| 00:01.38 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 00:03.07 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 00:06.52 | *** join/#asterisk ruied (n=ruied@bl9-255-200.dsl.telepac.pt) |
| 00:07.55 | *** join/#asterisk WindBack (n=Jorge@host209.190-136-228.telecom.net.ar) |
| 00:08.00 | WindBack | Hello |
| 00:10.42 | WindBack | I have a question, I'm trying to store the date and time that a call started and ended in a database. I'm using CDR(start) and CDR(end), when i see the value of CDR(end) is null, what can it hapen to me?' |
| 00:16.27 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 00:20.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:22.30 | *** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner) |
| 00:24.04 | Corydon76-dig | WindBack: are you using the option in cdr.conf to end the call prior to the execution of the "h" extension? |
| 00:24.24 | Corydon76-dig | WindBack: because the call is usually not ended at that time |
| 00:24.30 | WindBack | Corydon76-dig, ahhhh |
| 00:25.20 | WindBack | Corydon76-dig, what is the "h" extension?? |
| 00:25.46 | Corydon76-dig | WindBack: if it exists, dialplan execution continues at that extension when a hangup occurs |
| 00:25.49 | SwK | someone remind m |
| 00:25.50 | SwK | e |
| 00:26.05 | SwK | so dtmf debug in logger.conf include inband? |
| 00:26.13 | SwK | in relation to sip calls? |
| 00:26.36 | SwK | (and no I dont have the option of changing from in band) |
| 00:26.48 | Corydon76-dig | Yes, it should. |
| 00:27.12 | WindBack | Corydon76-dig, ahhh, ok so I have to store this values in the h extension |
| 00:27.22 | WindBack | Corydon76-dig, ok? |
| 00:27.57 | WindBack | Corydon76-dig, and I have to had this option enable in cdr.conf |
| 00:28.59 | Corydon76-dig | WindBack: You can't directly set CDR(end) |
| 00:29.13 | WindBack | Corydon76-dig, yes i know that |
| 00:29.31 | WindBack | Corydon76-dig, this functions (variables ) is read only |
| 00:29.44 | Corydon76-dig | Why aren't you using the CDR engine to store your data? |
| 00:29.56 | WindBack | Corydon76-dig, nop |
| 00:30.15 | Corydon76-dig | "Why" |
| 00:30.15 | WindBack | Corydon76-dig, I'm using a script in python with agi |
| 00:31.16 | *** join/#asterisk cleone (n=cleo@41.251.47.190) |
| 00:31.44 | WindBack | Corydon76-dig, because i Need to store more than the data that the engine store |
| 00:32.21 | WindBack | Corydon76-dig, And I need to have a diferent structure in the table |
| 00:32.25 | *** join/#asterisk dddbliss (i=639a6602@gateway/web/ajax/mibbit.com/x-8de65cd191d91791) |
| 00:33.48 | dddbliss | hi, i am having a problem getting this asterisk appliance box working, does anyone know how to enable debugging for digium asterisk appliances |
| 00:33.50 | dddbliss | ? |
| 00:34.05 | Corydon76-dig | WindBack: you might be interested in cdr_adaptive_odbc |
| 00:34.06 | WindBack | Corydon76-dig, so.. I need to pass CDR(end), CDR(start), CDR(duration), etc as arguments of my script |
| 00:34.31 | Corydon76-dig | http://svncommunity.digium.com/svn/tilghman/branches/1.4/ |
| 00:34.51 | Corydon76-dig | or http://svncommunity.digium.com/view/tilghman/branches/1.4/ |
| 00:35.32 | Corydon76-dig | dddbliss: please call support |
| 00:36.08 | dddbliss | i would if I had the information to register the thing.. my boss tossed the support subscription code |
| 00:36.11 | mvsn | <PROTECTED> |
| 00:36.21 | *** join/#asterisk beernuts (n=mattfox7@CPE-124-186-159-162.qld.bigpond.net.au) |
| 00:36.39 | nny_1 | any cisco 79x0 users here? |
| 00:36.44 | WindBack | Corydon76-dig, it is interesting |
| 00:37.19 | Corydon76-dig | Not just interesting. It is specifically what you're looking for |
| 00:37.50 | beernuts | Hello all, I have a question regarding Musiconhold for agents - how can i disable it? If i put in a bogus class it defaults to the default in moh.conf. Any ideas? |
| 00:38.06 | nny_1 | anyone wanna hit me in the junk for working with Cisco phones? |
| 00:38.56 | beernuts | I can put in a silence file in the class but there is a faint annoying buzz in the background |
| 00:39.28 | WindBack | Corydon76-dig, and... the extension h, where I have to put it?? |
| 00:39.54 | Corydon76-dig | WindBack: in the same context as the last extension which executed |
| 00:40.04 | *** part/#asterisk dddbliss (i=639a6602@gateway/web/ajax/mibbit.com/x-8de65cd191d91791) |
| 00:41.35 | WindBack | Corydon76-dig, There not exists somethings like exten => s-HANGUP like (exten => s-BUSY) instead the h extension?? |
| 00:41.51 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-42b6c0a401b7c5a1) |
| 00:41.55 | WindBack | Corydon76-dig, to use in a macro |
| 00:41.56 | WindBack | ? |
| 00:42.29 | *** join/#asterisk robbyb___ (n=rbartlet@203.63.126.9) |
| 00:44.27 | robbyb___ | Hi all, having a really strange intermittant fault with asterisk queues. Seemingly randomly, the queue app crashes (I think), when i do show queues on the console in one of these events it just returns a blank line. have searched, but havent found anything that seems relevant. does this sound familiar to anyway? able to point me in the right direction? |
| 00:44.58 | nny_1 | so i have a phone here, if i put the wrong user name in, asterisk says "foo@192.168.100.10" not found, but if I replace foo with the correct name, it doesn't register, nothing. sip show peers says UNREACHABLE, but if I try to dial out from the phone I get registration errors. All in all it's obvious the phone isn't sip registered, but it is trying. It is a cisco 7940. besides suicide, any advice? |
| 00:45.21 | seanbright | suicide? |
| 00:45.26 | seanbright | oh wait, you already said that. |
| 00:47.57 | nny_1 | ahh |
| 00:48.04 | nny_1 | may be the config for which codec |
| 00:48.14 | SwK | Corydon76-dig, any reason it would show up at all? (we're obviously getting some DTMF but the menus are firing wrong and the dialplan looks correct I think the carrier is screwing me over |
| 00:48.58 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:49.28 | Corydon76-dig | WindBack: Macros exit when a Hangup event occurs |
| 00:50.30 | Corydon76-dig | SwK: likely cause is that the packet path is interrupted, causing some DTMF to either not show up or be duplicated |
| 00:50.56 | Corydon76-dig | SwK: that's a very common problem with inband DTMF, especially over the wild and wooly Internet |
| 00:51.40 | Corydon76-dig | That's precisely why OOB DTMF was created |
| 00:52.24 | *** join/#asterisk really_phukt (n=chatzill@209.216.64.44) |
| 00:53.50 | *** join/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 00:54.00 | DrkShadow | Is there an easy way to log callerid information? |
| 00:56.00 | DrkShadow | *sigh* I think my IRC is broken again. |
| 00:56.07 | husimon | why? |
| 00:56.11 | husimon | you are typing on it |
| 00:56.22 | husimon | it's obviously working |
| 00:56.58 | DrkShadow | husimon: sometimes when I sign on to freenode, I can do things.. but I don't get any text back.. then a couple min later it reconnects on me. Obviously not now, but it is _very_ frequent. |
| 00:57.13 | husimon | what irc client are you using? |
| 00:57.20 | DrkShadow | chatzilla :-) |
| 00:57.31 | husimon | what os |
| 00:57.42 | husimon | could try a different client |
| 00:57.48 | husimon | after that it's a networking issue |
| 00:57.57 | DrkShadow | not concerned about the client. It doesn't matter to me that much. |
| 00:58.59 | DrkShadow | at the moment, only concerned with callerid logging.. |
| 01:00.55 | *** part/#asterisk drue (n=drue@stiff.therub.org) |
| 01:05.06 | nny_1 | getting 401 Unauthorized from a cisco phone, username and pass match, any advice? |
| 01:05.23 | seanbright | DrkShadow: AGI? |
| 01:05.45 | DrkShadow | sean: sure, whatever easy way there is to log caller ID info, I'm willing to consider it. |
| 01:05.54 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 01:06.11 | seanbright | DrkShadow: oh... easy... hmmm |
| 01:06.19 | DrkShadow | pre-made is pretty easy, too. |
| 01:07.10 | seanbright | i wonder if something as simple as System(echo ${CALLERID(num)} > logfile.txt) would work |
| 01:07.20 | seanbright | i'm guessing no |
| 01:07.25 | DrkShadow | that... might be a bit too simple ;-) |
| 01:07.52 | seanbright | well do you know perl or php? |
| 01:08.03 | DrkShadow | yeah |
| 01:08.04 | seanbright | or python |
| 01:08.22 | seanbright | ok, well than write a simple PHP agi using phpagi |
| 01:08.30 | seanbright | then just call it from the dialplan |
| 01:09.35 | nny_1 | husimon: can you offer any advie as to why I would be gettiing 401 from the cisco? |
| 01:13.08 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 01:14.14 | husimon | explain? |
| 01:14.23 | husimon | 401 where |
| 01:14.33 | husimon | brb |
| 01:16.37 | nny_1 | husimon: i have the phone updated. When I do a sip debug I get SIP/2.0 401 Unauthorized. The phone itself says "REGISTERING" and "REG auth failed". My username and pass match in SIP<MAC>.cnf and sip.conf, and I have duoble checked with sip show users and show config on the cisco... not sure what else would be causing it to fail reg |
| 01:21.37 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.213) |
| 01:23.04 | *** join/#asterisk Katty (n=The@adsl-71-153-49-37.dsl.stlsmo.sbcglobal.net) |
| 01:23.48 | Katty | o hai |
| 01:23.56 | Katty | i upgraded your pbx |
| 01:24.00 | nny_1 | lol |
| 01:24.24 | Nivex | i can haz chan_mobile? |
| 01:26.25 | Katty | and teh ceiling cat looked upon chan_mobile and it was good. |
| 01:26.32 | *** join/#asterisk mmurdock (n=tga@c-24-10-190-87.hsd1.co.comcast.net) |
| 01:26.38 | Katty | also, buggy. |
| 01:26.51 | nny_1 | ceiling cat knows your secrets! |
| 01:26.53 | Katty | and teh ceiling cat proseeded to say |
| 01:27.13 | Katty | first! shall jo compile teh chan |
| 01:27.22 | Katty | and den you run it overth wit teh mobile |
| 01:27.23 | drmessano | ~asteriskcat |
| 01:27.23 | jbot | [asteriskcat] not amused |
| 01:27.36 | nny_1 | please oh please i'll paypal someone 10 bucks to fly here and kick me in the junk for messing with cisco |
| 01:27.49 | Nivex | where is "here"? |
| 01:28.00 | nny_1 | South Carolina |
| 01:28.07 | drmessano | Where in SC? |
| 01:28.12 | Katty | i don't think i know anyone in SC |
| 01:28.23 | nny_1 | Hilton Head Island |
| 01:28.31 | Katty | how can you be on an island |
| 01:28.32 | drmessano | Thats 8 hours drive |
| 01:28.34 | drmessano | I will do it |
| 01:28.35 | nny_1 | lol |
| 01:28.38 | nny_1 | right on |
| 01:28.40 | Katty | 10 bucks to drive... |
| 01:28.42 | Katty | for 8 hours |
| 01:28.44 | Katty | and kick someone? |
| 01:28.48 | Katty | you been drinking too much. |
| 01:28.49 | drmessano | In the nads |
| 01:28.50 | Nivex | in the junk |
| 01:28.53 | *** part/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 01:28.53 | nny_1 | yeah but you get tokick someone, for using cisco! |
| 01:29.03 | drmessano | he ASKED to be kicked in the jimmy |
| 01:29.04 | nny_1 | i'd do it if i was flexible enough |
| 01:29.08 | drmessano | I can't pass that up |
| 01:29.13 | Katty | i just don't understand |
| 01:29.16 | Katty | but that's okay, i'm not a male |
| 01:29.16 | nny_1 | lol |
| 01:29.19 | Nivex | just jump down onto a sawhorse |
| 01:29.22 | Katty | i'm just not meant to i guess |
| 01:29.50 | drmessano | When a guy asked to be kicked in his bag o cherries, you do it |
| 01:29.51 | Katty | i'm curious about voice recognition, tho |
| 01:29.55 | Katty | and what it can do |
| 01:30.00 | Katty | also, how good it really is |
| 01:30.08 | drmessano | I think it recognizes voices |
| 01:30.15 | Katty | that's quite grand, that is |
| 01:30.29 | Katty | does it respond well to "Call Katty"? |
| 01:30.43 | Katty | or is it designed for less useful things. |
| 01:31.42 | drmessano | Depends |
| 01:31.46 | husimon | nny_1 i'm back |
| 01:31.53 | drmessano | If it a male, it will ignore it.. |
| 01:32.07 | drmessano | If it's a female, it will ask "Who the &%$% is Katty?" |
| 01:32.12 | nny_1 | k |
| 01:32.14 | husimon | nny_1, pastebin me your sip.conf and SEP files |
| 01:32.18 | nny_1 | k |
| 01:32.28 | sbingner | drmessano, aren't all females katty? |
| 01:32.35 | drmessano | Sorry, i'm not supposed to make any jokes in here |
| 01:32.42 | sbingner | lol |
| 01:32.44 | nny_1 | husimon: debug says some conf errors |
| 01:32.44 | sbingner | why? |
| 01:32.57 | Katty | today was weird. |
| 01:33.00 | Katty | i can use some jokes. |
| 01:33.12 | Nivex | girls? on the internet? </sarcasm> |
| 01:33.16 | drmessano | heh |
| 01:33.19 | Katty | crazy, isn't it? |
| 01:33.24 | Nivex | not really |
| 01:33.31 | sbingner | crazy? psh. myth more like it |
| 01:33.32 | DrkShadow | katty: have you ever heard of bash.org? |
| 01:33.52 | Katty | indeed. |
| 01:33.54 | nny_1 | husimon: http://pastebin.com/m43a9b176 |
| 01:34.00 | Katty | did you sumbit something? |
| 01:34.06 | DrkShadow | katty: then you've read the top 200 :-/ |
| 01:34.09 | Katty | doi get to be famous!? |
| 01:34.16 | Katty | oh :/ |
| 01:34.20 | Katty | oh well, i tried. |
| 01:34.24 | DrkShadow | hehe |
| 01:34.26 | Nivex | "Joanna, fire." |
| 01:34.50 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:34.51 | *** join/#asterisk yidiyuehan (n=yidiyueh@node04.ssg.qmax.com.sg) |
| 01:34.54 | sbingner | bash.org rocks |
| 01:34.56 | sbingner | lol |
| 01:34.59 | yidiyuehan | hi, good morning guys. |
| 01:35.23 | yidiyuehan | can asterisk be setup with a few phones through a switch only? (no router and broadband internet for it) |
| 01:35.26 | nny_1 | husimon: http://pastebin.com/m3e833bc2 |
| 01:35.53 | drmessano | yes yidiyuehan |
| 01:36.09 | Katty | yidiyuehan: you mean analog lines and voip phones? |
| 01:36.13 | Nivex | yidiyuehan: sure. they'd only be able to talk to each other, unless you have some other path to the PST |
| 01:36.14 | nny_1 | http://pastebin.com/m55c0b7dd |
| 01:36.16 | Nivex | PSTN |
| 01:36.32 | Katty | it'd make a nice expensive paging system |
| 01:36.38 | yidiyuehan | katty, yes |
| 01:36.39 | Katty | with a conference room, and hold music! |
| 01:36.43 | Katty | yidiyuehan: quite possible (= |
| 01:36.49 | Katty | yidiyuehan: lots of people do that, too |
| 01:36.59 | Nivex | hey, I have three extensions in my apartment, and I'm the only one there :) |
| 01:36.59 | yidiyuehan | nivex, i am using the server with a few ip phones and pstn lines |
| 01:37.14 | yidiyuehan | i have assigned static ips for server and all the phones. |
| 01:37.24 | Nivex | I do have a couple extensions on the other end of a VPN |
| 01:37.34 | Katty | and the server and phones are connected via little 5 port switch or somethin? |
| 01:37.43 | nny_1 | http://pastebin.com/m47836a74 |
| 01:37.45 | yidiyuehan | yes cat5 switch |
| 01:37.47 | Katty | mkay |
| 01:37.50 | yidiyuehan | just normal linksys switch |
| 01:37.51 | nny_1 | note:my sip password is not 1234 |
| 01:37.59 | Katty | nny_1: IT IS NOW |
| 01:38.02 | nny_1 | lol |
| 01:38.09 | DrkShadow | or rather, not any more... |
| 01:38.22 | nny_1 | husimon: my xml file was a fix from voip-info to get it to load |
| 01:38.26 | yidiyuehan | katty, is it fine? with normal linksys switch and static IP assigned? |
| 01:38.43 | nny_1 | i have a larger xml file from an example somewhere |
| 01:38.46 | Katty | yidiyuehan: should be yes. you can double check connectivity from server to phone by pinging the IP address ofthe phone from the server. |
| 01:39.00 | nny_1 | husimon: truth be told i had to hack those together without experience, couldn't find any example configs |
| 01:39.17 | yidiyuehan | katty:as now i encounter one problem, after 2 days, my IP Phone could not answer the incoming call from pstn lines although i can see there is an incoming call from the phone. |
| 01:39.37 | yidiyuehan | but outgoing call is fine from this phone, what's the possible cause? |
| 01:39.54 | Katty | let me make sure i've got this right... |
| 01:39.59 | Katty | you can make outgoing calls fine. |
| 01:40.11 | yidiyuehan | yes |
| 01:40.41 | Katty | when you get an incoming call, does your phone ring at all? |
| 01:40.52 | yidiyuehan | yes, and incoming calls could reach the phone, just no sound for the phones when picking up. |
| 01:41.00 | *** join/#asterisk jameswf-home (n=james@166.128.198.184) |
| 01:41.01 | Katty | are you using polycom phones by chance? |
| 01:41.10 | yidiyuehan | no, i am usiing GXP2000 |
| 01:41.13 | Katty | hrmm. |
| 01:41.31 | yidiyuehan | and the caller side just keeps ringing |
| 01:41.33 | Katty | i presume you've rebooted both the server and the phone |
| 01:41.58 | Katty | can you route the incoming call to another phone to test? |
| 01:42.05 | yidiyuehan | yes, rebooting of both the phone and the server with cables upplugged and plugged will solve the problem, temporarily |
| 01:42.13 | Katty | ah.. |
| 01:42.17 | Katty | hmmmm |
| 01:42.31 | Katty | that sounds very close to a polycom problem i had a while back |
| 01:42.37 | Katty | it would mysteriously kill audio after a few days |
| 01:42.44 | yidiyuehan | Katty, it's strange that i have a few phones with the server, first day it happens on one phone, and second day it happens on another phone. |
| 01:42.48 | Katty | and we had to reboot the phones on occasion. |
| 01:42.52 | jameswf-home | ~idnms |
| 01:42.52 | jbot | Why would a Wookiee, an eight-foot tall Wookiee, want to live on Endor, with a bunch of two-foot tall Ewoks? That does not make sense! But more important, you have to ask yourself: What does this have to do with this channel? Nothing |
| 01:43.20 | Katty | yidiyuehan: sounds a lot like the polycom issue we had awhile back. i think it ended up to be some sort of quirk with the firmware of the phone |
| 01:43.21 | yidiyuehan | katty, you mean it's a kind of bug of the phones |
| 01:43.36 | Katty | yidiyuehan: it was with ours. but we were using Polycoms, of course. |
| 01:43.48 | yidiyuehan | ic. at least i have some confidence that the server is fine:) |
| 01:43.55 | Katty | aye. |
| 01:44.14 | yidiyuehan | but with the same phone models and firmware, all works fine in another location. |
| 01:44.23 | yidiyuehan | the only difference is i have internet in another location |
| 01:44.27 | Katty | J4k3: lol |
| 01:44.33 | J4k3 | (the combination of polycom and grandsuck) |
| 01:44.39 | Katty | yidiyuehan: very odd indeed. |
| 01:44.53 | JT | yidiyuehan: have you used a different phone? |
| 01:45.01 | yidiyuehan | yes, i do. |
| 01:45.03 | Katty | yidiyuehan: i know that analog tdm cards have their quirks.. |
| 01:45.10 | Katty | yidiyuehan: using digium hardware? |
| 01:45.11 | husimon | nny_1, looking now |
| 01:45.33 | yidiyuehan | katty, i use openvox card for the server |
| 01:45.42 | nny_1 | husimon: k. I am up for new configs if needed. Those were horrible attempts based on numerous sources |
| 01:45.46 | JT | yidiyuehan: what other phones have you used? |
| 01:45.51 | Katty | yidiyuehan: and the other server as well? |
| 01:46.02 | yidiyuehan | katty, it never happens on other servers. |
| 01:46.11 | Katty | hrmm. |
| 01:46.12 | yidiyuehan | but all other servers have the internet connectivity |
| 01:46.34 | Katty | i don't suppose you have a spare card you could swap around in the server to give it a test |
| 01:46.35 | yidiyuehan | JT, i use atcom, softphone, and Grandstream phones. |
| 01:46.47 | husimon | nny_1, btw did you reload your asterisk when you changed the phone passwords? |
| 01:46.51 | JT | yidiyuehan: and they all experience the problem? |
| 01:47.10 | husimon | nny_1, that was the first mistake I made, not restarting asterisk after making changes to sip.conf |
| 01:47.12 | nny_1 | husimon: yes |
| 01:47.13 | yidiyuehan | JT, no for this troublesome server, i happens use only grandstream phone |
| 01:47.26 | husimon | nny_1, btw what does the cli say when a phone tries to register? |
| 01:47.31 | nny_1 | sb |
| 01:47.42 | JT | yidiyuehan: grandstreams are pretty buggy |
| 01:47.43 | yidiyuehan | and i have similar servers with same model TDM cards and gxp2000 and other phones, but no problem at all |
| 01:47.57 | nny_1 | http://pastebin.com/m9ce12e3 from sip debug |
| 01:48.03 | Katty | yidiyuehan: anything abnormal about the network cabling in that build? |
| 01:48.07 | husimon | btw |
| 01:48.09 | Katty | yidiyuehan: mayhaps extra long cable runs? |
| 01:48.13 | husimon | what is all that crap on the text? |
| 01:48.15 | nny_1 | husimon: i don't get any messages from verbose cli in regards to registration, which is stranfe |
| 01:48.22 | nny_1 | lol from an example config |
| 01:48.23 | yidiyuehan | katty, the cable is within 5 meters. |
| 01:48.25 | husimon | no |
| 01:48.29 | husimon | the /par stuff |
| 01:48.33 | nny_1 | i have no idea |
| 01:48.38 | nny_1 | that was also in the example config |
| 01:48.41 | yidiyuehan | but that maybe still the case, as the cable is done by a part-time contractor. |
| 01:48.41 | husimon | uh |
| 01:48.47 | husimon | that's not supposed to be there :) |
| 01:48.50 | nny_1 | husimon: i need a real set of example files |
| 01:48.51 | yidiyuehan | katty, but if it works once, does it mean the cable is fine |
| 01:48.53 | JT | yidiyuehan: try new firmware on the phones |
| 01:48.58 | JT | yidiyuehan: i doubt it's cabling |
| 01:48.58 | Katty | yidiyuehan: could be. we have in house people that can make me a cable right sometime... |
| 01:48.59 | husimon | here i'll pastebin you one of my phone configs |
| 01:49.01 | nny_1 | husimon: ok i'll delete them they looked stranfe |
| 01:49.02 | JT | it's ethernet |
| 01:49.05 | nny_1 | i am up for anything |
| 01:49.05 | JT | but you never know |
| 01:49.08 | yidiyuehan | JT, unluckily, i am using the latest firmware..... |
| 01:49.17 | JT | if it becomes disconnected, the phone could unregister |
| 01:49.23 | JT | yidiyuehan: unlucky brand of phone ;) |
| 01:49.43 | Katty | yidiyuehan: not sure to be honest, never really had a situation like that come up. |
| 01:50.00 | yidiyuehan | JT, yes I guess there maybe something wrong with the cable. and firmware, |
| 01:50.08 | dmz | nny_1 you in hhi? |
| 01:50.25 | dmz | i never run into anyone i know in the same town since leaving SF :) |
| 01:50.27 | yidiyuehan | JT & katty, So i can confirm that without an internet and router, asterisk will work as normal? |
| 01:50.50 | Katty | yidiyuehan: well it doesn't need internet For Sure |
| 01:50.51 | nny_1 | dmz: indeed |
| 01:51.02 | Katty | yidiyuehan: i can't think of any logicial reason about why you'd need a router if you're setting everything static |
| 01:51.06 | nny_1 | dmz: i used to live in socal |
| 01:51.10 | nny_1 | grew up there |
| 01:51.10 | Katty | yidiyuehan: you said you were using a linksys switch? |
| 01:51.22 | dmz | damn that's crazy |
| 01:51.28 | dmz | noone in hilton head is into asterisk ;) |
| 01:51.37 | nny_1 | dmz: nope :D |
| 01:51.45 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 01:51.45 | yidiyuehan | my setting of ifcfg-eth0: device=eth0, onboot=yes, bootproto=static, ipaddr=192.168.15.160, netmask=255.255.255.0, gateway=192.168.15.1 |
| 01:51.52 | yidiyuehan | katty, yes |
| 01:52.10 | yidiyuehan | i am wondering whether i need to set gateway as there is actually no gateway. |
| 01:52.22 | Katty | eh, shouldn't matter. |
| 01:52.27 | Katty | they're not really looking for it |
| 01:52.50 | Katty | when you reboot your server and phones, do you powercycle the switch as well? |
| 01:52.54 | husimon | hhi? |
| 01:53.02 | yidiyuehan | katty, no, |
| 01:53.03 | husimon | hilton head? |
| 01:53.05 | Katty | hrmm. |
| 01:53.26 | Katty | it does sound like a weird quirky networking issue |
| 01:53.42 | Katty | i think there's a way to do a sip debug from the CLI |
| 01:53.47 | yidiyuehan | katty. the case is, i reboot the phone, nothing happens, then i reboot the server, still not works, after that i unplug the computer cable from the phone and plug in again, then everything is fine. |
| 01:53.56 | Katty | kinda watch what the phone is doing, but i don't recall how to do it off the top of my head. |
| 01:54.11 | Katty | hrmmm. |
| 01:54.24 | Katty | do you have some premade cables? |
| 01:54.43 | yidiyuehan | yes, i do have. |
| 01:55.05 | husimon | nny_1, http://pastebin.com/m6cda59fe |
| 01:55.05 | yidiyuehan | the cables are DIY by the contractor, i am confused that if it works intially, does it mean the cable is fine? |
| 01:55.08 | Katty | it'd be quick and easy to swap one of the phone cat5 cables around and see what happens. |
| 01:55.17 | tzanger | oh katty's back |
| 01:55.17 | Katty | maybe the jack at the end is loose |
| 01:55.19 | tzanger | get it working/ |
| 01:55.22 | tzanger | er ? |
| 01:55.25 | Katty | or not crimped down right |
| 01:55.28 | Katty | tzanger: no i went home |
| 01:55.33 | jameswf-home | huh huh jack |
| 01:55.37 | yidiyuehan | katty, yes, i agree with you. |
| 01:55.37 | tzanger | haha I'm at the hotel |
| 01:55.37 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 01:55.49 | jameswf-home | hotel or motel |
| 01:55.51 | Katty | yidiyuehan: not hard to put a premade in there and see if the problem magically disappears |
| 01:56.03 | Katty | yidiyuehan: if it does, yay, if not, you can rule out the cable, and move forward |
| 01:56.13 | tzanger | yay my mom wants to learn C |
| 01:56.14 | jameswf-home | tzanger: you in SC? |
| 01:56.19 | yidiyuehan | katty, yes, i will try this first.;-) |
| 01:56.22 | tzanger | jameswf-home: ?? uh, no |
| 01:56.25 | tzanger | jameswf-home: MTL |
| 01:56.25 | yidiyuehan | thanks brother. |
| 01:56.33 | yidiyuehan | and JT, thanks to you as well. |
| 01:57.08 | husimon | nny_1, see if something like that works |
| 01:57.19 | Katty | tzanger: i try to ignore all things work related... |
| 01:57.23 | Katty | tzanger: don't ask me why i'm in here |
| 01:57.26 | tzanger | Katty: haha |
| 01:57.28 | Katty | tzanger: i'm relaxing. |
| 01:57.30 | tzanger | the conversation? |
| 01:57.32 | Katty | tzanger: dont' ask. |
| 01:57.35 | Katty | sure! |
| 01:57.39 | tzanger | intellectual stimulation? |
| 01:57.55 | Katty | is that a trick question? |
| 01:58.00 | Katty | oh wait. |
| 01:58.05 | Katty | is the answer, yes? |
| 01:58.13 | tzanger | haha |
| 01:58.19 | tzanger | that's the right answer, but necessarily the correct one |
| 01:58.26 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:58.41 | Katty | oh boy! lunc! |
| 01:58.45 | nny_1 | husimon: trying right now |
| 01:58.49 | Katty | we should introduce that break to our office too! |
| 01:58.58 | tzanger | ooh, I wonder if the rick mercer report is on now |
| 01:58.59 | tzanger | it's 9 |
| 01:59.13 | Katty | i'm not sure what i'm watching on tv. |
| 01:59.22 | tzanger | there are only a handful of english channels |
| 01:59.26 | Katty | mostly, i'm ignoring it, but i heard the term goons fly by a few minutes ago and looked up. |
| 01:59.30 | tzanger | and the ghosting reminds me of why I love my satellite |
| 01:59.40 | tzanger | Katty: yeah that's kind of what I"m doing |
| 01:59.45 | Katty | J4k3: background noise. |
| 01:59.55 | J4k3 | Katty: thats why god made music. |
| 02:00.02 | J4k3 | ;) |
| 02:00.05 | tzanger | J4k3: I listen to music all day |
| 02:00.09 | Katty | so THATs why all the bands suck |
| 02:00.14 | tzanger | I'm at a contract for a company that makes jukeboxes that run linux |
| 02:00.23 | Katty | we need to hire jesus for the next live proformance around here! |
| 02:00.30 | Katty | i knew there had to be reason!! |
| 02:00.39 | Katty | tzanger: oooh that's neat |
| 02:00.50 | tzanger | Katty: www.touchtunes.com |
| 02:00.53 | Katty | tzanger: do you have an open ssh key thingy were you can pipe it over to asterisk? |
| 02:01.08 | tzanger | ? |
| 02:01.09 | Katty | tzanger: like play music over speakers, ssh over lower the volume, overlay another audio file |
| 02:01.17 | Katty | tzanger: then up the volume, get out of ssh |
| 02:01.24 | Katty | tzanger: with uhh... xmms shell or something |
| 02:01.31 | tzanger | I have no idea what you're talking about |
| 02:01.32 | Katty | that'd be neat. |
| 02:01.37 | tzanger | look at the website |
| 02:01.40 | Katty | box 1 runs xmms All The Time |
| 02:01.40 | tzanger | that's what I'm designing |
| 02:01.44 | J4k3 | hmm, project for tonight: list crappy wm6 phone on ebay |
| 02:01.47 | Katty | box 2 runs asterisk All The Time |
| 02:01.56 | Katty | box 2 sshs over to box 1 and says BE QUIETER |
| 02:02.11 | tzanger | ohhhhhh I think I get what you're saying |
| 02:02.22 | Katty | box 2 spits out auto answer page stuffs |
| 02:02.24 | tzanger | seems like overkill |
| 02:02.26 | Katty | over to b1 input |
| 02:02.32 | tzanger | but yeah I hear what you're saying now |
