| 00:00.02 | Jaxxan | when i do a 'reload zapata' from the console, it's like nothing happens |
| 00:01.08 | russellb | there is no such command reload zapata |
| 00:01.13 | russellb | reload chan_zap.so |
| 00:02.10 | Jaxxan | still going to the wrong context )= |
| 00:02.23 | UdontKnow | what do you guys know about sipdiscount.com |
| 00:02.24 | UdontKnow | ? |
| 00:04.01 | Jaxxan | hrm |
| 00:05.40 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 00:05.51 | Jaxxan | when you make changes to zaptel.conf and zapata.conf do you have to unload/reload the modules with insmod ? |
| 00:08.07 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:10.44 | iGotNoTime | UdontKnow, is it simply a SIP client? |
| 00:10.59 | Jaxxan | from the console when i execute 'zap show channels' it displays the channels to [context a] even though i've specified them for [context b] in zapata.conf |
| 00:11.19 | Jaxxan | at this point am i just forced to restart asterisk ? |
| 00:12.26 | UdontKnow | iGotNoTime: its a sip provider |
| 00:12.50 | iGotNoTime | UdontKnow, will a vonage adapter work with them? |
| 00:13.19 | iGotNoTime | I like the website, it is very comfortable navigating |
| 00:13.24 | UdontKnow | iGotNoTime: well, if you unlock the adapter, you get trouble with vonage... but yes |
| 00:13.38 | UdontKnow | iGotNoTime: comfortable? heh. I find it sucky |
| 00:14.00 | UdontKnow | iGotNoTime: besides, I just tried to buy EUR 10 credit... and it ate my money and didnt give me credit |
| 00:14.05 | UdontKnow | haha |
| 00:14.13 | iGotNoTime | I just bought it an hour ago, it has never been plugged in to the net. Why woudl vonage make problems for me? |
| 00:14.44 | UdontKnow | iGotNoTime: no idea how vonage contract works |
| 00:15.24 | iGotNoTime | no big deal need to find a company that serves up unlocked hardware :D I am tired of spending money |
| 00:16.20 | *** join/#asterisk octothorpe_ (n=octothor@c-67-186-207-234.hsd1.ut.comcast.net) |
| 00:27.57 | *** part/#asterisk ikey (i=ikey@220.226.7.227) |
| 00:31.19 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 00:33.49 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net) |
| 00:41.43 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:42.23 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 00:47.57 | iGotNoTime | Is broadvoice an honest and reliable company? |
| 00:49.44 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 00:50.51 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-10-143.cybersurf.com) |
| 00:51.07 | pauldy | iGotNoTime: for the most part yes, they do have a level of compitence in their suport department that can leave a bit to be desired |
| 00:51.49 | pauldy | once your up and going however it usually smooth sailing |
| 00:51.53 | Qwell | heh, and their service is never working |
| 00:52.04 | Qwell | but other than that... |
| 00:52.51 | iGotNoTime | are they competive in pricing? |
| 00:53.06 | iGotNoTime | competitive |
| 00:53.06 | Qwell | $25/mo? |
| 00:53.07 | pauldy | no they are the highest out there |
| 00:53.10 | iGotNoTime | LOL |
| 00:53.12 | iGotNoTime | ok |
| 00:54.13 | iGotNoTime | Can you please suggest a decent provider to go with? |
| 00:54.24 | iGotNoTime | I have had my credit card out for an hour now |
| 00:54.44 | iGotNoTime | I just don't know which one to choose there are almost 400 links on my screen |
| 00:54.49 | pauldy | just give me the numbers and I will get you all setup |
| 00:55.14 | iGotNoTime | 4454 3432 7657 4883 expires 05/07 |
| 00:55.16 | iGotNoTime | LOL |
| 00:55.52 | iGotNoTime | seriously I even called voice pulse and they said they can't work with my adapters |
| 00:56.30 | pauldy | wth kinda adpter do you have? |
| 00:56.38 | iGotNoTime | I mean for $25 a month, why not simply use the Vonage account I cancelled yesterday? |
| 00:56.44 | iGotNoTime | pauldy, ^^^^^^ |
| 00:56.57 | [TK]D-Fender | iGotNoTime : Have you hacked it yet? |
| 00:57.11 | iGotNoTime | it is virgin, never been plugged in |
| 00:57.15 | sevard | http://phoenix.cc.edu/MegaFloppy.htm |
| 00:58.03 | iGotNoTime | If I turn the vonage back on.... can I run it through * then simply add my current SIP account to that? |
| 00:58.26 | pauldy | you cna but I wouldn't recomend it |
| 00:58.41 | *** join/#asterisk Wipe (n=louis_el@MTL-ppp-145267.qc.sympatico.ca) |
| 00:58.41 | iGotNoTime | pauldy, what do you suggest? |
| 00:59.13 | iGotNoTime | I don't care if I pay the same rate as my Vonage, as long as it is reliable and will work with * |
| 00:59.16 | mishehu | when using ztdummy in kernel 2.6.x, do you need to still edit the /etc/zaptel.conf and asterisk's zapata.conf files? |
| 00:59.25 | pauldy | just go with one of the open providers out there who support third party sip products |
| 00:59.49 | pauldy | broadvoice is pretty simple to get setup with and there are others |
| 01:00.03 | [TK]D-Fender | iGotNoTime : If I were you I'd either ditch the vonage box and get your money back, or hack it. If you ditch it, just get yourself something KNOWN and proven with *. |
| 01:00.05 | pauldy | they won't support your ata though unless you hack it |
| 01:00.14 | iGotNoTime | [TK]D-Fender, To do what pauldy just suggested the adapter must be unlocked 100% of the time |
| 01:00.24 | iGotNoTime | [TK]D-Fender, nevermind |
| 01:00.25 | *** part/#asterisk DarkFlib (n=DarkFlib@cpc4-nfds9-6-0-cust148.leic.cable.ntl.com) |
| 01:01.40 | pauldy | if you use vonage its vonage -> ata -> PSTN -> (ata|fx?) -> asterisk |
| 01:02.32 | pauldy | hakc your ata use another provider and you can be provider -> asterisk |
| 01:02.44 | pauldy | then you can use your ata for whatever if you hack it |
| 01:03.00 | iGotNoTime | So I take back this router I bought today, then simply buy a Sipura online is best? |
| 01:03.11 | pauldy | yup |
| 01:03.26 | iGotNoTime | is there an official site to buy that from |
| 01:03.37 | pauldy | unless your not into playing around with it and doing some of the kewl stuff you can do with asterisk and you just want voip |
| 01:03.47 | pauldy | then just keep it use vonage and be done with it |
| 01:04.03 | iGotNoTime | well I do want my SIP to be on it |
| 01:04.12 | pauldy | your SIP? |
| 01:04.25 | iGotNoTime | My SIP account yes |
| 01:04.42 | iGotNoTime | people instant message me from it |
| 01:05.06 | pauldy | you mena like skype,gremlin,fwd |
| 01:05.11 | CrashHD | what are your recommendations on sip tos? reliability or lowdelay? |
| 01:05.11 | iGotNoTime | like from their computer they call my wifi sipphone |
| 01:05.12 | iGotNoTime | yes |
| 01:05.13 | pauldy | err gizmo |
| 01:05.32 | iGotNoTime | yes gizmo |
| 01:05.37 | pauldy | well that isn't going to happen with straight vonage |
| 01:05.43 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 01:05.46 | iGotNoTime | correct |
| 01:05.56 | iGotNoTime | but it could with * right out the box |
| 01:06.03 | pauldy | so go the sipra route |
| 01:06.08 | iGotNoTime | will do :) |
| 01:06.20 | pauldy | yea I run gizmo to my asterisk box and broadvoice |
| 01:06.30 | iGotNoTime | frustrating too, I mean just paid $130 for a router that I can not use as I see fit |
| 01:06.34 | *** join/#asterisk xtrvd (n=j@d209-121-35-150.bchsia.telus.net) |
| 01:06.35 | pauldy | I have all 747 numbers routed to gizmo |
| 01:06.39 | iGotNoTime | how much more microsoft can you get?!!! |
| 01:06.57 | pauldy | there are some old school excs at vonage |
| 01:07.07 | pauldy | but they serve the mass market |
| 01:07.19 | pauldy | the people who just want to pick up the phone and hear dialtone |
| 01:07.41 | iGotNoTime | yes, businesswise I commend them, but I do the same with Bill Gates |
| 01:07.49 | willt | any good articles on clustering w/ asterisk? |
| 01:07.50 | iGotNoTime | doesn't mean we support them LOL |
| 01:08.03 | iGotNoTime | thanks for your time Pauldy |
| 01:08.05 | pauldy | I don't think anyone in here uses vonage |
| 01:08.07 | iGotNoTime | I am goign shopping now |
| 01:08.08 | pauldy | np |
| 01:09.29 | willt | good luck gotnotime |
| 01:12.23 | riddlebox | if I have exten =>*63,2,DISA(no-password|from-incoming) I should be able to to call in and dial *63 in my menu then get dial tone and dial a number right? |
| 01:13.06 | [TK]D-Fender | riddlebox : sure.. why not... |
| 01:13.07 | pauldy | probably may need a 1 instead of 2 |
| 01:13.28 | riddlebox | [TK]D-Fender, I get a busy tone immediatly when I try to dial a number? |
