ie how do you pay your phone bill, and who do you pay for long distance and that sort of thing |
| 16:44.09 | ZenMasta | ManxPower: which book? |
| 16:44.30 | ManxPower | Just remember that sending calls across the Internet is as reliable as the Internet. |
| 16:44.53 | Winkie | ZenMasta: asterisk doesn't inherently have any affiliation for particular phone line providers |
| 16:44.58 | Winkie | ZenMasta: are you looking for only inter-office calls? |
| 16:45.08 | Winkie | or do you have external numbers you send to individual offices? |
| 16:46.16 | ZenMasta | Winkie: we have 4 lines right now, but we only ever give out our 800 number. When customers call it rings our first line, if busy it rolls over to the next line. we are not looking for only inter-office |
| 16:46.40 | ZenMasta | Winkie: Basically we are interested in allowing people to work from home but still be able to answer calls as if they were at a cubical |
| 16:46.51 | ManxPower | Asterisk isn't really a PBX. It is a toolkit that allows you to build a PBX |
| 16:47.06 | ZenMasta | Winkie: this way we save money on rent, people save money not commuting etc |
| 16:47.48 | ZenMasta | thanks for the link to that book I just downloaded it |
| 16:49.08 | Winkie | ZenMasta: in that case you would likely buy either an asterisk appliance type device, or buy a digium (or similar) card to put in a PC that runs asterisk |
| 16:49.34 | Winkie | do you know anything about the phone system you currently use? |
| 16:49.52 | ZenMasta | Winkie: yeah, it's just off the shelf 4 line phones, nothing special |
| 16:50.37 | ZenMasta | Winkie: we dont have voice mail or anything, just an answering machine. so I realize a system like asterisk has voice mail and music on hold, that is also appealing to me |
| 16:50.43 | Winkie | ZenMasta: i'm confused, you say you have 6 offices, each with a 4 line phone |
| 16:50.50 | Winkie | does each office have 4 lines? |
| 16:51.26 | Winkie | (and only one phone*) |
| 16:51.37 | ZenMasta | Winkie: yes the goal would be for each office to be a mirror of the other, we would all here whatever phone line is ringing, additionally if someone picked up the same line from the 2nd office they could listen in. Each office might have more than one phone |
| 16:52.08 | Winkie | ZenMasta: yeah you have quite a few options here, it's probably best for you to read the book and get a good feel for how asterisk works |
| 16:52.13 | ZenMasta | Winkie: all *hear*, yes spelling is awesome |
| 16:52.21 | Winkie | :) |
| 16:52.57 | ZenMasta | Winkie: does it go over what costs are involved? cause I know someone needs to be paid for something |
| 16:53.01 | Winkie | you can certainly transfer calls with asterisk, but you most likely don't want to buy 6 individual phone systems to go in each office |
| 16:53.16 | Winkie | ZenMasta: there are no charges for asterisk, the only charge you will come across is the cost of hardware |
| 16:53.24 | Winkie | and the details on that are widely available |
| 16:53.37 | ZenMasta | Winkie: then who do you pay for your phone numbers and long distance? that sounds crazy lol |
| 16:54.00 | Winkie | ZenMasta: you can pay whoever you currently pay, they provide you with phone lines and there are hardware cards which allow asterisk to connect to those phone lines |
| 16:54.16 | Winkie | ZenMasta: think of asterisk as the router on your internet connection, you can buy any router you like without having to change your ISP |
| 16:54.19 | Winkie | asterisk works in a similar manner |
| 16:54.50 | ZenMasta | Winkie: i see, very informative. |
| 16:55.04 | ZenMasta | Winkie: when you say phone system, I assume that is different than just a "phone" |
| 16:55.26 | Winkie | ZenMasta: indeed, the book should give you a good idea of how things work |
| 16:55.39 | ZenMasta | I appreciate it, thanks all |
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| 17:36.54 | kamh | hi all |
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| 17:55.09 | kannan | hello, fotr web based faxes is asterisk fax stable (1.4.21.2) or is it still recmmended for iam modem + hylafax? |
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| 17:55.54 | kannan | ~fax |