| 02:02.32 | Katty | then kicks the volume back up |
| 02:02.37 | tzanger | I'm picking up what you're laying down |
| 02:02.44 | Katty | and then quietly goes back to doing what they each do best |
| 02:02.59 | Katty | Oh please. people build linux boxes to feed their cats. |
| 02:03.03 | Katty | don't tell me that's overkill :P |
| 02:03.06 | Nugget | Linux is poo. :) |
| 02:03.15 | Katty | Nugget: i'm going to tell drdink you said that. |
| 02:03.20 | Nugget | heh |
| 02:03.26 | husimon | it's on now nugget :P |
| 02:03.30 | drmessano | ~bsd |
| 02:03.31 | jbot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 02:03.42 | Nugget | bah ;) |
| 02:03.54 | drmessano | ~epoch time |
| 02:03.56 | tzanger | Katty: you could just do something like this: manager event watch looks for pages and either mutes or attenuates the audio |
| 02:04.08 | tzanger | then when the page ends, 2s later it restores normal volume |
| 02:04.13 | tzanger | no need for ssh or many boxes |
| 02:04.40 | Katty | just because you know more about linux doesn't mean i can't do it too. |
| 02:04.43 | tzanger | the nice thing about hockey is that it's the same in every language |
| 02:04.44 | Katty | <PROTECTED> |
| 02:05.34 | tzanger | rasp? |
| 02:05.59 | sbingner | Saying that Java is nice because it works on all OS's is like saying that anal sex is nice because it works on all genders |
| 02:06.03 | drmessano | ~epoch |
| 02:06.03 | jbot | 1 Jan 1970 - The expiration date on side of the carton of Unix |
| 02:06.13 | sbingner | everybody needs to see that one, it's perfect |
| 02:06.17 | [TK]D-Fender | J4k3, ditching your VX6700? |
| 02:06.42 | tzanger | sbingner: bwaaaaaaaaaaahahahahahahahahha |
| 02:06.58 | JT | sbingner: weird comparison |
| 02:07.03 | husimon | sbingner, hilarious |
| 02:07.07 | J4k3 | [TK]D-Fender: yep |
| 02:07.20 | Katty | tzanger: world of warcraft reference |
| 02:07.24 | J4k3 | [TK]D-Fender: once I removed enough software its now 'stable' |
| 02:07.28 | *** join/#asterisk guillote_GNU (n=guillote@host235.200-117-226.telecom.net.ar) |
| 02:07.29 | Katty | tzanger: $person makes rude gesture at $nick |
| 02:07.33 | Katty | tzanger: basically. |
| 02:07.46 | J4k3 | removing the windows live package made it go from 4-48 hours per reboot to >2 weeks between reboots |
| 02:07.51 | Katty | [TK]D-Fender: hai |
| 02:07.59 | J4k3 | gf spent half the day sunday playing sc2k on it. |
| 02:07.59 | tzanger | you didn't have a target |
| 02:08.00 | J4k3 | hehe |
| 02:08.02 | tzanger | just a rude gesture |
| 02:08.06 | tzanger | so to the chan in general? |
| 02:08.15 | Katty | <PROTECTED> |
| 02:08.16 | Katty | <PROTECTED> |
| 02:08.18 | Katty | <PROTECTED> |
| 02:08.26 | Katty | better? |
| 02:08.35 | *** join/#asterisk guillote_GNU (n=guillote@host235.200-117-226.telecom.net.ar) |
| 02:08.36 | sbingner | Katty, what is /rasp? |
| 02:08.40 | tzanger | what was that for :-) |
| 02:08.41 | Katty | tzanger: mock me again and i'll MAKE A MACRO |
| 02:08.55 | sbingner | Katty, you know you can have the target in the command in WoW... ie /bonk Katty |
| 02:09.00 | J4k3 | jameswf-home: rediculously high uptimes in the linux world = you're insecure ;) |
| 02:09.12 | husimon | please, no more about the most time wasting game evr |
| 02:09.20 | J4k3 | Katty: raspberry?!?! do you myspcaeim or something? |
| 02:09.25 | sbingner | husimon, apparently you never played EverCrack |
| 02:09.25 | Katty | 178 days :/ |
| 02:09.28 | Katty | husimon: you're right. |
| 02:09.29 | jameswf-home | I am not isecure i just have body image issues |
| 02:09.41 | Katty | J4k3: baroo? |
| 02:09.42 | husimon | sbingner, I did some, but didn't continue |
| 02:09.49 | tzanger | jameswf-home: that's almost as funny as sbinger's comment |
| 02:09.51 | Katty | J4k3: you do not parse |
| 02:10.06 | jameswf-home | ~wow |
| 02:10.07 | jbot | I have no life | Lets go raid! |
| 02:10.13 | J4k3 | RAID? |
| 02:10.16 | J4k3 | ~raid |
| 02:10.17 | jbot | hmm... raid is (Redundant Array of Inexpensive Drives) A manipulation of SCSI technology that allows you to combine several hard drives for use as one. The benefits can be either speed increase or failsafe (so that if one drive dies or crashes you don't lose your data), or a combination of both. |
| 02:10.38 | Katty | :< |
| 02:11.14 | Katty | jbot: you must be pre 2.3 patch |
| 02:11.18 | J4k3 | of course, I have outrageously great luck with HDDs as long as they don't say maxtor or hitachi on them |
| 02:11.37 | Katty | i'm skeered of most raids. |
| 02:11.41 | tzanger | I have had excellent luck with maxtor drives |
| 02:11.43 | Katty | mirror raids are pretty handy |
| 02:11.46 | sbingner | ~panty raid |
| 02:11.47 | drmessano | RAID: Random Array of Inferior Drives |
| 02:11.50 | Katty | but rebuilding them makes me wibble. |
| 02:11.54 | JoeShmo | kind of odd that the best raids were typically SCSI based, which is the most expensive drives... |
| 02:12.04 | sbingner | the best raids were panty based |
| 02:12.05 | Katty | more wibbling just happened. |
| 02:12.16 | Katty | sbingner: kings to you! |
| 02:12.20 | J4k3 | yeah, RAID1 is the only raid that doesn't totally make me cringe... cuz I can rip one HD out of a RAID1 and use it 'normally' and everything is ok |
| 02:12.30 | nny_1 | husimon: maybe its this firmware |
| 02:12.33 | nny_1 | husimon: 8.2 |
| 02:12.36 | husimon | nny_1, shouldn't matter |
| 02:12.39 | husimon | what problem do you have now? |
| 02:12.40 | nny_1 | husimon: k |
| 02:12.42 | husimon | anything new? |
| 02:12.43 | J4k3 | JoeShmo: and in my experience, in the modern day, scsi drives are some of the most unreliable drives you can get. |
| 02:12.44 | nny_1 | husimon: exact same :D |
| 02:12.49 | jameswf-home | ~drmessano |
| 02:12.50 | jbot | [drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway |
| 02:12.58 | J4k3 | JoeShmo: the drive controllers seem about as likely as the physical drives to fail |
| 02:12.58 | drmessano | HAHAH |
| 02:13.07 | husimon | J4k3, raid 6 is not bad |
| 02:13.07 | J4k3 | err on-drive controllers |
| 02:13.15 | JoeShmo | yup.. |
| 02:13.18 | SteveTotaro | hardware raid 5 is excellent |
| 02:13.20 | Katty | ~Katty |
| 02:13.21 | jbot | [katty] the only girl in the channel, so be nice to her |
| 02:13.24 | nny_1 | husimon: wiped the tftpboot dir, replaced the firmware stuff used your examples and changed what needed changing |
| 02:13.25 | husimon | f raid 5 |
| 02:13.25 | Katty | who put that in? |
| 02:13.26 | jameswf-home | ~uptime |
| 02:13.27 | husimon | raid 6 ftw |
| 02:13.29 | JoeShmo | seems alot of the "failed" drives, I've replaced the board on the drive to recover the data |
| 02:13.31 | nny_1 | let me telnet into the phone and debug |
| 02:13.31 | Nugget | telnet is eeeeeeevil! |
| 02:13.39 | Katty | jbot: who's feeding you this nonsense! |
| 02:13.41 | nny_1 | <PROTECTED> |
| 02:13.41 | J4k3 | raid6 = raid5 with a raid1'd parity drive, right? |
| 02:13.44 | SteveTotaro | ultra320 with six 18gb drives |
| 02:13.44 | nny_1 | <PROTECTED> |
| 02:13.47 | Katty | jbot: file? |
| 02:13.48 | jbot | it has been said that file is Joshua Colp, jcolp@digium.com |
| 02:13.50 | husimon | raid 6 = raid 5 except two disks can fail |
| 02:13.53 | sbingner | Katty, can I get might instead plz |
| 02:13.58 | tzanger | anyway, I should go |
| 02:14.02 | tzanger | ~tzanger |
| 02:14.03 | jbot | methinks tzanger is the raddest fcking dude alive |
| 02:14.03 | Katty | sbingner: sorry, my pally is only lvl 12 |
| 02:14.07 | SteveTotaro | raid 5 +1 |
| 02:14.07 | tzanger | haha |
| 02:14.09 | sbingner | Katty, salv? |
| 02:14.11 | tzanger | that's still in there |
| 02:14.19 | SteveTotaro | ~stevetotaro |
| 02:14.20 | jbot | well, stevetotaro is an IRC nub |
| 02:14.21 | tzanger | I forget who put that in there |
| 02:14.21 | Katty | sbingner: prot |
| 02:14.25 | sbingner | lol |
| 02:14.26 | Katty | sbingner: for now. |
| 02:14.33 | Katty | sbingner: going to be a pvp healadin! |
| 02:14.42 | Katty | sbingner: you can't kill me. |
| 02:14.44 | jameswf-home | ~unixdog |
| 02:14.44 | jbot | <unixdog> Everyone use BSD gah linux sux progress is overrated use my project gah |
| 02:14.47 | J4k3 | ~reboot |
| 02:14.47 | jbot | Not on your life cowboy :(, or are you using Window$? |
| 02:14.47 | Katty | sbingner: unless...you're a priest... |
| 02:14.52 | J4k3 | haha |
| 02:14.57 | Katty | please don't :< |
| 02:15.02 | SteveTotaro | ~asterisk |
| 02:15.03 | jbot | [asterisk] the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org |
| 02:15.03 | Katty | i hate dispell all :< |
| 02:15.12 | SteveTotaro | ~qwell |
| 02:15.12 | jbot | somebody said qwell was a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 02:15.16 | drmessano | RAID: Randomly Arsing Important Data |
| 02:15.27 | jameswf-home | ~dood |
| 02:15.27 | jbot | from memory, dood is a typo for d00d, or dutch for dead |
| 02:15.48 | jameswf-home | ~d00d |
| 02:15.49 | jbot | somebody said d00d was a dude, but in hax0r skript |nn|+, or asking a lot of questions today |
| 02:15.52 | SteveTotaro | i dunno, i pull my scsi drives pretty regularly just to test |
| 02:16.09 | jameswf-home | jbot: what's mine say |
| 02:16.10 | jbot | dude! ... What's Mine Say? |
| 02:16.20 | sbingner | Katty, 70 mage.. I pwn you |
| 02:16.29 | drmessano | SteveTotaro: Effin Wuss, pull two |
| 02:16.30 | Katty | sbingner: ..i have a 70 mage... |
| 02:16.34 | Katty | sbingner: an undead FROST mage |
| 02:16.37 | sbingner | Katty, foiled again |
| 02:16.41 | Katty | sbingner: and a 70 hunter |
| 02:16.51 | Katty | sbingner: and i have TRINKETS |
| 02:16.54 | Katty | sbingner: you shall not sheep. |
| 02:17.02 | sbingner | Katty, my mage was > pally at 60... not at 70... but I could take your mage |
| 02:17.05 | nny_1 | husimon: heh no errors in confs now at least, only missing ringlist.dat, which doesn't seem to matter |
| 02:17.09 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-939f9475c3008a13) |
| 02:17.13 | Katty | sbingner: probably. she's not epic yet. |
| 02:17.16 | *** join/#asterisk alrs (i=foobar@216.151.159.21) |
| 02:17.17 | drmessano | My RAID array has a +20 against drive failures, oh and magic missile |
| 02:17.30 | drmessano | I have a CLOUDSONG on my RAID arrays |
| 02:17.35 | husimon | nny_1, so when you do a "sip show peers" |
| 02:17.35 | Katty | if by drive failures you mean the main boss of ZA |
| 02:17.37 | husimon | does it show up? |
| 02:17.42 | Katty | who just might be ZA. |
| 02:17.48 | sbingner | Katty, you can still sheep ppl with trinkets... just takes proper timing and psyching out... |
| 02:17.51 | jameswf-home | I wonder if I crank up the wireless (cell) card on the plane if the faa will kill a kitten |
| 02:17.58 | nny_1 | husimon: sweet!@ |
| 02:17.59 | Katty | sbingner: true. |
| 02:18.03 | nny_1 | husimon: now it shows up :D |
| 02:18.15 | husimon | nny_1, make sure to not paste configs with RTF codes in them next time :P |
| 02:18.15 | Katty | sbingner: but it's very difficult to frost nova or sheep a kitty with bestial wrath on |
| 02:18.26 | Katty | sbingner: i imagine by the time you'd get to sheep, you'd be dead :/ |
| 02:18.33 | Katty | sbingner: but who knows! |
| 02:18.39 | jameswf-home | heh sheep |
| 02:18.40 | Katty | sbingner: i've been killed by many a mage! |
| 02:18.42 | nny_1 | husimon: ok |
| 02:18.43 | husimon | brain getting smaller..... |
| 02:18.55 | husimon | sex life shrinking |
| 02:18.57 | jameswf-home | dah ah ah dee |
| 02:18.58 | nny_1 | husimon: may i ask why? (I imagine it is a security risk) |
| 02:19.05 | jameswf-home | ~emo |
| 02:19.05 | jbot | /wrists |
| 02:19.12 | Katty | jbot: finding emo? |
| 02:19.20 | husimon | nny_1, because RTF codes are not part of the config syntax |
| 02:19.21 | Katty | jbot needs updating. |
| 02:19.29 | nny_1 | husimon: oh um, sorry man |
| 02:19.30 | jameswf-home | jbot find emo |
| 02:19.41 | husimon | they are garbled crap from someone pasting their config from an .rtf file instead of a real text file |
| 02:19.43 | jameswf-home | jbot wiki emo |
| 02:19.47 | drmessano | ~emo |
| 02:19.48 | jbot | I just cut myself |
| 02:19.57 | drmessano | heh |
| 02:20.03 | husimon | markup codes to describe bold, font size, etc. |
| 02:20.04 | nny_1 | husimon: indeed, i usually trim all my confs and keep the RTFs as examples, this was a 5 hour hack job on the cisco |
| 02:20.20 | drmessano | ~emo |
| 02:20.21 | jbot | I just cut myself :`( |
| 02:20.23 | sbingner | Katty, oh a hunter? I don't sheep them... I just blow them up |
| 02:20.47 | jameswf-home | jbot: go cry emo |
| 02:20.53 | Katty | sbingner: much more effective! |
| 02:21.00 | drmessano | How do you do <CR> with jbot, or can you? |
| 02:21.06 | Qwell | hunter? ice block. QQ |
| 02:21.13 | Katty | drmessano: i thought you typed cpr >.< |
| 02:21.36 | nny_1 | husimon: thanks again btw |
| 02:22.12 | drmessano | ~emo poem |
| 02:22.12 | jbot | The night is dark, can't feel my toes, my cat is staring at me, I want to die |
| 02:22.22 | drmessano | yay |
| 02:22.48 | Katty | halp, ferrets attacking me |
| 02:22.51 | jameswf-home | http://www.reallyfunnyclips.com/clips/weird/uber-emo-queen/ LMAO |
| 02:23.19 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 02:23.31 | nny_1 | alright time to go home, tty tomorrow |
| 02:23.34 | Katty | glater |
| 02:23.47 | husimon | np |
| 02:23.57 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 02:26.21 | jameswf-home | You diss I slit my wrist When you cause me pain I slit the left vain You break my heart That’s where it starts When you pretend . |
| 02:26.51 | jameswf-home | jbot cat /etc/resolv.conf |
| 02:26.51 | jbot | ACTION shoves a feral cat down /etc/resolv.conf's pants |
| 02:26.57 | jameswf-home | lmao |
| 02:27.16 | jameswf-home | jbot: cat drmessano |
| 02:27.17 | jbot | ACTION shoves a feral cat down drmessano's pants |
| 02:27.26 | drmessano | ha |
| 02:27.39 | drmessano | That makes two |
| 02:28.14 | jameswf-home | *these |
| 02:29.25 | Katty | feralas. |
| 02:34.31 | DrkShadow | agi show GET VARIABLE agi_callerid --> No such command 'show GET VARIABLE agi_callerid'. <-- how can I get the value of a variable through agi? :-/ (I'm at the asterisk console, but same thing in script) |
| 02:40.15 | [TK]D-Fender | ~emo |
| 02:40.16 | jbot | I just cut myself :`( |
| 02:40.22 | Katty | [TK]D-Fender: hai. |
| 02:40.29 | [TK]D-Fender | Katty, O HAI |
| 02:40.51 | Katty | [TK]D-Fender: ever heard of the count of monte crisco? |
| 02:41.00 | Katty | [TK]D-Fender: apparently it's a Good Film |
| 02:41.20 | DrkShadow | yep.. asterisk command line interface for AGI is completely broken. Even a copy + paste example won't work. |
| 02:41.20 | [TK]D-Fender | Crisco? Sounds like a porno ;) |
| 02:41.24 | DrkShadow | a scripts are no better off. |
| 02:41.35 | Katty | [TK]D-Fender: tho i'm wondering if mayhaps the poor person i was talking to had no idea they turned a perfectly good movie into ahorribly messed up porno film |
| 02:42.04 | [TK]D-Fender | Katty, That'd be "The Count of Monte Cristo" |
| 02:42.12 | Katty | indeed. |
| 02:42.23 | Katty | the typo made me giddy. |
| 02:42.32 | [TK]D-Fender | Katty, As opposed to a petroleum-based product from which only porn can "come" ;) |
| 02:43.09 | Katty | s/thinks/thanks/ |
| 02:44.18 | Katty | [TK]D-Fender: having a nice evening? |
| 02:46.45 | *** join/#asterisk mitcheloc (n=mitchel@68-190-7-103.dhcp.dctr.al.charter.com) |
| 02:47.27 | [TK]D-Fender | Katty, I'm wiped... jsut completely wiped |
| 02:47.54 | [TK]D-Fender | Katty, I'm rebuilding my server from a FUBAR incident and redoing apache config almost from the bottom up |
| 02:47.58 | [TK]D-Fender | PITA |
| 02:48.05 | [TK]D-Fender | Stupid version mod changed |
| 02:48.12 | *** join/#asterisk iPod-nano (n=david@c-68-43-60-193.hsd1.mi.comcast.net) |
| 02:48.23 | iPod-nano | Can someone tell me where to find the sound files on Ubuntu 7.10? |
| 02:48.42 | drmessano | Which sound files? |
| 02:49.06 | iPod-nano | Like tt-weasels and such. |
| 02:49.21 | Katty | [TK]D-Fender: sounds liked you were wiped for sure. |
| 02:49.33 | DrkShadow | someone must tell me.. When I do ${CALLERID(num)} of an incoming call.. I get the number they called: my DID. How, then, do I get _their_ number? |
| 02:49.36 | iPod-nano | Perhaps the better question would be where to put .gsm files in order to play them back, |
| 02:49.53 | drmessano | Um /var/lib/asterisk/sounds maybe |
| 02:49.59 | Katty | that souns right |
| 02:50.05 | Katty | mayhaps into custom |
| 02:50.12 | [TK]D-Fender | DrkShadow, ${CALLERID(num)} is their number |
| 02:50.26 | DrkShadow | hum. Maybe you're right. |
| 02:50.45 | iPod-nano | drmessano, I looked there, it doesn't exist. |
| 02:50.46 | Katty | help! i can't find { on my phone! |
| 02:50.49 | DrkShadow | arright, so how can I use an agi show command? any command.. I've not had a single one work. |
| 02:51.21 | iPod-nano | Will it work if I just put Playback(/path/to/file)? |
| 02:51.28 | Katty | should |
| 02:51.33 | Katty | don't put .gsm on there though |
| 02:51.54 | Katty | optionally use Background() if you want it interactive |
| 02:53.08 | drmessano | Asterisk seems a little beefy to be using as an iPod |
| 02:53.53 | Katty | beef in lots of... foil |
| 02:54.29 | drmessano | mmmm |
| 02:54.34 | drmessano | Now I want to make scouts |
| 02:54.44 | Katty | don't bother. kids are expensive. |
| 02:54.50 | drmessano | lol |
| 02:55.09 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 02:55.12 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
| 02:55.14 | drmessano | I'm not having kids |
| 02:55.20 | Katty | hi manx! |
| 02:55.33 | ManxPower | Hello, Katty |
| 02:57.21 | Katty | oh boy! cosmic radiation on the history channel! |
| 02:57.33 | ManxPower | Katty: How is your PRI doing? |
| 02:57.40 | Katty | i bet someone already has a patent on cosmic lead. |
| 02:57.47 | *** part/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 02:57.55 | Katty | ManxPower: i'm trying to ignore it for the evening and relax (= |
| 02:58.11 | ManxPower | lead acid, that is |
| 02:58.25 | b11d | i bet those bastards are heavy |
| 02:58.27 | ManxPower | Came out of a battery backup |
| 02:58.30 | Katty | ^_- |
| 02:58.37 | ManxPower | b11d: more than 500 lbs |
| 02:58.40 | b11d | aye |
| 02:58.44 | Katty | the ferret is trying to drag the food bowl behind the entertainment center |
| 02:58.55 | Katty | i don't think he's quite figured out it just Won't Fit |
| 02:58.59 | Katty | brb |
| 03:00.59 | [TK]D-Fender | OH YAY |
| 03:01.05 | Katty | oh? |
| 03:01.13 | [TK]D-Fender | apparently full web server recovery! |
| 03:01.32 | Katty | horay! |
| 03:01.59 | [TK]D-Fender | ~sipnat |
| 03:01.59 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT :<I have to rebuild my server *sigh*> , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:02.56 | [TK]D-Fender | ~sipnat |
| 03:02.57 | 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 |
| 03:03.06 | [TK]D-Fender | \o/ |
| 03:03.23 | Qwell | O.o |
| 03:04.07 | [TK]D-Fender | Qwell, whats hard to follow? |
| 03:04.30 | *** join/#asterisk PepOSX (n=angeldav@190.72.131.172) |
| 03:04.30 | Katty | <Qwell> christ's example. |
| 03:05.04 | [TK]D-Fender | Qwell, its a heavy burden to carry, at least you get nailed in the end ;) |
| 03:05.15 | [TK]D-Fender | oh wait... that sounds like another pr0n... |
| 03:05.27 | b11d | :P |
| 03:05.39 | Katty | oh geeze |
| 03:10.04 | b11d | it cracked me up |
| 03:11.42 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 03:11.56 | Wayhigh | w0rd |
| 03:13.23 | drmessano | l0l |
| 03:14.05 | drmessano | \o/ >'=^===< <-- SHARRRK!!! |
| 03:15.15 | b11d | \\n/ |
| 03:15.19 | *** join/#asterisk ruied (n=ruied@bl7-217-48.dsl.telepac.pt) |
| 03:15.29 | drmessano | Oh sorry |
| 03:15.33 | drmessano | I almost forgot |
| 03:15.46 | drmessano | \o/ >'=^===< <-- SHARRRK!!! (This is a JOKE, AKA "A LOL |
| 03:15.49 | drmessano | \o/ >'=^===< <-- SHARRRK!!! (This is a JOKE, AKA "A LOL") |
| 03:15.55 | b11d | haha LOL |
| 03:16.02 | drmessano | Not a REAL Shark |
| 03:16.16 | Qwell | not real? good, I was scared |
| 03:16.31 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-77e76f33ad6d2cfa) |
| 03:16.34 | drmessano | Just wanted to make sure there was no confusion |
| 03:20.12 | *** join/#asterisk bmg505 (n=leon@196-209-182-222-rndf-esr-5.dynamic.isadsl.co.za) [NETSPLIT VICTIM] |
| 03:20.12 | *** join/#asterisk nitram (i=nitram@superblob.com) [NETSPLIT VICTIM] |
| 03:20.12 | *** join/#asterisk Bosse (i=bosse@miraculix.klykken.com) [NETSPLIT VICTIM] |
| 03:20.51 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:20.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:21.21 | drmessano | ~shark |
| 03:21.22 | jbot | There are NO sharks in #Asterisk. We want to be perfectly clear on that. |
| 03:22.12 | drmessano | That was NOT the result of a shark attack.. move along pls |
| 03:23.59 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 03:24.54 | b11d | is that shark gone yet? |
| 03:24.58 | Nugget | Candygram! |
| 03:25.10 | drmessano | Feature suggestion: Set the NO_SHARK compiler option in 1.6 by default |
| 03:25.23 | drmessano | b11d: There was NO shark. Please be clear on that. |
| 03:25.38 | b11d | What? False shark alarm? |
| 03:25.52 | b11d | Falsifying Shark Alarms is a misdemeanor here in Minnesota. |
| 03:26.07 | drmessano | No, there was no false shark alarm.. It's been clearly documented from the start that there IS and WAS NO shark |
| 03:26.24 | b11d | oh.. what about SHARRRKs? |
| 03:26.38 | russellb | *CLI> core show sharks |
| 03:26.41 | drmessano | This is how people end up thinking there's sharks in here, and that's CLEARLY NOT true |
| 03:26.48 | b11d | I freaked out.. |
| 03:26.52 | b11d | <drmessano> \o/ >'=^===< <-- SHARRRK!!! |
| 03:26.55 | b11d | that scared me |
| 03:27.08 | drmessano | [22:14] <drmessano> \o/ >'=^===< <-- SHARRRK!!! (This is a JOKE, AKA "A LOL") |
| 03:27.12 | b11d | Ohh.. |
| 03:27.14 | b11d | OHHHH |
| 03:27.16 | b11d | Now I get it |
| 03:27.17 | drmessano | I noted that is was CLEARLY a joke |
| 03:27.17 | b11d | :) |
| 03:27.36 | b11d | still... freaked me out.. |
| 03:27.43 | russellb | all i know is that i just wandered in here, looking for casual asterisk discussion, and BAM |
| 03:27.43 | drmessano | I apologize |
| 03:27.44 | russellb | there was a shark |
| 03:27.48 | b11d | I thought I was up against some kind of shark attack in here |
| 03:27.49 | *** join/#asterisk iPod-nano (n=david@c-68-43-60-193.hsd1.mi.comcast.net) |
| 03:28.12 | b11d | russellb.. you know how crazy this place gets |
| 03:28.13 | iPod-nano | What would I put in my dial plan, that would allow me to take a call while the other end is listening to MP3Player()? |
| 03:28.17 | drmessano | russellb: Perhaps a topic change is in order.. There are clearly NO sharks in here |
| 03:28.30 | russellb | i don't know if there is room in the topic |
| 03:28.34 | b11d | Should the topic be set to "Shark Free Since 10:28pm EST" ? |
| 03:28.40 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.1 (2008/02/25), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: there are no sharks in this c |
| 03:28.40 | drmessano | LOL |
| 03:28.42 | russellb | damn |
| 03:28.51 | drmessano | FYI: No Sharks! |
| 03:28.56 | drmessano | That would be sufficient |
| 03:29.00 | SHARRRK | MUHAHAHAHA |
| 03:29.02 | SHARRRK | WATCH OUT!@!@! |
| 03:29.04 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.1 (2008/02/25), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: no sharks in this channel |
| 03:29.12 | drmessano | perfect.. Thank you |
| 03:29.22 | drmessano | Eliminates the confusion |
| 03:29.37 | russellb | iPod-nano: i don't understand what you're trying to do ... |
| 03:30.13 | b11d | hahaha |
| 03:30.36 | b11d | who coopered those barrels? the staves are all out of whack.. i can swim still |
| 03:30.39 | drmessano | HAHA |
| 03:30.46 | drmessano | "We're gonna need a bigger boat" |
| 03:30.49 | drmessano | Damn good line |
| 03:30.56 | iPod-nano | russellb, a caller calls in and is greeted with a "Welcome to..." greeting, and is then played an MP3 stream while he/she waits for me to pick up. |
| 03:30.59 | drmessano | RIP Mr Scheider |
| 03:31.19 | ecto | when did Roy Scheider die? |
| 03:31.27 | drmessano | Month ago, I think |
| 03:31.29 | drmessano | Maybe less |
| 03:31.30 | ecto | ah |
| 03:31.33 | russellb | iPod-nano: use Dial with the m() option to play musiconhold instead of ringing |
| 03:31.53 | b11d | on the 10th |
| 03:32.00 | iPod-nano | Will m() use mpg123? |
| 03:32.04 | russellb | you can use the format_mp3 module to play mp3's as MOH if you're too lazy to convert them to another format |
| 03:32.11 | russellb | but i will warn you: format_mp3 crashes a lot |
| 03:32.29 | iPod-nano | The problem is that this isn't a static file I'm trying to play. |
| 03:32.38 | iPod-nano | I'm trying to play a live stream. |
| 03:32.44 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 03:32.54 | russellb | iPod-nano: you can use a live stream as a MOH source |
| 03:33.07 | drmessano | chan_shark is gonna need a bigger processor |
| 03:33.34 | russellb | it's basically the older style MOH method ... where some other app provides the MOH stream |
| 03:33.43 | russellb | so use mpg123 or whatever to read the stream and dump it to asterisk |
| 03:35.59 | iPod-nano | How do I do this, though? |
| 03:36.27 | iPod-nano | What code do I write? |
| 03:36.42 | russellb | none ... |
| 03:36.59 | russellb | i don't think i can help any more than that |
| 03:37.02 | iPod-nano | What do I put in my dialplan to make all this happen? |
| 03:37.08 | russellb | i already told you! |
| 03:37.15 | russellb | Dial with the MOH option. |
| 03:37.20 | russellb | use the stream as a MOH source. |
| 03:37.31 | russellb | read up on MOH configuration, this has been done many times |
| 03:39.46 | Nugget | I just pay a high school kid to sit on a speakerphone all day and sing showtunes. |
| 03:40.00 | Nugget | then instead of putting callers on hold I stuff them in a meetme with him |
| 03:41.54 | iPod-nano | What about making it invoke mpg123 so it can play the MP3 stream? |
| 03:42.11 | [TK]D-Fender | iPod-nano, "core show application dial" |
| 03:43.36 | russellb | vim /etc/asterisk/musiconhold.conf |
| 03:48.20 | drmessano | make samples doesn't create shark.conf |
| 03:48.55 | drmessano | I'll submit a bug report |
| 03:50.18 | ManxPower | iPod-nano: you generaly don't want mp3 MoH |
| 03:50.31 | iPod-nano | ManxPower, it's a live stream. |
| 03:52.00 | ManxPower | iPod-nano: Best of luck. |
| 04:00.47 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 04:09.38 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:10.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:10.37 | iPod-nano | I'm still confused with mpg123 here. |
| 04:10.58 | iPod-nano | I'm not sure what comand line arguments I should give it. |
| 04:11.31 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 04:12.40 | ecto | iPod-nano: man mpg123 doesn't tell you anything? I'm not sure it can do streams anyway... |
| 04:13.07 | Qwell | mpg123 is a hack. don't use it. |
| 04:13.50 | J4k3 | hack hack hack your sack |
| 04:15.19 | russellb | Qwell: you missed the back story |
| 04:15.29 | russellb | Qwell: needs to use a stream as a MOH source, in which case, mpg123 is not a hack |
| 04:15.40 | Qwell | streamplayer : |
| 04:15.41 | Qwell | :D |
| 04:15.49 | iPod-nano | Mpeg stream, Qwell. |
| 04:15.51 | russellb | streamplayer only works for raw TCP streams |
| 04:16.00 | Qwell | silly |
| 04:16.05 | russellb | most streams are http initiated |
| 04:16.11 | russellb | which wouldn't be hard to add, but meh |
| 04:18.58 | iPod-nano | Um... would Mplayer work, perhaps? |
| 04:27.30 | ManxPower | It would be nice if the musiconhold.conf.sample had an example of mpg123 usage |
| 04:29.34 | ManxPower | Here is what 1.2 uses: |
| 04:29.36 | ManxPower | <PROTECTED> |
| 04:30.36 | ManxPower | You will, as pretty much anyone on this channel will tell you, have significant issues with dropped streams. |
| 04:32.41 | LemensTS | Hey at G729 codec, and 100 concurrent calls from VOIP provider what kind of internet connection you think we need/ |
| 04:33.02 | Qwell | LemensTS: one with QoS |
| 04:33.03 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-184-1.dsl.tul2ok.sbcglobal.net) |
| 04:33.26 | J4k3 | LemensTS: 4mbit synchronous/symmetrical would be a minimum, and yes, good qos or dedicated circuits for voip. |
| 04:34.25 | LemensTS | J4k3: what is that in mb? |
| 04:34.39 | JT | LemensTS: Mbit/s is how connection speed is measured |