| 01:13.33 | pauldy | not sure if that generates the tone though |
| 01:13.46 | [TK]D-Fender | riddlebox : Do you get the 2nd dial-tone? |
| 01:13.49 | riddlebox | pauldy,I have an answer line above it |
| 01:13.50 | Qwell | riddlebox: You're sending it out "from-incoming" |
| 01:13.57 | pauldy | go to console and set degbu 999 set verbose 999 |
| 01:14.06 | Qwell | Does "from-incoming" have an outbound trunk? I hope it doesn't |
| 01:14.07 | riddlebox | Qwell, I had it in from-internal |
| 01:14.08 | pauldy | err debug 999 |
| 01:15.51 | iGotNoTime | I found a sipura ATA for $69 |
| 01:16.14 | iGotNoTime | but the one for $99 say eliminate long distance charges ? |
| 01:16.20 | iGotNoTime | is that a sales pitch?> |
| 01:16.43 | Qwell | No, it's a flat lie |
| 01:16.53 | iGotNoTime | haha |
| 01:16.56 | iGotNoTime | A typical user calling from a land line or mobile phone will be able to reduce and even eliminate international and long distance telephone charges by first calling their SPA-3000 via a local phone number or by using a telephone connected directly to the unit. |
| 01:17.24 | [av]bani | http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7970/sipadmn/5_0/7970difs.htm |
| 01:17.27 | iGotNoTime | it is saying anyone can call me free of charge from a PSTN |
| 01:17.27 | [av]bani | ick. |
| 01:18.06 | riddlebox | pauldy, I see Invalid extension 'thenumberIdialed', but no rule 'i' in context 'default'? |
| 01:20.09 | Qwell | iGotNoTime: again - it's a flat out lie |
| 01:20.47 | *** join/#asterisk kostagr33k (n=opa@ool-43514353.dyn.optonline.net) |
| 01:20.55 | iGotNoTime | Qwell, I figured as much :) Is there a difference in a Sipura ATA for $69 and a Digium PCI card for $79? |
| 01:21.34 | Qwell | umm, yes |
| 01:21.39 | Qwell | and no digium pci card is $79 |
| 01:21.56 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 01:22.14 | [TK]D-Fender | iGotNoTime : For starters, the PCI way means you need to run wires right into your server and have to much around with IRQ's, etc. Seconds, there is no digium card for $70 w/ 2 FXS. Then theres the matter that Digium cards are by and large locking you into * for their use. |
| 01:23.35 | iGotNoTime | [TK]D-Fender, I am not second guessing you, I am clearly new to this but is this the case with all PCI? |
| 01:23.46 | iGotNoTime | [TK]D-Fender, this one looks pretty plug and play |
| 01:23.55 | iGotNoTime | [TK]D-Fender, http://www.thevoipconnection.com/store/catalog/product_16185_Digium_TDM400P_FXO_FXS_Interface_Card.html |
| 01:23.59 | Qwell | I'll say it again |
| 01:24.01 | Qwell | no digium pci card is $79 |
| 01:24.17 | Qwell | yeah, that comes with ZERO ports |
| 01:24.50 | iGotNoTime | Qwell, ok now I've really embarrased myself :D:D |
| 01:24.58 | Qwell | though, I wonder... |
| 01:25.04 | shido6 | heh |
| 01:25.05 | shido6 | :) |
| 01:25.05 | Qwell | can a bare tdm400p be used as a timer? |
| 01:25.11 | [TK]D-Fender | iGotNoTime : Plug&Pray more like..... |
| 01:25.25 | iGotNoTime | [TK]D-Fender, hehe |
| 01:25.35 | [TK]D-Fender | iGotNoTime : Trust me you're better off with an SPA-2002. |
| 01:25.55 | iGotNoTime | [TK]D-Fender, ok I will get that one then :) |
| 01:26.08 | [TK]D-Fender | iGotNoTime : No worring about compatibility, and it handles SIP hardphone functionality better than TDM's. |
| 01:26.11 | *** join/#asterisk Ridgeback (n=jircii@104.243.8.67.cfl.res.rr.com) |
| 01:26.20 | [TK]D-Fender | AND cheaper. |
| 01:26.24 | iGotNoTime | k |
| 01:26.25 | Ridgeback | hey guys, whats up |
| 01:26.45 | kostagr33k | heyy, can someone recommend any good calling card solutions please? |
| 01:26.59 | Qwell | kostagr33k: astcc |
| 01:27.10 | shido6 | TOYWY 2.0 |
| 01:27.29 | riddlebox | weird, I commented this line, cause it was the only thing I have changed in a while and it works |
| 01:27.34 | [TK]D-Fender | shido6 : you only get it right the 2nd time ;) |
| 01:27.35 | riddlebox | ;exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) |
| 01:27.37 | Ridgeback | curious, is anyone using Asterisk as a product in thier business? |
| 01:27.42 | kostagr33k | Thanks ill take a alook into that |
| 01:27.44 | shido6 | hehe |
| 01:27.59 | Qwell | Ridgeback: no, none of us :P |
| 01:28.25 | Ridgeback | Qwell lol |
| 01:28.26 | Qwell | I'd say a good half of the people in here, are using it at work |
| 01:28.28 | shido6 | Ridgeback, whats your real question? |
| 01:28.51 | Ridgeback | I use Asterisk quite a bit, nbot at work. and would like to deplot it for small businesses |
| 01:28.58 | shido6 | go for it |
| 01:29.05 | shido6 | got a business plan? :) |
| 01:29.40 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 01:29.50 | Ridgeback | i have onee fairly well setup business plan. the only factor is determining cost benefit for small businesses |
| 01:30.05 | Ridgeback | i want to make sure they see a good return on thier investment |
| 01:30.11 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 01:30.45 | brodiem | anyone have a recommendation for a SIP (or IAX) phone that supports cordless handsets, or just any cordless/wifi phone that's decently priced? I only know of the 480i that offers the cordless handsets, and haven't seen any wifi phones that were well priced and good reviews |
| 01:31.15 | [TK]D-Fender | brodiem : Current Wifi phones SUCK. Better off with ATA's and real cordless phones. |
| 01:31.36 | MRH2 | hi ne1 know why i can;t seem to pull svn updates for 1.2 stable since 8th March (I'm stuck at rev12455) |
| 01:31.43 | shido6 | should I get some bacon? |
| 01:31.49 | brodiem | that was the consenceus I got when reading reviews |
| 01:32.06 | *** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net) |
| 01:32.19 | brodiem | [TK]D-Fender, anything with just 2.4/5.8ghz cordless? I really like the 480is but wondered if there was something a bit cheaper |
| 01:34.07 | brodiem | [TK]D-Fender, and do you know if it's a problem to use a splitter on a single port ATA to use two analog phones? |
| 01:34.16 | [TK]D-Fender | brodiem : ATA =$70 tops for 2 ports, phone costs whatever the phone costs.... 5.8 ghz? Sure.. I run Uniden flip-phones on mine |
| 01:34.53 | brodiem | [TK]D-Fender, you mean uniden analog phones? |
| 01:35.07 | [TK]D-Fender | brodiem : no problem on most. The factor you are thinking about is called REN (ringer equilvalency number). That is the limit to the # of phones supported on a given circuit. |
| 01:35.27 | [TK]D-Fender | Most any ATA would support at least 2-3 phones for sure. Decent onces handle around 7 |
| 01:35.34 | brodiem | cool |
| 01:35.46 | [TK]D-Fender | brodiem : yes, uniden analog flip-phones (ther ONLY one of its kind I've ever seen) |
| 01:36.04 | brodiem | [TK]D-Fender, got a URL? |
| 01:36.14 | [TK]D-Fender | www.uniden.com :) |
| 01:36.22 | [TK]D-Fender | its the ELT-560 I believe. |
| 01:36.22 | brodiem | lol |
| 01:36.55 | groogs | hey those are pretty cool.. |
| 01:36.59 | groogs | bluetooth too.. |
| 01:38.18 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 01:38.25 | CrashHD | with sip trunks from another provider (getting choppy call quality) is there anything I should have them change/setup on their end? |
| 01:38.26 | brodiem | yeah, sharp lookin |
| 01:38.45 | CrashHD | and is there anyway to determine sip rtp loss? |
| 01:39.03 | brodiem | CrashHD, your latency from you to them? |
| 01:39.08 | CrashHD | 80ms |
| 01:39.12 | CrashHD | average |
| 01:39.13 | CrashHD | pretty stable |
| 01:39.53 | joe | any of you know of scripts available to configure many polycom-301/501 so you don't have to do it manually? |
| 01:40.20 | brodiem | joe, does it not support downloading a config via tftp? |
| 01:41.11 | *** join/#asterisk welles (n=welles@61.150.43.114) |
| 01:42.15 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 01:42.22 | joe | brodiem: yes I was looking for scripts to generate the config files :) |
| 01:43.58 | brodiem | ah |
| 01:44.11 | brodiem | it'd be a pretty easy bash script |
| 01:44.58 | joe | brodiem: yeah, just a matter of figuring out the syntax ... which I have no idea about atm and the reason I was looking for someone who had done it already :) |
| 01:45.05 | *** join/#asterisk Samoied (n=Samoied@201.3.221.73) |
| 01:45.12 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 01:45.35 | brodiem | like, use a template file with the customizable values like: sipproxy1: ##siproxy1## could be one line of the file: cat template | sed s/##sipproxy1##/192.168.1.1/ > newfile |