| 17:55.55 | jbot | i heard fax is The honor of designing the first fax *service* in actual use goes to Giovanni Caselli, an Italian abbot, born in Siena in 1815, who turned his hand to science and was, by 1849, editing a scientific magazine. In 1856 he claimed that he had developed a device, which he called a "pantelegraph," that could send facsimiles of images and text. Napoleon III did not come up with the idea, he merely backed it. |
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| 18:59.35 | mne | Hi. Is it possible to call SIP addresses (e.g. username@server) with asterisk ? So far I configured extensions so that each SIP user has a separate number (e.g. "5002"). |
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| 19:03.52 | GameGamer43 | mne: yes |
| 19:04.10 | mne | GameGamer43, how ? ;) |
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| 19:06.14 | mne | Do you have an example configuration or a link to some documentation that explains how one can set up asterisk so that one can call SIP addresses instead of extension numbers ? |
| 19:07.29 | rob0 | core show application dial |
| 19:08.51 | rob0 | Dial(SIP/username@server) ... in fact, that's how SIP extensions are made! |
| 19:09.33 | mne | yes, that's what I'm using here. Thus I have lines like "exten => 5001,1,Dial(SIP/username, 20)" |
| 19:10.14 | mne | Thus a SIP user can call the user "username" by dialing the extension "5001" |
| 19:10.22 | rob0 | if the @server is omitted, it looks for a match in sip.conf |
| 19:10.51 | mne | Hmm, so I could just call "username" ? |
| 19:11.24 | mne | Doesn't seem to work. It gives me a 404 not found |
| 19:12.26 | rob0 | A few hundred reasons why it might not. Wiki's outdated, but read the SIP config stuff there. |
| 19:13.53 | mne | the voip-info wiki ? |
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| 19:30.13 | mne | Hmm, I think I have to create alphanumeric extensions ? |
| 19:31.00 | mne | Thus something like "exten => username,1,Dial(SIP/username)" |
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| 19:44.07 | mne | Cool, that works :)) |
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| 19:52.36 | kannan | i read on the wiki, Digium itself says not remmended to use their cards with fax? is that hold true? |
| 19:53.11 | saftsack | which cards? and how do you build up your environment? |
| 19:54.57 | kannan | digium E1 card -> asterisk -> email |
| 19:55.03 | kannan | and vice versa |
| 19:55.16 | mvanbaak | you can use digium cards for that, no problem |
| 19:55.30 | mvanbaak | just make sure to not use echo cancel on fax |
| 19:55.39 | saftsack | thats true but whats about software? |
| 19:55.46 | kannan | is it still recommended to use iaxmodem with hylafax, or |
| 19:55.52 | kannan | any other good options |
| 19:56.04 | saftsack | eicon diva cards they have a true hardware tty interface |
| 19:56.29 | kannan | saftsack, thanks i will look those up |
| 19:57.00 | saftsack | but this cards are very expensive so you have to decide for yourself how important reliable and scalable faxing is |
| 19:57.21 | Greek-Boy | is there a way to permanently set the caller id of a user/phone from the dial plan? (caller is not just set for that call, I need to set the default caller ID for that SIP user or Phone) |
| 19:57.22 | kannan | hmm , ok want to try efax as a business |
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| 19:57.46 | saftsack | kannan, then look for eicon cards |
| 19:58.00 | kannan | digium E1 card -> fax machine with Linksys works fine , always , and vice versa |
| 19:58.09 | kannan | saftsack, i will |
| 19:59.06 | Greek-Boy | and while we're on the topic of faxing I recently attempted to install asterfax. It works for outgoing fax but incoming doesn't seem to work... |
| 19:59.37 | kannan | Greek-Boy, isnt that a commercial one? |
| 19:59.42 | kannan | for more than 1 channel |
| 19:59.54 | Greek-Boy | yeah |
| 20:00.24 | Greek-Boy | kannan have you setup hylafax successfuly? |
| 20:03.25 | kannan | Greek-Boy , yes |
| 20:03.29 | kannan | actually doing |
| 20:03.42 | Greek-Boy | how's it going? |
| 20:04.25 | kannan | the iaxmodems is fine, and passes it to faxrcvd |
| 20:04.33 | kannan | but i cant use avantfax , interface |
| 20:04.54 | kannan | or yet figured out how to do email routing based in Callerid appearanvce |
| 20:05.02 | kannan | havent tried to send out a fax yet |