| 04:34.47 | ManxPower | LemensTS: IAX2 or SIP? |
| 04:34.51 | JT | LemensTS: no-one measures connection speed in megabytes per second |
| 04:34.52 | LemensTS | SIP |
| 04:34.59 | ManxPower | quite a bit. |
| 04:35.04 | ManxPower | ~bandwidth |
| 04:35.05 | jbot | somebody said bandwidth was This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus. |
| 04:35.16 | ManxPower | ~codecs |
| 04:35.17 | jbot | somebody said codecs was http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works. Anyone that tells you to use 'allow=all' is an idiot as it usually causes audio problems, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 04:35.26 | J4k3 | ~g729 |
| 04:35.27 | jbot | from memory, g729 is an ITU-standard voice codec which operates at 8kbps and offers quality very similar to GSM. G.729 is patent-encumbered; those wishing to use it with Asterisk must buy a license from Digium. |
| 04:35.44 | LemensTS | Yes $10 channel |
| 04:35.55 | iPod-nano | I've found out that Asterisk, at least when using MP3Player() has little to no difficulty with streams, so long as they're 32 kbps. |
| 04:36.01 | ManxPower | There is a web site that will help you calculate the numbers, but I don't recall *where* |
| 04:36.22 | Qwell | 13k * 100 |
| 04:36.27 | ManxPower | packeteer or something |
| 04:36.27 | Qwell | the math isn't that hard :p |
| 04:36.42 | J4k3 | Qwell: but you can't always expect everything to move smoothly. |
| 04:36.47 | J4k3 | sip isn't exactly a high-fiber diet. |
| 04:36.56 | ManxPower | Qwell: I was looking for something a tad more precise |
| 04:37.21 | LemensTS | Qwell: yea that 13k includes header overlay too. I think just G729 is 8kbps right |
| 04:37.29 | Qwell | yes |
| 04:37.37 | ManxPower | LemensTS: you can decrease bandwidth usage by increasing the sample size, but you will add latency |
| 04:37.40 | J4k3 | Qwell: that calculation says you can do 100 g729 calls over a single T1 circuit, and that just ain't gonna happen with sip |
| 04:38.06 | J4k3 | can you adjust sample size in asterisk? |
| 04:38.18 | ManxPower | I believe you can in 1.4 |
| 04:39.10 | LemensTS | ManxPower: oh yea? What kind of increase are we talking? |
| 04:39.32 | LemensTS | Manxpower: Latency would = echo right |
| 04:40.06 | JT | no |
| 04:40.12 | JT | latency is not echo |
| 04:40.27 | JT | but it can exacerbate pre-existing echo |
| 04:40.58 | JT | additional latency of using 30ms rtp instead of 20ms pales in insignificance in comparison to Internet connection latency |
| 04:42.27 | ManxPower | correct, but additional latency of using 60ms rtp is significant |
| 04:42.44 | J4k3 | depends |
| 04:42.55 | J4k3 | if you're a direcway user trying to use voip, you'd never notice the difference. |
| 04:43.09 | Qwell | J4k3: the 13k figure is sip+rtp overhead, at 20ms packetization |
| 04:45.19 | JT | ManxPower: most gear doesn't work with 60ms rtp |
| 04:46.19 | J4k3 | yeah, its not the sample size that'd be adjusted, it'd be the packetization size |
| 04:46.29 | J4k3 | ie - 1/2/4/8 samples per packet |
| 04:47.52 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 04:48.08 | J4k3 | packetization=60 ; put 60 samples in each RTP packet <- that? |
| 04:48.08 | sun_moon | hello friends |
| 04:48.29 | JT | J4k3: nah, 60ms |
| 04:48.37 | JT | 60ms of audio in a packet |
| 04:48.41 | J4k3 | yeah |
| 04:48.51 | J4k3 | that'd be 3 g729 samples per packet |
| 04:49.27 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 04:50.10 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 04:51.08 | *** join/#asterisk bmg505 (n=leon@196-209-182-222-rndf-esr-5.dynamic.isadsl.co.za) |
| 04:51.08 | *** join/#asterisk nitram (i=nitram@superblob.com) |
| 04:51.09 | *** join/#asterisk Bosse (i=bosse@miraculix.klykken.com) |
| 04:51.46 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
| 05:06.37 | J4k3 | so... asterisk under freebsd is viable? |
| 05:06.59 | drmessano | Sort of |
| 05:07.22 | drmessano | If you set up Asterisk on BSD, you become one of "Them people" |
| 05:07.25 | drmessano | Is it worth it? |
| 05:07.44 | J4k3 | its worth the ~$15/month in electricity to throw it on my entirely underused new server |
| 05:07.50 | J4k3 | that I'm having to run anyways |
| 05:08.00 | J4k3 | instead of running the old P3-800 beast I'm using now. |
| 05:08.21 | drmessano | Oh wait |
| 05:08.31 | drmessano | "HAHA THAT WAS A JOKE, LOL J/K SIKE" |
| 05:08.40 | drmessano | Now that I got that out of the way |
| 05:08.42 | drmessano | "cool" |
| 05:08.45 | JT | not as viable as linux if you have zap hardware |
| 05:09.06 | Corydon76-dig | drmessano: that would be "psych" |
| 05:09.34 | drmessano | I'm using the preferred english slang version of the word, tyvm |
| 05:09.52 | drmessano | http://www.urbandictionary.com/define.php?term=sike |
| 05:10.17 | J4k3 | I got a couple ppc 405gp@266 boards, how many calls could they take? :D |
| 05:17.12 | drmessano | ~sike |
| 05:17.13 | jbot | [Sike!] The preceeding statement was intented to be humourous, and may have involved false statements used in a humorous manner, or an exaggeration of a real life situation with humourous intent, but in no way should be taken as pure fact or not considered questionable at best. |
| 05:37.11 | *** join/#asterisk nirz (n=nir@IGLD-83-130-102-40.inter.net.il) |
| 05:39.34 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:48.22 | J4k3 | I think what I'll do is take one of my spare boxes, load fbsd 7.0-rc2 on it, and try running * on it |
| 05:48.30 | J4k3 | then if it works well, set * up on the real server |
| 05:48.38 | J4k3 | cuz, to be honest, I've never done a clean * install |
| 05:48.45 | J4k3 | :/ |
| 05:49.39 | drmessano | What have you done then? |
| 05:50.15 | J4k3 | old trixbox, switchbox, asterisknow, worked on other peoples somewhat-preconfigured self-installs... thats about it |
| 05:50.22 | drmessano | Ah ok |
| 05:50.49 | J4k3 | I know whats going on but not a lot about how it works, pretty much |
| 05:51.25 | drmessano | First time is a bitch... As you step down the list and something doesn't FAIL, write it down |
| 05:51.29 | drmessano | Then 2nd time will be easy |
| 05:51.40 | J4k3 | yeah |
| 05:52.12 | drmessano | I failed a failboat to failtown the first time I installed.. but documented as I went |
| 05:52.19 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3876966ca028ff8d) |
| 05:54.56 | drmessano | J4k3, you ever hear the 500 mile email story? |
| 05:56.09 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 05:56.31 | J4k3 | negative |
| 05:56.36 | drmessano | http://www.ibiblio.org/harris/500milemail.html |
| 05:56.39 | Robba | hi guys |
| 05:56.42 | drmessano | Its worth the read |
| 05:58.01 | Robba | when dialling out on a zaptel card originally it was dialling out on the DID numbers now its dialling out and the CID is the main number on our ISDN circuit, any ideas how to change this back to display the DID numbers? |
| 06:00.26 | J4k3 | drmessano: wtfbbq hahhaa thats great |
| 06:00.59 | drmessano | yep |
| 06:01.41 | drmessano | While users may always be wrong, sometimes you just never know |
| 06:05.08 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 06:08.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:10.15 | adeel | anyone have any experience with vicidial? |
| 06:16.40 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 06:16.47 | joelsolanki | <PROTECTED> |
| 06:17.43 | drmessano | http://downloads.digium.com/pub/asterisk/g729/ works |
| 06:20.01 | joelsolanki | hmm |
| 06:32.15 | J4k3 | what the world needs is an ilbc-like protocol with some loss tolorance but g729-ish quality |
| 06:32.57 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 06:33.44 | Robba | soo no ideas? |
| 06:42.10 | awk | hmm, if asterisk is started without ANSI Colors, why does asterisk -rn not display it? |
| 06:47.26 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.213) |
| 06:51.19 | *** join/#asterisk nirz (n=nir@mail2.tikalnetworks.com) |
| 06:55.24 | *** join/#asterisk Sweet-P (n=me@ip68-111-191-246.sd.sd.cox.net) |
| 06:55.41 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 06:56.02 | Sweet-P | Alright, I'm so sick of my Linksys WIP330 and no outbound audio...can anyone recommend a wifi sip handset that I can use while traveling? |
| 06:56.52 | endre | nokia e51 |
| 06:56.55 | endre | works for me well |
| 06:57.12 | Sweet-P | is that a gsm phone as well? |
| 06:57.15 | endre | yeah |
| 06:57.22 | endre | with native sip support |
| 06:58.44 | Sweet-P | ahh, i already have a phone that I love, for *now* I'm just looking for wifi only |
| 06:58.44 | Sweet-P | but it sounds like I may have to switch |
| 06:59.31 | endre | you'd switch if you try this |
| 07:00.08 | Sweet-P | i'm checking it out on nokia's site. my girlfriend bought me an iphone and I really like it. but I'm also moving to Italy from the uS in a couple of months |
| 07:03.51 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:04.04 | endre | uhh italy |
| 07:04.07 | endre | good luck |
| 07:04.57 | Sweet-P | i'm looking forward to it, i've been there before and I'll be making a ton of money so that's a nice thing too |
| 07:05.21 | Sweet-P | these nokia's are looking really attractive...too bad i don't see these in teh states, they'd probably do well |
| 07:06.16 | Sweet-P | i like the e51 and the e65...other than physical looks, it seems the only major diff between the two is 3G |
| 07:08.23 | Sweet-P | wow, the e51 is going for about US$300 |
| 07:09.54 | endre | is that expensive? |
| 07:10.15 | Sweet-P | about 150 euros |
| 07:10.25 | Sweet-P | wait.. |
| 07:11.14 | Sweet-P | 200 euros |
| 07:11.46 | craigk | anybody ever had a problem where bulk voice mail messages are generated - all blank. I am litterally getting one every minute. It started with a real call where a 10second blank messages was left, and then the messages just keep appearing even though the call has terminated :/ |
| 07:12.07 | J4k3 | Sweet-P: when you move to the US will you be living in a city? |
| 07:12.12 | J4k3 | or rural? |
| 07:12.31 | J4k3 | in many places in the US, GSM carriers tend to lack 'solid' coverage |
| 07:12.35 | Sweet-P | J4k3, I live in the US now, i will be moving to italy (Naples) in a few months |
| 07:12.42 | J4k3 | ahh |
| 07:12.48 | Sweet-P | but I know what you mean, ATT 3G is so-so |
| 07:12.53 | J4k3 | so unlock your iphone and take it with you |
| 07:13.22 | Sweet-P | i'm att customer, but I unlocked so that I could take it, but the iphone voip application is *currently* REALLY lacking |
| 07:13.32 | J4k3 | yeah |
| 07:13.33 | Robba | can someone help cause i need to get this issue sorted ASAP |
| 07:13.57 | Sweet-P | i wanted to run asterisk at home and use my wifi handset while in the house, then be able to take it to the cafe or hotel, etc |
| 07:14.00 | drmessano | My boss.. Who is a cool guy.... except doesnt know a lot of tech.. |
| 07:14.21 | drmessano | Says "You would think a new $500 phone could do all the latest and greatest" |
| 07:14.31 | drmessano | One word "iphone" |
| 07:14.33 | J4k3 | Sweet-P: wifi-in-your-ear doesn't tend to work very well from what I've found |
| 07:14.53 | Sweet-P | hehe...nah...gotta pay for upgrades to the apple-monopoly phone |
| 07:15.25 | J4k3 | I've owned a couple dedicated voip-o-wifi handsets and a wm6 pocketpc with wifi, and they all suck about equally |
| 07:15.25 | drmessano | He couldnt believe the iPhone doesn't work with MS Exchange email |
| 07:15.25 | Sweet-P | j4k3, i used a softphone (xlite) on my laptop and the service was great |
| 07:15.34 | J4k3 | Sweet-P: your laptop has a real antenna and isn't crammed against your head. |
| 07:15.43 | drmessano | Exactly |
| 07:15.49 | Sweet-P | i had an HTC wm6 phone and skype (i know bad word) and it worked great over wifi |
| 07:15.56 | J4k3 | and about 5x the transmit power of a typical wifi equipped handset |
| 07:15.58 | Sweet-P | i know it's not sip, but it sounded fine |
| 07:16.07 | drmessano | It's also not using 30% CPU and half it's battery drain to keep the Wifi going |
| 07:16.13 | Sweet-P | true true |
| 07:16.17 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 07:16.25 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 07:16.31 | Sweet-P | i guess i'm trying to over-geek myself |
| 07:16.39 | J4k3 | yeah, my 6700 gets 4 hours wifi-on with a new high capacity battery |
| 07:16.49 | J4k3 | er xv6700 (htc apache) |
| 07:16.51 | drmessano | Handheld + Wifi is like a peanut butter and squid sandwich |
| 07:17.02 | Sweet-P | hehe...yuk |
| 07:17.38 | J4k3 | Sweet-P: get a good bluetooth earpiece and a good usb bluetooth adapter for your desktop, and a usb extension cable to get the bluetooth away from your computer for better rf coverage |
| 07:18.08 | J4k3 | run a softphone or attach asterisk directly to the earpiece and figure out some method to use it (voice dial software?) |
| 07:18.16 | drmessano | J4k3, im gonna put a USB dongle on the end of a 10 foot extension and tape it to the ceiling |
| 07:18.18 | J4k3 | but, you'll be happier |
| 07:18.21 | J4k3 | or a DECT phone |
| 07:18.31 | J4k3 | drmessano: that works well |
| 07:18.36 | drmessano | Yep |
| 07:19.00 | Sweet-P | my voip provider is sending me an adapter (linksys SPA-2102) and I'll get a dect phone since I've heard so much good stuff about them |
| 07:19.14 | Sweet-P | that's for home |
| 07:19.22 | J4k3 | I do the same thing with usb wifi adapters. got a bunch of good 10' usb2 extension cables for $0.99/ea |
| 07:19.24 | drmessano | The one I have on my * box isn't too bad now, but I am a too much of a ham to settle for "plugged in the back" |
| 07:20.04 | J4k3 | some of that strip velcro + wall = good coverage |
| 07:20.18 | Sweet-P | i've got asterisk on one machine, trixbox on another and 3cx on yet another and i'm trying to see which one "works better", so far I can't really tell the differenc |
| 07:20.41 | J4k3 | drmessano: if its a metal chassis, you'll see like 5-10x the coverage getting it out of the back of the PC |
| 07:20.41 | drmessano | I'll settle when I scan for devices and I pick up 3 or 4 of my neighbors bluetooth devices on their tables next to their beds at night |
| 07:20.46 | Sweet-P | my setup is extremely simple task for any of them anyway |
| 07:21.14 | J4k3 | drmessano: hack your bt adapter open and/or get a bt radio with an antenna plug, and get a nice 2.4 ghz 'wifi' antenna |
| 07:21.18 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 07:21.19 | drmessano | Its a Dell with the indented back.. so its damn near surrounded on 3 sides by metal |
| 07:21.34 | Sweet-P | for bluetooth, make sure to get one of those class c (?) models for super range |
| 07:21.35 | J4k3 | oh lord, thats definetly more on the 10x side ;) |
| 07:21.40 | drmessano | Oh yes |
| 07:21.58 | J4k3 | Sweet-P: yep, and get a good earpiece. |
| 07:22.07 | *** part/#asterisk Robba (n=rob@203.56.181.15) |
| 07:22.22 | J4k3 | only buy stuff that lists its tx and rx performance in standard units |
| 07:22.45 | Sweet-P | recommendation? |
| 07:22.53 | drmessano | J4k3: You can't convert decibels to fluid ounces? |
| 07:23.04 | Sweet-P | the only bluetooth i have now is one of those jawbones that she got me with my iphone |
| 07:23.14 | J4k3 | I've seen bt earpieces with txpower anywhere from -10 dBm to +10 dBm, and rx performance from -74 (awful!) to -90 dBm (quite good) |
| 07:23.46 | J4k3 | I've seen usb adapters with txpower from +4 dBm to +20 dBm, and similar rx as the earpieces |
| 07:23.57 | J4k3 | typically 6 dB is a doubling in coverage |
| 07:24.15 | J4k3 | so the more txpower and less required rxpower, the better off you are |
| 07:24.46 | J4k3 | and make sure you buy a bt earpiece that supports modern protocols |
| 07:25.16 | J4k3 | you can still find bt1.1 stuff on the market cheap, but it eats its battery in 4 hours and only works for 2 meters. |
| 07:25.42 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:27.18 | *** join/#asterisk techie (n=techie@adsl-76-214-12-55.dsl.lsan03.sbcglobal.net) |
| 07:27.50 | siya | how can I best debug this: |
| 07:27.53 | siya | chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call |
| 07:27.56 | drmessano | You can find cheap 2.0 stuff on eBay now though, so it's not so bad |
| 07:28.05 | drmessano | Avoid Big Lots and flea markets |
| 07:28.13 | J4k3 | yep |
| 07:28.14 | siya | it gives me an id but on the same line it says "Giving up" |
| 07:28.18 | drmessano | Where the 1.1 stuff ends up |
| 07:28.20 | J4k3 | just keep your eyes open and be careful what you buy |
| 07:28.25 | Sweet-P | good to know |
| 07:28.34 | siya | so I can't display that sessions anymore so have no idea which IP/ext/etc originates it |
| 07:28.46 | J4k3 | Sweet-P: I believe the iphone can work as a sip client |
| 07:28.55 | J4k3 | err sip earpiece |
| 07:29.08 | Sweet-P | how so? |
| 07:29.28 | Sweet-P | sip client app still sucks for now |
| 07:29.40 | J4k3 | you can connect it to another phone or a pc running a softphone with a bt adapter, and use its mic/earpiece |
| 07:29.49 | Sweet-P | ahh |
| 07:29.54 | J4k3 | I'm talking bt client, not sip client. let the pc do the sip grunt work |
| 07:30.13 | Sweet-P | might be interesting |
| 07:30.58 | drmessano | I've used my cell as an earpiece for X-Lite before |
| 07:31.04 | drmessano | It's not bad |
| 07:32.31 | Sweet-P | so is this asteriskNOW like "asterisk for dummies"? |
| 07:32.47 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:32.48 | J4k3 | yeah |
| 07:32.56 | J4k3 | its a quick way to learn to hate asterisk |
| 07:33.00 | J4k3 | :) |
| 07:33.00 | Sweet-P | cool, i'll give it a test |
| 07:33.05 | Sweet-P | hehe |
| 07:33.34 | *** join/#asterisk sl00 (n=sl00@static-213-115-8-8.sme.bredbandsbolaget.se) |
| 07:34.47 | sl00 | Hi. My employees are complaining about a silence at the beginning of each call. The other end picks up but my employees cannot hear anything for like 1-2 seconds. What could be the problem? Where are only using softphones. |
| 07:35.11 | J4k3 | sl00: I've seen this on rare occasion and not sure why it occurs. |
| 07:35.55 | J4k3 | usually on calls from cellular phones, I've suspected an echo cancelation issue |
| 07:36.16 | J4k3 | after anywhere from 2 to 15 seconds, the problem goes away and the call works fine |
| 07:37.07 | sl00 | J4k3: Yes. Before it happened sometimes but now it seems like it occurs all the time. Yesterday and the day before we had problems with the phones suddenly just dropped the connection. Then one customer said they heard us all the time so it seems our audio just dropped for some reason. So weird. It have been working fine for month ad nothing has changed in config or anything. |
| 07:37.49 | sl00 | So yesterday I created my own config files from scratch and really basic. But could the 1-2 second ilence really be a extension config issue? |
| 07:38.14 | drmessano | What kind of phones? |
| 07:39.03 | sl00 | J4k3: I just checked a call and it was not a cellularphone target. |
| 07:39.17 | sl00 | drmessano: Ekiga on Linux/Slackware and X-Lite on Windows. |
| 07:39.52 | sl00 | Ekiga showed the audio status as GSM then it went away for a second and reappeared. |
| 07:39.54 | sl00 | Weird. |
| 07:40.06 | J4k3 | codec negotiation issue |
| 07:40.34 | sl00 | Okey? |
| 07:41.51 | sl00 | How can I investigate that further? |
| 07:42.00 | *** join/#asterisk oej (n=olle@213.89.171.130) |
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| 07:52.06 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:53.21 | sl00 | Can I reload the config without Asterisk dropping ongoing calls? |
| 07:54.41 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 07:59.12 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.77.211) |
| 08:00.36 | mvanbaak | sl00: yeah, just issue 'reload' on the cli |
| 08:01.36 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 08:02.05 | drmessano | Is a SRV record for IAX2 _iax2._udp ? |
| 08:03.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:04.35 | *** join/#asterisk frk2 (n=fkhan@117.20.28.50) |
| 08:04.51 | frk2 | anybody using res_config_ldap from 1.6.0beta 2/4 ? |
| 08:05.39 | frk2 | guess everyone is sleeping :P |
| 08:09.12 | *** join/#asterisk rvhi (n=chatzill@udp186710uds.hawaiiantel.net) |
| 08:09.37 | rvhi | when i do a 'ps aux | grep asterisk', i only see one process. is it normall? |
| 08:09.46 | drmessano | I found a patch from 06 for adding SRV to chan_iax2.c |
| 08:10.03 | drmessano | But I can't find any reference to it in chan_iax2.c now |
| 08:17.52 | drmessano | Ok, bingo |
| 08:19.22 | *** join/#asterisk rati (n=chatzill@124.125.76.195) |
| 08:20.35 | rati | Hey I have installed asterisk successfully, and configure it but I want to make my own gUI. so what should I consider as requirement? |
| 08:32.50 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.77.211) |
| 08:33.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:41.24 | J4k3 | hrm FreeBSD 7.0 went release? |
| 08:41.24 | J4k3 | heh |
| 08:43.32 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-184-1.dsl.tul2ok.sbcglobal.net) |
| 08:43.42 | sl00 | Gaaah!! |
| 08:43.46 | *** join/#asterisk zagaya (n=almeli@LPointe-a-Pitre-151-14-9.w81-248.abo.wanadoo.fr) |
| 08:44.16 | sl00 | Now my connection to the provider just died. It tries to register but fails. I called them and it seems they get a register reguest and respond with a auth request but the Asterisk is not sendinth any auth ?!? |
| 08:44.27 | sl00 | I have not changed anything in the config files so how can this just happend like this? |
| 08:44.46 | sl00 | I have tried to restart the Asterisk aswell but no help. |
| 08:46.55 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 08:48.38 | tzafrir | rvhi, yes. there's indeed only one Asterisk process. |
| 08:48.53 | tzafrir | Did you expect to see a line per thread? |
| 08:49.02 | rvhi | ya |
| 08:49.33 | tzafrir | rvhi, in linux this is something that has changed between linux 2.4 and linux 2.6 - the threading implementation (NPTL) |
| 08:49.50 | tzafrir | To see threads, press H in top |
| 08:50.18 | rvhi | how about in ps? |
| 08:50.52 | tzafrir | ps eLf or amux |
| 08:51.12 | tzafrir | hmm... any better word? |
| 08:51.35 | rvhi | :) that works! |
| 08:52.04 | tzafrir | amux, maxu, maux, umax . None of them is a real word :-( |
| 08:52.34 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 08:53.39 | frk2 | I'm hitting the wall with res_config_ldap- anybody tried it on their system with the new asterisk 1.6.0 beta 4? |
| 08:55.55 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 08:57.27 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
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| 09:19.04 | tsubasafr | hi ! |
| 09:19.34 | tsubasafr | Someone know how to debug this in asterisk ? [Feb 27 10:19:05] WARNING[6534]: chan_sip.c:6705 determine_firstline_parts: Bad request protocol Packet |
| 09:19.42 | tsubasafr | I have this warning every 4-5 secondes |
| 09:27.12 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
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| 10:08.53 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:10.40 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 10:12.30 | rati | Hey I have setup asterisk and its working fine. now I want to put it on live. so what should I do? means how can I make calls to other country or otherthan my machine users |
| 10:18.39 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:20.40 | *** join/#asterisk [JFET] (n=alma@this.is.very.insecure.hu) |
| 10:21.12 | *** part/#asterisk techie (n=techie@adsl-76-214-12-55.dsl.lsan03.sbcglobal.net) |
| 10:22.22 | rati | Anyone tell me how can I achive that? |
| 10:23.34 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 10:36.39 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
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| 11:03.48 | budol | hi rati |
| 11:04.56 | budol | rati : do you have Public IP or your * box behind the firewall? |
| 11:08.59 | *** join/#asterisk guillote_GNU (n=guillote@host26.201-253-58.telecom.net.ar) |
| 11:12.52 | *** join/#asterisk steliosk (n=Stelios@athedsl-117347.home.otenet.gr) |
| 11:16.59 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
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| 11:21.26 | *** join/#asterisk FreezeS (n=bruno@82.208.157.125) |
| 11:21.34 | FreezeS | hey guys |
| 11:21.52 | FreezeS | how can I make asterisk hang up calls if a sip client hasn't updated its registration ? |
| 11:22.23 | [JFET] | in extensions.conf --> 10 call(sip,somebody) 20 hangup() |
| 11:22.36 | [JFET] | or something like that. append a hangup at the end. |
| 11:22.56 | FreezeS | my problem is that sometimes a client crashes and doesn't send the hangup message |
| 11:23.16 | FreezeS | and if it has a call going on, that call takes forever |
| 11:28.57 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 11:41.39 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 11:45.57 | budol | FreezeS : what do you mean a channel lock? |
| 11:46.10 | *** join/#asterisk dewd (n=dewd@sexy.demon.nl) |
| 11:47.23 | dewd | Hi I am using cdr in an psql db but billsec doesn't seem right, when looking at an cdr record for an call with an estimated lenght of 3 secs it says 36 |
| 11:47.31 | dewd | what could this be ? |
| 11:51.56 | RoyK | dewd: is that billsec or duration? |
| 11:54.03 | dewd | billsec |
| 11:54.29 | dewd | the doc says it should be seconds |
| 11:55.47 | dewd | btw it would be both |
| 11:57.56 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:05.21 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 12:14.39 | FreezeS | budol: if I dial an IVR number that never hangs up and then I kill the softphone, the channel remains on, even after the registration times out |
| 12:15.07 | FreezeS | I need to hangup all the calls of a sip user when the registration times out |
| 12:17.12 | budol | FreezeS : do you have zaptel card or something? |
| 12:17.46 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 12:19.50 | FreezeS | budol: yes, I've got a PRI card |
| 12:19.58 | FreezeS | but this is a SIP issue |
| 12:20.21 | FreezeS | I need to detect that a client hasn't re-registered and hangup all his calls |
| 12:20.35 | dewd | so what could this issue with the cdr seconds be ? has something changed or someting ? |
| 12:20.44 | dewd | looks like it is secs * 10 |
| 12:22.28 | FreezeS | what exactly does "defaultexpirey" does ? |
| 12:22.33 | FreezeS | *do |
| 12:22.35 | tzanger | morning |
| 12:27.56 | Dovid | how would i set CALLER ID on an E1 ? the same as i set it for SIP (SETCALLRID(NUM) ? |
| 12:28.33 | FreezeS | Dovid: it's something like set(CALLERID(num)=12345) |
| 12:29.05 | Dovid | FreezeS: Ok. I did it that way and my carrier says they are not getting anything on their end |
| 12:30.57 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 12:32.12 | FreezeS | Dovid: are you sure you reloaded the config file ? |
| 12:32.31 | FreezeS | this is how I do it and it works just fine... |
| 12:32.38 | *** join/#asterisk whymarkwhy (n=koos@196.211.34.2) |
| 12:32.49 | lmadsen | there isn't anything more special than that I've ever seen... |
| 12:33.05 | lmadsen | I've forgotten a dialplan reload or two in my time after making changes though :) |
| 12:33.21 | Dovid | FreezeS: posittive |
| 12:34.03 | FreezeS | Dovid: can you paste the console output for a call ? |
| 12:34.08 | FreezeS | in pastebin |
| 12:37.14 | whymarkwhy | anyone tried voicetronics cards |
| 12:37.16 | whymarkwhy | ? |
| 12:37.39 | whymarkwhy | and cards for pika tech?? |
| 12:40.15 | Dovid | FreezeS: http://pastebin.ca/920447http://pastebin.ca/920447 |
| 12:41.00 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 12:41.42 | *** join/#asterisk ocgltd (n=support@CPE004063e0ee74-CM00159a010632.cpe.net.cable.rogers.com) |
| 12:42.55 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:43.04 | ocgltd | Can someone help with a parkedcalls problem? I have included parkedcalls context in my dialplan, and I can successfully transfer calls to the parking lot (700). However, trying to pickup a call from the parking lot results in matching 701 to my general dialout extension. How can I force the parkedcall context to be matched first? |
| 12:43.06 | FreezeS | Dovid: from what I see, everything looks OK. Maybe you should play around with zaptel.conf |
| 12:43.23 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.77.211) |
| 12:44.26 | zeeesh | using xlite , using DeadAGI, dialing out, when receiver disconnect the call ..but xlite does't disconnect until i disconnect ... means if receiver hangup but no notification at caller side???? |