| 01:46.14 | orlock | brodiem: smeserver comes with its own templating language for that sort of thing |
| 01:46.17 | brodiem | basically just do that and string along a bunch of sed pipes to replace all values, or use $1 instead of 192.168.1.1 to use the first cmd line argument passed to it |
| 01:46.30 | brodiem | $2 2nd arg, $3, etc |
| 01:48.07 | brodiem | i'm about to do a voip deployment of 30 phones and i'm just generating the templates based on a text file contains the mac addr and extensions needed for each phone |
| 01:48.59 | orlock | ahh |
| 01:49.12 | orlock | yeah, the smeserver stuff is probably more hassle than you need then |
| 01:49.57 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 01:51.34 | jeebusroxors | so im currently having a problem i think is because of bad up speeds - i can send and recieve calls through my junction networks, but i only hear audio from the POTS side for about 1 second. then nothing. i always have audio from the VOIP side though... |
| 01:51.35 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 01:51.51 | x86 | [TK]D-Fender: anything yet man? |
| 01:51.53 | x86 | :P |
| 01:52.09 | brodiem | anyone know how the spa-1001 claims it supports 2 service lines? does it mean you need to split the pair out of the rj11 for each line? I mean, does it actually separate the two phones plugged into the FXS port so that htey aren't sharing the same dial tone? |
| 01:52.12 | jeebusroxors | that sound like bad up speed to you guys? |
| 01:52.13 | [TK]D-Fender | x86 : Its at work :) |
| 01:52.24 | CrashHD | brodiem any ideas about my question from before? |
| 01:52.43 | [TK]D-Fender | brodiem : No, what it means is it supports 2 calls at a time (call-waiting / 3way calling) |
| 01:52.45 | brodiem | CrashHD, maybe you could test it with a lossier codec to see if it still persists? |
| 01:52.50 | brodiem | ahh |
| 01:53.17 | brodiem | just cause both spa-1001 and spa-2002 say 2 service lines |
| 01:53.27 | [TK]D-Fender | brodiem : and I believe it also means that it suppotrs 2 independat registrations and dial-plans to map to them. |
| 01:53.39 | [TK]D-Fender | More likely the latter. |
| 01:54.15 | CrashHD | brodiem: such as 729? |
| 01:54.26 | [TK]D-Fender | Polycom provisioning is a snap. Typically you jsut hand-build the sip.conf main, then make 3-4 templates max and then its a copy+changeuser+pas job.... |
| 01:54.27 | CrashHD | brodiem: using 711 atm |
| 01:55.30 | brodiem | [TK]D-Fender, so then I would be able to use two phoens independantly if 1 phone was in use, then use the dial prefix on the 2nd to use the other sip registration? |
| 01:55.42 | brodiem | CrashHD, yeah try something that uses less b/w like 729 or gsm |
| 01:56.05 | brodiem | brb |
| 01:56.25 | [TK]D-Fender | brodiem : nope. |
| 01:56.45 | [TK]D-Fender | brodiem : You are working with a 1 port FXS. that means 1 voice strem out the back, period. |
| 01:57.17 | [TK]D-Fender | The SPA-2002 has 2 FXS ports, each working independant of the other. |
| 01:59.28 | CrashHD | alright brodiem I'll give that a whirl |
| 01:59.54 | CrashHD | you think that could be it even with a 100mbit internet connection (tested to multiple internet servers at atlest 40mbit)? |
| 01:59.54 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 02:00.11 | robin_sz | eek, bkw |
| 02:00.54 | [av]bani | :D |
| 02:01.57 | robin_sz | and .. with that ... |
| 02:02.00 | robin_sz | im off :) |
| 02:02.39 | Jaxxan | i'd like to write a bit of information to a mySQL database when a user calls a certain number. how would i go about doing that ? |
| 02:03.03 | [TK]D-Fender | Jaxxan : res_mysql. |
| 02:04.25 | Jaxxan | so i need to upgrade from 1.0.9 to 1.2 then? |
| 02:08.43 | [TK]D-Fender | Jaxxan : There are dozens of ways of doing these sort of things... |
| 02:09.05 | [TK]D-Fender | Jaxxan : you could sysmply do a System call to pump it in from CLI if that'd work |
| 02:11.35 | Wipe | anyone here have than setup for 1100 agents ? |
| 02:13.29 | [TK]D-Fender | Wipe : I don't think * would like you much for trying that many... |
| 02:13.58 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 02:14.15 | x86 | Wipe: you'd be better off building an asterisk cluster to handle that many agents |
| 02:14.57 | Chotaire | dudes, I got one really stupid question, just that I never did that before... I want an extension whatever@my.host.name to answer a phone regardless of registration, it shall connect the caller to an extension immediately. |
| 02:15.16 | Chotaire | like if you call vmb@my.host.name you will immediately be connected to comedian. |
| 02:15.16 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 02:15.20 | Chotaire | (SIP) |
| 02:15.22 | Wipe | i read some where 512 Simultaneous Calls with Digital Recording with NFS was success |
| 02:15.37 | Qwell | nfs? |
| 02:15.56 | [av]bani | Wipe: s/NFS/ramdisk/ |
| 02:16.06 | Wipe | with 20gig ram disk |
| 02:16.08 | [TK]D-Fender | Chotaire : Very easy. |
| 02:16.10 | Chotaire | probably, yeah. |
| 02:16.13 | Wipe | yes |
| 02:16.26 | Chotaire | just give me a hint how to write the entry in sip.conf ;) |
| 02:16.53 | [TK]D-Fender | Chotaire : just set a context in [general] and allowguest=yes. |
| 02:17.32 | [TK]D-Fender | then guest calls will fall under that context and you can do whatever like "exten => fred,1,Dial(SIP/100,20)" |
| 02:17.36 | Chotaire | ah I see.. and then in that context in extensions.conf I define the extension? |
| 02:17.45 | Chotaire | superb... it was as easy as I thought ;) |
| 02:17.55 | Chotaire | thanks for the help... |
| 02:18.59 | Wipe | im looking for some one who can help me in this project he will be paid and im in montreal |
| 02:19.54 | [TK]D-Fender | why so many agents? |
| 02:20.17 | Wipe | it s big corp |
| 02:20.34 | [TK]D-Fender | Tried calling Aheeva? |
| 02:20.44 | Wipe | they have 400 agents in call center |
| 02:20.52 | Wipe | and the rest in corp |
| 02:21.30 | Wipe | you think * can't do job ? |
| 02:23.08 | [TK]D-Fender | Wipe : well.... what are they using now |
| 02:23.12 | Wipe | they have 6 PRI |
| 02:23.23 | Wipe | meridian |
| 02:23.47 | [TK]D-Fender | And what doesn't it do for them that they would want? |
| 02:24.26 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 02:25.21 | Wipe | PBX start crashing |
| 02:25.29 | Wipe | and old |
| 02:26.25 | [TK]D-Fender | Hmmm.... could be a job for Citel Handset Gateways + AudioCodes PRI gateways... |
| 02:26.48 | Wipe | everything is tighten together. one component cannot be upgraded if the other one isn't and vice-versa so they want to change every thing |
| 02:27.20 | [TK]D-Fender | They could keep their phones potentially with the gateways I suggested |
| 02:27.51 | Wipe | yes i want to go step bye step |
| 02:27.53 | *** join/#asterisk futura (n=user@12-210-203-61.client.insightBB.com) |
| 02:29.32 | [TK]D-Fender | Wipe : Could be doable, but there are questions about the kind of reporting that you'd need as well... anything special? out-bound predictive or inbound basic (but large)? |
| 02:30.37 | Wipe | in out for call center IVR |
| 02:30.39 | [TK]D-Fender | * queue's don't support categorization natively (though I've had plans on how to cheat it). |
| 02:30.43 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 02:30.58 | [TK]D-Fender | Wipe : queue for OUTBOUND as well? |
| 02:31.00 | *** join/#asterisk yxa (n=diablo@58.185.90.110) |
| 02:31.24 | Wipe | no |
| 02:31.41 | Wipe | it s more in than out |
| 02:32.03 | _Sam-- | Wipe: you are crazy if you want to setup your call center of 400 on asterisk, and you wont be able to support it |
| 02:32.08 | _Sam-- | personal opinion |
| 02:32.10 | yxa | anyone used h.323 extensively for * and tell me how good it is? |
| 02:32.13 | Wipe | call center to support there products |
| 02:32.28 | _Sam-- | * works fine, if you are a computer geek |
| 02:32.33 | _Sam-- | and can devote time to supporting it |
| 02:32.38 | [TK]D-Fender | wipe : ok, * could do the job. I'd strongly suggest you stick with all VoIP gear on the back-end ike the ones I suggested and only pass the traffic through *. recording may be an issue though... |
| 02:33.12 | [TK]D-Fender | yxa : buggy and trouble. Avoid unless NECESSARY |
| 02:33.59 | Wipe | Digital Recording with NFS can help |