| 20:06.07 | Greek-Boy | i c |
| 20:07.15 | mrsci | since we are on faxing, I am actually researching it now. Are there any good sip providers that support t.38? Is this still the best way to go with a plain old fax machine |
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| 20:12.51 | kannan | not very encouraging reading at -> http://www.soft-switch.org/foip.html |
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| 20:14.00 | TellarHK | I need to decide if today I'm going to clean my desk, or if I'm going to roll my own * install instead of using Trixbox. |
| 20:14.31 | rob0 | That sounds like an easy decision. |
| 20:14.44 | TellarHK | Not with my desk -this- bad. |
| 20:15.09 | rob0 | scrape it off into a box, then get to work |
| 20:15.21 | TellarHK | Hmm, there's a plan. |
| 20:15.26 | mrsci | that's how I clean my desk |
| 20:15.45 | mrsci | now there are stacks of boxes |
| 20:15.49 | rob0 | speaking of which, mine's about due too, but I have some * hacking calling me. |
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| 20:17.01 | Khratos | good afternoon |
| 20:21.13 | kannan | * version 1.6 seems to have fax support t.38 , any one tried this? |
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| 20:40.33 | JoseBravo | Why commands after rxfax dosnt execute? |
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| 20:42.06 | JoseBravo | I have a dialplan to get faxes to email. rxfax create the tif perfect, but Im doing a system later but it never executed |
| 20:42.35 | mvanbaak | JoseBravo: rxfax is like dial. it stops executing dailplan |
| 20:42.45 | mvanbaak | JoseBravo: use the h exten to mail it |
| 20:44.01 | JoseBravo | mvanbaak exten => h,4,system... ? |
| 20:44.29 | mvanbaak | exten => h,1,.... |
| 20:44.39 | JoseBravo | mvanbaak thank you |
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| 20:49.27 | JoseBravo | mvanbaak plesaee see my dialplan: http://www.pastebin.ca/1323854 |
| 20:49.37 | JoseBravo | But the system is never executed |
| 20:51.03 | kannan | Greek-Boy, i am going to go for 1.6 version rather than stay with hylafax |
| 20:51.19 | kannan | or rather try it first |
| 20:56.27 | kannan | does the span DSP need patching, i remeber reading quite some months ago? |
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| 21:03.48 | JoseBravo | mvanbaak are you there? |
| 21:04.02 | mvanbaak | sometimes |
| 21:05.16 | JoseBravo | mvanbaak I tried with the h exten. But rxfax is juming my system. |
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| 21:06.44 | mvanbaak | I dont know. we use iaxmodem + hylafax |
| 21:07.42 | JoseBravo | mvanbaak I tried to use it, it works from exten to hylafax. But passing the fax from my analog card the fax arrvied with very bad quality. |
| 21:08.35 | mvanbaak | yeah, analog is bad |
| 21:09.49 | *** join/#asterisk cvnet (n=dahitler@24.156.136.205) |
| 21:10.15 | cvnet | hello |
| 21:10.47 | cvnet | if i calls comes in SIP and you want to send it out via H.232 would the * translate it ? |
| 21:13.45 | cvnet | anyone awake? |
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| 21:14.57 | [TK]D-Fender | cvnet: Every call is just a call. |
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| 21:15.38 | cvnet | [TK]D-Fender: how do you use h.232 ? like in sip you do sip/IP/Number would it be h.232/ip/number ? |
| 21:15.55 | [TK]D-Fender | cvnet: h.232? |
| 21:15.58 | cvnet | ya |
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| 21:16.16 | cvnet | i got a test account set up with MCI but never used h.232 |
| 21:16.17 | [TK]D-Fender | cvnet: that looks a little backwards... |
| 21:16.45 | JoseBravo | [TK]D-Fender may you help me please? Im trying to get a fax to email. The rxfax receive good the fax and create the file. But I need to execute a system for send the email, but the rxfax skip the system. Any idea? |
| 21:16.48 | cvnet | is it h.323 ? |
| 21:16.56 | [TK]D-Fender | cvnet: what do YOU think? |
| 21:17.09 | [TK]D-Fender | JoseBravo: Here's an idea... PASTEBIN |
| 21:18.12 | cvnet | ya i just double checked its h.323 |
| 21:18.30 | cvnet | from * if u want to call using h.323 u do h.323/ip/numbe ? |
| 21:19.05 | [TK]D-Fender | cvnet: go read the channel driver config files. |
| 21:19.37 | JoseBravo | [TK]D-Fender http://www.pastebin.ca/1323873 |