| 12:44.45 | Dovid | FreezeS: what i want to change in zaptel.conf ? |
| 12:45.43 | FreezeS | Dovid: actually in zapata.conf, hidecallerid = no |
| 12:47.43 | Dovid | FeeezeS: Already have that |
| 12:47.44 | Dovid | usecallerid=yes |
| 12:47.44 | Dovid | hidecallerid=no |
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| 12:50.29 | Dovid | FreezeS: in the PRI debug I get Presentation: Number not available (67) '0788187000' ] |
| 12:50.37 | Dovid | that is what i am sending as CID |
| 12:51.38 | FreezeS | Dovid: did your telco assign you that number ? when I tried CLI spoofing, it didn't work with my telco |
| 12:53.29 | Dovid | FreezeS: Yes. that numbe is mine. |
| 12:53.38 | Dovid | is that error message coming from the telco ? |
| 12:54.03 | FreezeS | that's what I assume |
| 12:54.16 | FreezeS | try setting the cli without the prefix |
| 12:56.14 | Dovid | FreezeS: No matter what I send them they are saying they are not getting anything for the CLI |
| 12:56.38 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:56.57 | Dovid | FreezeS: Think its something on their end ? |
| 12:57.23 | lmadsen | ocgltd: you need to have a context that you start your dialing in, place anything in that context within another context, then include it AFTER the parkedcalls context |
| 12:57.40 | Dovid | http://pastebin.ca/920460 |
| 12:57.48 | lmadsen | ocgltd: anything within the context will match FIRST, because includes are only checked after attempting to match in the starting context first |
| 12:57.51 | Dovid | trying to figure out if the 67 error is from asterisk or from the telco |
| 12:58.11 | Dovid | lmadsen: the error in my pb (67) is that coming from asterisk or my carrier ? |
| 12:58.25 | lmadsen | Dovid: I have no idea, I don't hardware |
| 12:58.31 | lmadsen | I don't do hardware* |
| 12:59.25 | whymarkwhy | anyone used the webvmail before?? |
| 12:59.46 | yang | <PROTECTED> |
| 13:00.29 | FreezeS | Dovid: pridialplan=local |
| 13:00.33 | FreezeS | is it setup like that ? |
| 13:01.02 | FreezeS | also, do you know your switchtype ? |
| 13:01.25 | yang | hey lmadsen my congrats for the good asterisk book ! |
| 13:01.35 | lmadsen | heh, thx :) |
| 13:01.37 | lmadsen | glad you like it |
| 13:02.05 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.111.94) |
| 13:02.06 | yang | lmadsen: I slowly started reading it, however when I manage to get some time :) |
| 13:02.08 | Dovid | pridialplan=local |
| 13:02.13 | Dovid | gets set in zapata.conf ? |
| 13:02.17 | lmadsen | time is always the factor for sure |
| 13:02.27 | lmadsen | or else I'd already have written a cookbook and a 3rd edition of TFoT... |
| 13:02.41 | yang | ;) |
| 13:02.50 | FreezeS | Dovid: yes |
| 13:02.52 | lmadsen | the cookbook should already be done, and we haven't even started, sadly enough |
| 13:03.25 | yang | lmadsen: Are all O'Reilly books freely available on the internet or just those which have authors permission ? |
| 13:03.47 | endre | it's a good question |
| 13:03.51 | lmadsen | yang: just those which the authors have asked O'Reilly to publish freely, or which are very old :) |
| 13:03.56 | Dovid | FreezeS: i forgot whats next. just an asterisk restart or do i need to modprobe again as well ? |
| 13:04.04 | lmadsen | we're certainly the exception, not the rule |
| 13:05.06 | yang | lmadsen: you know the Jokes about scottsmans, how they are tight with the money, we call them "Gorenjci" its a district in our country, and my boss comes from there, otherwise I would tell him to order the printed copy :) |
| 13:05.49 | lmadsen | heh... ya, I'm cheap too, which is why I do real work instead of writing books because I make a lot more per hour doing consulting than I do writing books |
| 13:05.54 | lmadsen | which is unfortunate |
| 13:06.08 | Dovid | FreezeS: That didnt help :( |
| 13:06.16 | *** join/#asterisk lirakis (i=lirakis@pr0tected.us) |
| 13:06.23 | lmadsen | $2k over a period of 3 months... or $2k over a period 1 week.... that's basically the balance |
| 13:06.32 | yang | lmadsen: you don't have to tell me, my father is a writer also... |
| 13:07.10 | yang | In our country you cannot survive by being an author (sadly) |
| 13:07.12 | lmadsen | if I had like 4-5 books on the go at a time, then I'd be all set... the problem is spending the initial year or so full time with no money to write 5 books so you can get that buffer built up |
| 13:07.38 | lmadsen | yang: I'm not sure how many countries you can survive being an author in, unless you're writing fiction, and are VERY good at it |
| 13:07.50 | lirakis | lmadsen: or get on oprah |
| 13:07.50 | lmadsen | like... Stephen King I'm sure does alright :) |
| 13:07.52 | lirakis | ;P |
| 13:07.53 | yang | Yeah something like Harry Potter |
| 13:08.11 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:08.58 | lmadsen | I've always wondered if I'd make more money buying a bunch of books at author discount, and then selling them on my website |
| 13:09.33 | lmadsen | never bothered since the hassle in doing a shipment run for 1 book wouldn't really make it worth it I don't think |
| 13:09.43 | lirakis | lmadsen: i guess it depends on the author discount... and .. there is probably some stipulation of no resale |
| 13:09.49 | coppice | only write book for dummies. it ensures a large potential audience |
| 13:09.58 | yang | My father is selling some of his books to students |
| 13:10.05 | lmadsen | lirakis: that is entirely plausible |
| 13:10.24 | lmadsen | coppice: why do you think I picked Asterisk? :D |
| 13:10.27 | yang | And I do run a publishing company, so all the money stays in the family |
| 13:10.46 | lirakis | lmadsen: do you thinik atfot sells less b/c its available as a pdf? |
| 13:10.58 | yang | But releasing a book every 3 years, doesnt make any significiant income |
| 13:11.50 | lmadsen | lirakis: nope, it actually sells very well, but when you have to split royalties 3 ways, that doesn't help, and tech books don't make you rich to begin with. TFoT1 was actually in the top 10 best selling books for O'Reilly last year |
| 13:11.52 | lirakis | lmadsen: im curious about alt. media and distribution in publishing... i know oreilly is pretty far out in front with that.. I wondered about the commercial implications |
| 13:11.53 | coppice | it sells less because its called the future of telephony, so people are waiting until it arrives and they need the book |
| 13:11.54 | lmadsen | and I still didn't buy a house or anything :) |
| 13:12.10 | lmadsen | lirakis: we looked at it, and it didn't seem to hurt sales, and maybe even helped |
| 13:12.12 | lirakis | lmadsen: i know i bought a copy (of both editions) :D |
| 13:12.25 | lirakis | lmadsen: yeah... i have a safari subscription |
| 13:12.31 | lirakis | lmadsen: and i bet it only makes me buy more books |
| 13:12.48 | lmadsen | lirakis: exactly -- people don't really want to read a book online anyways... if they were going to buy it at all, they'll buy it even if a PDF is available |
| 13:13.05 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 13:13.08 | lirakis | lmadsen: b/c the online format is okay... but it really just helps me find out what book i need faster |
| 13:13.11 | lmadsen | it's like piracy... I might pirate WinXP, but if I couldn't, I still wouldn't buy it |
| 13:13.25 | lmadsen | lirakis: that's what I figure, and it's nice when you want to search for something |
| 13:13.35 | coppice | I've come to hate printed book. they are easily readable on the couch or the train, but the search function sucks |
| 13:13.35 | lirakis | lmadsen: if a pdf were available at a discounted price, id just buy that.. as long as it came with single printing rights |
| 13:13.44 | lirakis | lmadsen: thats my biggest gripe with safari. |
| 13:14.05 | lmadsen | coppice: :D |
| 13:14.25 | Dovid | FreezeS: may i PM ? |
| 13:14.59 | lirakis | lmadsen: I mean.. i pay $400/yr ... and you cant buy a pdf... only the dead tree book... and the "discounted" safari price is still more expensive than amazon's price... forget used/new prices... those are WAY lower |
| 13:15.00 | lmadsen | lirakis: heh ya... I wonder how much something like that would be... although usually printing something like that would cost you more than buying the book,... no? (once you factor in personal printing costs?) |
| 13:15.04 | FreezeS | yes |
| 13:15.39 | lirakis | lmadsen: it depends. I've got a effecient laster printer. With toner and paper costs figured in, its below $0.02 per page double sided |
| 13:15.54 | lmadsen | lirakis: gotcha... well that's convenient :) |
| 13:16.02 | lirakis | lmadsen: and i can bind books too... but i dont bother with any thing over 500 pages really.. so there is a limit |
| 13:16.35 | lmadsen | welp, I'm off to get ready for my on-site meeting with some people who tend to waste my time at these stupid meetings because they don't know what they should ask, and when they do get a question right, they tend to ask it over and over again, even when I give the same answer |
| 13:16.38 | lmadsen | </rant> |
| 13:16.46 | coppice | even war and peace was originally printed in 4 volumes :-) |
| 13:16.46 | lirakis | lmadsen: later |
| 13:17.09 | lmadsen | lates |
| 13:20.18 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:20.41 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.204) |
| 13:21.24 | tzanger | lmadsen: nice |
| 13:24.22 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 13:26.30 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 13:26.44 | Nivex | http://www.10dailythings.com/2007/10/20/funny-computing-image-you-computer-is-at-risk/ |
| 13:30.17 | dandre | hello, |
| 13:31.14 | dandre | Is there any option to pass to zaptel module during modprobe for international (french) installation? |
| 13:32.05 | dandre | I have fase ring events sent by my wx24xxx driver |
| 13:32.56 | tzafrir | dandre, ring detection or ring generation? |
| 13:33.25 | dandre | ring detection on my fxo |
| 13:34.20 | dandre | I have lines such as: |
| 13:34.26 | dandre | RING on 1/7! |
| 13:34.26 | dandre | NO RING on 1/7! |
| 13:34.32 | dandre | after hungup |
| 13:35.14 | tzafrir | I think that there was a fix to that driver yesterday about that |
| 13:35.48 | tzafrir | http://svn.digium.com/view/zaptel?view=rev&revision=3883 |
| 13:36.55 | tzafrir | Right after you pick up or hang up the current is still not steady. |
| 13:37.04 | *** join/#asterisk synthetiq (n=roger@unl201395.nl.customer.alter.net) |
| 13:37.50 | synthetiq | is there a way i can send a certain failed sip call that got specifically 482 loop detected to another context, where as all other errors go to my voicemail cotnext? |
| 13:38.06 | tzafrir | And it changes from + to -. Might look like a ring. |
| 13:38.36 | tzafrir | Isn't analog telephony fun? :-( |
| 13:39.59 | tzafrir | synthetiq, don't call yourself |
| 13:43.04 | synthetiq | tzafrir im changing the uri with ser |
| 13:43.18 | synthetiq | rahter the domain of it |
| 13:43.35 | whymarkwhy | anyone used the webvmail for asterisk voicemail before?? |
| 13:43.37 | rati | budol: sorry I was not at my desk. but I dont have public ip, my box is behind firewall |
| 13:44.06 | synthetiq | so i call user@domain.com to ser, if ser doesnt find that use it returns the call to asterisk user@voicemail.domain.com |
| 13:44.51 | synthetiq | i hacked the chan_sip.c file before in 1.2, but when the calls ended asterisk seg faulted, wondering if there is somethign enw in 1.4 to handle this |
| 13:52.39 | dandre | ok tzafrir, I'll try that |
| 13:53.56 | Dovid | morning TK |
| 13:54.00 | Dovid | am i glad to see you ;) |
| 13:54.55 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:55.13 | *** join/#asterisk Duplicator (n=Duplicat@proxy.fmpa.com) |
| 13:55.28 | stansmith | ming_zym: did you try Read() ? |
| 14:02.59 | stansmith | ${CHANNEL} corresponds to zap/1, zap/2, sip2342342, whatever channel is currently being used?? [confirm/deny] |
| 14:05.03 | FreezeS | can anyone tell me how can I hangup a sip client's calls when his registration times out ? |
| 14:05.53 | [TK]D-Fender | stansmith: that is an old var, use the CHANNEL function. |
| 14:05.54 | *** join/#asterisk nighty^ (n=nighty@p9120-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 14:06.19 | stansmith | [TK]D-Fender: thats dank |
| 14:06.26 | stansmith | (thanks) |
| 14:06.27 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:06.42 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:07.07 | [TK]D-Fender | stansmith: Dank : Disagreeably damp or humid. See Synonyms at wet. |
| 14:07.10 | Katty | is anyone from the las vegas area? |
| 14:07.16 | Katty | [TK]D-Fender: i have a job offer. |
| 14:07.25 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 14:07.30 | [TK]D-Fender | Katty: Showgirls, my favourite!!!! |
| 14:07.35 | Katty | [TK]D-Fender: not quite. |
| 14:07.36 | [TK]D-Fender | :p |
| 14:07.41 | Katty | [TK]D-Fender: more like citibank corporatein las vegas. |
| 14:07.51 | Katty | [TK]D-Fender: i'm skeered. |
| 14:07.58 | stansmith | see synonyms: great things that are green |
| 14:09.26 | *** join/#asterisk ddunavant (n=David@pool-71-191-24-37.washdc.east.verizon.net) |
| 14:11.34 | stansmith | ${CHANNEL(channeltype)}, in case anyone else runs into the same issue as me |
| 14:12.09 | [TK]D-Fender | stansmith: What "issue"? |
| 14:12.12 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:12.29 | stansmith | ${CHANNEL} var vs ${CHANNEL()} function |
| 14:12.46 | stansmith | looking at voip-info.org, i would of had no clue that function is new in 1.4 |
| 14:12.54 | *** join/#asterisk DJF6 (n=DJF5@84-105-201-37.cable.quicknet.nl) |
| 14:12.56 | [TK]D-Fender | stansmith: That isn't an issue, thats a list of the "players". |
| 14:13.08 | [TK]D-Fender | stansmith: And the WIKI is often outdated. |
| 14:13.16 | stansmith | now you tell me |
| 14:13.28 | [TK]D-Fender | stansmith: Common knowledge |
| 14:13.38 | stansmith | common knowledge isnt really all that common |
| 14:13.59 | [TK]D-Fender | stansmith: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+channel |
| 14:14.05 | ManxPower | stansmith: "core show applications" "core show functions" |
| 14:14.09 | [TK]D-Fender | stansmith: Gets/set various pieces of information about the channel. (New in 1.4) |
| 14:14.14 | stansmith | [TK]D-Fender: already read it, thanks |
| 14:14.16 | [TK]D-Fender | stansmith: Looks pretty blatant to me. |
| 14:14.39 | stansmith | http://www.voip-info.org/wiki-Asterisk+variables |
| 14:14.49 | stansmith | thats what i was looking at before you said use the function |
| 14:15.04 | ManxPower | stansmith: no, you would read about channel variables from channelvariables.txt in the Asterisk source code. |
| 14:15.50 | DJF6 | if the wiki is outdated... where does everybody get their info? |
| 14:16.20 | [TK]D-Fender | DJF6: Bits from there, more from the docs included with the source, and here. |
| 14:16.28 | stansmith | i mean...you cant really blame me for using google... |
| 14:16.33 | [TK]D-Fender | DJF6: And of course... THE BOOK |
| 14:16.39 | DJF6 | hm, yes |
| 14:16.48 | ManxPower | DJF6: there's an entire directory in the Asterisk source code called "docs" |
| 14:16.50 | DJF6 | i've read some pages of the book |
| 14:17.19 | DJF6 | but there are many (shortnames?) like ABC or DEF and i don't know what they mean |
| 14:17.32 | ManxPower | stansmith: If I could wipe the entire Wiki and start over, I would. |
| 14:17.37 | DJF6 | ah, source :) i've apt-getted it |
| 14:17.38 | tzafrir | ~wiki DEF |
| 14:17.40 | [TK]D-Fender | DJF6: huh? |
| 14:17.49 | stansmith | i think he meant apt-got |
| 14:18.09 | tzafrir | DJF6, now you know what DEF stands for |
| 14:18.14 | DJF6 | yeah, english isn't my 1st language |
| 14:18.22 | DJF6 | i ment long words shortened |
| 14:18.24 | stansmith | french? |
| 14:18.30 | DJF6 | i don't know the word |
| 14:18.31 | DJF6 | dutch |
| 14:18.35 | [TK]D-Fender | DJF6: please give some sort of example for what you mean by "shortnames" |
| 14:18.39 | DJF6 | uhm |
| 14:18.42 | ManxPower | DJF6: "abbreviations" |
| 14:18.49 | DJF6 | yes! |
| 14:18.53 | tzafrir | that is: wikipedia is generally a good reference for Telephony terms |
| 14:19.04 | ManxPower | or 'Three Letter Acronyms' |
| 14:19.17 | DJF6 | yes, there are a lot in that book |
| 14:19.25 | DJF6 | most of them don't ring a bell at all :( |
| 14:19.38 | DJF6 | *as a way of speaking ;) |
| 14:19.56 | ManxPower | "est". is an abbreviation for "established" IBM is an acronym for International Business Machines |
| 14:19.57 | [TK]D-Fender | DJF6: Thats what the glossary is for. |
| 14:21.43 | stansmith | svn = subversion |
| 14:24.24 | stansmith | there was this dude in california that got caught prank calling 911. he did it a couple thousand times, and when they caught him, they asked him why he did it so much and he said "cause its free" |
| 14:24.28 | stansmith | classic |
| 14:24.38 | DJF6 | lol |
| 14:24.49 | DJF6 | just remove your SIM card :p |
| 14:25.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:28.56 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:29.09 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:33.29 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 14:34.18 | yang | how can i temporary unload zaptel from asterisk |
| 14:34.49 | awk | module unload |
| 14:34.56 | Dayver | any idea why "ring all" in queue is loading server? |
| 14:35.03 | SteveTotaro | rmmod zaptel |
| 14:35.07 | awk | yes, queue / asterisk |
| 14:35.22 | awk | if you rmmod zaptel when asterisk is running you can get kernel panic.. |
| 14:35.24 | stansmith | you might have to stop asterisk first yang |
| 14:35.25 | awk | not wise... |
| 14:35.45 | stansmith | yang: stop asterisk, stop zaptel, start asterisk |
| 14:35.46 | awk | Dayver what version of asterisk... but when it comes to queues and asterisk,.. they just fucked... |
| 14:36.01 | awk | or listen to what i said first.. module unload ... |
| 14:36.08 | awk | module unload chan_zap.so |
| 14:36.18 | Dayver | any solution? I have 40 ext. to ring ..... it's 2.4 |
| 14:36.30 | SteveTotaro | check this deal for an older IBM eServer 325 http://www.geeks.com/details.asp?invtid=8835Y11-R enter promo code 8835DEAL to get it for $219 plus shipping. I just ordered two. |
| 14:36.41 | awk | Dayver: well it could quite easly be a bug.. if you using 2.X use 2.24 |
| 14:36.51 | awk | its quite stable. |
| 14:37.06 | Dayver | awk: thanks |
| 14:37.25 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 14:37.27 | SteveTotaro | 8835Y11-R -- IBM eServer 325 Dual Opteron 2.0GHz 1GB 12 2 $219.99 $439.98Â |
| 14:37.46 | *** join/#asterisk chigital (n=chigital@213.131.236.199) |
| 14:37.56 | *** join/#asterisk BenGR (n=Ben_@79.131.52.155) |
| 14:38.43 | Datax | good deal :) |
| 14:38.47 | Dayver | awk: it's actually 1.2.26.1 svn rev 79171 run's on Dell 1950 2 quad xeon's and 8 GB memory |
| 14:38.48 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:39.01 | awk | well thats svn... |
| 14:39.03 | BenGR | hello, I want to ask about the poor quality of the IVR system. Although original samples are professional high quality recordings I had to downgrade to 8000khz 16bit to make it compatible with asterisk. the quality is really awful. any ideas how can I improve this?? |
| 14:39.07 | awk | I don't trust development trunks... |
| 14:39.13 | awk | 1.2.24 is stabl'ish |
| 14:39.34 | awk | they most probabbly fscked it up... like 1.4.18.. its actually a joke how broken 1.4.18 is |
| 14:39.40 | Daviey | gah, 1.2 / 1.4 / 1.6 are all development trunks :( |
| 14:39.44 | SteveTotaro | BenGR: what codec, translation, zatpel timing |
| 14:39.45 | awk | and 1.4.17 iax doesn't work... so if i where u dont use 1.4 at all... |
| 14:39.47 | SteveTotaro | ? |
| 14:39.50 | ManxPower | BenGR: 8000Khz is what the telco uses for phone calls. I suspect your conversion program did not convert properly |
| 14:39.54 | Dayver | Thanks |
| 14:39.59 | file | awk: iax doesn't work? |
| 14:40.11 | awk | file: its very buggy. and you run out of io threads... |
| 14:40.23 | SteveTotaro | iax is teh suck except for very specific work arounds |
| 14:40.33 | file | most of those issues have been solved, and if not we would really like to know |
| 14:40.38 | SteveTotaro | nats behind nats |
| 14:40.46 | BenGR | SteveTotaro what you mean codec? translation? zaptel timing? :\ |
| 14:41.17 | ManxPower | BenGR: WAV, ulaw, and alaw should have the same quality as PSTN calls |
| 14:41.18 | SteveTotaro | exactly that, are you using ulaw/alaw (uncompressed) |
| 14:41.18 | awk | file: what move to 1.4.18 where constant crashes? zap channel recording... |
| 14:41.27 | BenGR | im using PCM |
| 14:41.34 | file | awk: have you filed bug reports? |
| 14:41.36 | ManxPower | BenGR: PCMU or PCMA? |
| 14:41.40 | SteveTotaro | did you load ztdummy or other zaptel hardware |
| 14:42.01 | awk | zap channel recording, totally out of sync.. but 1 person speaks then another person speaks... timing is completely out |
| 14:42.09 | ManxPower | BenGR: if you listen to the sounds outside of Asterisk do they still sound bad? |
| 14:42.10 | BenGR | On Adobe Audition I select "Save as ACM Waveform" then I select filter PCM and then attributes 8,000khz; 16bit; Mono; 15kb/sec |
| 14:42.14 | awk | ive gone right up to 1.4.18-5 |
| 14:42.25 | SteveTotaro | just take the original sound files and use sox to resample |
| 14:42.33 | file | -5 doesn't exist, but I know there is an issue open in regards to sync on Zap channels with Mixmonitor |
| 14:42.35 | SteveTotaro | it is very well documented on the wiki |
| 14:42.39 | BenGR | ManxPower yes of course. But I cannot increase quality because asterisk cannot accept it |
| 14:42.42 | ManxPower | BenGR: do you have the option of 64kb/sec ? |
| 14:42.44 | awk | file is does with me... scopserv... |
| 14:42.47 | awk | rpm package... |
| 14:42.56 | awk | cant remeber what version I compiled... |
| 14:43.01 | BenGR | i dont have access to the root platform. is it a convertor for windows? |
| 14:43.09 | awk | 1.4.17 1 svn trunk was stable! but no iax.. but the rest of the system was stable.. |
| 14:43.12 | BenGR | ManxPower: maximum is 44,100khz |
| 14:43.14 | ManxPower | BenGR: what you are experiencing is not common |
| 14:43.25 | awk | now 1.4.18 is broken again.. |
| 14:43.32 | ManxPower | BenGR: not khz, but the 15kb/sec |
| 14:43.40 | BenGR | ManxPower if I try to increase quality it says that 22050 freqency is not supported or 'invalid header" etc |
| 14:44.22 | ManxPower | BenGR: no, You said "select filter PCM and then attributes 8,000khz; 16bit; Mono; 15kb/sec" And I asked if instead of using 15kb/sec you try 64kb/sec |
| 14:44.25 | BenGR | sox convertor maintains good quality? can anyone makes me convertion plz? just for testing purposes |
| 14:44.28 | *** part/#asterisk zamba (i=marius@sveigde.hih.no) |
| 14:44.29 | awk | easy solution.. dont use TDM cards with asterisk... untill zaptel is finally stable which I doubt.. use a quintum unit and get rid of 90% of your problems.... |
| 14:44.48 | BenGR | ManxPower no there is not such thing. all those atrributes are one line attributes, not seperate options |
| 14:44.54 | [TK]D-Fender | BenGR: * can't use 22khz so stop trying. Set it to 8khz Mono as you were instructed |
| 14:45.12 | ManxPower | BenGR: there are two possible problems, 1) the conversion was not good or 2) you have some audio issues with Asterisk |
| 14:45.31 | BenGR | but the quality is poor outside asterisk. what im asking is how can I achieve the best quality with asterisk |
| 14:45.34 | ManxPower | BenGR: How are you calling into the IVR? PSTN? SIP phone? |
| 14:45.37 | BenGR | asterisk plays what it takes, a bad wav file |
| 14:45.53 | BenGR | im calling from outside line to isdn modem and then to asterisk and then to sip |
| 14:45.57 | coppice | you need to use a green marker pen on the edge of the disk, if you want to improve the sound quality |
| 14:45.58 | BenGR | call -> isdn modem -> asterisk -> sip phone |
| 14:46.09 | ManxPower | BenGR: the voip-info.org wiki may have suggestions with how to use "sox" to convert files. |
| 14:46.14 | coppice | and use very very thick cables |
| 14:46.21 | BenGR | ManxPower I dont have access to that sox utility, only web interface :( |
| 14:46.26 | ManxPower | BenGR: what specific brand of SIP phone? |
| 14:46.29 | tzafrir | BenGR, what channel with the ISDN modem? What ISDN modem? |
| 14:46.44 | BenGR | linksys spa921 |
| 14:46.48 | ManxPower | BenGR: AHA! |
| 14:47.04 | BenGR | i dont think sip or asterisk is my fault. the wav file is 100kb when original is 2mb |
| 14:47.20 | BenGR | which makes very very big compression. can you please send me an example of a wav file to compare the quality?? |
| 14:47.38 | ManxPower | in the Linksys phone, can you find the option called somet thing like "samples per packet". If it is set to 0.30, change it to 0.20 and that will help. |
| 14:48.42 | awk | bug report!!!!! |
| 14:48.43 | awk | Since upgrading to 1.4.18-rc4, MixMonitor seems to be producing files that are out of sync. You hear the caller and then several seconds later, you hear the response. |
| 14:48.48 | awk | file there.... |
| 14:48.48 | BenGR | RTP Packet Size: 0.30 |
| 14:48.53 | awk | everyone is expericing this problem. |
| 14:49.04 | file | awk: yes, I said I knew of that bug report |
| 14:49.07 | BenGR | can someone please send me a wav file converted with SOX?? please |
| 14:49.42 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:49.53 | tzafrir | BenGR, why don't you convert it yourself? |
| 14:50.04 | ManxPower | BenGR: http://www.fnords.org/~eric/ERASTIRLING.wav It is a big file so it may take time to download |
| 14:50.13 | tzafrir | BenGR, what does the utility 'file' say on the sound file you have? |
| 14:50.28 | BenGR | I DONT HAVE ACCESS TO ROOT :) |
| 14:50.34 | BenGR | only internet explorer web interface |
| 14:50.38 | BenGR | so there is no sox there |
| 14:50.40 | ManxPower | BenGR: change the RTP packet size to 0.20 That may fix the problem |
| 14:50.51 | ManxPower | BenGR: we won't be able to help you much if you do not have access to the system |
| 14:51.07 | BenGR | ManxPower yes for inuse, what about when someone is calling me from his phone which is not linksys?? |
| 14:51.21 | ManxPower | BenGR: we solve one problem at a time here. |
| 14:51.27 | BenGR | why music on hold supports mp3 and ivr doesnt support mp3?? |
| 14:51.36 | *** join/#asterisk goodines (n=goodines@201.48.5.105) |
| 14:51.52 | ManxPower | BenGR: mp3 support is removed from Asterisk because of licensing |
| 14:51.58 | tzafrir | BenGR, building sox only takes a combination of System and Curl ;-) |
| 14:52.09 | tzafrir | no, that was not serius |
| 14:52.27 | zeeesh | i m registering a sip peer .. when its type=peer ... its registered when type=friend its registered .. but when its type=user ... it does't register???? |
| 14:53.14 | Daviey | ogg support would be handy :) |
| 14:53.26 | SteveTotaro | that whole peer/friend/user is confusing |
| 14:53.33 | BenGR | Daviey does it support ogg? |
| 14:53.37 | tzafrir | Daviey, you mean: ogg/vorbis support is handy? |
| 14:53.37 | SteveTotaro | i think it is being rewritten |
| 14:54.02 | Daviey | tzafrir: yes |
| 14:54.29 | tzafrir | BenGR, yes |
| 14:54.50 | ManxPower | zeeesh: users can't register |
| 14:54.57 | ManxPower | as they never ever ever receive an incoming call. |
| 14:55.10 | goodines | im working with asterisk using h323 and ss7. * receives a h323 call and forwards to E1. But i have a problem in this scenario. I took off g729 and I tried to receive a h323 call using 729. But * ACCEPTED the call. It returned a 'Alerting' and a 'Connect' to GK. Would it be a bug? Im using * 1.4 |
| 14:55.11 | ManxPower | registration just notifies asterisk what user/pass is associated with a specific IP address |
| 14:55.18 | SteveTotaro | sox is cross compatible with windows http://sox.sourceforge.net/ |
| 14:55.54 | SteveTotaro | ss7 and h323 sounds like a real challenge |
| 14:55.59 | zeeesh | <ManxPower>: ohh thanks .. got it ... |
| 14:56.22 | tzafrir | BenGR, format_mp3 is available from addons. However there is really no reason to use mp3 for on-hold music |
| 14:56.25 | ManxPower | SteveTotaro: no, ss7 and h323 sounds like something out of Dante's Hell to punish bad geeks |
| 14:56.56 | Daviey | people _still_ use h323. :O |
| 14:57.12 | ManxPower | Daviey: Asterisk users don't unless they have NO other choice. |
| 14:57.31 | goodines | I wouldnt like to use, but the customers..... are customers |
| 14:57.58 | SteveTotaro | yeah but h323 with ss7, i would like to play with that but not with a production use in mind |