| 02:34.23 | [TK]D-Fender | NFS = mistake.... better to do it loca then copy off the server. |
| 02:34.30 | _Sam-- | uh...you dont want to do a damn thing with NFS |
| 02:34.52 | Wipe | why? |
| 02:35.09 | [TK]D-Fender | NFS = No File Security and doesn't survive flooding etc, IIRC.... |
| 02:35.29 | _Sam-- | plus nfs is the mounted point becomes available can be buggy for the server that mounted it |
| 02:35.44 | [av]bani | thats true of any network filesystem |
| 02:35.54 | _Sam-- | linux nfs was particularly bad |
| 02:35.59 | _Sam-- | when the mounted host would die |
| 02:36.01 | [av]bani | the biggest problems are NFS is slow |
| 02:36.07 | _Sam-- | nfs is a crap protocol |
| 02:36.12 | [av]bani | _Sam--: you havent seen how HPUX blows up then... |
| 02:36.14 | [TK]D-Fender | Just do it local OK!?>!?! |
| 02:36.26 | Wipe | great |
| 02:36.36 | [av]bani | _Sam--: the only company who does nfs even remotely well is Sun |
| 02:36.38 | [TK]D-Fender | nd copy them off on a cron job or something... |
| 02:36.40 | yxa | [TK]D-Fender even for openh323? |
| 02:36.50 | Wipe | any one from montreal? |
| 02:36.52 | Chotaire | thanks again fender... |
| 02:36.56 | Chotaire | n8 |
| 02:36.58 | [TK]D-Fender | yxa : Just avoid... H323 is DEAD. |
| 02:37.10 | [av]bani | [TK]D-Fender: tell that to televantage |
| 02:37.32 | [av]bani | they think thats all there is |
| 02:37.36 | [av]bani | sip? whats that? |
| 02:37.45 | Wipe | thanks D-fender & sam |
| 02:38.23 | _Sam-- | Wipe: you will have your hands full if you are trying to setup a 400 seat call center, dont know asterisk, and want to do it with 1 or 2 people |
| 02:38.28 | _Sam-- | that is a big big job |
| 02:38.32 | yxa | [TK]D-Fender our incumbent is using that |
| 02:38.38 | *** join/#asterisk castrom (n=mcastro@200-122-36-229.dsl.prima.net.ar) |
| 02:38.48 | [av]bani | yay another reason to hate snom |
| 02:38.49 | *** join/#asterisk loko (n=rbrown@c-67-172-54-135.hsd1.pa.comcast.net) |
| 02:38.54 | [av]bani | they fucking blew off my bugreport |
| 02:39.16 | pigpen | Question: I am having troubles getting the Meetme to work, and I am getting: "chan_zap.c:915 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory" |
| 02:39.19 | [TK]D-Fender | Who cares who's still using it... VOODOO I tell you.... dead protocol's being reanimated by companies looking to lock you in... |
| 02:39.32 | pigpen | I do have ztdummy...do I have to have a "real" zap device for this to work? |
| 02:39.32 | castrom | can anybody help me with agi script ? |
| 02:39.37 | [TK]D-Fender | Wipe : it might work... how much recording would you be doing? |
| 02:39.42 | CrashHD | castrom what kind of agi? |
| 02:40.04 | _Sam-- | Wipe : as cheap as external USB drives are, you could probably consider maybe something like that |
| 02:40.27 | [TK]D-Fender | Wipe : Call Aheeva up and ask if their solution can scale to your needs then get the back-end hardware somewhere else. |
| 02:40.59 | castrom | i'm writing a script for calling card application, i need to see the ip addrees from agi script |
| 02:41.29 | CrashHD | you need to see which ip address? and what needs to see it? |
| 02:41.32 | loko | this question is directed to anyone using SixTel - do they have to set the caller id on their end so that brivia comm does not show up when I make calls? |
| 02:41.32 | Wipe | they want to exp 500k in this setup |
| 02:41.45 | _Sam-- | 500k canadian? |
| 02:41.51 | Wipe | yes |
| 02:41.52 | [TK]D-Fender | _Sam-- :: clearly |
| 02:41.59 | pigpen | Pesos? |
| 02:42.21 | [TK]D-Fender | Wipe : easily do-able. |
| 02:42.22 | _Sam-- | sound doable by far |
| 02:42.30 | [TK]D-Fender | Wipe : with plenty of change.... |
| 02:42.35 | Qwell | I'll do it for 600k |
| 02:42.38 | Qwell | USD |
| 02:42.39 | _Sam-- | i think if you could do it yourself, you could do it for 300 per seat easy |
| 02:42.46 | castrom | i need to know how to get the ip addrees from agi script, i didn't find any functions to get ip addrees |
| 02:43.37 | [TK]D-Fender | castrom : Take the channel name from AGI and the do a CLI call through AMI "show channel [thechannel]" |
| 02:43.41 | _Sam-- | should get a group from #asterisk to come do the install |
| 02:43.50 | Wipe | im looking for some one good in that to be part with i have 2 more place's |
| 02:44.18 | _Sam-- | Wipe: if you want good keep looking, but if you want great, you are in the right place :) |
| 02:44.29 | _Sam-- | <not that i am the great one...but there are some here> |
| 02:44.33 | Wipe | hahaha |
| 02:44.38 | [TK]D-Fender | Wipe : I'm local and available after-hours |
| 02:44.55 | *** part/#asterisk loko (n=rbrown@c-67-172-54-135.hsd1.pa.comcast.net) |
| 02:44.57 | Wipe | where local |
| 02:45.02 | Wipe | montreal |
| 02:45.02 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 02:45.05 | [TK]D-Fender | Pointe-Claire |
| 02:45.18 | [TK]D-Fender | Assez-proche? :) |
| 02:45.22 | castrom | i'll try with show channel, thks in advance |
| 02:45.31 | Wipe | tres bien |
| 02:45.55 | Wipe | can i page you |
| 02:46.38 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:46.43 | [TK]D-Fender | 1st step, ask Aheeva what they'd chard. They are downtown and do large call centers with *. Then deterime what you'd need to do on tp of the base and consult a bit of that away. |
| 02:46.44 | Ariel_ | hello everyone |
| 02:46.50 | Wipe | send me email with your phone louis_el@hotmail.com |
| 02:46.52 | [av]bani | http://www.consumerist.com/consumer/office-max/the-office-hax-guarantee-159488.php |
| 02:46.53 | [TK]D-Fender | Wipe : page how? |
| 02:47.05 | [TK]D-Fender | charge* |
| 02:47.28 | [TK]D-Fender | Wipe : PM |
| 02:47.40 | _Sam-- | [TK]D-Fender : if you need help with wipe's project i can help configure some phones or something! |
| 02:47.49 | _Sam-- | just tell me what to do...400 seats is alot for 1 man! |
| 02:48.08 | [TK]D-Fender | _Sam-- : he's running a meridian right now and I'd likely suggest Citel gateways... nothing to do there! |
| 02:48.33 | _Sam-- | with a 500k budget maybe they could get some new phones! |
| 02:49.28 | [TK]D-Fender | _Sam-- : I'm betting they have headsets they'd want to re-use which may be a problem on certain phones, and I'm SURE wiring is a real limitations. |
| 02:49.59 | _Sam-- | those phones dont use cat5 type wiring? |
| 02:50.01 | Wipe | phone m2009 |
| 02:50.08 | Wipe | meridian |
| 02:50.13 | Wipe | nortel |
| 02:50.22 | [TK]D-Fender | _Sam-- : bet on cat3 single pair running on BIX... |
| 02:50.57 | [TK]D-Fender | _Sam-- : so it be a cross-connect job to bridge to telecom style stuff instead of LAN. |
| 02:51.10 | [TK]D-Fender | _Sam-- : the joy of digital sets.... |
| 02:51.34 | _Sam-- | cat3 cant do 10baset? |
| 02:52.05 | [TK]D-Fender | _Sam-- : continue to the SINGLE PAIR comment..... |
| 02:52.15 | _Sam-- | i see...sorry |
| 02:52.58 | [TK]D-Fender | No-one in those days cared to run a full RJ45 cat3/5 to a jack... drove the cost up too much... (who needs it they said... and back THEN they were right) |
| 02:53.31 | [TK]D-Fender | besides you don't WANT 400 phones like that.... your switching needs would royally suck.... |
| 02:53.51 | *** join/#asterisk tubaman (n=ryan@gateway.britestream.com) |
| 02:54.10 | [TK]D-Fender | What you'd rather have is a mass gateway witha single fat trunk to GBIT to minimize your back-end infrastructure |
| 02:54.21 | [TK]D-Fender | Which in his case looks doable. |
| 02:54.47 | *** join/#asterisk yxa (n=diablo@58.185.90.102) |
| 02:55.32 | [TK]D-Fender | Shit this doesn't seem to support the m209 .... http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-47071457280.htm |
| 02:56.03 | _Sam-- | seems like the cost of the switches needed would be cheaper than the ports the current phones would need to plug into |
| 02:56.25 | *** part/#asterisk tubaman (n=ryan@gateway.britestream.com) |
| 02:56.57 | [TK]D-Fender | _Sam-- : 2332$ / 24 = $97 / port and no phones or switches to buy. |
| 02:57.14 | _Sam-- | switched 100 ports are a lot less than that |
| 02:57.14 | xtrvd | All from headsets? |
| 02:57.34 | _Sam-- | switched 100 port plus a crappy phone = 97 |
| 02:57.54 | [TK]D-Fender | _Sam-- : this lets them use their old phones! What phone + switch bundle would you suggest for $97 totl cost per port? |
| 02:57.57 | xtrvd | Why get rid of the phones though? |
| 02:58.12 | xtrvd | If it ain't broke, don't fix it. =) |