| 21:19.41 | [TK]D-Fender | cvnet: Because it depends which one you're using. |
| 21:19.53 | kannan | what goes for 1.6 in place of zaptel.conf |
| 21:20.01 | [TK]D-Fender | kannan: DAHDI |
| 21:20.04 | [TK]D-Fender | ~dahdi |
| 21:20.04 | jbot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 21:20.24 | kannan | ty |
| 21:21.00 | [TK]D-Fender | JoseBravo: I don't see a problem in there... |
| 21:21.06 | JoseBravo | [TK]D-Fender It did the rxfax but never the system |
| 21:21.08 | Micc | Someone tried to register with my server using different passwords. They scanned my box and found the one extension between 1000 and 9999, then attempted a brute force password attack on that extension. |
| 21:21.16 | Micc | Is there any way to prevent this in the future? |
| 21:21.40 | Micc | Is there something that I can say only allow X number of registration attempts from one IP. |
| 21:21.40 | [TK]D-Fender | JoseBravo: that isn't a call, and that isn't a dilaplan dump for * CLI that I can prove was loaded or called like that bit you did just show me. |
| 21:22.00 | [TK]D-Fender | Micc: No |
| 21:22.14 | Micc | TK, does SER have something like that? |
| 21:22.40 | rob0 | with a firewall, only allow in the SIP or IAX peers you know. |
| 21:23.15 | kannan | [TK]D-Fender , may i know your opinion of ip faxing with 1.6 rather than with 1.4 (using hylafaxthru iaxmodem)? |
| 21:24.05 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 21:24.13 | JoseBravo | [TK]D-Fender its the output of my CLI: http://www.pastebin.ca/1323880 |
| 21:26.19 | [TK]D-Fender | kannan: I don't do T.38 |
| 21:26.28 | [TK]D-Fender | Micc: Quite possibly |
| 21:28.46 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 21:29.28 | [TK]D-Fender | JoseBravo: You lack understanding of how Macro's and cotext includes work |
| 21:29.31 | [TK]D-Fender | JoseBravo: -- Executing [h@from-internal:1] Macro("SIP/110-092854e8", "hangupcall") in new stack |
| 21:29.47 | [TK]D-Fender | JoseBravo: You didn't even see WHICH "h" was getting executed |
| 21:30.10 | [TK]D-Fender | JoseBravo: [h@from-internal:1] |
| 21:31.41 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 21:31.45 | cvnet | by default h323 is not compiled ha? |
| 21:32.06 | [TK]D-Fender | cvnet: Depends what version |
| 21:32.13 | cvnet | 1.4 |
| 21:32.25 | [TK]D-Fender | cvnet: Go look. |
| 21:32.39 | cvnet | im reading the readme file |
| 21:33.16 | [TK]D-Fender | cvnet: Worthless. Look in the "channel" folder in the source. do YOU see an h323 chann driver .c file? |
| 21:34.43 | cvnet | . /usr/src/asterisk-1.4.22/channels/chan_h323.c <--- thats the only c file i c |
| 21:35.07 | [TK]D-Fender | cvnet: then yes, * 1.4 comes with at least 1 H.323 channel driver options. |
| 21:35.22 | JoseBravo | [TK]D-Fender the problem is that the hangup interrupt my dialplan? |
| 21:35.26 | [TK]D-Fender | cvnet: that of course means there is a SAMPLE CONFIG for it |
| 21:36.06 | cvnet | then why [Jan 31 16:35:58] WARNING[12067]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'H323' (cause 66 - Channel not implemented) |
| 21:36.27 | [TK]D-Fender | JoseBravo: Actually....[macro-faxreceive] <- this isn't even being CALLED in your CLI output <- |
| 21:36.36 | [TK]D-Fender | JoseBravo: You are showing me 2 completely different things |
| 21:36.57 | JoseBravo | [TK]D-Fender yes, im sorry. |
| 21:37.29 | [TK]D-Fender | cvnet: is the channel driver loaded? Where is tis confi file? Whats in it? |
| 21:37.51 | [TK]D-Fender | cvnet: What format are you supposed to do your "dial" in? |
| 21:37.56 | cvnet | . /usr/src/asterisk-1.4.22/channels/chan_h323.c <-- thats the location of it |
| 21:38.04 | [TK]D-Fender | cvnet: NO |
| 21:38.27 | [TK]D-Fender | cvnet: Where is hte CONFIG FILE in your CONFIG FOLDER? Whats in it? |
| 21:38.49 | cvnet | which config file you referring to? |
| 21:39.00 | JoseBravo | [TK]D-Fender dialplan: http://www.pastebin.ca/1323890 output http://www.pastebin.ca/1323880 |
| 21:39.03 | [TK]D-Fender | cvnet: your H.323 channel driver config file |
| 21:39.47 | [TK]D-Fender | JoseBravo: -- Executing [h@from-internal:1] Macro("SIP/110-092854e8", "hangupcall") in new stack |
| 21:39.53 | [TK]D-Fender | JoseBravo: HUNG UP <---------------- |