| 14:59.06 | goodines | my customer is putting away his ciscos to use our machines with asterisk/digium cards. |
| 14:59.58 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 15:00.15 | goodines | but it is not the problem. The problem is when asterisk accept a 729 call when it doenst have the codec to translate. |
| 15:01.45 | goodines | any ideas? :) |
| 15:01.51 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-18fa39ccf3945329) |
| 15:02.00 | goodines | avoid h323 is not the right answer :) |
| 15:02.03 | Qwell | goodines: either disallow g729, or buy codec licenses |
| 15:02.27 | Daviey | I managed to RMA a second hand ATA because it was H.323.. the only reason! |
| 15:03.06 | goodines | Yes, we will buy them. But I just wondered why * accepted the call.. |
| 15:03.22 | zeeesh | i hv 2 peeer like 100 and 200 .. 100 is behind the nat, and 200 is registered by using live ip ... both can call each other when they just registered .. but after a sometime just 100 can dial 200 ... but 200 is unable to dial 100, although both registered??? |
| 15:03.32 | goodines | it doesnt have (yet) the 729 codecs. |
| 15:03.49 | Qwell | goodines: sounds like a bug in the h323 channel driver you're using |
| 15:04.24 | goodines | it is not good... |
| 15:04.25 | stansmith | naaw |
| 15:04.31 | goodines | I will try to update the version |
| 15:04.33 | Qwell | h323? no, it's not :) |
| 15:05.28 | *** join/#asterisk jomarie (n=asterisk@158.59.192.218) |
| 15:07.33 | ManxPower | zeeesh: qualify=9000 for the phone that is NAT'd |
| 15:07.47 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:08.06 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-122-84.bltmmd.east.verizon.net) |
| 15:08.27 | zeeesh | <ManxPower>: 9000 .. is that seconds or what ?? |
| 15:09.34 | zeeesh | ok milliseconds ... i just ... will asterisk will get acknowledgement after 9000 or what ? |
| 15:10.22 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 15:10.29 | ManxPower | zeeesh: if asterisk does not get an ACK in 9 seconds it will consider the peer offline. This makes the NAT translation in the router not close the connection |
| 15:10.45 | BenGR | ok I think i handled a better quality by playing with Adobe Audition. Its much better but not the quality I can hear with I calling to big companies like mobile companies. They probably use another more sophisticated PBX technology |
| 15:10.45 | ManxPower | well, the extra traffic will |
| 15:11.10 | ManxPower | BenGR: did you listen to the sample WAV file I gave you the URL to? |
| 15:11.18 | Daviey | BenGR: you should be able to get great quality |
| 15:11.38 | BenGR | ManxPower yes i listened. not so impressed |
| 15:12.21 | ManxPower | BenGR: that file was recorded over the phone connected to asterisk |
| 15:12.42 | BenGR | then its very good quality for that recording. I have mp3 samples from professional studio and professional voice |
| 15:13.11 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 15:13.13 | ManxPower | BenGR: that is great, but you are limited to 8000hz 16bit for any phone call because that is what the phone company uses |
| 15:13.43 | coppice | if you are playing audio over a phone line, it ain't gonna be Hi-Fi. professional studio recording is largely irrelevant |
| 15:13.57 | BenGR | yea |
| 15:14.11 | *** join/#asterisk chigital (n=chigital@91.90.144.102) |
| 15:14.32 | BenGR | does asterisk support playing background on hold music and on top speaking the IVR messages? because now it stops and then music starts again... |
| 15:14.42 | ManxPower | BenGR: no |
| 15:15.07 | BenGR | ok you saved me from a lot of research :) |
| 15:15.30 | Daviey | erm, could you not do Background()? |
| 15:16.27 | BenGR | first of all I have to bypass the problem I have with the music on hold. it doesnt start from scratch when someone is calling. it continues from the last point |
| 15:16.34 | BenGR | im currently working on it |
| 15:16.35 | BenGR | pfff |
| 15:20.26 | ManxPower | Daviey: Background() is for playing an audio file while listening for DTMF |
| 15:20.33 | SteveTotaro | there is a switch for that |
| 15:20.46 | SteveTotaro | to play audio from the beginning |
| 15:20.59 | BenGR | SteveTotaro that is what Im seeking for |
| 15:21.07 | ManxPower | SteveTotaro: not if you are using mpg123 for MoH |
| 15:21.17 | ManxPower | BenGR: then you need to look at musiconhold.conf |
| 15:22.05 | SteveTotaro | Format_MP3, Asterisk's mpg123 replacement, will play the MOH file from the beginning each time a caller is put on hold or each time the musiconhold command is invoked from extension.conf. |
| 15:22.22 | BenGR | there's nothing here. 4 lines only :) |
| 15:22.54 | ManxPower | BenGR: then you are not looking at the sample config file. Look in the asterisk source code directory in the "configs" subdirectry |
| 15:24.12 | SteveTotaro | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 15:24.18 | SteveTotaro | looks mostly accurate |
| 15:24.24 | ManxPower | SteveTotaro: that is prolly outdated |
| 15:24.40 | SteveTotaro | is he using 1.4? |
| 15:25.03 | BenGR | 1.2.26.2 here |
| 15:25.07 | BenGR | yes i know old enough |
| 15:25.34 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:25.43 | SteveTotaro | then it should not be that "outdated" |
| 15:25.58 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
| 15:26.06 | SteveTotaro | the wiki is slammed regularly but still my top place to look for answers |
| 15:26.56 | ManxPower | SteveTotaro: without version info you can never know of what it tells you applies to what you are using |
| 15:27.11 | *** join/#asterisk fukz (n=basti@p4FC2E763.dip.t-dialin.net) |
| 15:27.34 | SteveTotaro | speak for yourself, but i can |
| 15:27.53 | BenGR | i was already reading that article SteveTotaro but im still trying to find that restart option. i'm still on it:) |
| 15:28.07 | fukz | hello all. I have a problem while authenticate IXA2 channel via rsa. CLI shows "I don't know how to authenticate" when I try to get a connection. |
| 15:28.14 | ManxPower | BenGR: 1.2 does not have a restart option |
| 15:28.18 | fukz | If I use plain, all works |
| 15:28.41 | BenGR | ManxPower oh okie. so the music on hold always plays from the last point, right? |
| 15:28.58 | ManxPower | no, it plays continously regardless of anyone being on hold. |
| 15:29.00 | SteveTotaro | this is your clue "Format_MP3, Asterisk's mpg123 replacement, will play the MOH file from the beginning each time a caller is put on hold or each time the musiconhold command is invoked from extension.conf." |
| 15:29.15 | ManxPower | SteveTotaro: have you used format_MP3 with 1.2? |
| 15:29.23 | SteveTotaro | yes |
| 15:29.31 | SteveTotaro | for a rather large call center |
| 15:29.40 | SteveTotaro | 15k calls a day |
| 15:29.41 | ManxPower | SteveTotaro: since he does not have access to the system, I doubt he can build the module |
| 15:29.52 | ManxPower | anyway, off to work for me |
| 15:30.16 | SteveTotaro | offering answers, the rest is not my problem |
| 15:31.02 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 15:32.19 | BenGR | hehe |
| 15:32.33 | drmessano | Always pleasant |
| 15:32.37 | *** join/#asterisk ph0ne (n=ph0ne@dsl-207-112-93-203.tor.primus.ca) |
| 15:34.27 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:36.18 | BenGR | altough I have build the IVR menu when someone press a digit on the phone the IVR doesnt respond. that means hardware (isdn card) setting issue. right? |
| 15:37.04 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 15:39.48 | ph0ne | BenGR: are you using regular analog telelephones for your tests? |
| 15:40.16 | ph0ne | I'm no expert by the way! |
| 15:45.28 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 15:45.33 | BenGR | yes |
| 15:45.39 | BenGR | or my mobile for testing |
| 15:46.55 | scooby2 | anyone seen this neat error before? |
| 15:46.58 | scooby2 | <PROTECTED> |
| 15:47.39 | ph0ne | dtmfmode=rfc2883 |
| 15:47.44 | [TK]D-Fender | scooby2: looks like packet-loss that exceeds jitter buffers |
| 15:48.41 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:48.49 | ph0ne | BenGR: dtmfmode=rfc2883 |
| 15:49.08 | scooby2 | i stop getting inbound calls on my t1's and the only error is that one |
| 15:49.11 | ph0ne | you must put that into sip.conf |
| 15:49.19 | scooby2 | over and over thousands of times until i 'restart now' |
| 15:51.14 | BenGR | ph0ne in which file? |
| 15:51.21 | BenGR | oops |
| 15:51.22 | BenGR | sip.conf |
| 15:51.23 | BenGR | thanks |
| 15:51.24 | BenGR | let me check :D |
| 15:51.41 | [TK]D-Fender | BenGR: and that should be "rfc2833" |
| 15:52.58 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 15:52.59 | ph0ne | [TK]D-Fender: my bad sorry |
| 15:53.06 | [TK]D-Fender | BenGR: And if you are having problems on a IVR for calls coming in via ISDN, sip.cofn will have nothing to do with it. |
| 15:53.18 | [TK]D-Fender | BenGR: I would make sure your IVR is even properly designed |
| 15:53.18 | BenGR | yes that is my problem, incoming calls |
| 15:53.40 | BenGR | well I'm using the sample from the web interface and it has to work :( |
| 15:53.49 | [TK]D-Fender | BenGR: PASTEBIN the incoming call attempt at verbose 10, and include your actual dialplan. |
| 15:53.50 | [TK]D-Fender | ~pb |
| 15:53.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:53.54 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:54.12 | [TK]D-Fender | BenGR: No, it doesn't "have to work". and GUI's are not supported here. |
| 15:54.19 | scooby2 | do I need ztdummy if I am usinig th wct4xxp module? |
| 15:54.25 | [TK]D-Fender | scooby2: No |
| 15:54.32 | BenGR | yes i know that is why Im not talking about guis. to help me do it in conf files if possible |
| 15:54.51 | [TK]D-Fender | BenGR: Please provide what I have requested |
| 15:55.06 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-ea1159f7fafacf24) |
| 15:55.45 | BenGR | ok let me try |
| 15:57.23 | BenGR | does the debug level 999999999999 works for you? |
| 15:57.38 | [TK]D-Fender | BenGR: Fine |
| 15:57.42 | BenGR | ok i set it |
| 15:57.51 | BenGR | now you want the dialplan |
| 15:57.52 | Qwell | 99999999999 != 10 |
| 15:58.18 | BenGR | debug level 10 ? |
| 16:00.40 | BenGR | when the incoming call is coming i cant see any debug info besides a 2 warning messages about variables |
| 16:00.55 | [TK]D-Fender | BenGR: PASTEBIN |
| 16:01.36 | BenGR | you will not believe it but that console doesn't have any way to select text and copy paste :) i have to type it. hehe |
| 16:01.48 | [TK]D-Fender | BenGR: Log in via SSH |
| 16:02.20 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 16:03.03 | BenGR | that would be great but impossible. pffff. im trying |
| 16:03.32 | [TK]D-Fender | BenGR: Impossible? And why would that be? |
| 16:06.00 | BenGR | because this asterisk is something like a PBX in a box. it comes from a company that setups it and they give you web interface + access to conf with some restrictions |
| 16:06.10 | BenGR | i dont know why, they probably afraid of stealing their "work" ? |
| 16:07.00 | Katty | anyone here in the las vegas area? |
| 16:07.16 | [TK]D-Fender | BenGR: Then you are screwed. They own your ass, go suck up for support. |
| 16:07.24 | Katty | [TK]D-Fender: btw, i talked with my boss. |
| 16:07.32 | Katty | [TK]D-Fender: he had sad eyes. |
| 16:07.43 | [TK]D-Fender | Katty: Is that to say you're taking up the offer? |
| 16:07.50 | Katty | [TK]D-Fender: i am seriously considering it. |
| 16:07.58 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:08.07 | Katty | [TK]D-Fender: i need to have an in depth conversation (not an interview) with the person making the offer |
| 16:08.26 | Katty | [TK]D-Fender: amazingly, nearly everything is falling into place. |
| 16:08.35 | tzafrir | BenGR, Can you dial to a custom extension? |
| 16:08.39 | Katty | [TK]D-Fender: i've been guaranteed a place to stay, for free, for a few months... |
| 16:08.39 | BenGR | yes that is the last option before I can handle things by myself. that's why my question is general: if it something common when keys are ignored when someone is calling outside and he is on the IVR menu. maybe something common that many users may face in the beggining |
| 16:08.56 | Katty | [TK]D-Fender: the people i'd be working with all know Ryan... |
| 16:08.57 | BenGR | tzafrir what you mean by that? from SIP phone to SIP phone? |
| 16:09.11 | Katty | [TK]D-Fender: his best friend, his best friend's dad, and the best friend's dad's best friend... all in the same shift i'd be in |
| 16:09.18 | tzafrir | actually, do you have access to the manager interface? |
| 16:09.24 | BenGR | tzafrir yes |
| 16:09.25 | Katty | [TK]D-Fender: a big company. guranteed job for a year, by contract |
| 16:09.35 | Katty | [TK]D-Fender: a substancial pay raise |
| 16:09.49 | [TK]D-Fender | Katty: Actually NOT a thing I'd like to be encountering. If something were to change between the two of you there are wired repercussions |
| 16:09.59 | Katty | [TK]D-Fender: indeed. |
| 16:10.06 | Katty | [TK]D-Fender: one of the things i'm thinking about. |
| 16:10.16 | tzafrir | BenGR, so you can practically run custom commands as the user asterisk |
| 16:10.24 | BenGR | yes it has a CLI> console |
| 16:10.29 | Katty | [TK]D-Fender: but the good of it, are so far, far outweighing the bad. |
| 16:10.40 | Katty | [TK]D-Fender: and i've been looking for an excuse to get out of this area for a good while. |
| 16:10.43 | tzafrir | BenGR, I meant: custom shell commands |
| 16:10.56 | BenGR | give me a shot. name a command to test |
| 16:11.38 | Katty | [TK]D-Fender: it's hard to leave this company... |
| 16:11.50 | Katty | [TK]D-Fender: i'm very involved here, and while they annoy me, they're almost like family. |
| 16:12.02 | Katty | [TK]D-Fender: my mother is the other biggest factor :/ |
| 16:12.35 | Katty | [TK]D-Fender: very difficult time stomaching the idea of being so far away from her. |
| 16:12.57 | tzafrir | BenGR, set up an extension that runs System(${EXTENSION}) in some custom context, and originate a call to a strange-looking extension in that context |
| 16:13.11 | Katty | [TK]D-Fender: far more opportunity in las vegas tho, for sure. |
| 16:13.15 | Katty | [TK]D-Fender: far far greater. |
| 16:13.26 | BenGR | tzafrir that is too complicated for me :( |
| 16:13.42 | [TK]D-Fender | Katty: Little girl gonna make it big.... |
| 16:13.52 | Katty | [TK]D-Fender: it is a dream i have... |
| 16:14.11 | *** join/#asterisk GrumpManAtWork (n=meanderi@pool-72-78-182-204.phlapa.east.verizon.net) |
| 16:14.12 | dmz | morning y'all, i have a cisco 7960 connected to a local * box via sip, that then has an openvpn connection to our colo and an IAX connection to our * box in the colo which then dials out via sip to our voip provider. latency from home -> colo is 95ms. Everything is great except when calling conference calls i can never get remote conference systems to recognize my key touches to join conferences. any suggested changes/settings i s |
| 16:14.13 | dmz | hould consider to get key tones to work better? -- sorry to interrupt job change thoughts :) |
| 16:14.17 | Katty | [TK]D-Fender: i'd make a hell of an example at a high school reunion ;) |
| 16:14.42 | Katty | [TK]D-Fender: the snobby goth chick from high school now kicks your life's tail! |
| 16:15.01 | Katty | [TK]D-Fender: what is it with small town people never leaving small town. |
| 16:15.40 | [TK]D-Fender | Katty: U R A FEERD |
| 16:16.03 | Katty | i would have to leave my deary pipsqueak, i think )= |
| 16:16.08 | Katty | at least for a time. |
| 16:16.13 | Katty | that induces teariness just thinking about it. |
| 16:16.14 | stansmith | wtf@php+postgres |
| 16:16.26 | Katty | stansmith: mine's working okay |
| 16:16.28 | GrumpManAtWork | I have sip peers and I was wondering if there was a way to see if that extension is on a call ? |
| 16:16.29 | stansmith | :-/ |
| 16:16.38 | Katty | stansmith: asterisk-stat,php,gd,postgres |
| 16:16.46 | Katty | it's the cdr that has me confuzzled |
| 16:16.49 | stansmith | ~asterisk-stat |
| 16:16.50 | Katty | i'm not getting any data |
| 16:17.05 | Katty | stansmith: http://angela.sleekgeek.org/2008/02/20/asterisk-stat-v20/ |
| 16:17.09 | stansmith | danke |
| 16:17.12 | Katty | np |
| 16:17.19 | Katty | [TK]D-Fender: hypothetical question. |
| 16:17.21 | BenGR | ok i fixed it. i had to enable a softdtmf checkbox |
| 16:17.30 | [TK]D-Fender | GrumpManAtWork : "core show channels concise" |
| 16:17.38 | Katty | [TK]D-Fender: apache, postgres, php, gd, pgsql.conf, cdr.conf all configured |
| 16:17.47 | Katty | [TK]D-Fender: asterisk no putith data into pgsql. |
| 16:17.51 | Katty | [TK]D-Fender: but! they are connected properly |
| 16:17.57 | Katty | [TK]D-Fender: it's like asterisk doesn't know to DOO EEET |
| 16:18.04 | Katty | [TK]D-Fender: enable=yes! |
| 16:18.06 | GrumpManAtWork | [TK]D-Fender: I ment in a dial plan. |
| 16:18.19 | Katty | [TK]D-Fender: what would you suggest? |
| 16:18.23 | Katty | i'm confused. |
| 16:18.30 | dmz | katty, what's with small town people leaving town but then coming back 20 years later :) |
| 16:18.31 | [TK]D-Fender | GrumpManAtWork: "core show application chanisavail" |
| 16:18.34 | Katty | it's like there's some magic GO button i'm not seeing. |
| 16:18.43 | [TK]D-Fender | Katty: No clue, I suck at DB's |
| 16:18.48 | Katty | dmz: you have a point there (= |
| 16:18.56 | Katty | [TK]D-Fender: i do too, apparently. |
| 16:19.16 | stansmith | my issue is fusebox, i can see perl passing the asterisk GET variables to the php script :-/ |
| 16:19.45 | Katty | so, dear channel, dear channel |
| 16:19.57 | Katty | is there some sort of module loading command you have to do to tell asterisk to dump to a database? |
| 16:20.06 | Qwell | dump what to a database? |
| 16:20.14 | Katty | cdr inflimation |
| 16:20.20 | Katty | i have pgsql.conf configured... |
| 16:20.34 | Katty | with the same info as my php, and connection in php is good.. so i presume asterisk can say hi too |
| 16:20.39 | Katty | cdr.conf is enable=yes |
| 16:20.45 | Katty | but nothing's /going there/ |
| 16:20.54 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:20.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:20.58 | Katty | it's like asterisk just keeps saying "hi! how are ya" |
| 16:21.10 | FreezeS | Katty: check the debug log |
| 16:21.12 | Qwell | Katty: there's a cdr show command, or something, to show which engines are configured |
| 16:22.08 | Qwell | or, rather, which engines it thinks are configured |
| 16:22.26 | Katty | maybe 'cdr status'? |
| 16:22.36 | Qwell | something like that |
| 16:22.40 | Katty | lemme pastebin |
| 16:22.44 | FreezeS | exactly that |
| 16:22.46 | FreezeS | cdr status |
| 16:23.02 | Katty | Qwell: http://pastebin.ca/920770 |
| 16:23.17 | Katty | this cdrmode: simple concerns me |
| 16:23.19 | Qwell | yeah, it don't like it |
| 16:23.23 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:23.23 | Katty | there was nothing simple about that postgres setup i did! |
| 16:23.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:23.30 | FreezeS | but I had problems at the beginning with the library, it didn't accept my password, that's why I'm suggesting you to check the debug logs |
| 16:23.52 | stansmith | ahhh! it worked |
| 16:23.56 | stansmith | well i'll be god damned |
| 16:24.04 | drmessano | now you will |
| 16:24.22 | Katty | that's so bash.orged. |
| 16:24.35 | cesar_CR | hi guys any impresion about buying a dell or hp server for an asterisk ??? |
| 16:24.46 | FreezeS | cesar_CR: how about IBM ? |
| 16:24.53 | stansmith | cesar_CR: be careful of Asterisk + HP + HP Proliant drivers |
| 16:24.57 | Katty | Qwell: do you have to tell asterisk to load a module to handle cdr to pgsql? |
| 16:25.08 | Katty | Qwell: or should configuring cdr.conf and pgsql.conf magically do it? |
| 16:25.15 | Qwell | Katty: just the pgsql cdr module |
| 16:25.19 | Katty | ... |
| 16:25.19 | drmessano | Packard Bells are good with Asterisk |
| 16:25.21 | Katty | which i didn't do |
| 16:25.22 | Qwell | which should be loaded automatically |
| 16:25.29 | Katty | how can i test that |
| 16:25.33 | Katty | show module.. |
| 16:25.44 | Qwell | cdr_<tab> |
| 16:25.57 | cesar_CR | FreezeS, stansmith , well I need and advice from someone who has that kind of exp. do you ?? IBM ?? |
| 16:26.26 | Katty | Qwell: from CLI? |
| 16:26.30 | Qwell | yeah |
| 16:26.30 | FreezeS | cesar_CR: my point was that not the brand is important but the hardware itself |
| 16:26.40 | Qwell | module show cdr_<tab>, then find what it's called |
| 16:26.41 | Katty | Qwell: either what you said didn't parse... |
| 16:26.41 | cesar_CR | drmessano, wich model have you used ? |
| 16:26.44 | Katty | ah |
| 16:26.46 | Katty | transposed it |
| 16:27.50 | cesar_CR | FreezeS, so if I find and hardware that fits, I can use it without problems.. |
| 16:27.55 | Katty | meh, it refuses to give it to me.. |
| 16:27.59 | Katty | it will show me all of my modules |
| 16:28.03 | Katty | so many.. |
| 16:28.19 | FreezeS | check the ones that start with cdr_ |
| 16:28.34 | Katty | i'm using putty |
| 16:28.36 | RoyK | http://www2.b3ta.com/host/creative/13/1204118301/etchasketchquake.jpg |
| 16:28.38 | Katty | it dun like to show me that much |
| 16:28.41 | Katty | i'll go look at the server |
| 16:28.44 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:28.51 | FreezeS | you can set the line buffer |
| 16:28.54 | FreezeS | settings-window |
| 16:28.59 | stansmith | cesar_CR: I am testing asterisk for deployment on HP, it took me a couple days to figure out that HP Proliant drivers were interfering with my asterisk install |
| 16:29.00 | FreezeS | I usually set it to 20000 |
| 16:29.26 | FreezeS | cesar_CR: I'm using a dell among many no-names and it's the only one I've had problems with |
| 16:29.49 | FreezeS | but the problems were something among the "you need our custom compiled kernel stooopid !" |
| 16:30.48 | stansmith | cesar_CR: the ML350 G5 HP server im testing on only has pci-x and pci-express slots too |
| 16:31.04 | stansmith | cesar_CR: make sure your pci slots are 2.2 compliant |
| 16:31.53 | Katty | Qwell: i have cdr_csv, cdr_manager, and cdr_custom |
| 16:31.59 | Katty | Qwell: but no cdr_pgsql thingy |
| 16:31.59 | Qwell | no cdr_pgsql? |
| 16:32.05 | Katty | no :< |
| 16:32.14 | FreezeS | then you found your problem :) |
| 16:32.19 | Katty | yeah. |
| 16:32.20 | Katty | i found it |
| 16:32.23 | stansmith | ballin' |
| 16:32.31 | Katty | my problem is communicating with asterisk to fix it! |
| 16:32.38 | Katty | i've never had to manually tell it to load a module before. |
| 16:32.43 | FreezeS | really ? |
| 16:32.48 | Katty | really really. |
| 16:32.49 | FreezeS | how long have you been working with it ? |
| 16:32.53 | Katty | 3 years |
| 16:32.57 | FreezeS | me too |
| 16:33.00 | Qwell | Katty: check /etc/asterisk/modules.conf |
| 16:33.07 | FreezeS | and during my second day I needed to do that :) |
| 16:33.35 | Katty | hey |
| 16:33.37 | Katty | how bout that |
| 16:33.44 | Katty | Qwell: do i need to reload, or stop now? |
| 16:33.53 | Qwell | for what? |
| 16:34.01 | FreezeS | thing is, you can actually load the module manually |
| 16:34.02 | *** join/#asterisk ACiDV (n=acidv@24-122-213-230.dr.cgocable.ca) |
| 16:34.02 | Katty | to get the settings to asterisk |
| 16:34.09 | Qwell | from modules.conf? |
| 16:34.12 | Katty | yes |
| 16:34.16 | FreezeS | try something like "load module cdr_pgsql" |
| 16:34.21 | Katty | nono |
| 16:34.23 | Qwell | it'll just work next time you start asterisk |
| 16:34.24 | Katty | i mean.. |
| 16:34.28 | Katty | okay |
| 16:34.33 | Katty | but am i preloading? |
| 16:34.34 | Katty | or loading |
| 16:34.35 | Qwell | if the module is loaded now, no need to restart |
| 16:34.37 | FreezeS | that was in order to avoid restarting |
| 16:34.43 | Qwell | load, if you have autoload=no |
| 16:34.47 | stansmith | lol@chaos |
| 16:34.50 | Katty | autoload=yes |
| 16:34.56 | Katty | so, preload? |
| 16:34.59 | Qwell | then it should have loaded on it's own.. |
| 16:35.08 | Katty | ^_- |
| 16:35.11 | Qwell | did you restart asterisk after building/installing that module? |
| 16:35.24 | Katty | the whole machine has been restarted since then |
| 16:35.42 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
| 16:35.47 | Qwell | can you load the module yourself? |
| 16:35.48 | Katty | but...installing the module.. |
| 16:35.52 | Katty | let me try |
| 16:36.08 | Qwell | it may give you some clue as to why it didn't load |
| 16:36.31 | nvrpunk | can anyone point me to a good ovice prompt howto? The one in the asterisk book is not working for me |
| 16:36.52 | [TK]D-Fender | nvrpunk: care to clarify that a bit... |
| 16:37.04 | Katty | http://pastebin.ca/920785 <- Qwell |
| 16:37.22 | Katty | Qwell: does that mean that asterisk and addons were compiled with out pgsql support? |
| 16:37.37 | nvrpunk | [TK]D-Fender, followed the books example and m voice prompt system never asks for an extension, it just says goodbye and hangs up |
| 16:37.42 | Qwell | Katty: possible. check menuselect |
| 16:37.58 | Qwell | oh, .so |
| 16:38.15 | Katty | same message |
| 16:38.16 | Katty | also |
| 16:38.21 | Qwell | right |
| 16:38.22 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:38.22 | Katty | asterisk or asterisk-addons? |
| 16:38.28 | Qwell | asterisk I think |
| 16:38.31 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 16:38.45 | FreezeS | asterisk |
| 16:38.48 | Qwell | Katty: check in `make menuselect`, see if it's checked |
| 16:38.51 | [TK]D-Fender | nvrpunk: pastebin the whole context |
| 16:38.52 | [TK]D-Fender | ~pb |
| 16:38.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:38.55 | [TK]D-Fender | ^^^^^^^^^^^ |
| 16:39.38 | Katty | ah HA! |
| 16:40.22 | *** join/#asterisk oej (n=olle@cust-IP-11.data.tre.se) |
| 16:40.46 | FreezeS | space ? |
| 16:40.56 | FreezeS | then <- then s |
| 16:41.01 | tzanger | coppice: I discovered something |
| 16:41.10 | tzanger | perhaps it is one of those "duh" things for you |
| 16:41.18 | Katty | it's asking for pgsql e |
| 16:41.21 | Katty | but i don't know where it's act |
| 16:41.23 | Katty | at. |
| 16:41.37 | Qwell | Katty: got the pgsql -devel package(s) installed? |
| 16:41.45 | FreezeS | I think it's in one of the latest menu items |
| 16:42.07 | coppice | tzanger: don't worry. someone will already have patented it |
| 16:42.13 | Katty | i think so |
| 16:42.16 | Katty | let me double check |
| 16:42.25 | FreezeS | (rant) I really hate it when dome distros call the packages "devel" and others "dev" |
| 16:42.56 | *** join/#asterisk NeonLevel (n=NeonLeve@189.169.146.12) |
| 16:43.19 | Katty | postgresql-dev is already the newest version. <- it claims i have it |
| 16:43.35 | Qwell | Katty: there might be another package it needs too |
| 16:43.36 | FreezeS | and what other dependencies does it require ? |
| 16:43.42 | NeonLevel | hello good morning, anyone in Mexico using Protel IP lines? |
| 16:43.54 | Katty | FreezeS: that's all it says. pgsql(e) |
| 16:44.01 | drmessano | I install all the dev and perv packages |
| 16:44.12 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:44.22 | FreezeS | and the line is disabled ? |
| 16:44.23 | Katty | drmessano: perv? |
| 16:44.26 | stansmith | LOL |
| 16:44.31 | Katty | FreezeS: ic an't change it, if that's what you're asking |
| 16:44.49 | FreezeS | hmm, try a make clean before |
| 16:45.00 | FreezeS | maybe you ran it before installing psql-dev |
| 16:45.03 | drmessano | If you're going to be a deviant, may as well be a pervert.. thats what grandpappy used to say |
| 16:45.06 | drmessano | Oh wait |
| 16:45.09 | drmessano | ~sike |
| 16:45.09 | jbot | [Sike!] The preceeding statement was intented to be humourous, and may have involved false statements used in a humorous manner, or an exaggeration of a real life situation with humourous intent, but in no way should be taken as pure fact or not considered questionable at best. |
| 16:45.14 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 16:45.48 | Katty | possibility |
| 16:45.56 | Katty | i did add the postgres stuff after i complied asterisk |
| 16:45.58 | Katty | that seems logical |
| 16:46.05 | drmessano | There you go |
| 16:46.11 | Qwell | Katty: ./configure |
| 16:46.54 | Simon-- | hrm. crash in 1.4.18 immediately following "Channel 'IAX2/vancouver-22' may not have been hung up properly" ... any known bugs fixed since release? |
| 16:47.25 | *** join/#asterisk cjk (i=ldidelot@d212-66-81-88.cust.tele2.lu) |
| 16:47.29 | Katty | Qwell: that won't overwrite my conf files unless i tell it to make samples, right? |