| 02:58.17 | *** join/#asterisk freat (n=freat@h-72-244-84-46.chcgilgm.covad.net) |
| 02:58.24 | *** join/#asterisk fugitivo (n=ajf@201.255.176.98) |
| 02:58.46 | _Sam-- | apparently it is broke...that is why they want to fix it |
| 02:58.56 | [TK]D-Fender | _Sam-- : not considering the fact they don't have the wiring for it.... |
| 02:58.59 | xtrvd | The phones are broke?... |
| 02:59.10 | [TK]D-Fender | no, their central SYSTEM is broke. |
| 02:59.53 | rpm | what is a decent voip company in canada providing DID's in alberta? i have been waiting 2 days for link2voip.com to re-activate my account and still have no phone. |
| 03:01.08 | kostagr33k | d |
| 03:03.26 | [TK]D-Fender | rpm : check the wiki for links.... |
| 03:03.38 | [TK]D-Fender | rpm : there are 3-4 I remember running into there. |
| 03:04.07 | rpm | chan_iax2.c:7398 socket_read: Registration of '.........' rejected: 'Registration Refused' from: '139.142.184.136' |
| 03:04.13 | rpm | awful awful company |
| 03:04.34 | _Sam-- | [TK]D-Fender : the Citellink thing, it registers to asterisk as just one device, then all the phones are controlled through that unit? |
| 03:07.54 | *** join/#asterisk iq (n=iq@71-38-79-126.omah.qwest.net) |
| 03:08.27 | [TK]D-Fender | _Sam-- : Typically each port would register with the single IP. |
| 03:08.33 | *** join/#asterisk ast_freak|Laptop (n=jesse@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 03:09.19 | [TK]D-Fender | They unit just makes the digital phones do what they have to to act as much like an IP phone as possible. The Norstar one is pretty neat. |
| 03:09.24 | _Sam-- | is there much config on the citellink unit? or mostly *? |
| 03:09.27 | _Sam-- | or equal? |
| 03:10.29 | [TK]D-Fender | pretty basic, just SIP accout info + button layout. |
| 03:10.32 | iq | Hi All |
| 03:10.49 | [TK]D-Fender | I believe the units can be provisioned which would make it a cust & paste deal. |
| 03:12.19 | _Sam-- | do they have any competition, or they are the only ones who make those types of devices? |
| 03:12.28 | justinu | bleh |
| 03:13.18 | [TK]D-Fender | _Sam-- : Intel has lower port density models, but frankly its all Citel's game right now.... |
| 03:14.01 | heison | does anyone here use HKBN with Asterisk? |
| 03:14.03 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 03:14.14 | [TK]D-Fender | Intel's gear can act as EITHER side though and is useful for VM integration, etc though. |
| 03:14.17 | _Sam-- | [TK]D-Fender : if they had wiring that supported ip phones, would you still lean towards the citellink? |
| 03:15.05 | heison | i'm getting SIP 403 Forbidden even following the instructions on the wiki... |
| 03:15.34 | [TK]D-Fender | _Sam-- : well.... its still <100$ port and they certainly have headsets whose compatability would have to be considered. |
| 03:16.22 | [TK]D-Fender | _Sam-- : my next suggestion would be PoE + Polycom IP301's but that'd jack the cost up and involve retraining. |
| 03:16.51 | *** join/#asterisk alephcom (n=darren@host75.net14.mcsnet.ca) |
| 03:19.02 | _Sam-- | [TK]D-Fender : those types of installations that keep older legacy phones are common at call centers of that size? |
| 03:19.13 | [TK]D-Fender | _Sam-- : so that'd be $134 for the phone + 20$ for the PoE port (on the cheap side) = 400 * 55$ = 22,000 more + retraining, and more. |
| 03:20.04 | [TK]D-Fender | _Sam-- : not sure.. I don't know the market that well.. jsut that at that size previous infrastructure screams to be given due consideration. |
| 03:21.01 | _Sam-- | i wasnt here when when that was discussed....just out of curiosity, why arent there any hardware phones that are wireless besides the portable ones |
| 03:21.34 | _Sam-- | like why couldnt he get rid of all the wiring and go wireless IP phones, if someone made them. |
| 03:21.44 | _Sam-- | im just wondering why nobody makes them |
| 03:21.57 | _Sam-- | like i have a silly little portable one that works fine |
| 03:22.01 | _Sam-- | why not a desktop model |
| 03:22.06 | [TK]D-Fender | There are WIFI phones... they just SUCK :/ |
| 03:22.16 | _Sam-- | i dont know of any desktop wifi phones |
| 03:22.19 | _Sam-- | but i dont know alot. |
| 03:22.32 | *** part/#asterisk Samoied (n=Samoied@201.3.221.73) |
| 03:22.42 | [TK]D-Fender | _Sam-- : Zultys makes one I believe.. there are 3-4 I've seen, but they are FUGLY..... |
| 03:23.46 | _Sam-- | im just trying to understand functionally why nobody really offers them |
| 03:23.52 | _Sam-- | my portable works fine |
| 03:23.57 | mrbuzz | is mpg123 required for music on hold? |
| 03:24.09 | _Sam-- | its not required if you use can use native moh |
| 03:24.13 | mrbuzz | ooh how |
| 03:24.21 | [TK]D-Fender | _Sam-- : DEMAND <- the great deciding factor. |
| 03:24.45 | _Sam-- | i would argue that there would be demand, at least from the residential segment. |
| 03:24.48 | _Sam-- | and some small biz. |
| 03:25.02 | _Sam-- | mrbuzz : http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 03:25.05 | mrbuzz | thanks |
| 03:25.26 | _Sam-- | check that part under this heading: Using native Asterisk format_mp3 for Music on Hold* |
| 03:26.36 | *** join/#asterisk iq (n=iq@71-38-79-126.omah.qwest.net) |
| 03:26.49 | _Sam-- | you may need the asterisk-addons format_mp3 |
| 03:26.52 | _Sam-- | i forget |
| 03:27.37 | _Sam-- | if you do, you just download asterisk-addons, untar it, and cd to format_mp3 |
| 03:27.45 | _Sam-- | then make install and that should take care of that |
| 03:28.11 | _Sam-- | ya, i do believe you need it for native mp3 moh |
| 03:33.22 | CrashHD | so there is no support for jitter buffer on sip channels with asterisk? |
| 03:33.32 | CrashHD | what about: http://lists.digium.com/pipermail/asterisk-dev/2005-September/015472.html ??? |
| 03:33.44 | *** join/#asterisk I-MOD (i=opticron@68.62.165.168) |
| 03:33.48 | _Sam-- | CrashHD : i dont think there is any JB for SIP...but someone said its in the works |
| 03:36.11 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-143-33.dsl.irvnca.pacbell.net) |
| 03:38.02 | CrashHD | does ooh323c have jitter support? |
| 03:39.54 | trixter | yes it will jitter all day |
| 03:41.29 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:44.20 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-18-172-7.nwrk.east.verizon.net) |
| 03:48.42 | *** join/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com) |
| 03:51.31 | xtrvd | Quick ?: How many Zap calls will a P2 233mhz support at a time? (Best guesses are also appreciated) |
| 03:51.41 | *** join/#asterisk bmg505 (n=leon@dsl-146-37-97.telkomadsl.co.za) |
| 03:53.49 | tmccrary | I could be off, but I think it can handle about 88.8 jiggawatts |
| 03:54.03 | tmccrary | my math is a little rusty |
| 03:54.58 | CrashHD | how do I set jitter buffer on an ooh323c channel? |
| 03:55.11 | xtrvd | tmccrary: I hate you. |
| 03:55.12 | xtrvd | =) |
| 03:55.39 | xtrvd | Or is that "I love you"... I can't remember, my engrish is a little rusty. |
| 03:56.00 | *** join/#asterisk xtr (i=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 03:56.16 | *** join/#asterisk iq|tablet (n=iq@71-38-79-126.omah.qwest.net) |
| 03:57.52 | CrashHD | how can I see available config options for a module? |
| 03:59.25 | *** join/#asterisk dofear (n=arodef@146-142-222-203.rev.techex.net.au) |
| 03:59.37 | dofear | Anyone using Asterisk with Quintum Gateways? |
| 04:01.12 | [TK]D-Fender | dofear : I've heard of those who have here |
| 04:02.41 | CrashHD | [TK]D-Fender does ooh323c have jitter buffer support? |
| 04:02.43 | CrashHD | I can't find an answer |
| 04:02.52 | CrashHD | I see oh323 does |
| 04:03.13 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:03.59 | *** join/#asterisk Vitux (n=vituz@cable-63-135-21-193.sudbury.dyn.personainc.net) |
| 04:05.18 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
| 04:08.03 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 04:08.06 | dofear | How good is h323 support in asterisk? |
| 04:09.10 | Nugget | slightly less awful than the state of linux gaming. |
| 04:10.20 | CrashHD | lol |
| 04:10.23 | [TK]D-Fender | "slightly" |
| 04:10.30 | dofear | Thants not a good news for interoperability between Asterisk and Quintum |
| 04:10.32 | CrashHD | it's the only means by which jitter can be achieve currently |
| 04:10.37 | dofear | Does Quintum support SIP? |
| 04:11.00 | dofear | CrashHD: You are working with H323 on Asterisk? |