| 21:39.54 | mrsci | micc, I know there is a program that limits the number of sip attempts, the name is escaping me. It would modify your iptables rules, to block the ip address where the attempt came from. I will try to find it |
| 21:39.57 | cvnet | [TK]D-Fender: i have no idea, whats the name of it h323.conf ? |
| 21:40.19 | [TK]D-Fender | cvnet: Go look in your SAMPLE CONFIGS. |
| 21:40.43 | cvnet | . /usr/src/asterisk-1.4.22/configs/h323.conf.sample <-- there it is |
| 21:41.11 | mrsci | micc, fail2ban should do the trick |
| 21:42.08 | JoseBravo | [TK]D-Fender and I can do somenthing in my dialplan to do the system after the hangup ? Its a trixbox and it have more Macro´s includes. |
| 21:42.14 | cvnet | h323.conf.sample has lots of commented out lines |
| 21:44.00 | jaytee | silly wabbit, trixboxs are for kids |
| 21:44.05 | [TK]D-Fender | JoseBravo: Go learn how macro's deal with overlapping extens ("h" in your case) |
| 21:44.24 | JoseBravo | OK TK, thank you. |
| 21:44.24 | *** join/#asterisk unpaidbill (n=bill@alteredbeastiality.org) |
| 21:44.34 | [TK]D-Fender | JoseBravo: And Trixbox is NOT SUPPORTED HERE |
| 21:44.47 | JoseBravo | Yes I know, sorry. |
| 21:44.52 | [TK]D-Fender | JoseBravo: the complimentary K-Y is by the door |
| 21:45.25 | rob0 | goes over to the door |
| 21:46.07 | jaytee | rofl |
| 21:46.19 | cvnet | [TK]D-Fender: /usr/src/asterisk-1.4.22/configs/h323.conf.sample got lots of lines, but all commented out, what do i do with it? |
| 21:46.29 | [TK]D-Fender | cvnet: Go read |
| 21:47.20 | jaytee | cvnet, a great man once said, "Necessity is the mother of invention." |
| 21:47.33 | rob0 | Frank Zappa? |
| 21:49.27 | denon | Plato |
| 21:49.33 | jaytee | no, Zappa said things like "Is that a real poncho or a Sears poncho", "Can I interest you in some zircon encrusted tweezers?" and everyone's favorite "Cuz you're an asshole, an asshole, an asshole so cram, cram, cram, cram it up your poopchute. Cuz your an asshole!" |
| 21:49.36 | Micc | mrsci, thanks. |
| 21:49.51 | cvnet | I'm reading h323.conf.sample, now once i change everything, do i keep it in /usr/src/asterisk-1.4.22/configs/ or move to a different folder? |
| 21:50.04 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 21:50.11 | aiksa[LV] | hello everybody |
| 21:50.22 | [TK]D-Fender | cvnet: where do * config files normally go? |
| 21:50.35 | cvnet | in /etc/asterisk/ ? |
| 21:50.39 | Micc | mrsci, will fail2ban scan my asterisk messages log? |
| 21:50.46 | rob0 | If they lived a good life, they go to Heaven. |
| 21:50.58 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 21:50.58 | rob0 | along with all dogs, I guess |
| 21:51.15 | aiksa[LV] | does everyone know if backbort of new AMI funcionality to 1.4 exist? |
| 21:51.22 | aiksa[LV] | anyone, sorry |
| 21:51.40 | [TK]D-Fender | jaytee: FYI.... I'm at the office, have been since 11am, and Novell = BAI BAI :) |
| 21:51.51 | [TK]D-Fender | aiksa[LV]: Very unlikely |
| 21:51.55 | jaytee | [TK]D-Fender, way cool!!! |
| 21:52.01 | cvnet | in /etc/asterisk/ i already ahve a h323 file, so i just edit that, correct ? |
| 21:52.15 | [TK]D-Fender | jaytee: Just stripped the client off all the PC's and am jsut cleaning up stuff now. |
| 21:52.16 | jaytee | [TK]D-Fender, about Novell I mean, not about you having to be at the office :-) |
| 21:52.23 | aiksa[LV] | [TK]D-Fender: because it was related to a number of other changes? |
| 21:52.28 | rob0 | "make samples" to the rescue |
| 21:52.29 | [TK]D-Fender | jaytee: s'ok... this is good. |
| 21:52.38 | aiksa[LV] | and would technically be near to impossible to backport? |
| 21:52.59 | [TK]D-Fender | jaytee: shaves almost 2H off my nightly backup, makes daily performance so much higher and consolidates on a Mac-friendly server platform. |
| 21:53.53 | aiksa[LV] | I would really need to have transfer event in manager interface, but as I understand I can forget about it under 1.4 |
| 21:54.51 | aiksa[LV] | so the logical question in seq. would be - how stable is 1.6 ? |
| 21:54.58 | [TK]D-Fender | aiksa[LV]: Yup, thats like saying you'll backport... well.. all of 1.6. May as well install 1.6 and slap an "* 1.4" sticker on it |
| 21:55.12 | aiksa[LV] | [TK]D-Fender: :))) |
| 21:55.19 | [TK]D-Fender | aiksa[LV]: VERY stable. Especially when it comes to a complete stop, then NOTHING budges it |
| 21:55.40 | aiksa[LV] | :) |