| 16:47.34 | NeonLevel | can anyone please help to debug why am i not able to connect to a sip provider even though usging a LInksys PAP2 with the same configuration i can connect without problem, i have debug and have the info, anyone please? |
| 16:47.42 | FreezeS | Katty: correct |
| 16:47.43 | Qwell | Katty: right |
| 16:47.45 | Katty | uber. |
| 16:47.46 | cjk | hi, is there an asterisk application that does nothing except indicate a ringing for X seconds |
| 16:48.09 | Katty | make install next? |
| 16:48.09 | Simon-- | cjk: Ringing Wait(X) |
| 16:48.20 | Katty | or try to do make menuselect |
| 16:48.25 | cjk | Simon--, thanks |
| 16:48.32 | Qwell | Katty: menuselect, make sure it's enable |
| 16:48.33 | Qwell | d |
| 16:48.47 | FreezeS | it's: make clean; ./configure; make menuselect; make; make install |
| 16:48.50 | nvrpunk | [TK]D-Fender, http://www.pastebin.ca/920792 |
| 16:49.03 | Katty | magically, it's enabled! |
| 16:49.19 | *** join/#asterisk maze (i=wijnand@unaffiliated/tehmaze) |
| 16:49.24 | maze | hello |
| 16:49.43 | Katty | next make and make install? |
| 16:49.46 | Katty | but no make samples |
| 16:49.46 | FreezeS | no, it's "automagically" |
| 16:49.47 | Katty | right? |
| 16:49.49 | NeonLevel | can anyone please help to debug why am i not able to connect to a sip provider even though usging a LInksys PAP2 with the same configuration i can connect without problem, i have debug and have the info, anyone please? http://pastebin.com/d1b1de5cc |
| 16:49.50 | FreezeS | exactly |
| 16:49.53 | Katty | k |
| 16:50.03 | maze | I have a problem in my AGI script, I cannot interrupt 'STREAM FILE welcome "1234567890" 0' when pressing any digit, is there any known situation why this shouldn't work? |
| 16:50.33 | stansmith | make -j3 |
| 16:50.33 | [TK]D-Fender | nvrpunk: And please pastebin the complete CLI output of a failed call at verbose 10 |
| 16:50.35 | *** join/#asterisk shinao1 (n=shinao1@67.151.234.38) |
| 16:50.35 | Netgeeks | Good morning Katty, did you get your cable/sangoma card problem resolved? |
| 16:50.39 | tzanger | I recorded 1000Hz 0dBm (manual milliwatt tone) and I see a few frame slips here and there (maybe 5 in an 8-second clip) |
| 16:50.48 | Katty | Netgeeks: i did find out that the adtran equipment is good. |
| 16:50.58 | tzanger | coppice: however just for shits and giggles, I ran a fourier analysis on the waveform |
| 16:51.00 | Katty | Netgeeks: but my loopback cable couldn't make the sangoma card turn green. |
| 16:51.03 | tzanger | I have my 1000Hz as you'd expect |
| 16:51.08 | tzanger | but I also have this little 3000Hz spur |
| 16:51.13 | Katty | Netgeeks: i'm blaiming the t1 crossover cable... since the same guy made the loopback |
| 16:51.17 | tzanger | everything else is perfectly clean |
| 16:51.18 | Qwell | 1000Hz is bad |
| 16:51.37 | Katty | Netgeeks: i'm thinking mayhaps he can't make a cable with a good ocnnection, cause i really had to push that loopback cable in to make the adtran equipment work |
| 16:51.44 | [TK]D-Fender | NeonLevel: SIP/2.0 409 Conflict |
| 16:51.46 | Katty | Netgeeks: soon, a premade t1 crossover cable will be here, and i'll go from there. |
| 16:51.53 | coppice | tzanger: well, it didn't look all that clean, but the main energy is at 1k and 3k |
| 16:51.59 | [TK]D-Fender | NeonLevel: Looks like you're fighting with your PAP2 for the right to register with them |
| 16:52.05 | drmessano | Katty: Some guys just can't make cables |
| 16:52.33 | NeonLevel | [TK]D-Fender: i have turned off the PAP2 |
| 16:52.44 | Netgeeks | Katty: Yeah, making cables is an art, and you can screw up if you happen to use the wrong end (an end for solid versus non-solid wire, etc.) |
| 16:52.46 | tzanger | coppice: no on this new one where I am playing a digital stream (1e 0b 0b 1e 9e 8b 8b 9e) |
| 16:52.50 | [TK]D-Fender | NeonLevel: you might have to wait for it to expire |
| 16:52.52 | NeonLevel | [TK]D-Fender: could be the some sort of cache? |
| 16:52.59 | [TK]D-Fender | NeonLevel: possibly |
| 16:53.06 | Netgeeks | katty: or a cheap crimp tool will ruin your day |
| 16:53.09 | tzanger | those are the only ulaw codes coming in, and the odd slip where the next expected one is skipped |
| 16:53.20 | [TK]D-Fender | NeonLevel: But its not a config problem from what we can see... its actively refusing you due to some "conflict" |
| 16:53.27 | Katty | Netgeeks: tell me about it :/ |
| 16:53.29 | tzanger | is there something in digital audio theory that would explain why the odd frame slip gets you a 3rd harmonic? |
| 16:54.32 | nvrpunk | http://www.pastebin.ca/920800 |
| 16:54.43 | *** join/#asterisk Angel-D (n=angeldav@190.79.252.29) |
| 16:54.54 | *** join/#asterisk webar7 (n=webart@CPE0080c8f208a5-CM001371173cf8.cpe.net.cable.rogers.com) |
| 16:54.55 | nvrpunk | looks like its a codec issue? |
| 16:55.13 | nvrpunk | hrmm |
| 16:55.16 | nvrpunk | or the file doesnt exist |
| 16:55.20 | Katty | okay, asterisk has been made and madeinstalled and all that fun stuff |
| 16:55.23 | Katty | moment of truth! |
| 16:55.41 | lirakis | Katty: EXPLOSION! |
| 16:55.42 | NeonLevel | [TK]D-Fender: well this configuration is driving me nuts, one more thing, they gave me proxy 200.76.111.57 but i recieved the response from 200.52.137.115, maybe could that be a problem ? |
| 16:55.42 | lirakis | lol |
| 16:56.12 | Katty | lirakis: SUGGESTION OF INSANITY |
| 16:56.23 | [TK]D-Fender | NeonLevel: make sure your peer is "nat=no" |
| 16:56.29 | stansmith | http://filebin.ca/srmwag <--- nice clip, no? |
| 16:56.38 | *** part/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 16:56.42 | [TK]D-Fender | nvrpunk: File isn't there |
| 16:56.59 | nvrpunk | [TK]D-Fender, is it an addons package or such? or? |
| 16:57.02 | NeonLevel | well here's the problem i have no peer defined yet, i only have register sentence |
| 16:57.21 | nvrpunk | [TK]D-Fender, or something with my asterisk.conf pointing to the wrong place? |
| 16:57.37 | [TK]D-Fender | nvrpunk: Its jsut a file, Nobody said that particular file exists that I know of. You telling me you typed the name out of thin air assuming it was some pre-made wonder? |
| 16:57.54 | tzanger | heh, googling for third harmonic related documents brings up lots of induction motor theory |
| 16:58.06 | nvrpunk | [TK]D-Fender, i was just following the book, so yes I assumed that file existed |
| 16:58.19 | [TK]D-Fender | nvrpunk: Poor assumption. |
| 16:58.21 | cesar_CR | stansmith, ok thanks a lot... |
| 16:58.21 | [TK]D-Fender | ~assume |
| 16:58.22 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 16:58.23 | [TK]D-Fender | ^^^ |
| 16:58.36 | [TK]D-Fender | nvrpunk: Go record your own. |
| 16:58.55 | [TK]D-Fender | nvrpunk: "core show application record" |
| 16:59.24 | Katty | OH BOY! |
| 17:00.01 | [TK]D-Fender | stansmith: lol |
| 17:00.19 | zeeesh | how to check that zaptel is installed or not ? |
| 17:00.51 | [TK]D-Fender | zeeesh: "cat /proc/interrupts" "ztcfg -vvvv", "zap show status", etc |
| 17:00.51 | dewd | Hi I am still having problems with the cdr records in postgres (I think) the following record shows me calling and saying "Hi I am hanging up" and actually doing that: |
| 17:00.52 | FreezeS | Katty: got cdr ? |
| 17:01.01 | Katty | FreezeS: ^_^ |
| 17:01.08 | dewd | duration | billsec |
| 17:01.08 | dewd | ----------+--------- |
| 17:01.08 | dewd | <PROTECTED> |
| 17:01.09 | FreezeS | weee :) |
| 17:01.21 | Katty | now if i only that stupid card would work *grump* |
| 17:01.29 | dewd | I would expect this to be about 3 seconds .. |
| 17:01.31 | Katty | but who cares, i'll probably be in vegas in 2 week sanyway |
| 17:01.35 | FreezeS | now try playing with asterisk realtime :) |
| 17:01.40 | Katty | FreezeS: oh, i do |
| 17:01.44 | Katty | FreezeS: that's our production box |
| 17:01.50 | Katty | FreezeS: this is the new one server, to replace it |
| 17:01.51 | FreezeS | and dundi |
| 17:02.01 | Katty | one thing at a time, one thing at a time |
| 17:02.04 | FreezeS | and registrations on multiple servers :D |
| 17:02.28 | Katty | i'm sure your skills are quite wonderful |
| 17:02.35 | FreezeS | no, actually I suck |
| 17:02.45 | FreezeS | I've read those in tutorials :)) |
| 17:02.52 | dewd | is this normal or not the docs say this would be seconds |
| 17:03.18 | [TK]D-Fender | nvrpunk: Aside from that it looks like its waiting for your dial an ext even with the failed prompt |
| 17:03.26 | FreezeS | for example, I can't really make asterisk hang up when a client's registration expires |
| 17:03.42 | FreezeS | and for that we had 2 calls to germany that lasted about 2 days each |
| 17:03.44 | nvrpunk | [TK]D-Fender, yeah that goes by really fast |
| 17:04.31 | dewd | FreezeS: isn't it possible to do that via an external process ? |
| 17:04.54 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 17:04.54 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 17:05.19 | FreezeS | of course it is, I'm expecting there is a flag somewhere called "hang_up_calls_when_registration_expires" that I missed when reading the docs |
| 17:05.45 | [TK]D-Fender | FreezeS: Nope.... you can have a call without registration. the 2 events don't have anything implicitly to do with each other. |
| 17:06.08 | FreezeS | is there a way to trigger a script when the registration expires ? |
| 17:06.23 | FreezeS | or should I check every minute or so ? |
| 17:06.30 | [TK]D-Fender | FreezeS: You should make a polling process to scan for dead peers and active channels running on like a 5 min interval or so. |
| 17:06.38 | FreezeS | damn |
| 17:06.44 | FreezeS | that's exactly what I was trying to avoid |
| 17:06.44 | [TK]D-Fender | FreezeS: Sure there is.... you have the source... |
| 17:06.58 | FreezeS | yes, I also have my genetic code... |
| 17:07.21 | dewd | yeah seems logical, I don't know anything about registrations but if they can be read you'd have an external loop checking them and kicking the expired ones |
| 17:07.40 | FreezeS | (btw, I tried modifying things in the sourcecode several times, but I find it really cumbersome) |
| 17:08.14 | FreezeS | I never know if my change will affect other things and unfortunately this is a production server and I can't really test my new "features" properly |
| 17:08.59 | dewd | FreezeS: I am not really an expert with asterisk but I have been tempted to change code a few times as well as I find that not all things are how one would expect them to be |
| 17:09.11 | dewd | but there is always an solution around it |
| 17:09.14 | FreezeS | dewd: exactly |
| 17:10.04 | dewd | now If i just knew what kind of value was in those cdr records I'd know what to calculate to make it seconds which they are not obviously ? |
| 17:10.23 | *** join/#asterisk alrs (i=foobar@216.151.159.21) |
| 17:10.43 | FreezeS | but I do think there should be a flag like: only_allow_calls_from_registered_users |
| 17:11.07 | dewd | FreezeS: but most of that comes from seeing that you are working within one thread of the app |
| 17:11.53 | cesar_CR | can I install a TDM800P in a PCI-X slot ??? |
| 17:12.06 | Qwell | cesar_CR: should work |
| 17:12.16 | dewd | I think the same applies to your situation, everyone who's connected will just loop around in the contexts you create, nothing exists there that will do such checks |
| 17:12.33 | Qwell | cesar_CR: PCI-X is just PCI with some extra stuff thrown on top |
| 17:12.42 | dewd | unless you do :-) |
| 17:13.09 | cesar_CR | Qwell, ok thanks ...got it! |
| 17:13.11 | stansmith | cesar_CR: on my HP, I had to install a Proliant driver, otherwise it would not work (on RHES4) |
| 17:13.59 | dewd | so nobody writes apps that actually bill people for the seconds they call and stuff ? can't really believe that ? |
| 17:14.10 | stansmith | dewd, dude.. |
| 17:14.20 | dewd | stansmith: yeah |
| 17:14.21 | cesar_CR | stansmith, and was that difficult ?? |
| 17:14.33 | cesar_CR | cause I like this one.. http://h18004.www1.hp.com/products/quickspecs/12570_na/12570_na.html |
| 17:14.33 | FreezeS | dewd: there are a lot of apps like that |
| 17:14.37 | stansmith | cesar_CR: no, not at all, just had to find out the hard way that that is what had to be done |
| 17:14.47 | FreezeS | search for calling card apps |
| 17:14.53 | stansmith | a lot of trial and error + head scratching .... "well it works on the old hardware wtf" |
| 17:15.08 | dewd | FreezeS: great so my question about the billsec value in the cdr record is very easy to awnser |
| 17:15.27 | dewd | that's what I want to hear, and the awnser offcourse |
| 17:15.56 | dewd | the doc says seconds in integer, my db field is in integer, a call of about 3 secs gives me 38 |
| 17:16.12 | FreezeS | and one of a minute ? |
| 17:16.40 | dewd | should I assume this is 3.8 then or something ? is it a bug? the docs say .. seconds, even the field says seconds |
| 17:17.05 | FreezeS | on my machine they look normal |
| 17:17.09 | FreezeS | let me do a test |
| 17:18.05 | dewd | I'd have to experiment but that seems awkward stopwatch, phone, how much lag is in my eye/hand coordination and the gsm network .. I would think they should be seconds |
| 17:18.06 | FreezeS | duration: 22, billsec 7 |
| 17:18.19 | FreezeS | I've let it ring for a while and after several seconds I hang up |
| 17:18.37 | FreezeS | so the duration means the whole duration, including the ring time |
| 17:18.43 | dewd | ok so the systems I work on do not behave properly |
| 17:18.48 | FreezeS | and billsec means actual voice time |
| 17:19.00 | FreezeS | most probably |
| 17:19.01 | dewd | yeah with me it is alway same duration/billsec as well |
| 17:19.10 | dewd | which seems weird to me |
| 17:19.12 | FreezeS | what version are you using ? |
| 17:19.15 | dewd | as it rings as well |
| 17:19.18 | dewd | checking |
| 17:19.21 | Qwell | are you answering your calls right away? |
| 17:19.33 | FreezeS | and, are you running the call through a pra/isdn/tdm line ? or just sip-sip ? |
| 17:19.41 | dewd | 1.4.13 |
| 17:19.46 | dewd | through zap |
| 17:20.00 | FreezeS | weird... |
| 17:20.22 | dewd | yeah really I was thinking maybe the db layout is not ok |
| 17:20.25 | FreezeS | try upgrading, maybe you stumbled across a bug |
| 17:20.38 | dewd | but doc says integer, db = integer .. |
| 17:21.19 | dewd | hmm again this would not be a machine I can upgrade I am just installing the application I made more stuff is running there |
| 17:21.21 | dewd | damnit |
| 17:21.35 | dewd | not my machine as well |
| 17:21.42 | FreezeS | I think there were also some security patches a while ago |
| 17:21.55 | FreezeS | so if the machine is not properly firewalled, you can get owned |
| 17:22.17 | dewd | hmm well dinnertime I will install the other version somewhere else first |
| 17:22.43 | dewd | kinda annoying last time I needed to update as well for something .. |
| 17:22.57 | FreezeS | this always happends to me :) |
| 17:25.01 | cesar_CR | stansmith, and they have linux "drivers" ?? |
| 17:25.05 | *** join/#asterisk jeffj (n=jeff@jeffjirsa.net) |
| 17:25.14 | *** join/#asterisk NeonLevel (n=NeonLeve@189.169.146.12) |
| 17:27.28 | stansmith | HP? yea |
| 17:27.57 | stansmith | the drivers werent needed for fedora 8, centos 5, or arch linux tho |
| 17:28.10 | stansmith | only RHES 4 :-/ |
| 17:28.13 | jeffj | anyone know off the top of their head if the cisco 7936's have a sip capable image, or some other trick to get it to work with asterisk? |
| 17:28.18 | ACiDV | hmm, commit 104593 (main/file.c) break all my audio on server :( |
| 17:28.23 | ACiDV | playback prompt |
| 17:29.26 | cesar_CR | stansmith, I am planing a Debian server.. |
| 17:29.59 | cesar_CR | any impressions about it ? |
| 17:30.14 | FreezeS | debian rules :) |
| 17:30.48 | mvsn | anyone know if its possible to have an extension call another # that has a pbx, and automatically select say ext 22 from the other pbx? |
| 17:31.03 | stansmith | cesar_CR: i use debian for dev work, but never used it with asterisk |
| 17:31.11 | FreezeS | mvsn: depends on how you select the other extension |
| 17:31.24 | FreezeS | if it's via DTMF, you can send them after dialing |
| 17:31.31 | ACiDV | mvsn: use D(called:calling) option on Dial apps |
| 17:32.10 | mvsn | thank you, gives me a better direction :) |
| 17:32.32 | *** join/#asterisk bkw__ (n=brian@76.211.64.159) |
| 17:33.45 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 17:35.04 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:35.05 | cesar_CR | nobody here uses asterisk with debian ??? |
| 17:35.15 | FreezeS | I use it ONLY with debian :) |
| 17:37.11 | russellb | the entire asterisk dev team at digium runs debian or some variant :) |
| 17:37.25 | Daviey | where variant = ubuntu? |
| 17:37.33 | russellb | basically. |
| 17:37.34 | Qwell | Daviey: only the nubs |
| 17:37.38 | Daviey | heh |
| 17:37.46 | Qwell | ow |
| 17:37.51 | coppice | I think the best thing to run is a bath |
| 17:38.02 | Qwell | kinda hard to code in a bath |
| 17:38.04 | Daviey | slowlaris? |
| 17:38.20 | coppice | that's what notebooks are for |
| 17:38.32 | stansmith | what bout gentoo |
| 17:38.56 | tzanger | coppice: is there some "rule" or explanation for frame slips causing 3rd harmonics? |
| 17:38.58 | Daviey | yeah, i heard the ones using gentoo are still emerging from last year |
| 17:39.09 | stansmith | LOL |
| 17:39.14 | cesar_CR | ok ok thanks guys I got the idea... I do not want to start a distro-deathmatch :) |
| 17:39.20 | FreezeS | why ? |
| 17:39.28 | FreezeS | they're soo fun |
| 17:39.33 | ACiDV | If I revert patch r104593 (main/file.c), all my prompts are now working. Does I need to change something on my sounds directory ? else I get error like fileexists_core: buffer too small, allocating larger buffer |
| 17:39.37 | stansmith | there is no distro-deathmatch, we already know Arch is the best :) |
| 17:39.52 | tzanger | hmm these frame slips are periodic too, not random |
| 17:39.55 | FreezeS | before Debian, obviously |
| 17:39.58 | stansmith | lol |
| 17:40.08 | cesar_CR | : ) |
| 17:40.19 | *** join/#asterisk TheCops (i=henri@206-248-146-30.dsl.teksavvy.com) |
| 17:40.41 | stansmith | I like Arch cause it promotes the "linux" way, rather then the "debian" or "gentoo" etc way |
| 17:40.48 | stansmith | but this is for a diff IRC channel |
| 17:40.48 | tzanger | ha |
| 17:40.51 | tzanger | there is no 'linux way' |
| 17:40.53 | TheCops | There's a way to mix Ventrilo server and Asterisk ? |
| 17:40.53 | coppice | tzanger: Its not obvious to me that slips should look like third harmonic. gear in mind that it might not be third harmonix. there are some odd effects where you get an image as far from the end of the spectrum as the real signal is from the beginning. with a 1K tone that would also be 3k |
| 17:40.54 | tzanger | that's your first mistake |
| 17:41.04 | jeffj | the "linux way" is a mishmash of crap |
| 17:41.11 | stansmith | linux way = transferable to other distros of linux |
| 17:41.19 | tzanger | [TK]D-Fender: haha |
| 17:41.27 | jeffj | userland out of sync with kernel land, all sorts of nonsense that works by chance rather than planning |
| 17:41.29 | FreezeS | actually, linux is only the kernel, we should be talking about GNU ways here |
| 17:41.37 | Qwell | [TK]D-Fender: why marshmallows? let's roast RH users |
| 17:41.48 | tzanger | coppice: oh yes I remember reading about this, what the hell was the phenomenon called |
| 17:41.52 | [TK]D-Fender | Qwell: I'm a derivative... I wouldn't want to be next ;) |
| 17:41.57 | jeffj | right, that's a linux-ism, which is a fundamental flaw ... freebsd does it right, userland is all released and maintained by the same team |
| 17:42.09 | jeffj | in any case, sip with cisco7936, possible or no? |
| 17:42.13 | stansmith | lol |
| 17:42.20 | Qwell | jeffj: pretty sure it's not |
| 17:42.23 | coppice | [TK]D-Fender: best to use an acoustic guitar. nobody around here would trust anything electronic |
| 17:42.27 | tzanger | basically you could sample an 'image' of the signal you were interested in by placing it in your sampling window thorugh careful selection of sample rate |
| 17:42.30 | Qwell | jeffj: check ccm |
| 17:42.38 | Qwell | erm...cco? |
| 17:42.38 | stansmith | ~BSD |
| 17:42.39 | jbot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 17:42.46 | Qwell | the online cisco thingie to download firmware |
| 17:42.51 | jeffj | cco, yea |
| 17:42.59 | *** part/#asterisk TheCops (i=henri@206-248-146-30.dsl.teksavvy.com) |
| 17:43.01 | file | ACiDV: fixed in 104665 |
| 17:43.02 | tzanger | i.e. if you didn't need dc-fs, you could choose a band, place the image in that band and sample |
| 17:43.07 | ACiDV | thanks file |
| 17:43.16 | Daviey | stansmith: i mean BSD userspace tools, not GNU |
| 17:43.27 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 17:43.34 | Daviey | still linux kernel |
| 17:43.59 | coppice | tzanger: are you refering tp the fact that the shannon/nyquist rate doesn't define a maximum frequency, but a maximum bandwidth? |
| 17:44.02 | nvrpunk | ok, stupid question... but I can't find the info anywhere. If I get a package from say pbxprompts will it be encoded in g729 or are the only premades in g729 available from digium? |
| 17:44.07 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:44.21 | tzanger | coppice: I think so, yes. it was a neat effect |
| 17:44.25 | Qwell | nvrpunk: you'll have to ask them |
| 17:44.44 | nvrpunk | Qwell, hmm time vs info :/ wish it was on their site |
| 17:44.53 | Qwell | you could always convert it yourself |
| 17:45.13 | Qwell | assuming you have at least one g729 license, you can use the convert CLI command in asterisk |
| 17:45.41 | nvrpunk | Qwell, I do, do conversions come out well? |
| 17:45.50 | nvrpunk | and not like the on the fly reencoding |
| 17:46.00 | nvrpunk | er transcoding |
| 17:46.01 | nvrpunk | :) |
| 17:46.25 | nvrpunk | normal transcoding if not converted sounds very metallic |
| 17:46.53 | nvrpunk | digium being high quality than the open source implementation but both don't sound good :P |
| 17:47.10 | tzanger | coppice: another related question... when doing the fft of the ulaw miliwatt signal, it (audacity) is showing +25dBm, not 0dBm... is that just a sampling/analysis error? I would have thought osmething that loud would be very distorted |
| 17:48.19 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 17:48.42 | coppice | it can't possibly be 25dBm. the loudest single spectral line would be +3 0 or -3, depending which view of 0db they use |
| 17:48.58 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:49.27 | tzanger | coppice: yeah, so it is a software bug :-) |
| 17:49.50 | coppice | funky scaling, probably |
| 17:54.23 | Qwell | coppice: got any suggestions on where to research audio...stuff? |
| 17:54.45 | coppice | a university? :-\ |
| 17:54.51 | Qwell | too lazy |
| 17:54.53 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 17:55.20 | Qwell | Qwell and school don't get along |
| 17:55.23 | mintee | anyone gotten iaxmodems to stick around on asterisk 1.4.18 longer than 2-3 days? |
| 17:56.49 | flujan | hi all |
| 17:57.14 | flujan | guy, does asterisk have a command to Pickup channels? |
| 17:57.14 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 17:57.22 | flujan | the PickUp just works with extensions right? |
| 17:57.59 | coppice | pika are finally trying to sell their fax software for *. that was supposed to appear a couple of years ago |
| 17:58.19 | Qwell | is it open source? probably not |
| 17:59.20 | flujan | does PickupChan is part of asterisk or I need to install additional stuff for it? |
| 17:59.25 | Qwell | I was trolling on pika at astricon... they were talking about their channel drivers being open source, so I asked where to download it right then and there. |
| 17:59.34 | coppice | "Unlike Span DSP, PIKA Fax works independently from zaptel." - haha, they seem to be clueless |
| 17:59.49 | Qwell | coppice: didn't you know? that Steve Underwood guy made it require zaptel |
| 18:00.32 | Qwell | (I fully realize that you are one in the same) |
| 18:01.07 | coppice | it seems like theirs is the one tied to zaptel, as it works in the kernel with the drivers |
| 18:01.27 | Qwell | yeah, their skype one uses a kernel module too I think |
| 18:01.27 | Qwell | brb |
| 18:01.45 | *** join/#asterisk Leiste (n=m@217.70.133.157) |
| 18:03.26 | Leiste | Hey guys, has anybody some information about xpdif? I'm trying to get the ring status working with a polycom IP650! Polycom said it works, but we have to change something |
| 18:04.00 | stansmith | ou a riddle! |
| 18:04.14 | Leiste | in the chan_sip.c |
| 18:05.19 | Leiste | yes, i think it is one... |
| 18:05.36 | [TK]D-Fender | Leiste: *'s ringing is picked up fine for Aastra, Polycom uses a slightly different messaging varient for this IIRC (since 2.0) |
| 18:06.31 | *** join/#asterisk nirz (n=nir@mail2.tikalnetworks.com) |
| 18:06.39 | Leiste | yes, the messaging variant is xpidf |
| 18:07.33 | *** join/#asterisk qdk_ (n=qdk@195.242.194.42) |
| 18:07.58 | Leiste | the polycom guy said, that one of his customrs has it working...he changed the busy state with the ringin state...but that seems a bit confusing - it wasn't a technican who said that |
| 18:19.13 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:19.53 | *** join/#asterisk Sweet-P (n=me@ip68-111-191-246.sd.sd.cox.net) |
| 18:20.31 | *** join/#asterisk ruied (n=ruied@bl7-217-48.dsl.telepac.pt) |
| 18:21.14 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 18:22.46 | *** part/#asterisk ph0ne (n=ph0ne@dsl-207-112-93-203.tor.primus.ca) |
| 18:22.50 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:24.02 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-75-33.vif.net) |
| 18:24.09 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 18:24.28 | Yourname`` | WARNING[18546]: cdr.c:831 ast_cdr_init: CDR already initialized on '**Unknown**' -> Not that this error looks like a big problem, but what is it exactly? |
| 18:32.42 | coppice | has anyone seen this Pika FAX for asterisk thing? |
| 18:32.47 | *** part/#asterisk maze (i=wijnand@unaffiliated/tehmaze) |
| 18:37.23 | *** join/#asterisk guillote_GNU (n=guillote@host26.201-253-58.telecom.net.ar) |
| 18:38.31 | Yourname`` | I know someone who is using it, I think. |
| 18:38.36 | Yourname`` | And is loving it. |
| 18:39.11 | _ShrikE | is it T.38? |
| 18:40.16 | coppice | they don't mention T.38 |
| 18:40.40 | coppice | their press release is clueless. |
| 18:41.21 | _ShrikE | I didnt see it in there either |
| 18:42.47 | coppice | they say its in the kernel, and works for all the zaptel cards, but is not tied to zaptel. on the other hand they say spandsp is tied to zaptel while its actually independent of any hardware |
| 18:43.52 | *** join/#asterisk yenno (n=yenno@92.104.83.122) |
| 18:44.11 | _ShrikE | I'm sticking with callweaver for my faxing |
| 18:44.11 | yenno | hi, somebody knows a channel where talking about sip is allowed? :) |
| 18:44.15 | coppice | it sounds like this is the stuff they were supposed to have delivered two years ago, but back them it was only for their own cards. |
| 18:46.16 | *** join/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
| 18:46.53 | tzafrir | what's the standard way on ISDN to ask the provider to disable caller ID? |
| 18:47.21 | coppice | why would you want to? |
| 18:48.09 | tzafrir | I'm asking for a friend |
| 18:48.18 | stansmith | if ( $ARGV[$a] =~ /(.*)=(.*)/ ) ... teehee |
| 18:48.21 | stansmith | silly perl |
| 18:48.27 | yenno | hmm.. i always thought that ACK after the 200 OK are sent directly to the called party, not over asterisk - why do they still send the ack to the asterisk even there's a Contact header field? |
| 18:48.52 | *** join/#asterisk huey23 (n=yea@64.192.209.34) |
| 18:49.26 | tzafrir | For home users here there's a standard service (*42 IIRC) to block CID |
| 18:49.49 | huey23 | [TK]D-Fender: you around? |
| 18:49.57 | coppice | these things vary around the world |
| 18:50.17 | [TK]D-Fender | huey23: yup |
| 18:50.25 | martha | <PROTECTED> |
| 18:51.01 | huey23 | [TK]D-Fender: i think we are ready to take a giant step and upgrade from 1.0.7 |
| 18:51.26 | [TK]D-Fender | huey23: Yay, so much great stuff awaits you in 1.0.8! |
| 18:51.35 | *** join/#asterisk neoalex (n=chatzill@h-68-167-22-138.nycmny83.covad.net) |
| 18:52.11 | neoalex | hi guys I just installed the asterisk-mobile rpm from fedora but there's no chan_mobile in /usr/lib/asterisk/modules where it should be |
| 18:52.59 | huey23 | [TK]D-Fender: is there anything i need to know before i do this? will it affect anything after the update? |
| 18:53.58 | [TK]D-Fender | huey23: Yup, tons of stuff changed |