| 04:11.15 | CrashHD | started playing with it yesterday |
| 04:11.22 | CrashHD | I needed jitter support |
| 04:11.29 | CrashHD | can't find another solution |
| 04:11.30 | dofear | What are you trying to do? |
| 04:11.46 | dofear | Trying to make Asterisk to speak to a H323 device? |
| 04:11.57 | CrashHD | nothing special really, I have 64 or so ip trunks which have choppy call quality |
| 04:12.02 | CrashHD | running over the internet |
| 04:12.15 | CrashHD | hoping using h323 with a little jitter buffer will help |
| 04:12.38 | dofear | Trunk between Asterisks? |
| 04:12.43 | CrashHD | nah |
| 04:12.48 | CrashHD | trunk from voip vendor to us |
| 04:12.54 | CrashHD | they are using cisco equipment I believe |
| 04:13.21 | dofear | So what you have found, no jitter buffer support in Asterisk H323 at all? |
| 04:13.33 | CrashHD | oh323 has jitter |
| 04:13.38 | CrashHD | but not a lot of support in * |
| 04:13.47 | CrashHD | ooh323c is what I'm trying now |
| 04:13.54 | CrashHD | but I don't see a clear way to utilize the buffer |
| 04:13.58 | russellb | no, oh323 doesn't have jitter, your network does. |
| 04:13.58 | CrashHD | so unless it's just built in |
| 04:14.00 | dofear | What support you are looking for which is not on oh323? |
| 04:14.18 | CrashHD | oh323 I didn't actually compile and install |
| 04:14.22 | CrashHD | the process was too in depth |
| 04:14.28 | wundaboy | is there a problem with dtmf in asterisk? |
| 04:14.37 | CrashHD | I figure anything that is that pieced together is not worthy of production machines |
| 04:15.09 | dofear | I am running through a similar problem. I need to make Asterisks to speak to some H323 Gateways |
| 04:15.18 | CrashHD | ooh323c comes packages with * (addons) and was a breeze to install |
| 04:15.23 | dofear | But I am constntly being told that it's not a good idea |
| 04:15.27 | CrashHD | try it |
| 04:15.42 | CrashHD | first hand knowledge is the only knowledge worth talking about |
| 04:16.11 | tmccrary | It could be network latency somewhere on your network (or say, high amounts of traffic through to your head end) |
| 04:16.22 | CrashHD | ya could be lots of things |
| 04:16.26 | tmccrary | You may want to look into trying quality of service |
| 04:16.28 | dofear | So you are receiving those calls from Internet in your Asterisks with H323 already? Or you are heading there? |
| 04:16.29 | CrashHD | I'm tackling it on all fronts |
| 04:16.56 | CrashHD | tmccrary: tos lowdelay or reliability? what are your thoughts (lowdelay are mine) |
| 04:17.03 | CrashHD | headed there |
| 04:17.13 | CrashHD | it's a pain to get the provider to test things with me |
| 04:17.17 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 04:17.46 | dofear | Can anyone recommend me a low cost but good IP Phone? Supports H323, SIP, IAX |
| 04:18.12 | tmccrary | the second cheapest sipura is fairly decent |
| 04:18.13 | [TK]D-Fender | dofear : What functionality do you need? |
| 04:18.17 | tmccrary | the cheapest is.... scary |
| 04:18.21 | dofear | Also I can pass message to the display of the IP Phone from Asterisk (i.e: Balance left ) |
| 04:18.56 | *** join/#asterisk BattleBridge (n=BB@h-67-103-43-82.snfccasy.covad.net) |
| 04:19.02 | [TK]D-Fender | dofear : Pushing info to an IP phone will require a higher end phone. You're lookin at $200USD + |
| 04:19.15 | [TK]D-Fender | Likely more for a Cisco |
| 04:19.35 | dofear | I have a IAX DID from local carrier terminating to my Asterisk box. I do not have any FXS card, so I am planning to get an IP Phone to receive the IAX DID |
| 04:19.43 | [TK]D-Fender | Sipura phones (Linksys) just aren't worth it.... |
| 04:19.43 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 04:20.03 | ctooley | Mar 10 04:17:21 WARNING[22643]: app_meetme.c:461 build_conf: Unable to open pseudo channel - trying device ... I'm trying to use ztdummy, is this a problem? |
| 04:20.06 | dofear | Also I want to play with H323 and SIP trunks too |
| 04:20.11 | [TK]D-Fender | dofear : You could get an ATA to allow you to use an analog phone which would work jsut fine. |
| 04:20.15 | tmccrary | IMO cheap voip phones in general are not worth it |
| 04:20.21 | ctooley | I don't remember having to do anything special but it's been a long time since I've used ztdummy |
| 04:20.38 | [TK]D-Fender | ctooley : It means ZTDUMMY didn't load and its nuking your room.... |
| 04:21.12 | ctooley | [TK]D-Fender, lsmod shows ztdummy as loaded |
| 04:21.51 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 04:22.18 | [TK]D-Fender | ctooley : Something is wrong with it... I had the same.... |
| 04:22.23 | dofear | CrashHD: So you are saying oh323 is better than ooh323c, but you need to build it yourself? |
| 04:22.51 | CrashHD | oh323 seems more feature rich, but I get an unstable vibe from it |
| 04:23.14 | CrashHD | ooh323c is said to be more stable but 10-20 % higher cpu load |
| 04:24.07 | *** join/#asterisk Djeli (n=djeli@ppp157-243.static.internode.on.net) |
| 04:24.23 | ctooley | And let me guess. MeetMe isn't going to work without a zt module |
| 04:24.45 | dofear | what is the best and most mature billing software to build a pre-paid system on asterisk? any idea? |
| 04:24.49 | [TK]D-Fender | meetme is a timer dependent app... so yeah, you're DOA without one... |
| 04:28.34 | tainted_ | dofear what's your budget |
| 04:29.23 | dofear | looking for something opensource tainted_ |
| 04:29.42 | tainted_ | u get what u pay for |
| 04:31.22 | dofear | is that why you are here in asterisk? |
| 04:31.26 | Abydos313 | what does it normally cost |
| 04:35.19 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
| 04:38.39 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 04:41.44 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 04:41.49 | Winkie | jeebusroxors: thanks for reminding me |
| 04:43.03 | tainted_ | dofear what are you implying? |
| 04:46.51 | Zipper_32 | How many Zap calls will a P2 233mhz support at a time?... |
| 04:50.27 | *** join/#asterisk mphill (n=mphill@CPE-70-92-245-5.wi.res.rr.com) |
| 04:57.44 | dofear | tainted_: i am implying asterisk is free too |
| 05:00.55 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 05:01.04 | astra^^ | hello all |
| 05:01.11 | *** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3742615.sympatico.ca) |
| 05:01.54 | astra^^ | i need some help settin up my asteriskpbx.. |
| 05:03.10 | astra^^ | any help.. ? |
| 05:03.22 | theorem_ | stuid questions get stupid answers |
| 05:03.25 | Yashy | just doing a random poll if you can get help? |
| 05:03.26 | JunK-Y | asking a specific question would be better. |
| 05:03.27 | theorem_ | ask something specific |
| 05:03.39 | JunK-Y | theorem_: u got it! |
| 05:03.40 | JunK-Y | :) |
| 05:03.50 | astra^^ | i get an error in pbx -dindi. |
| 05:03.54 | astra^^ | while installin |
| 05:04.02 | theorem_ | pastebin |
| 05:04.07 | astra^^ | i get an error in pbx -dundi. |
| 05:04.09 | theorem_ | let's see it |
| 05:04.12 | Yashy | http://www.yashy.com/help/index.php/IRC:1 |
| 05:04.37 | JunK-Y | pastebin ur error. |
| 05:05.04 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:05.24 | astra^^ | pbx_dundi.c:1408: warning: implicit declaration of function `compress' |
| 05:05.24 | astra^^ | pbx_dundi.c:1409: `Z_OK' undeclared (first use in this function) |
| 05:05.24 | astra^^ | make[1]: *** [pbx_dundi.o] Error 1 |
| 05:05.24 | astra^^ | make[1]: Leaving directory `/usr/src/asterisk/pbx' |
| 05:05.24 | astra^^ | make: *** [subdirs] Error 1 |
| 05:05.45 | theorem_ | PASTEBIN |
| 05:05.51 | *** join/#asterisk welles (n=welles@61.150.43.114) |
| 05:06.13 | theorem_ | http://pastebin.com/ |
| 05:06.27 | JunK-Y | astra^^: which * version? |
| 05:06.35 | *** join/#asterisk wellng (n=welles@61.150.43.114) |
| 05:07.41 | *** part/#asterisk welles (n=welles@61.150.43.114) |
| 05:13.45 | astra^^ | JunK-Y:1.2.3 |
| 05:16.40 | JunK-Y | astra^^: pastebin few lines before too. |
| 05:17.26 | astra^^ | sir i dint get u .. sir.. pastbin.. ? |
| 05:17.39 | JunK-Y | ~pb |
| 05:17.41 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 05:17.57 | *** join/#asterisk masked (n=masked@static-203-87-16-192.vic.chariot.net.au) |
| 05:18.02 | masked | yo |
| 05:18.04 | [TK]D-Fender | Wow, jbot just nagged me! |
| 05:18.38 | masked | is anyone here familiar with setting up an SPA-3000 to forward calls to a sip phone? |