| 21:55.49 | [TK]D-Fender | aiksa[LV]: Honest "word of mouth" = pretty good. |
| 21:56.25 | [TK]D-Fender | aiksa[LV]: a few hangups like every other version, not not at all "plagued" like 1.2 / 1.4 were at the start. Matured uicker. |
| 21:56.32 | [TK]D-Fender | quicker* |
| 21:56.48 | aiksa[LV] | [TK]D-Fender: ok. I guess I will install it on the test box, and give it a try. |
| 21:57.15 | aiksa[LV] | The only problem is - much of the problems with early 1.4 displayed themselves at higher loads |
| 21:57.17 | [TK]D-Fender | aiksa[LV]: read up on the changes.... |
| 21:57.28 | aiksa[LV] | [TK]D-Fender: I surely will. |
| 21:58.00 | aiksa[LV] | As a matter of fact I have done some reading upo. Mostly on the functionality I was missing before |
| 21:59.02 | jaytee | yeah, early 1.6 versions are light years more stable than early 1.4 version were. I don't think 1.4 really "quieted down and quit being so cranky" until 1.4.11 or so. |
| 21:59.08 | aiksa[LV] | I really like the whole concept of joining all Bridge type events under same roof. And of course - Transfer event , thing I have been craving for for a long time. |
| 22:02.01 | aiksa[LV] | oh. just forgot. gotta ping tzafrir_laptop regarding 1.6 and xpd_usb support there |
| 22:02.08 | [TK]D-Fender | wonders how long until CentOS updates to 5.3 |
| 22:03.40 | [TK]D-Fender | jaytee: Oh, in other news I grabbed a .deb for Lotus Notes 8.5 Beta 2 and installed it at home... I should f'n terrorize my US IT dept with a screen-shot showing my SSH'd into the office and tunningling my LN client through, accessing their AS/400 and cruising our file shares off my home Ubuntu box :) |
| 22:03.57 | [TK]D-Fender | jaytee: They don't know Linux for shit :p |
| 22:04.42 | jaytee | [TK]D-Fender, hahahaa. Just don't get canned! Economy sucks ass lately. |
| 22:04.55 | aiksa[LV] | [TK]D-Fender: before I started my current job I though AS/400 were long time dead. |
| 22:05.09 | aiksa[LV] | thought* |
| 22:05.10 | [TK]D-Fender | jaytee: Canned? We've converted... if I'm gone, who'll support it down there? They'd be shooting themselves int he foot :) |
| 22:05.28 | jaytee | Mark Spencer has an AS400 terminal keyboard on his desk at Digium, or at least it looks like one. |
| 22:05.29 | [TK]D-Fender | jaytee: And my office supportd this conversion 100% :) |
| 22:05.50 | [TK]D-Fender | jaytee: My bosses wanted progress, and we just got it... |
| 22:06.41 | [TK]D-Fender | jaytee: I'd been looking to ditch it for 2.5 years...... I'm clearly "rushing" things! |
| 22:07.30 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 22:07.39 | jaytee | [TK]D-Fender, who is saying you're rushing things? |
| 22:07.48 | [TK]D-Fender | jaytee: they will :) |
| 22:08.13 | [TK]D-Fender | jaytee: It's all good... they'll see I'm "interested" in Lotus Notes and take that as the good with the bad :) |
| 22:08.26 | [TK]D-Fender | jaytee: I (and the rest of us) LOATH it. |
| 22:08.45 | [TK]D-Fender | jaytee: I'm just proving superior platform interop. |
| 22:10.38 | jaytee | [TK]D-Fender, later on or tomorrow if you have the time I'd like you to take a look at a pastebin of a failed call with SIP debug, sip.conf and dialplan stuff. I've been trying to set up a SIP "trunk" between two asterisk servers and I'm running into a problem with authentication. It's been awhile since the class :-) |
| 22:10.54 | [TK]D-Fender | jaytee: Sure |
| 22:11.15 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 22:11.49 | jaytee | it's not an urgent thing. I'm just looking at possibly replacing an IAX trunk I have working as bandwidth isn't a problem and calls through the trunk sometimes get hungup for no apparent reason and debug shows zilch. |
| 22:13.13 | aiksa[LV] | jaytee: the most obvious reason could be - that asterisk is forwarding original SIP auth. it received from originating user agent. |
| 22:13.29 | jaytee | aiksa[LV], that's the case exactly |
| 22:13.54 | jaytee | I think I need fromuser= in my sip.conf for that server's connection. |
| 22:14.36 | jaytee | it's seeing my account in the From and the server's account in the digest |
| 22:15.07 | aiksa[LV] | let me remember, I had some fights with this issue a while ago. I guess there was an option in the peer/user definitions in sip.conf which cured this |
| 22:15.25 | jaytee | fromuser= ? |