| 18:54.34 | huey23 | [TK]D-Fender: OK, is there a way to go from 1.08 to 1.2 then 1.4 without losing settings? |
| 18:55.21 | *** join/#asterisk shinao1 (n=shinao1@67.151.234.38) |
| 18:56.29 | [TK]D-Fender | huey23: Get reading... |
| 18:57.05 | huey23 | [TK]D-Fender: i just wanted to know if that was possible before i started reading |
| 18:58.24 | huey23 | [TK]D-Fender: i am going to have to read eventually but unfortunately i don't have the time right now |
| 18:58.44 | [TK]D-Fender | huey23: of course its possible, just a question of how long it'll take to adapt |
| 18:58.59 | [TK]D-Fender | huey23: For everything else there's consulting :) |
| 18:59.22 | huey23 | [TK]D-Fender: we have that guy but he got a real job |
| 19:01.05 | e` | Can someone take a look at my post, and maybe help me out with an iax issue i'm having? http://forums.digium.com/viewtopic.php?t=20813 |
| 19:06.16 | flush | grssshhhh |
| 19:06.34 | flush | i wanna download and install asterisk right NOW with a TDM400P card |
| 19:06.39 | flush | asterisk.org ??// |
| 19:07.44 | huey23 | http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.18.tar.gz |
| 19:08.08 | flush | omg copy |
| 19:08.27 | huey23 | that will download Asterisk right NOW |
| 19:09.13 | scooby2 | so I have a backtrace of a deadlocked asterisk. what the heck do I do now? |
| 19:09.37 | *** join/#asterisk shinao1 (n=shinao1@67.151.234.38) |
| 19:10.52 | *** part/#asterisk huey23 (n=yea@64.192.209.34) |
| 19:13.56 | *** join/#asterisk bartpbx (n=bartpbx@217.24.210.210) |
| 19:13.59 | *** join/#asterisk sp00k3y (n=asterisk@74.202.4.2) |
| 19:14.02 | *** part/#asterisk bartpbx (n=bartpbx@217.24.210.210) |
| 19:15.56 | [TK]D-Fender | flush: Last I checked you couldn't download anything with that card.... |
| 19:16.54 | flush | http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.18.tar.gz tahts good ? |
| 19:18.44 | Yourname`` | Hello. Is there an init script for 1.4.18? /etc/init.d does not have 'asterisk' in it. |
| 19:19.44 | sp00k3y | HI, Can someone explain to me how the phonebook works? I try entering the first 3 letters of an entry but I always get a "No directory entries match your search" |
| 19:19.48 | Qwell | Yourname``: make config |
| 19:20.06 | [TK]D-Fender | flush: Better direct link : http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.18.tar.gz |
| 19:20.30 | flush | aight |
| 19:21.28 | [TK]D-Fender | sp00k3y: Directory looks up according to the voicemail.conf context specified. Please note that if you did changes to your conf's, that you'll have to do "module reload app_voicemail.so" to take on your changes |
| 19:22.29 | sp00k3y | i see, lemme check on that |
| 19:23.30 | Yourname`` | Qwell: After make install? or before? |
| 19:23.38 | scooby2 | chan_zap.c: Ring requested on channel 0/2 already in use or previously requested on span 2. Attempting to renegotiating channel. |
| 19:23.51 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 19:25.16 | *** join/#asterisk NeonLevel (i=NeonLeve@200.52.142.184) |
| 19:25.57 | *** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 19:26.00 | nny_1 | I am starting to experiment with the follow me app on asterisk. My basic idea is when a call comes in to ring my cell and desk phone at the same time, and depending on which answers, relay the conversation to that channel (Zap/SIP). I also am interested in having a way to have it so if I was to walk in the office, and am still on the call to my cellphone, I could pick up the deskphone and take the conversation over. Is this possible/ been implemented/ any |
| 19:26.41 | nny_1 | lol the googles they do nothing, i found "fine me" in voip-info |
| 19:26.43 | [TK]D-Fender | nny_1: Can be done with some sizable trickery |
| 19:27.05 | stansmith | and a lil bit of black magic |
| 19:27.05 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-164-21.dsl.teksavvy.com) |
| 19:27.17 | *** part/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 19:27.23 | JenniferAkemi | does anyone know what "mexico equal access" means? |
| 19:27.27 | nny_1 | and a herring! |
| 19:27.32 | stansmith | lol |
| 19:28.03 | sp00k3y | so if say the entry in voicemail.conf is "1234 => 1234,User,user@local.com,etc... |
| 19:28.24 | sp00k3y | in the directory, if i would have to type in USE to find that entry? |
| 19:28.37 | nny_1 | [TK]D-Fender: stansmith i see via voip-info.org's find me example that you can have it call both http://www.voip-info.org/wiki/view/Asterisk+tips+findme , the second part (re establish call on the deskphone) seems to be the trickery part |
| 19:29.21 | stansmith | nny_1: i was told today not to follow anything voip-info says |
| 19:30.07 | nny_1 | stansmith: hehe seems reasonable.. i use it as a reference for what I have done wrong or what I can do wrong |
| 19:37.59 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 19:39.42 | [TK]D-Fender | nny_1: Just dial your cell with "tT" and you can jsut transfer the call off. |
| 19:40.13 | [TK]D-Fender | sp00k3y: Sounds about right, though you should have first & last names... |
| 19:41.30 | LemensTS | Anyone use solegy? |
| 19:41.40 | sp00k3y | yeah that is set in Freepbx, im just a little confused on how to navigate it with the phone.... |
| 19:42.01 | *** join/#asterisk VaNNi (n=VaNNi@216.70.165.200) |
| 19:42.24 | [TK]D-Fender | sp00k3y: you dial the last name, which means you should have a first name. And for any other problems related to this : |
| 19:42.27 | [TK]D-Fender | ~freepbx |
| 19:42.27 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:42.38 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 19:43.32 | *** join/#asterisk tobias (n=tobias@rrcs-70-62-101-155.midsouth.biz.rr.com) |
| 19:48.42 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 19:49.09 | VaNNi | newbie question... what do i need to run voip in my company? equipments, lines, etc. |
| 19:49.20 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:49.34 | docelmo | say [TK]D-Fender you have a sec I wanted to bounce a couple thoughts off ya and see if you had any ideas |
| 19:49.50 | [TK]D-Fender | docelmo: Sure, fire away |
| 19:49.52 | docelmo | VaNNi no.. You can go TDM/VOIP hybrid or not use asterisk at all.. Your choice |
| 19:50.39 | stansmith | ~stansmith |
| 19:50.46 | [TK]D-Fender | VaNNi: Everything depends on what parts of your system you want to use VoIP for, and which parts you don't |
| 19:50.48 | stansmith | "a str8 G" |
| 19:50.57 | docelmo | ok here's scenario.. Vendor -> OpenSER -> Asterisk (what I spoke on at astricon.. ) works fine with EVERYONE except this one vendor.. When I send my packets back from asterisk via openser the vendor tries to send sip packets directly to my asterisk GW behind openser. |
| 19:51.06 | docelmo | The only thing that should bypass openser is RTP |
| 19:51.12 | docelmo | but somehow his sip is.. |
| 19:51.15 | NeonLevel | hello good day i'm having a weird problem, I have two lines with my sip provider, i double checked the user password, etc. and on one line i can register perfectly, but on the other line, i cannot login, i have debuged it and the only difference is that the response of my provider comes from a different IP address that they gave me to register, that is the say a.a.a.a is the host you'll register but the answer comes from b.b.b.b... now using other de |
| 19:52.04 | [TK]D-Fender | docelmo: I'm guessing you should be having SER attempt to B2BUA between these 2. Or make sure you are properly rewriting the SIP packets |
| 19:52.17 | [TK]D-Fender | docelmo: I have no technical experience in this however |
| 19:52.43 | *** part/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:53.24 | docelmo | I personally believe its being done correctly.. However.. You know where's mcnamara.. He's a OpenSer GURU |
| 19:53.41 | VaNNi | docelmo, so we really need a voip provider, internet line, asterisk or similar? |
| 19:54.10 | docelmo | VaNNi voip provider == cost savings over TDM do you have to have it no.. |
| 19:54.17 | docelmo | But if you have money to burn ship some my way |
| 19:55.09 | docelmo | VaNNi I consult if you are looking for a pbx I can set your company up with one |
| 19:55.13 | [TK]D-Fender | VaNNi: What do you want to do exactly? |
| 19:55.26 | _ShrikE | docelmo: sounds like that provider may not be respecting the record routing |
| 19:56.05 | docelmo | The thing is they are using asterisk to send to me.. I dont get it.. its quite mind boggling.. |
| 19:56.22 | docelmo | its possible.. |
| 19:56.30 | NeonLevel | hello good day i'm having a weird problem, I have two lines with my sip provider, i double checked the user password, etc. and on one line i can register perfectly, but on the other line, i cannot login, i have debuged it and the only difference is that the response of my provider comes from a different IP address that they gave me to register, that is the say a.a.a.a is the host you'll register but the answer comes from b.b.b.b... now using other de |
| 19:56.53 | docelmo | NeonLevel dude.. rephrase your question.. it makes no sense.. |
| 19:59.20 | NeonLevel | ok sorry, i do register => 523338391548:secret:auth@200.76.111.57, th reply i get in the debug SIP/2.0 409 Conflict the weird part is the response arrives from different IP 200.52.137.115 |
| 19:59.27 | JenniferAkemi | can someone help me figure out why i can't connect to asterisk with -r? I'm getting Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 19:59.42 | stansmith | JenniferAkemi: try "asterisk -cvvv" |
| 19:59.44 | JenniferAkemi | I'm logged in as root |
| 19:59.51 | docelmo | JenniferAkemi try just asterisk |
| 19:59.53 | docelmo | then asterisk -r |
| 19:59.58 | stansmith | ^ works too |
| 20:00.03 | docelmo | or safe_asterisk if your not using a VPS |
| 20:00.16 | JenniferAkemi | that works, but shoudl'nt asterisk already be running? (it used to be) |
| 20:00.19 | stansmith | -r = reconnect to an existing asterisk service, if asterisk is not running, u will get that error |
| 20:00.26 | JenniferAkemi | when i do ps -A, safe_asterisk is in teh list |
| 20:00.45 | stansmith | does /var/run/asterisk.ctl exist? |
| 20:00.47 | JenniferAkemi | yeah |
| 20:00.50 | stansmith | ouch |
| 20:01.05 | JenniferAkemi | i googled it but most people getting that error aren't runing as root |
| 20:01.09 | JenniferAkemi | and it's permission based |
| 20:04.33 | LemensTS | Anyone here use solegy? |
| 20:04.56 | [TK]D-Fender | LemensTS: probably just as many people as the last time you asked |
| 20:05.25 | Yourname`` | 0 |
| 20:05.29 | stansmith | LOL |
| 20:05.31 | stansmith | pwn |
| 20:05.34 | LemensTS | TKD-Fender: Why arent they answring then |
| 20:06.08 | [TK]D-Fender | LemensTS: Are you expecting 150 personal "No" answers from everyone in this channel? |
| 20:06.19 | LemensTS | TKD-Fender: I must have not see thier replies i had to run to the bank |
| 20:06.47 | [TK]D-Fender | LemensTS: I'd get tired if I had to any EVERYBODY's questions for which I don't have the answer they're looking for... |
| 20:06.51 | JenniferAkemi | heh. naming my server asterisktwo makes it very difficult to grep my logs for "asterisk" :( |
| 20:06.55 | stansmith | i mean, sometimes i ask the same question a couple times in the same day becuase new people come and go |
| 20:07.06 | [TK]D-Fender | LemensTS: There WERE no replies, and thats what scroll-back is for. |
| 20:07.38 | LemensTS | TKD-Fender: I know i was being a smartass |
| 20:07.52 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-67-188.vif.net) |
| 20:08.01 | [TK]D-Fender | stansmith: Yes, but < 1/2 hour with few joins = waste of time |
| 20:08.23 | [TK]D-Fender | LemensTS: First step is admitting yuo have a problem... only 11 to go... |
| 20:08.28 | cpm | hint, it is rare that a question that begins with 'Anyone' will have any replies. |
| 20:08.48 | LemensTS | Anyone want a million dollars i can paypal it |
| 20:08.59 | stansmith | i kept asking people if they had used app_swift (cepstral + asterisk)..in the time it took someone to respond (no one did) i was able to figure it out myself |
| 20:09.18 | cpm | nice! |
| 20:11.13 | *** join/#asterisk hfb (n=hfb@pool-72-67-142-193.lsanca.dsl-w.verizon.net) |
| 20:11.42 | tzanger | stansmith: that's mine1 |
| 20:11.46 | tzanger | I dropped it at that place |
| 20:11.48 | stansmith | o yea? |
| 20:11.55 | stansmith | u love taco bell's in columbus ohio? |
| 20:13.13 | J4k3 | stansmith: I know a guy who found a $10k bank stack the other day |
| 20:13.15 | J4k3 | on the street |
| 20:13.21 | J4k3 | consecutive serials, all clean |
| 20:13.39 | J4k3 | he went to the bank and was like "uhm, I found this" and they were like "shh its yours" |
| 20:14.10 | stansmith | LOL |
| 20:14.14 | stansmith | thats pretty sick |
| 20:14.23 | stansmith | im thinking about investing my 100 bucks |
| 20:14.37 | stansmith | but 10K? i would of bought a new hyundai |
| 20:16.46 | J4k3 | haha |
| 20:16.56 | J4k3 | I didn't get $10k and I bought a new hyundai anyways |
| 20:17.10 | stansmith | lol, hyundais are were its at |
| 20:17.37 | J4k3 | http://www.intrastar.net/~jsuter/Twins.JPG |
| 20:17.51 | J4k3 | luckily neither of those were harmed when I vehicular manslaughtered a deer the other night |
| 20:18.05 | J4k3 | I used my work truck (old dodge ram 1500) for that. |
| 20:18.17 | stansmith | haha |
| 20:18.22 | Yourname`` | I'd invest the 10k in NASA. |
| 20:18.25 | JenniferAkemi | cpm, how do you think the best way to phrase a question to get replies is |
| 20:18.40 | J4k3 | Yourname``: I'd use the $10k to bet against NASA. |
| 20:18.49 | J4k3 | whats the odds on a shuttle disaster? 1 in 75? |
| 20:18.53 | J4k3 | :D |
| 20:19.12 | J4k3 | odds of finding shuttle parts in my back yard: prolly about the same |
| 20:19.19 | cpm | JenniferAkemi, http://catb.org/~esr/faqs/smart-questions.html |
| 20:20.30 | JenniferAkemi | cpm: i try not to ask the easy questions, but when i have a question that can't be answered by google it seems i get no replies either because no one else knows either, or because it's too complicated or too isolated a problem etc. |
| 20:20.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:21.14 | Yourname`` | lol |
| 20:21.36 | stansmith | i learned this time last year how to "ask the right question" |
| 20:21.47 | JenniferAkemi | i need that skill :) |
| 20:22.04 | stansmith | JenniferAkemi: in google, when you search, you can do stuff like "site:www.yoursite.com" as part of the query to JUST search in that site |
| 20:22.18 | Qwell | ~smart questions |
| 20:22.18 | jbot | ACTION plops a heatsink on questions's head so questions's brain so it doesn't overheat. |
| 20:22.20 | JenniferAkemi | yeah. i know. thanks |
| 20:22.22 | Yourname`` | J4k3: Fine, Google? |
| 20:22.24 | Qwell | silly bot |
| 20:23.03 | Yourname`` | JenniferAkemi: I love Teksavvy. |
| 20:23.07 | stansmith | that was the craziest thing when i found out u can do that |
| 20:23.17 | JenniferAkemi | hey so do i :) screw bell ;) |
| 20:23.25 | Yourname`` | Amen! |
| 20:23.54 | Yourname`` | And their stupid support lines in God knows which country. |
| 20:24.30 | *** join/#asterisk DirtyD (n=DirtyD@ool-18bddaa0.dyn.optonline.net) |
| 20:24.58 | JenniferAkemi | i've rarely if ever called their support, but their billing is horrible. 5 months in a row bell kept messing up different things before we finally left |
| 20:25.19 | J4k3 | ~sexualchocolate |
| 20:25.20 | jbot | I told you that boy could sing |
| 20:25.22 | flush | what distro to run asterisk ? |
| 20:25.25 | stansmith | LOL |
| 20:25.34 | DirtyD | Hello. I havfe a T200P and a TDM2400P installed on the same box. My T200 is up and running.. However, I'm having trouble with the TDM2400P.. First of all, is there a tool of sort that I can use to query the board to see how it's the modules are configured? |
| 20:25.37 | flush | lulz |
| 20:25.38 | stansmith | flush: ask 10 people get 10 diff answers |
| 20:25.50 | Yourname`` | That's what I mean actually. Billing. |
| 20:25.51 | flush | if i install backtrack |
| 20:26.02 | stansmith | backtrack is debian based? [confirm/deny] |
| 20:26.08 | JenniferAkemi | the best was that they screwed up again at the end after we left by sending us a $90 cheque (we had pricematched teksavvy originally which they translated into 12 monthly $10 credits) to cover the remaining (into the future) credits on our account |
| 20:26.12 | flush | deny |
| 20:26.14 | stansmith | pwn |
| 20:26.19 | flush | pwntz |
| 20:26.20 | stansmith | i always thought it was |
| 20:26.31 | stansmith | flush: asterisk will work on just about any distro |
| 20:26.47 | stansmith | gnu/linux that is |
| 20:26.51 | stansmith | dunno bout others |
| 20:26.53 | flush | wonderful |
| 20:26.58 | Yourname`` | Bell sucks. Bigtime! |
| 20:27.34 | J4k3 | ring ding dong, ring a ding ding ding dong |
| 20:27.38 | *** join/#asterisk nirz (n=nir@84.228.109.118) |
| 20:27.40 | DirtyD | How can I probe a TDM2400 I don't have physical access to the machine.. I need to check what types of FXO/S modules are installed on the board. |
| 20:27.52 | stansmith | lsmod |
| 20:27.56 | stansmith | lsmod | grep zap |
| 20:27.58 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 20:28.04 | flush | yo |
| 20:28.05 | stansmith | dmesg | tail |
| 20:28.19 | DirtyD | thanks! |
| 20:28.25 | flush | i have tdm400p and a voxzone x100p |
| 20:28.34 | flush | can i have both card plugged with no problem |
| 20:28.56 | flush | have the 2 cards slotted in but use tdm400p only |
| 20:29.33 | *** join/#asterisk jmacz (n=jmacz@190.90.35.212) |
| 20:29.38 | jmacz | Hi everyone |
| 20:30.10 | jmacz | are there any know issues in trying to connect an Asterisk 1.2 box with an Asterisk 1.4 box through IAX2? |
| 20:30.15 | *** join/#asterisk Havokmon (n=Havokmon@h69-128-237-204.69-128.unk.tds.net) |
| 20:30.25 | Qwell | jmacz: should work fine |
| 20:31.03 | jmacz | Qwell, great. Thank you for answering |
| 20:31.15 | stansmith | flush: a computer doesnt know how to use the device for which it doesnt have a driver for |
| 20:31.21 | stansmith | typically |
| 20:32.31 | tzafrir | flush, was that a statement or a question? |
| 20:34.05 | Havokmon | I have 11 Aastra 9133i |
| 20:34.07 | Havokmon | doh |
| 20:34.13 | jmacz | Do you guys have any suggestion (tried Google already) where I can find documentation about: "chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1" and "chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1" errors? |
| 20:34.22 | jmacz | The best I've found is: http://asteriskguru.org/tutorials/hdlc_bad_fcs.html |
| 20:34.28 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 20:34.54 | waverly360 | Any of you familiar with the new SAS feature in the AudioCodes 11X devices? |
| 20:35.35 | jmacz | I have some calls dropped and have no hard dissk r/w related troubles nor shared interrupts and zttest is doing just great (only some inbound calls are dropped) |
| 20:36.47 | jmacz | But I got a lot of those HDLC-related messages (tried changing the PRI card, and the box with same result). |
| 20:37.55 | *** join/#asterisk amorphka (n=amorphka@netacc-gpn-2-202.gprs.pgsm.hu) |
| 20:38.23 | stansmith | perl xml parsing is tits |
| 20:38.28 | stansmith | wrong channel sorry |
| 20:39.50 | [TK]D-Fender | jmacz: pastebin "cat /proc/interrupts" , "dmesg" , "ztcfg -vvvv" |
| 20:42.44 | Havokmon | I have 11 9133i's that randomly show 'No Service'. I've wiped the configs clean, removed unnecessary programming (buttons/xml), changed from DNS to IP based registration, and removed the use of the hub for a PC - but I can't get rid of that error. Other than the random 'No Service' blip, they work fine. Anyone run into this? Xlite seems fine, but I don't know that it would behave the same... |
| 20:44.50 | *** part/#asterisk shtoom (n=godson@59.93.116.244) |
| 20:46.21 | *** join/#asterisk chavigny (n=nrp@c-67-171-147-26.hsd1.or.comcast.net) |
| 20:52.08 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 20:52.29 | *** join/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
| 20:53.02 | bancus | Hey guys, I just rebooted the box running Asterisk and now I'm getting an "inappropriate ioctl (25)" error when trying to bring up my TDM card |
| 20:53.16 | *** join/#asterisk Q3Man (n=Steve@wsip-24-234-155-62.lv.lv.cox.net) |
| 20:53.29 | bancus | It's only happening on the FXO port. |
| 20:53.35 | Q3Man | anyone know if there is any kind of problem with on-demand recording on zap channels? |
| 20:53.41 | bancus | Google is being somewhat unhelpful. |
| 20:53.48 | DirtyD | ok. how to I reference the channels on my TDM2400 when my TE200 already takes up channels 1-24? |
| 20:53.51 | errr | does anyone know the max value for volgain= in voicemail.conf? |
| 20:53.52 | bancus | I see a couple other people with similar problems, but no solutions. |
| 21:01.26 | LemensTS | Dirty: 25-Whatever |
| 21:08.59 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:09.06 | fujin | anyone had any experiencing HAing Asterisk? |
| 21:09.13 | fujin | i'm using heartbeat v1 but it doesn't cut the mustard. |
| 21:09.17 | fujin | I need something better, and stable |
| 21:09.24 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 21:10.19 | supjigator | fujin: What exactly are you wanting to do? |
| 21:10.38 | grandpapadot | FYI: no sharks in this channel --- what's a shark? |
| 21:10.39 | jmacz | fujin, I haven't but otrher Asterisk guys at my company do, and they moved to heartbeat2 (xml syntax), after lots of trouble with heartbeat 1 |
| 21:10.40 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 21:11.17 | fujin | supjigator: have hot/warm failover |
| 21:11.34 | fujin | jmacz: I see.. heartbeat 1 appeared to work correctly in testing |
| 21:11.38 | supjigator | tdm of just sip? |
| 21:11.39 | jmacz | fujin, yes. TDM failover with a Redfone |
| 21:11.40 | fujin | but now that we have it in production I'm seeing some issues with it |
| 21:11.49 | fujin | supjigator: nah - just SIP failover |
| 21:12.33 | supjigator | real failover or just killing the active calls on failure. |
| 21:12.49 | fujin | failover where if one of the boxes dies the other comes up |
| 21:12.55 | fujin | virtual IP address, asterisk started |
| 21:13.01 | *** join/#asterisk BugBlue (n=mendel@clamans.mobach.nl) |
| 21:13.02 | fujin | my as5400 takes care of the call redirection etc |
| 21:13.06 | supjigator | so just hot spare |
| 21:13.13 | fujin | (it's cofnigured to point at the virtual IP address) |
| 21:13.24 | BugBlue | evening * |
| 21:13.29 | supjigator | an auto hot spare that is |
| 21:13.42 | fujin | yeah, hot/warm.. so if asterisk01 dies (or the asterisk process dies) asterisk02 steals the IP addresses and fires itself up |
| 21:13.49 | supjigator | fujin: what is asterisk doing sipwise? |
| 21:13.55 | fujin | accepting calls |
| 21:14.09 | supjigator | vm or something more complicated? |
| 21:14.21 | *** join/#asterisk mrdigital (i=mrdigita@65-78-101-179.c3-0.drf-ubr2.atw-drf.pa.cable.rcn.com) |
| 21:14.23 | fujin | huh? |
| 21:14.26 | fujin | they're real hardware |
| 21:14.27 | mrdigital | is cisco 7970 a good phone |
| 21:14.37 | supjigator | vm = voicemail |
| 21:14.38 | grandpapadot | mrdigital: Internet telephony or LAN based? |
| 21:14.40 | fujin | oh |
| 21:14.44 | fujin | yes, I'll need to share voicemail between them |
| 21:14.45 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:14.53 | fujin | and some user-recorded alaw files too |
| 21:15.16 | mrdigital | grandpapadot: huh? |
| 21:15.19 | mrdigital | lol its connecting to asterisk |
| 21:15.24 | mrdigital | or will be once my guy at cisco ships it |
| 21:15.27 | JenniferAkemi | After a fresh reboot, I can not connect to asterisk with "asterisk -r". I get the message "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)". If i type asterisk -cvvv I get a ton of scroll that goes by too fast to read. Does the fact that asterisk -cvvv started asterisk mean that it was not running in the first place? |
| 21:15.32 | grandpapadot | mrdigital: Are you going to use the Cisco over the Internet or with a LAN based system? |
| 21:15.39 | mrdigital | Lan |
| 21:15.43 | *** join/#asterisk timeshell (n=timeshel@gw.lusi.on.ca) |
| 21:15.45 | supjigator | we don't use asterisk for vm anymore but when we did we used openser boxes to a bank of asterisk servers |
| 21:15.47 | grandpapadot | JenniferAkemi: permissions w/your asterisk.ctl |
| 21:15.48 | timeshell | Ok... I need help |
| 21:15.53 | mrdigital | 7970 >>> switch >> Asterisk\ |
| 21:15.54 | timeshell | I just started get this error: |
| 21:15.56 | JenniferAkemi | grandpapadot: I'm running as root |
| 21:16.03 | grandpapadot | mrdigital: Yes, it's a good phone, then. |
| 21:16.04 | stansmith | JenniferAkemi: yes you are correct |
| 21:16.13 | mrdigital | i got it free |
| 21:16.15 | mrdigital | good price? |
| 21:16.23 | grandpapadot | JenniferAkemi: Ick! Running as root is bad, m'kay ... |
| 21:16.30 | grandpapadot | mrdigital: lol, yea |
| 21:16.43 | JenniferAkemi | stansmith: ok, so now that I know it was not running after bootup, i need to figure out why not |
| 21:16.49 | mrdigital | i got 4 |
| 21:16.51 | timeshell | sip_write: Asked to transmit frame type 4, while native formats is 0x1 (g723)(1) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
| 21:17.03 | stansmith | JenniferAkemi: what distro? |
| 21:17.06 | timeshell | What the heck does that mean? APart from being a codec mismatch? |
| 21:17.24 | JenniferAkemi | centos |
| 21:17.29 | timeshell | My codec's are matched properly between the service providers and my phones |
| 21:17.30 | stansmith | did you compile? |
| 21:17.33 | timeshell | So why am I getting this? |
| 21:17.40 | JenniferAkemi | Linux asterisktwo 2.6.18-53.1.13.el5 |
| 21:17.41 | BugBlue | mmm what does crash... asterisk 1.6.0-beta2 or my N810 SIP client with h263p video (h263 works fine..) |
| 21:17.58 | JenniferAkemi | grandpapadot: yeah. I figured once I got everything working as root I would tackle that problem. |
| 21:18.02 | timeshell | And I only get it when calling Italy. When calling Ethiopia it works fine. |
| 21:18.04 | stansmith | JenniferAkemi: typing "make config" will set up the start up scripts |
| 21:18.09 | stansmith | err |
| 21:18.09 | stansmith | wait |
| 21:18.16 | JenniferAkemi | stansmith: i did that already |
| 21:18.17 | timeshell | Through the same providers |
| 21:18.21 | stansmith | o |
| 21:18.28 | JenniferAkemi | stansmith: should i go through and do the whole make clean make install etc thing? |
| 21:18.33 | bkruse | :] |
| 21:18.37 | grandpapadot | bkruse: What's the public IP of your asterisk box? I'll show you something cool ... |
| 21:18.41 | JenniferAkemi | heh |
| 21:18.43 | stansmith | JenniferAkemi: you can, but not neccessary |
| 21:18.55 | timeshell | SO, anyway, can anyone offer any suggestions on this? |
| 21:19.17 | mrdigital | do i need the license to use it with asterisk? |
| 21:19.17 | bkruse | grandpapadot: If you only actually could. |
| 21:19.46 | *** join/#asterisk nirz (n=nir@IGLD-83-130-102-40.inter.net.il) |
| 21:19.51 | bkruse | grandpapadot: And if you are holding out some 'mad 1337' exploit from the community, also, kind of lame. Which, I HIGHLY doubt is the case |
| 21:20.02 | JenniferAkemi | stansmith: hmmm ok. i typed chckconfig --list asterisk, and it's in the list and "on" for 2, 3, 4 and 5 |
| 21:20.29 | BugBlue | sadly it's a local bug and you have to be root (or the user asterisk is running under) |
| 21:20.39 | bkruse | BugBlue: killall -9 asterisk? |
| 21:20.42 | BugBlue | and more sadly it's a bug, but I can't figure it out.. |
| 21:20.48 | grandpapadot | bkruse: lol, no, dude, just messing around. Running as root is bad, in any case. |
| 21:20.49 | BugBlue | bkruse: that's a feature... |
| 21:21.06 | BugBlue | bkruse: start asterisk, kill -9 it, start asterisk and it hangs |
| 21:21.06 | JenniferAkemi | grandpapadot: but it means that my problem probably isn't permission based right? |
| 21:21.15 | bkruse | grandpapadot: I agree :]. I am trusting my people in IT. |
| 21:21.22 | bkruse | BugBlue: I was just kidding, but you are serious |
| 21:21.24 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 21:21.31 | BugBlue | bkruse: not for long... |
| 21:21.48 | grandpapadot | It's sooooo easy to run as a user, why run as root? |
| 21:21.52 | BugBlue | bkruse: as long as my girlfriend has no videophone yet I don't *need* videophone support... |
| 21:22.08 | BugBlue | and it works somehow.. |
| 21:22.15 | bkruse | grandpapadot: I agree, just easier as in permissions (eg webserver, sound files, mailboxes). If you do not install the correct way, you could mess it up, but I agree, non the less |