| 05:19.13 | *** join/#asterisk Isaiah (n=test@208-187-93-4.br1.hnv.mi.frontiernet.net) |
| 05:20.32 | astra^^ | JunK-Y: si r |
| 05:20.46 | astra^^ | <PROTECTED> |
| 05:25.22 | JunK-Y | astra^^: see ur msg... |
| 05:25.36 | *** join/#asterisk coppice (n=chatzill@83.162.17.210.dyn.pacific.net.hk) |
| 05:29.12 | orlock | man |
| 05:29.30 | orlock | our shit ciscocentric voip provider has been down for the past 3 hours |
| 05:29.37 | orlock | from 1:30pm till 4:30pm |
| 05:29.39 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-143-33.dsl.irvnca.pacbell.net) |
| 05:29.40 | orlock | and still down |
| 05:35.11 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 05:36.38 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 05:39.49 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 05:59.46 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:01.29 | *** part/#asterisk tmccrary (n=tmccrary@d47-69-35-227.try.wideopenwest.com) |
| 06:06.11 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 06:11.41 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 06:12.08 | Zipper_32 | I just finished installing asterisk and I was wondering how it is able to determine the incoming zap channel and automatically route it to the 'demo' on the dialplan? |
| 06:17.19 | shido6 | zaptel.conf |
| 06:17.24 | shido6 | and zapata.conf |
| 06:17.43 | *** join/#asterisk b0xii (n=here@cpe-70-116-68-157.houston.res.rr.com) |
| 06:19.03 | *** join/#asterisk Eggplant (i=No@dsl-237.cascadeaccess.com) |
| 06:37.39 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:39.14 | *** join/#asterisk clive- (n=pirch@dsl-145-37-221.telkomadsl.co.za) |
| 06:41.51 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 06:44.58 | rpm | anyone use link2voip or freeworldtel and stuff able to dialout :P |
| 06:49.48 | *** join/#asterisk Qwell[] (n=north@unaffiliated/qwell) |
| 07:00.34 | wellng | hi all. how to turn off the native transfer ? |
| 07:01.59 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-59.claranet.co.uk) |
| 07:02.14 | *** join/#asterisk north (n=north@unaffiliated/qwell) |
| 07:04.20 | wellng | hi _Paulo_ |
| 07:05.52 | *** join/#asterisk Qwell[] (n=north@unaffiliated/qwell) |
| 07:11.42 | *** join/#asterisk leopardus (n=leopardu@217.22.179.144) |
| 07:12.15 | leopardus | help : which ports I should open to run asterisk? |
| 07:12.45 | leopardus | help : I know I need to have 5060, but I think I need more |
| 07:13.51 | Zipper_32 | SIP - 5060 Iax - 4569 |
| 07:14.11 | Zipper_32 | and the rtp ports from /etc/asterisk/rtp.conf |
| 07:15.43 | leopardus | Zipper_32 : I have rtpstart= 100000 & rtpend=20000 |
| 07:15.58 | leopardus | Zipper_32 : can u tell me how to do that with iptables?? |
| 07:18.20 | leopardus | ... |
| 09:17.14 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 09:17.14 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.5 released! (March 3, 2006) -=- Asterisk-addons 1.2.2 now available (March 6, 2006) |
| 09:17.26 | joelsolanki | I am planning for 70 to 100 calls at a time. |
| 09:17.29 | fourcheeze | joelsolanki: is there some reason why you wouldn't keep the codec the same? |
| 09:17.32 | joelsolanki | which hardware do u recommend |
| 09:18.03 | joelsolanki | Fourcheeze i will need to buy licenses for g729 for softphone :) |
| 09:18.10 | fourcheeze | joelsolanki: http://www.digium.com/index.php?menu=product_category&category=codec |
| 09:18.25 | fourcheeze | woah rewind |
| 09:18.41 | fourcheeze | you're starting of at gsm or ilbc at the softphone |
| 09:18.50 | fourcheeze | why do you want it to end up as g729? |
| 09:19.16 | joelsolanki | because our service provider supports g729 |
| 09:19.27 | joelsolanki | so i have to end up with g729. |
| 09:19.34 | joelsolanki | is there any solution ? |
| 09:19.37 | abusenode | probably to save bandwidth |
| 09:19.47 | fourcheeze | service provider doesn't support anything else? |
| 09:19.49 | joelsolanki | yes save bandwidth too. |
| 09:19.50 | abusenode | i used to transcode to 729 to do 2 calls over single 64k isdn channel |
| 09:20.13 | fourcheeze | never found one that didn't support ulaw/alaw |
| 09:20.14 | joelsolanki | no they support g711/g729 |
| 09:20.22 | abusenode | 711 is 64kbit/sec though |
| 09:20.24 | fourcheeze | is bandwidth a consideration for you? |
| 09:20.27 | abusenode | 70*that is a lot. |
| 09:20.38 | fourcheeze | more like 80kbit/sec |
| 09:20.58 | shiznatix | in the zapata.conf file, will asterisk ignore the [whatever_stuff_here] ?? |
| 09:21.01 | joelsolanki | yes bandwidth is also main concern coz users will call from there dialup connection. and here dialup connection gets maximum 40 kbps bandwidth :) |
| 09:21.14 | masked | ilbc is the way to go |
| 09:21.20 | fourcheeze | joelsolanki: but your users aren't going to be using g729 |
| 09:21.21 | joelsolanki | hmm |
| 09:21.28 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 09:21.37 | *** part/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 09:21.53 | fourcheeze | if you have a maximum of 40kbits/sec gsm is going to be a struggle |
| 09:22.03 | joelsolanki | fourcheeze: no bcoz i will have to pay for g729 licenses for softphone which i dont want. |
| 09:22.25 | fourcheeze | joelsolanki: it might work out cheaper |
| 09:22.33 | masked | stopkeylogger |
| 09:22.43 | joelsolanki | how ? |
| 09:22.51 | fourcheeze | you won't need to do any transcoding and so you can back to average hardware |
| 09:22.57 | fourcheeze | licenses are just a 1-off fee |
| 09:23.32 | masked | use ilbc. |
| 09:23.38 | joelsolanki | hmm ok. i will check with g729 rates too. |
| 09:23.48 | joelsolanki | masked: y ilbc / |
| 09:23.49 | joelsolanki | ? |
| 09:24.01 | masked | it uses 3kb |
| 09:24.05 | fourcheeze | yeah, better still twist your providers arm and make them support ilbc |
| 09:24.13 | masked | only fair chance you have for dialup |
| 09:24.27 | joelsolanki | hmm ok. |
| 09:24.28 | masked | sounds fine, works great |
| 09:24.38 | fourcheeze | masked: why doesn't everyone use it? |
| 09:24.39 | masked | gsm struggles on my adsl |
| 09:24.48 | masked | most people do. |
| 09:24.55 | joelsolanki | so i do transcoding for 70 calls which machine do i need ? p4 with 3ghz ? |
| 09:25.00 | masked | particuarally here in australia |
| 09:25.03 | fourcheeze | I find it hard to find a hardware phone with ilbc support |
| 09:25.22 | fourcheeze | see that link I posted |
| 09:25.22 | masked | i haven't found a ITSP that doesn't support it yet. |
| 09:25.32 | orlock | masked: OZZIE OZZIE OZZIE! |
| 09:25.36 | masked | OI OI OI |
| 09:25.42 | fourcheeze | they are suggesting dual 2.8GHz xeons for 80 calls |
| 09:25.45 | orlock | masked: OZZIE OZZIE OZZIE! |
| 09:25.48 | masked | OI OI OI |
| 09:25.51 | joelsolanki | ok |
| 09:25.53 | orlock | OZZIE! |
| 09:25.55 | masked | OI |
| 09:25.58 | orlock | OZZIE! |
| 09:25.59 | masked | OI |
| 09:26.02 | orlock | OZZIE! |
| 09:26.04 | masked | oi oi oi |
| 09:26.18 | masked | err |
| 09:26.19 | orlock | doh, fucked up there :) |
| 09:26.25 | masked | hahah |
| 09:26.36 | fourcheeze | pastebin? |
| 09:26.40 | austinnichols101 | ~seen opsys |
| 09:26.45 | jbot | opsys <n=opsys@68-235-141-52.miamfl.adelphia.net> was last seen on IRC in channel #asterisk, 25d 3h 45m 43s ago, saying: 'betaboi" true'. |
| 09:26.45 | masked | got other more important things to do ay |
| 09:26.45 | orlock | sorry for that everybody, but its almost 8:30pm friday night here |
| 09:26.47 | masked | like dinner |
| 09:26.53 | masked | bbl |
| 09:27.17 | orlock | and if you ever see a pair of auzzies do that in real life, get scared. |
| 09:27.49 | fourcheeze | I should point out that the original is "Oggie Oggie Oggie" |
| 09:28.44 | FuriousGeorge | hey all |
| 09:29.02 | fourcheeze | hey |
| 09:29.06 | *** join/#asterisk propagandhi (n=opera@d58-105-125-107.dsl.nsw.optusnet.com.au) |
| 09:29.10 | X-Rob | bugger |
| 09:29.13 | X-Rob | I missed the chant |
| 09:29.13 | orlock | fourcheeze: it is? |
| 09:29.36 | orlock | heh |
| 09:29.40 | X-Rob | (which is, I point out, a valid excuse for missing the aussie-aussie-aussie cry) |
| 09:29.46 | fourcheeze | orlock: http://www.ananova.com/news/story/sm_96221.html?nav_src=newsIndexHeadline |
| 09:30.26 | orlock | man, putty needs an "open in new browser on hotkey" function |
| 09:30.29 | X-Rob | ~seen a good movie |
| 09:30.32 | jbot | i haven't seen 'a good movie', X-Rob |