| 22:15.44 | aiksa[LV] | i think it was something else. |
| 22:16.04 | aiksa[LV] | Let me check if I can access that server from my home computer. |
| 22:16.50 | *** join/#asterisk ITguru (n=ITGuru@5ac10610.bb.sky.com) |
| 22:17.33 | *** join/#asterisk Mog (n=mog@c-68-62-170-174.hsd1.al.comcast.net) |
| 22:17.33 | *** mode/#asterisk [+o Mog] by ChanServ |
| 22:18.07 | aiksa[LV] | fck. I cant. And dont have conf. backup here either. |
| 22:18.40 | aiksa[LV] | I know i stumbled on this when trying to get callweaver and asterisk to play nicely togeather on the same box. |
| 22:19.13 | aiksa[LV] | jaytee: if I remember how I got around this, I will tell you. |
| 22:19.32 | jaytee | aiksa[LV], thanks |
| 22:19.34 | *** join/#asterisk ITguru (n=ITGuru@5ac10610.bb.sky.com) |
| 22:25.42 | aiksa[LV] | jaytee: looking at the wiki it loioks like fromuser could be the option you need. |
| 22:25.59 | aiksa[LV] | but I somehow have a memories of diiferent method applied in my case. |
| 22:26.58 | jaytee | aiksa[LV], I'll mess with it and if I'm still stuck I'll pester [TK]D-Fender for help tomorrow or sometime. If you remember and are in channel during the next few days give me a shout out. |
| 22:27.36 | [TK]D-Fender | jaytee: I promise I'll forget... just surprise me with it later :) |
| 22:28.07 | jaytee | [TK]D-Fender, hehehe, rest assured. |
| 22:29.39 | aiksa[LV] | jaytee: latest the monady (when back in the office) I will have access to that box and can check on actual example |
| 22:30.18 | jaytee | aiksa[LV], that's cool. I'm usually in channel most of the time. I don't login till around 9 EST in the US |
| 22:35.35 | *** join/#asterisk docelmo (n=vircuser@pool-151-199-178-15.lyn.east.verizon.net) |
| 22:36.18 | *** join/#asterisk D3b|4n (n=D3b_4n@unaffiliated/lynxnica) |
| 22:36.31 | *** join/#asterisk adnc (n=adnc@unaffiliated/adnc) |
| 22:38.02 | riddlebox | does anyone use teliax? |
| 22:38.59 | doug | yup |
| 22:39.07 | doug | someone does |
| 22:39.49 | beek | riddlebox: Just dumped then and went to Vitelity |
| 22:39.55 | beek | s/then/them/ |
| 22:40.10 | riddlebox | beek, did they display your callerid? |
| 22:40.20 | doug | name, or number? |
| 22:40.25 | beek | riddlebox: On outgoing calls? yes |
| 22:40.46 | riddlebox | beek, I just signed up last week for them and it doesnt display callerid on outgoing calls for me |
| 22:40.56 | path_ | exten => s,1,GotoIf($["${VMCOUNT(${ARG1})}" != "0"]?3) |
| 22:41.01 | doug | beek: were you setting that on your client, or via teliax's interface? |
| 22:41.02 | path_ | what 's' stands for there ? |
| 22:41.12 | beek | You set your callerid on Asterisk? |
| 22:41.17 | beek | Or you set it at their site? |
| 22:41.23 | path_ | s -> catch all your calls ? |
| 22:41.35 | [TK]D-Fender | path_: No. |
| 22:41.38 | beek | doug: I set mine myself. |
| 22:41.40 | [TK]D-Fender | ~stdextens |
| 22:41.59 | jbot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 22:42.00 | beek | doug: I did that so that calls coming on my POTS would ring my cell phone and have the correct CallerID |
| 22:42.06 | riddlebox | beek, how did you set it? |
| 22:42.32 | riddlebox | I set it in sip.conf but it still isnt passing it |
| 22:42.53 | beek | riddlebox: Set(CALLERID(all)=... |
| 22:42.56 | D3b|4n | i have big problem no can call |
| 22:43.02 | beek | :q |
| 22:43.03 | D3b|4n | http://pastebin.com/m5f5df81e |
| 22:43.10 | D3b|4n | <PROTECTED> |
| 22:43.36 | riddlebox | beek, in extensions,conf I guess |
| 22:43.42 | beek | Yes |
| 22:43.47 | beek | Just before I place the call. |
| 22:44.23 | D3b|4n | ? |
| 22:45.18 | doug | beek: why'd you dump them? |
| 22:45.32 | [TK]D-Fender | D3b|4n: please provide actual CLI output for your failed call with SIP DEBUG enabled. Logs = WORTHELESS |
| 22:45.47 | thehar | ztcfg |
| 22:45.50 | thehar | er, hi. |
| 22:45.55 | beek | doug: Because they went to freeswitch and in doing so managed to totally screw up. I couldn't get decent DTMF recognition, thus tutally screwing up my use of an IVR |
| 22:46.14 | eppigy | NEIN |
| 22:46.14 | beek | If I sent 1234567890 to my pager, I'd receive 11233445667789900 |
| 22:46.32 | D3b|4n | the debug is active |
| 22:46.43 | doug | nice |
| 22:46.51 | beek | doug: They weren't very reponsive to the problem and I couldn't wait any longer. |