| 21:22.20 | stansmith | JenniferAkemi: who owns /etc/rc.d/init.d/asterisk ? |
| 21:22.21 | jmacz | [TK]D-Fender, http://pastebin.ca/921164 |
| 21:22.29 | BugBlue | my voicemail DOES record video.. |
| 21:22.30 | jmacz | (sorry for the late response) |
| 21:22.38 | BugBlue | however it does not play it back :-( |
| 21:22.38 | JenniferAkemi | stansmith: root |
| 21:22.41 | grandpapadot | Just use groups for that... add www-data to your asterisk user's group ... |
| 21:23.17 | bkruse | grandpapadot: Asterisk web server directory is most likely != apache2 web server directory, and/or, permissions |
| 21:23.39 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 21:23.40 | grandpapadot | I've just never had problems running as a user, even when I'm doing stuff via php through apache2 |
| 21:24.19 | [TK]D-Fender | jmacz: Ok, can't comment on it being a Rhino card. |
| 21:24.27 | JenniferAkemi | i'm not very good with linux, but does the fact that asterisk.pid has one line which is 19113 and then when i type ps 19113 i get back no proceesses help? |
| 21:24.32 | grandpapadot | Actually, I take that back, one issue I've had is app_voicemail setting restrictive modes on new voicemails.. I did a quick patch to voicemai.c on 1.2 and allis well |
| 21:24.40 | jmacz | [TK]D-Fender, it was a TE120P, with same behavior :( |
| 21:24.42 | Robba | when dialling out from and extension how can i make it display the DID of the phone i am calling from? |
| 21:25.11 | JenniferAkemi | robba: i think you need to set the callerid field but i'm not sure. |
| 21:25.16 | jmacz | [TK]D-Fender, as I write before, we tried changing the card/trademark and the box without luck |
| 21:25.26 | [TK]D-Fender | Robba: set a callerid for that device |
| 21:25.32 | Robba | yeah i tried that but it only allows a static number |
| 21:25.49 | [TK]D-Fender | Robba: then jsut set the CID in the dialplan |
| 21:25.52 | stansmith | hey you can trust me, im just like you |
| 21:26.24 | Robba | TK i'm sorry i don't exactly follow |
| 21:26.31 | pkunkra | robba, unless you can figure out a way to make that callerid field dynamic... you'll have to hard code set callerid into your dialplan |
| 21:26.40 | JenniferAkemi | could asterisk not be starting on bootup because of my odbc database or something? |
| 21:26.48 | JenniferAkemi | i recently changed it to use odbc |
| 21:26.57 | JenniferAkemi | and the odbc database is not on the same machine |
| 21:27.11 | [TK]D-Fender | Robba: set the callerid before you dial |
| 21:27.12 | JenniferAkemi | so i guess if the network wasn't up yet... it might fail? |
| 21:27.38 | Robba | does this mean i have to set callerid on every device? |
| 21:27.46 | Corydon76-dig | JenniferAkemi: have you checked your logs yet? |
| 21:28.00 | JenniferAkemi | Corydon76-dig: i'm not sure what logs i should be checking |
| 21:28.15 | JenniferAkemi | Corydon76-dig: do you mean centos logs or asterisk logs |
| 21:28.17 | Corydon76-dig | /var/log/asterisk/messages |
| 21:29.03 | Corydon76-dig | If it doesn't spew anything to the logs, then check that your startup script is activated for your default init level |
| 21:29.31 | JenniferAkemi | ok. i am rebooting again after clearing my messages and i will look at them |
| 21:30.00 | [TK]D-Fender | Robba: Means you need to get a clue about what it is you want to do exactly. |
| 21:30.28 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 21:30.39 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583194.dsl.bell.ca) |
| 21:31.08 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
| 21:31.13 | Robba | ok when dialling out from a sip device i want it to display the DID of the extesion, currently its displaying the main number of our 100 number range on our ISDN circuit |
| 21:31.31 | DirtyD | ok. how to I reference the channels on my TDM2400 when my TE200 already takes up channels 1-24? |
| 21:32.17 | *** part/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 21:34.13 | JenniferAkemi | ok i looked at my logs, and it is doing the same thing over and over |
| 21:34.33 | *** join/#asterisk duckz (n=duckz@81-31-157-49.vm.dnshosting.it) |
| 21:34.49 | JenniferAkemi | http://www.pastebin.ca/921193 |
| 21:35.59 | *** join/#asterisk oej (n=olle@212.17.143.194) |
| 21:37.42 | JenniferAkemi | hm i have an idea. i set my ODBCINI in my .bash_profile but that only happens when i login right. maybe it's not set on bootup thing |
| 21:38.04 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 21:41.47 | *** part/#asterisk bancus (i=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
| 21:41.47 | husimon | greets |
| 21:42.32 | husimon | has anyone ever gotten 7935 or 7936 cisco phones to work with asterisK? |
| 21:43.06 | Qwell | husimon: I've been trying to get my hands on one for quite some time, so I can test them |
| 21:43.20 | husimon | ah |
| 21:43.26 | husimon | yeah I'm having problems with the tftp stuff |
| 21:43.42 | husimon | they keep cycling grabbing the same config file |
| 21:43.45 | Qwell | tftp isn't where I would expect problems |
| 21:43.56 | Qwell | though, the tftp is pretty dumb in them |
| 21:43.57 | husimon | yeah |
| 21:44.04 | Qwell | they're...finicky |
| 21:44.36 | husimon | yeah I spent a week trying to figure out what messed up configs the 7940s require |
| 21:45.05 | Qwell | very similar configs would work with the 7935, I would imagine |
| 21:45.05 | husimon | they were on some really old sccp image so it took some finesse to get them to sip 8.3 |
| 21:45.14 | Qwell | ahh, yeah |
| 21:45.15 | husimon | Qwell, yeah I'm trying the same xml files |
| 21:45.31 | husimon | the 7935 is asking for SEPXXXXXXXXX.cnf which is some binary cisco file |
| 21:45.36 | husimon | so i have no idea how i'll even upgrade that one |
| 21:45.43 | husimon | instead of the normal asking for cnf.xml |
| 21:45.49 | Qwell | ha |
| 21:45.58 | Qwell | same as the...umm...12sp+ |
| 21:46.02 | nny_1 | has anyone actually seen a sip client for a cell phone/wifi handset? |
| 21:46.06 | husimon | weak |
| 21:46.11 | nny_1 | ex: crackberry etc |
| 21:46.39 | Qwell | the 7936, however, should use the xml configs |
| 21:46.53 | husimon | yeah it does |
| 21:46.58 | Qwell | but? |
| 21:47.06 | husimon | kept looping asking for the SEPXXXXXXXXXXXX.cnf.xml |
| 21:47.09 | husimon | even though it exists |
| 21:47.21 | Qwell | yeah, it's probably failing for other reasons. |
| 21:47.25 | husimon | I need to play with it a little more |
| 21:47.26 | Qwell | find a 7961 config, and try that |
| 21:47.34 | husimon | how come 7961? |
| 21:47.36 | Qwell | 7961, and NOT 7960.. |
| 21:47.41 | Qwell | because it's the "new-style" config |
| 21:47.52 | husimon | you think the 7936 are the new style too? |
| 21:47.57 | Qwell | yes |
| 21:48.30 | husimon | It is sad how lame the cisco docs are on this stuff |
| 21:48.38 | husimon | well hopefully that will work, I'll try that. |
| 21:48.46 | husimon | guess I have two 7935 bricks |
| 21:48.49 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:48.53 | Qwell | bricks? |
| 21:48.58 | husimon | well i can't use them with asterisk |
| 21:49.02 | husimon | might as well be |
| 21:49.10 | Qwell | send one my way, I'd give it a good home :) |
| 21:49.13 | husimon | unless you know a way to generate that .cnf file |
| 21:49.27 | husimon | any idea? |
| 21:49.43 | Qwell | well, I imagine the config is the same as the 12sp+. it was a fairly simple config |
| 21:49.43 | outtolunc | i'll take the other <G> |
| 21:49.56 | SwK | h0 h0 h0 |
| 21:50.03 | Qwell | just had to hex edit some file, and add like...what was it, 17 bytes? |
| 21:50.14 | Qwell | all 17 bytes were understood, afaik |
| 21:50.27 | Qwell | it was like IP, port, version, and a few other things |
| 21:50.38 | SwK | anyone have a need for real CNAM dips? and not the whacky google scripts that fail 3/4s of the time? msg me |
| 21:51.04 | Qwell | husimon: the 12sp+ page on the wiki had a sample, I think |
| 21:51.18 | SwK | whats up qwell |
| 21:51.20 | husimon | Qwell, interesting I'll look that up, hopefully I can upgrade it to a newer sccp image that won't ask for .cnf |
| 21:51.30 | husimon | or does it not matter and you are stuck with .cnf |
| 21:51.36 | Qwell | brb |
| 21:56.52 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
| 21:56.56 | ManxPower | ~efo |
| 21:56.58 | ManxPower | ~ecfo |
| 21:56.59 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly ... |
| 21:59.42 | *** join/#asterisk OuterSpace (n=noidedud@201.230.161.178) |
| 22:01.37 | JenniferAkemi | well i edited the asterisk one in /rc5.d and added an export ODBCINI and also did my asa config script thing in there and now it works :D |
| 22:01.42 | *** part/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
| 22:01.42 | OuterSpace | hi, i have a pbx to use with digital phones, is there any digium card to avoid a SATA converter before fxo module? (something like a digital fxo module?) |
| 22:01.54 | *** part/#asterisk timeshell (n=timeshel@gw.lusi.on.ca) |
| 22:02.06 | JenniferAkemi | thanks for pointing me to the log files Corydon76-dig |
| 22:02.07 | husimon | Qwell, with the 12sp+ how do you update the firmware? |
| 22:02.18 | *** part/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
| 22:02.50 | Corydon76-dig | JenniferAkemi: glad it worked for you |
| 22:03.36 | husimon | OuterSpace, sata? |
| 22:04.04 | SwK | CLI> cnam 2564286000 |
| 22:04.05 | OuterSpace | to convert digital to analog |
| 22:04.12 | SwK | Status Number CallerName |
| 22:04.12 | SwK | 200 2564286000 DIGIUM, INC |
| 22:04.19 | Robba | ATA? |
| 22:04.59 | OuterSpace | yes |
| 22:05.16 | J4k3 | ecfo... sounds like every 5th call or so I make through verizon's network. |
| 22:05.38 | husimon | OuterSpace, so you have digital phones using sip, and you have an asterisk server |
| 22:05.47 | husimon | and you wan't to not use an ata? |
| 22:05.57 | *** part/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 22:06.00 | husimon | yes there are fx0 cards digium makes |
| 22:06.26 | [hC] | SwK: what module provides cnam? |
| 22:06.54 | SwK | [hc] one I just wrote and am waiting on some answers for before I can release it |
| 22:07.03 | [hC] | oh nice. I like it. :) |
| 22:07.24 | tzanger | coppice: www.mixdown.ca/~andrew/dump/foo.wav <-- those are my frame slips (in the milliwatt tone) ... every 2s, to the tick it seems |
| 22:07.31 | OuterSpace | but those fxo need an analog line and my pbx dont have any. im using ata converters before asterisk pbx. |
| 22:07.31 | tzanger | lock-step, I wonder if it's a DMA issue |
| 22:07.59 | SwK | [hc] I have a live CNAM service now.... its open for people like you to get access to... we're giving the CNAM module away (and are addign LNP SMS800 and LIDB support) |
| 22:08.35 | SwK | the hold up on us releasing the res module is a GPL issue cause of one of the LIBs we had to use |
| 22:08.55 | SwK | but I'll be releasing an AGI in about an hour for it |
| 22:09.07 | SwK | (i would have it done already but we just got it finnished |
| 22:10.04 | husimon | OuterSpace, what does your pbx provide then? |
| 22:10.13 | husimon | OuterSpace, some digital trunk? |
| 22:10.26 | *** join/#asterisk ph0ne (n=ph0ne@dsl-207-112-92-149.tor.primus.ca) |
| 22:11.04 | OuterSpace | yes |
| 22:11.14 | husimon | what protocol |
| 22:12.15 | Qwell | husimon: not sure.. been a long time since I did anything with my 12sp |
| 22:13.10 | OuterSpace | dont know, its a panasonic digital pbx |
| 22:13.29 | husimon | OuterSpace, uh well you need to figure that out in order to get the asterisk box to talk to it |
| 22:13.37 | OuterSpace | i just want to remove ata converters before fxo module |
| 22:13.51 | OuterSpace | so i supose i need a digital fxo module or something like that |
| 22:15.03 | OuterSpace | but cant find such module on digium |
| 22:15.11 | husimon | OuterSpace, your atas work now ? |
| 22:15.17 | OuterSpace | yes |
| 22:15.45 | J4k3 | are all the vonage pap2s unlockable? |
| 22:15.58 | *** join/#asterisk RoyK (n=roy@ip-21-60-149-91.dialup.ice.no) |
| 22:16.00 | husimon | OuterSpace, can you just throw out the pbx and use a direct pri interface card to asterisk? or are you just trying to add on |
| 22:16.06 | Qwell | J4k3: get a -NA |
| 22:16.07 | J4k3 | err 'unbranded' |
| 22:16.24 | *** part/#asterisk LemensTS (i=CustomGT@adsl-70-238-191-87.dsl.stlsmo.sbcglobal.net) |
| 22:16.35 | *** part/#asterisk amorphka (n=amorphka@netacc-gpn-2-202.gprs.pgsm.hu) |
| 22:16.50 | husimon | OuterSpace, so you have PRI->PBX->ATA->SIP->ASTERISK->SIP->PHONE |
| 22:16.51 | husimon | ? |
| 22:17.40 | OuterSpace | pstn -> pbx -> ata -> asterisk -> sip phone |
| 22:17.52 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 22:17.53 | JT | OuterSpace: sata is for hard drives ;) |
| 22:18.47 | J4k3 | Qwell: http://www.officedepot.com/a/products/123982/Digital-Expandable-VoIP-Cordless-Phone-With/ |
| 22:18.52 | J4k3 | interesting. |
| 22:19.09 | J4k3 | but if I'ma spend that kind of money, it'll be DECT |
| 22:19.37 | husimon | OuterSpace, so just pick any analog fx0 digium card? |
| 22:19.38 | husimon | http://store.digium.com/productview.php?product_code=1TDM404EF |
| 22:19.51 | OuterSpace | i dont want ata converter |
| 22:19.55 | husimon | that's a card |
| 22:20.05 | OuterSpace | gets stuck like once a week, tried 2 diferent ata |
| 22:20.16 | husimon | what are you using? |
| 22:20.35 | husimon | digium pci fx0 cards are a lot better then external stuff. |
| 22:20.45 | Qwell | fxo* |
| 22:20.46 | OuterSpace | im using digium pci fxo |
| 22:20.52 | husimon | sorry Qwell :) |
| 22:20.58 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 22:21.10 | husimon | OuterSpace, which model? |
| 22:21.17 | OuterSpace | tdm400p |
| 22:22.07 | *** join/#asterisk adihash (n=hendrix@c911f198.bhz.virtua.com.br) |
| 22:22.21 | husimon | i guess that model is close to 5 years old now |
| 22:22.26 | husimon | might be just a crap card |
| 22:22.30 | husimon | i'm not familiar with it |
| 22:22.48 | Qwell | OuterSpace: what problems are you having with it? |
| 22:23.07 | OuterSpace | the card is fine |
| 22:23.34 | OuterSpace | i have problems with ata converter, it gets stuck, so i trying to find a way to avoid'em |
| 22:24.39 | husimon | why do you need an ata converter if you have an fxo pci card |
| 22:24.50 | husimon | ... |
| 22:25.03 | OuterSpace | because the pbx gives digital extensions |
| 22:25.12 | husimon | and your fxo understands that? |
| 22:25.13 | OuterSpace | so i need an ata to convert from digital to analog |
| 22:25.14 | husimon | sorry ata |
| 22:25.27 | husimon | what ata is that? |
| 22:26.18 | OuterSpace | what you use to convert a digital extension from a pbx to an analog line ? |
| 22:26.47 | husimon | what is your ata? |
| 22:26.52 | husimon | model |
| 22:27.47 | Qwell | OuterSpace: digital extension for a digital phone? you don't... |
| 22:28.40 | husimon | yeah i'm having a hard time understanding how that is working now |
| 22:28.50 | husimon | you have some ata that has a digital fxo port? |
| 22:28.52 | OuterSpace | no model, the guy from panasonic gave it to me |
| 22:29.02 | husimon | ok that makes more sense now |
| 22:29.15 | OuterSpace | its a converter from digital to analog, just that |
| 22:29.20 | OuterSpace | 4 wires to 2 wires |
| 22:29.40 | *** part/#asterisk adihash (n=hendrix@c911f198.bhz.virtua.com.br) |
| 22:31.02 | husimon | OuterSpace, seems like you need better atas |
| 22:31.17 | Qwell | or get rid of the digital pbx |
| 22:32.12 | husimon | yeah that's what i would try to do |
| 22:32.24 | husimon | you could also put asterisk in the front |
| 22:32.27 | husimon | instead of the other way around |
| 22:32.36 | husimon | throw two t1 pri cards in there |
| 22:32.41 | husimon | Qwell, would that work? |
| 22:33.02 | OuterSpace | so only way to connect a digital pbx to a digium fxo is with digital to analog converters ? |
| 22:34.07 | husimon | yeah i'd say so, you might be able to find a better converter though |
| 22:34.30 | Katty | jbot: voice recognition? |
| 22:34.54 | Katty | jbot: recognition? |
| 22:34.57 | Katty | jbot: :< |
| 22:34.58 | jbot | from memory, < is redirection of stdin to a program |
| 22:34.58 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 22:35.04 | husimon | OuterSpace, do you still have to use the pbx? can you just throw it out? |
| 22:35.15 | husimon | OuterSpace, are you just trying to extend the current system somehow? |
| 22:35.15 | Katty | jbot: chan_voicerecognition? |
| 22:35.18 | Katty | jbot: app? |
| 22:35.19 | jbot | Algebraic Typing and Pattern Matching Preprocessor for C++. URL: http://www.primenet.com/~georgen/app.html |
| 22:35.21 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 22:35.26 | Katty | jbot: you're not being nice, mister! |
| 22:35.27 | jbot | no *you're* not being nice, mister! |
| 22:35.34 | Katty | jbot: that's miss to you. |
| 22:35.44 | tzanger | heh |
| 22:35.48 | tzanger | Katty: get the t1 up?\ |
| 22:35.54 | Katty | tzanger: waiting on cable. |
| 22:36.00 | Katty | we're getting a premade |
| 22:36.12 | husimon | Didn't want to make one to test with? |
| 22:36.31 | Katty | someone did make it |
| 22:36.35 | Katty | and it did fail to work properly. |
| 22:36.49 | Katty | and same person made loopback, which also failed to work in a timely manor. i'm skeered. |
| 22:36.59 | outtolunc | hehe |
| 22:37.22 | husimon | maybe that person sucks at making cables |
| 22:37.22 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
| 22:37.30 | Katty | does anyone have asterisk -gvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvc in rc.local? |
| 22:37.40 | Katty | husimon: my thoughts exactly. |
| 22:37.45 | husimon | Katty, so make it yourself |
| 22:37.53 | Katty | we've been through this. |
| 22:37.54 | husimon | ;) |
| 22:37.58 | husimon | oh I wasn't aroud |
| 22:37.59 | Katty | well, maybe not YOU and me |
| 22:38.07 | Katty | they do not have proper grippies for females. |
| 22:38.13 | husimon | grippies? |
| 22:38.15 | Katty | i'm all GRRR wreched leverage! |
| 22:38.20 | Qwell | Katty: grippies? |
| 22:38.20 | husimon | lol |
| 22:38.28 | Katty | Qwell: that's the technical term. |
| 22:38.35 | Qwell | for what? O.o |
| 22:38.41 | husimon | Katty, just tape some big extensions onto the handles |
| 22:38.41 | Katty | Qwell: crimpers. |
| 22:38.46 | Katty | haha |
| 22:38.46 | husimon | give you about 1 foot of leverage |
| 22:38.47 | husimon | ;) |
| 22:38.51 | Katty | next you'll tell me to run cable with a coat hanger. |
| 22:39.12 | Qwell | Katty: do what I do - use two hands and push down on the table :P |
| 22:39.24 | Katty | sadly, the tech that made that network cable for me... has done that before )_= |
| 22:39.24 | Qwell | put some body weight into it! |
| 22:39.26 | husimon | or just work out your grip ;) |
| 22:39.27 | Katty | i don't have any |
| 22:39.38 | Qwell | Katty: I don't have much either |
| 22:39.45 | Katty | Qwell: you sit on it. |
| 22:39.47 | outtolunc | just use a flatscrew driver and a hammer !!! |
| 22:39.49 | Qwell | no ty |
| 22:39.51 | outtolunc | haha |
| 22:39.59 | Qwell | jbot: outtolunc++ |
| 22:40.05 | husimon | but seriously, I bet you could duct tape some extensions to the crimper handles |
| 22:40.26 | Katty | screw that. |
| 22:40.33 | Katty | i'll pay 10 bucks and get a premade. |
| 22:40.40 | husimon | yeah I know, i'm just pondering |
| 22:41.50 | Katty | you've missed so many technical katty terms. |
| 22:42.55 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 22:45.34 | outtolunc | (this was before they came out with that small fishing line pistol one you see at graybar) |
| 22:45.59 | husimon | Qwell, lol i just had an idea |
| 22:46.04 | Katty | holy craptacular. |
| 22:46.06 | Katty | it's 4:48! |
| 22:46.09 | Qwell | hmm? |
| 22:46.49 | Katty | (i go home at 5) |
| 22:46.58 | Katty | and last time i looked, it was like 2. |
| 22:47.03 | husimon | Qwell, the 7935s used to be used on our old call manager system, I still have the config files from that tftp server, the digital cnf files. I wonder if I could just use them and redirect the old DNS entries for the ccm servers to my asterisk box :P |
| 22:47.10 | husimon | Katty, it's 12:48pm here |
| 22:47.11 | Qwell | eww |
| 22:47.28 | husimon | Qwell, hehe, the problem is that I don't know if the format on the SEPXXXXXXxxx.cnf is the same as the SEPDefault.cnf |
| 22:47.45 | Qwell | if you have a valid SEPDefault, use it |
| 22:47.55 | Qwell | it should be only like 17 bytes |
| 22:48.14 | husimon | Qwell, the problem is it isn't asking for one |
| 22:48.17 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 22:48.21 | husimon | Qwell, it's only asking for SEPXXXXXXXXXXXx.cnf |
| 22:48.33 | Qwell | husimon: it should ask for Default after <mac> |
| 22:48.37 | husimon | and it's not even working with the one from the cisco config, i'm not sure wtf is going on. |
| 22:48.38 | Qwell | if <mac> doesn't exist |
| 22:48.45 | husimon | ah maybe i'll try that |
| 22:48.52 | husimon | so just delete it then create the other one |
| 22:49.03 | husimon | I found a perl script that generates the SEPDefault.cnf |
| 22:52.16 | husimon | omg magic |
| 22:52.16 | husimon | thanks Qwell hehe |
| 22:52.25 | husimon | now, lets watch it crash my asterisk box because sccp sux |
| 22:52.54 | husimon | grr its still just cycling asking for SEPDEFAULT.cnf |
| 22:52.57 | husimon | what the crap |
| 22:53.00 | Qwell | check case |
| 22:54.18 | husimon | did. |
| 22:54.23 | husimon | it's getting the right ip now |
| 22:54.29 | husimon | shows up in the config |
| 22:54.55 | trnzmeta | oh this conversation is coming up again |
| 22:55.06 | husimon | hehe yea |
| 22:55.13 | husimon | wanna tell me how you got yours working? |
| 22:55.19 | husimon | i'm stuck on the tftp part |
| 22:56.01 | husimon | couldn't get a hold of you last time |
| 23:01.43 | *** join/#asterisk abaci (n=IceChat7@ool-4b7fc532.static.optonline.net) |
| 23:01.48 | husimon | seriously... these phones are lame. You NEED to have something in the skinny.conf for them to stop cycling |
| 23:01.55 | husimon | unlike the 7940s |
| 23:02.09 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:03.16 | jeffj | husimon: would you be terribly offended if i asked for copies of those two configs (SEPMAC and SEPDefault) for the 7935? |
| 23:03.25 | jeffj | have a 7936 that's giving me nightmares |
| 23:04.31 | *** part/#asterisk jmacz (n=jmacz@190.90.35.212) |
| 23:04.40 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:04.55 | husimon | nope not at all |
| 23:04.59 | husimon | i don't have a sepmac config though |
| 23:05.28 | husimon | what i'll do is point you at the page I found a perl script to generate the SEPDEFAULT.cnf from |
| 23:05.29 | husimon | http://www.voip-info.org/wiki/view/cisco+79xx |
| 23:05.34 | husimon | go down near the bottom |
| 23:06.07 | Qwell | husimon: did it end up being that 17 bytes or whatever? |
| 23:06.24 | husimon | husimon, yeah that script just puts it in the right place |
| 23:06.28 | husimon | laugh |
| 23:06.28 | Qwell | cool |
| 23:06.29 | husimon | i mean Qwell |
| 23:06.37 | husimon | i'm testing dialin gnow |
| 23:07.08 | husimon | hmm |
| 23:07.11 | husimon | in extensions.conf |
| 23:07.18 | husimon | is it Skinny, or SCCP, or SKINNY/NUM |
| 23:07.24 | Qwell | Skinny/line@device |
| 23:07.38 | Qwell | for chan_skinny |
| 23:08.05 | husimon | oh |
| 23:11.16 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 23:12.27 | Qwell | work? |
| 23:12.43 | husimon | yep |
| 23:12.47 | Qwell | nice |
| 23:12.48 | husimon | talking to myself on my cell phone |
| 23:12.52 | husimon | thanks for your help |
| 23:13.11 | Qwell | somebody needs to get me access to a 7985 for a while... |
| 23:13.19 | Qwell | that would be awesome to support |
| 23:14.11 | husimon | seems almost silly heh |
| 23:14.16 | Qwell | why? |
| 23:14.20 | husimon | i mean the phone |
| 23:14.23 | *** join/#asterisk nirz (n=nir@77.125.24.166) |
| 23:14.24 | Qwell | it is |
| 23:14.32 | Qwell | but it doesn't make it any less awesome :D |
| 23:14.34 | husimon | hehe |
| 23:14.42 | husimon | let me know when you port doom and quake to it |
| 23:14.47 | Qwell | it's insanely cost prohibitive... |
| 23:14.49 | Qwell | > $3k |
| 23:15.17 | husimon | yeah |
| 23:15.45 | husimon | i'd probably just buy a $400 laptop with a webcam and skype |
| 23:16.02 | husimon | and tape it something |
| 23:16.03 | husimon | heh |
| 23:16.14 | *** join/#asterisk anthm (n=anthm@mb20736d0.tmodns.net) |
| 23:16.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:17.58 | husimon | now to see if the 7936 works |
| 23:18.31 | eric2 | anyone use the t38 modem - h323 fax client ? |
| 23:18.47 | eric2 | I'm still trying to get a fax setup going |
| 23:19.19 | husimon | hmm |
| 23:19.32 | husimon | wonder if I can get my polycom vsx 8000 sip to talk to asterisk |
| 23:24.22 | *** join/#asterisk ber___ (i=brad@neu.cow.org) |
| 23:24.46 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.98.177) |
| 23:27.25 | J4k3 | ~sexualchocolate |
| 23:27.25 | jbot | I told you that boy could sing |
| 23:31.02 | *** join/#asterisk ProtocolAgnostic (n=PA@S010600032f362ce0.cg.shawcable.net) |
| 23:32.43 | SteveTotaro | did anyone else get a dollar from redhat in the mail? |
| 23:33.04 | SteveTotaro | a James Monroe coin |
| 23:33.13 | [hC] | for what purpose? |
| 23:33.40 | [hC] | husimon: im not sure of any hardware video conferencing that speaks directly to asterisk... |
| 23:34.08 | husimon | it speaks sip |
| 23:34.14 | husimon | it has a soundstation 1000 |
| 23:34.19 | husimon | was wondering if I could use it as a phone |
| 23:34.27 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 23:34.39 | husimon | SteveTotaro, hmm I didn't |
| 23:34.56 | [hC] | oh. interesting. so with two vsx 8000's you can talk over video through asterisk via sip? |
| 23:34.59 | husimon | SteveTotaro, what kind of support do you have? I think the stupid people at the main office pay for it so they probaby ge the coin |
| 23:35.09 | ProtocolAgnostic | Evening All. Anyone has few minutes to discuss a problem I have in ringback in SIP channels? |
| 23:35.29 | husimon | [hC] not sure, but I know you can somehow pair the vsx with ccm |
| 23:35.39 | [hC] | husimon: hm. |
| 23:35.46 | husimon | would be neat if the soundstation could be used as a phone |
| 23:35.51 | husimon | it's current the mic system for the polycom |
| 23:36.08 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 23:36.31 | [hC] | husimon: im really curious to know if it would do video properly through asterisk. ive been looking for a supported video conf product |
| 23:37.24 | SteveTotaro | i have no support from redhat, lol, why?!? |
| 23:38.00 | [hC] | SteveTotaro: so you have no idea why they mailed you a coin? didnt they include a letter? |
| 23:38.01 | SteveTotaro | i haven't even opened it yet but the coin is visible through the plastic covering and it says "Dollar for your thoughts?" |
| 23:38.02 | J4k3 | asshat |
| 23:38.20 | SteveTotaro | who are you calling asshat? |
| 23:38.26 | J4k3 | redhat |
| 23:38.28 | J4k3 | asshat |
| 23:38.36 | J4k3 | sounds the same |
| 23:38.37 | J4k3 | haha |
| 23:39.00 | SteveTotaro | i was about to go "yo momma" on you |
| 23:39.03 | J4k3 | oh snap |
| 23:39.25 | SteveTotaro | it is a customer survey |
| 23:39.49 | SteveTotaro | if i fill out the survey i will get three redhat golfballs!!! |
| 23:40.06 | J4k3 | I dunno, I'm quite happy with two balls myself |
| 23:40.16 | J4k3 | I'd never take them golfing |
| 23:40.23 | SteveTotaro | standard survey, how large is your company, what is your position, what role do you play in purchasing.... |
| 23:40.45 | husimon | SteveTotaro, did you magically get the coin or did you fill our a survey for that? |
| 23:40.49 | SteveTotaro | the mailer is top quality though and the coin is a very nice touch |
| 23:41.02 | husimon | i better go check my mail |
| 23:41.03 | SteveTotaro | it is glued to the survey |
| 23:41.20 | SteveTotaro | first James Monroe coin i have seen |
| 23:41.38 | SteveTotaro | WTH is James Monroe anyway ;) |
| 23:43.15 | `Sauron | Yay, I worked around getting all those bloody political spam calls |
| 23:44.04 | `Sauron | A few playbacks in the incoming sip context for press X to continue" |
| 23:44.17 | `Sauron | and exten => whatever,1,dowhatineedtodo |
| 23:45.04 | OuterSpace | husimon: thanks for the help |
| 23:46.52 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 23:51.26 | husimon | `Sauron, do you seriously get that many? |
| 23:58.18 | *** join/#asterisk seanbright (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