| 22:47.02 | [TK]D-Fender | eppigy: TRABAJO |
| 22:47.58 | drmessano | TRAJABO |
| 22:48.13 | drmessano | EL CHUPACABRA |
| 22:48.52 | doug | beek: have you had a chance to test out vitelity's responsiveness? |
| 22:49.27 | beek | doug: I sent a question about their service and the response was within a few hours. Other than that things have worked fine and I have had no issues. |
| 22:49.29 | path_ | ok, this is mi issue: I need to know how to assign an extension to a proper SIP account |
| 22:49.33 | doug | nice |
| 22:49.35 | path_ | where should I start ? |
| 22:49.38 | doug | i'll givvem a try. |
| 22:49.40 | D3b|4n | ? |
| 22:49.43 | path_ | s/mi/my |
| 22:50.07 | path_ | Inbound calls are falling to the same extensions, thats my main problem |
| 22:50.12 | [TK]D-Fender | path_: by clarifying your definition of "assign an extension". |
| 22:50.27 | [TK]D-Fender | path_: Pastbin it and your dialplan and we'll see |
| 22:52.25 | path_ | [TK]D-Fender, http://pastebin.com/d47d521ee |
| 22:52.30 | riddlebox | beek, ok got it working, is there a way to get the name to come across, all I got was number |
| 22:52.55 | riddlebox | I tried Set(CALLERID(all)="My Name"<number>) |
| 22:53.13 | beek | riddlebox: Coming across to where? |
| 22:53.22 | beek | riddlebox: What kind of number are you calling? |
| 22:53.28 | beek | e.g. POTS, etc... |
| 22:53.31 | riddlebox | beek, I am calling my other land line |
| 22:53.33 | riddlebox | POTS |
| 22:53.54 | beek | The CallerID name lookup is done by the telco who is delivering the call to you. |
| 22:53.58 | [TK]D-Fender | path_: I asked for debug.... |
| 22:54.20 | path_ | ok, doing the asterisk -r |
| 22:54.33 | riddlebox | beek, I got forbidden |
| 22:54.37 | riddlebox | in the cli |
| 22:54.52 | beek | riddlebox: when you did what? Set the callerID information? |
| 22:55.25 | riddlebox | I set it and then made a call out of asterisk, and back into asterisk so I could see the info in cli |
| 22:55.54 | beek | riddlebox: Pastebin the console output |
| 22:56.01 | D3b|4n | nothing help |
| 22:56.49 | beek | riddlebox: and while your at it, pastebin the relavent part of your dialplan. |
| 22:58.30 | riddlebox | beek, http://pastebin.ca/1323946 |
| 23:00.13 | beek | riddlebox: Put it in the format: "Your name" < 8885551212 > |
| 23:00.28 | beek | riddlebox: Make sure the spaces are there |
| 23:01.42 | riddlebox | beek, same thing |
| 23:02.37 | beek | riddlebox: If you don't set this the call completes okay? |
| 23:04.40 | riddlebox | yup |
| 23:05.19 | beek | riddlebox: And you're receiving NO caller ID information or the number only without setting this? |
| 23:05.32 | riddlebox | beek, comes up unavailable |
| 23:05.43 | beek | riddlebox: Please pastebin the console output on a completed call |
| 23:06.05 | dlewis | riddlebox: what version of asterisk do you use? |
| 23:06.22 | riddlebox | dlewis, 1.4.22.1 |
| 23:07.07 | dlewis | ok |
| 23:07.51 | riddlebox | should I maybe try to break it down to Set(CALLERID(name)=<name>)? |
| 23:08.11 | beek | riddlebox: give it a try... |
| 23:08.28 | riddlebox | I will have to tomorrow, I gotta get ready to go to a friends for chili |
| 23:08.41 | riddlebox | thanks for getting me started though |
| 23:09.06 | beek | riddlebox: Enjoy the indigestion. |
| 23:09.10 | riddlebox | btw I cant believe that teliax lets you set this yourself, it seems like it could cause problems |
| 23:09.29 | beek | riddlebox: Most VoIP providers do. |
| 23:09.46 | beek | I hope that they never change that as I rely on it to pass on incoming callerid to my cell phone. |
| 23:09.51 | riddlebox | broadvoice, didnt, but I am not happy with them so I will probably never use them again |
| 23:10.23 | [TK]D-Fender | [17:55]<D3b|4n>nothing help [18:03]|<--D3b|4n has left irc.freenode.net () <--- yup, retard |
| 23:10.28 | [TK]D-Fender | NEXT!@!!!@@! (c) BKW |
| 23:10.29 | riddlebox | I just got a letter today from AT&T, I can get a line from them for $16 a month |
| 23:11.02 | beek | [TK]D-Fender: evening [TK]D-Fender |
| 23:11.10 | riddlebox | using charter right now, its like 42 a month, hence the move to teliax |
| 23:11.14 | [TK]D-Fender | beek: y0 |
| 23:11.38 | beek | riddlebox: I can see why you'd be interested. |
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