| 09:30.37 | fourcheeze | ho ho |
| 09:30.38 | X-Rob | perhaps you should, jbot. |
| 09:30.44 | fourcheeze | the old ones are the best |
| 09:31.39 | orlock | mad max |
| 09:31.40 | orlock | :) |
| 09:31.51 | orlock | its oldish |
| 09:32.04 | X-Rob | bloody good movie |
| 09:32.28 | orlock | yeah, xa/xb's are the only decent fords |
| 09:32.39 | X-Rob | the law, in australia, says that after watching Mad Max (I, the original) you must drink 1/2 a slab of beer (eg, 12 cans) and then go doing burnouts around your neighbourhood in a v8. |
| 09:32.56 | X-Rob | if you don't own a V8, you're outta the country. |
| 09:32.58 | orlock | my gf's lj torana has a pair of xa gt bonnet scoops on it |
| 09:33.16 | X-Rob | 'pair'? |
| 09:33.17 | orlock | hey, bathurst was won by 6 cyl's in two different decades! |
| 09:33.35 | X-Rob | twin turbo gtr's do _not_ count as 6 cyl vehicles |
| 09:33.38 | fourcheeze | X-Rob is there any connection plot-wise between mad max and mad max II ? |
| 09:33.45 | orlock | ahh yes they do! |
| 09:33.46 | propagandhi | does anybody have an idea how i can eliminate a 2 second pause when asterisk picks up the call from PSTN |
| 09:33.56 | orlock | fourcheeze: you mean are any other characters related? |
| 09:34.02 | X-Rob | fourcheeze, well. It's post apocolyptic.. And, uh, they've got cars. And wozzisface, the actor. |
| 09:34.03 | orlock | fourcheeze: there are a few possibilities |
| 09:34.06 | orlock | www.madmaxmovies.com |
| 09:34.06 | fourcheeze | I mean is there any continuity at all |
| 09:34.22 | orlock | oh, one and two.. not really |
| 09:34.28 | fourcheeze | except for whatsisface |
| 09:34.35 | orlock | airplane guy? |
| 09:34.38 | X-Rob | mel gibson, there ya go |
| 09:34.43 | X-Rob | I knew it would come to me |
| 09:34.48 | X-Rob | more booze, Ithink. |
| 09:35.13 | orlock | theres an actor that played a pilot in both 1 and 2, in pretty non-minor roles too |
| 09:35.13 | fourcheeze | there's no sense of what happens between the 2 |
| 09:35.16 | X-Rob | BTW; I was moved from 'directly underneath the access point' to 'other side of the hotel away from the access point' |
| 09:35.20 | X-Rob | so I'm sitting outside, freezing. |
| 09:35.27 | X-Rob | oooh yeah |
| 09:35.29 | X-Rob | wierd-face dude |
| 09:35.45 | orlock | fourcheeze: he has his car. |
| 09:35.48 | X-Rob | he was in a pile of movies in the 80's |
| 09:36.11 | X-Rob | I was getting booze. |
| 09:36.14 | fourcheeze | I have to say that I watched them in reverse order |
| 09:36.18 | fourcheeze | and it makes no sense that way either |
| 09:36.21 | fourcheeze | ;-) |
| 09:36.22 | orlock | fourcheeze: ahh |
| 09:36.34 | fourcheeze | but it's fairly easy to see the continuity between 2 and 3 |
| 09:36.50 | orlock | wel, the world isnt completely stufed in #1, nuke war happens between #1 and #2 |
| 09:36.53 | digime | anyone recommend a good incoming did provider |
| 09:37.05 | fourcheeze | digime: which country? |
| 09:37.09 | digime | usa |
| 09:37.24 | orlock | man |
| 09:37.35 | orlock | the poor cars that were hurt during the making of that move |
| 09:38.04 | austinnichols101 | digime: we're using didx.org and telasip |
| 09:39.23 | orlock | he's too pissed |
| 09:39.29 | orlock | he tripped over the cable i bet |
| 09:40.02 | Zipper_32 | ... isn't he on wireless? |
| 09:40.07 | orlock | ahh |
| 09:40.08 | orlock | yeah |
| 09:40.08 | Zipper_32 | He probably just fell over... |
| 09:40.11 | orlock | that would explain it :) |
| 09:40.15 | orlock | hahahahaha |
| 09:40.34 | Zipper_32 | Or closed the laptop lid because he didn't want somebody to see his alt-tabbed window of pr0n... |
| 09:40.38 | orlock | man |
| 09:40.39 | orlock | wireless |
| 09:40.46 | orlock | goddamn, its the bane of my life i swear |
| 09:40.53 | Zipper_32 | And he keeps forgetting that he has his powermanagement on hibernate or something when he closes the lid... |
| 09:41.01 | Zipper_32 | orlock: Why so? |
| 09:41.09 | orlock | wisp's |
| 09:41.25 | orlock | all of them i have dealt with have had issues |
| 09:41.26 | Zipper_32 | Fockers... |
| 09:41.36 | Zipper_32 | * & wisps? |
| 09:41.48 | orlock | heh, we wish |
| 09:41.53 | orlock | they aint good enough for citrix |
| 09:42.02 | orlock | quality voip, haha, joke. |
| 09:42.10 | Zipper_32 | It's a shame... |
| 09:42.14 | orlock | yeah |
| 09:42.27 | orlock | we currently use cisco voip phones |
| 09:42.33 | orlock | we are about to move to *+sip |
| 09:42.44 | orlock | put the sip image on a few cisco phones |
| 09:42.51 | orlock | works well, * is very cool |
| 09:43.00 | Zipper_32 | I sure like *; lots of features. =) |
| 09:43.03 | orlock | we are trialling some cheaper phones now |
| 09:43.09 | orlock | Grandstreams seem shit |
| 09:43.22 | Zipper_32 | Ahh, The cheapo's, =) |
| 09:43.37 | Zipper_32 | I have... 12 cheapos. |
| 09:43.38 | *** join/#asterisk X-Rob (n=rob@dsl-220-235-230-122.vic.westnet.com.au) |
| 09:43.39 | orlock | yeah, the protocol is nice, its easy to grasp if you have ever looked at any standard traffic dumps |
| 09:44.00 | orlock | we have a 3/4 full 48 port switch of cisco's |
| 09:44.11 | Zipper_32 | Mmmmm, yumm. |
| 09:44.17 | orlock | i've got a cisco at home too, work provides me with dsl |
| 09:44.26 | orlock | our dsl provider also does voip |
| 09:44.46 | orlock | and the company that owns our dsl provider is a major player in the pabx market |
| 09:44.47 | Zipper_32 | We have a bunch of these: http://azatel.com/ipcall104.htm, they're cheap, but rich in features. |
| 09:45.08 | orlock | so when we use them+their dsl.. its damn good quality |
| 09:45.15 | X-Rob | Now |
| 09:45.24 | X-Rob | I was boozing before my wireless was rudely walked inbetween |
| 09:45.27 | Zipper_32 | X-Rob: You trip over the cable? |
| 09:45.39 | X-Rob | Nah.. I've been moved away from the access point |
| 09:45.42 | orlock | 20ms to the sip gateway from my place |
| 09:45.53 | X-Rob | and some bastard parked a big 4wd between me and it |
| 09:45.53 | Zipper_32 | X-Rob: Ahh, there was suspicion that you managed to fall over and hurt yourself... |
| 09:46.04 | X-Rob | A valid thought |
| 09:46.12 | Zipper_32 | X-Rob: I sure hope you showed him who was boss and meanderd your way around his large vehicle. |
| 09:46.12 | orlock | X-Rob: bet it doesnt have dent or mud on it either, fuckers |
| 09:46.13 | X-Rob | but I'm wireless here |
| 09:46.26 | X-Rob | so nothing to trip over |
| 09:46.35 | orlock | dunk wires |
| 09:46.39 | Zipper_32 | X-Rob: We realize that, but we also realize that you're inebreated. |
| 09:46.42 | orlock | you trip over them, no matter where they are |
| 09:46.43 | orlock | :) |
| 09:46.55 | Zipper_32 | Anything is possible. |
| 09:47.03 | orlock | X-Rob: where in vic are you? |
| 09:47.10 | X-Rob | Leongatha |
| 09:47.16 | *** join/#asterisk Altair256 (n=icechat5@tn-greenback1a-12.rhmdky.adelphia.net) |
| 09:47.28 | orlock | wheres that? |
| 09:47.32 | orlock | n/s/e/w? |
| 09:47.37 | X-Rob | um |
| 09:47.52 | orlock | traffic is stuffed around here |
| 09:47.55 | orlock | cops everywhere |
| 09:47.56 | abusenode | stopspy |
| 09:47.57 | orlock | weird shit |
| 09:48.02 | Zipper_32 | Gah! What's with you AU's taking our City Names? Victoria and Richmond are close to me... in Canada. |
| 09:48.03 | X-Rob | near phillip island sorta |
| 09:48.06 | orlock | abusenode: +++ATH0 |
| 09:48.12 | orlock | ahh |
| 09:48.14 | fourcheeze | ahem |
| 09:48.16 | orlock | nice area |
| 09:48.30 | abusenode | more like DCC SEND "abusenode" amirite? |
| 09:48.37 | fourcheeze | Zipper_32: you think there was a Richmond in canada before London? |
| 09:48.54 | orlock | ahh, my bad, i was thinking about the keylogger thing |
| 09:48.57 | Zipper_32 | fourcheeze: Hmm, I guess I better keep quiet about New Westminster... |
| 09:49.02 | fourcheeze | hehe |
| 09:49.12 | fourcheeze | actually why don't you just cut off london and float it over there |
| 09:49.17 | orlock | Zipper_32: hey, you took our coders! |
| 09:49.28 | Zipper_32 | orlock: You took our girls with your cheesy accents!! |
| 09:49.33 | Zipper_32 | Wait |
| 09:49.34 | Zipper_32 | no |
| 09:49.37 | orlock | one of them was a girl! |
| 09:49.47 | orlock | she went to work at mitel |
| 09:49.48 | Zipper_32 | fourcheeze has the cheezey accent... how could I be mistaken. |
| 09:49.51 | *** join/#asterisk Altair256 (n=icechat5@tn-greenback1a-12.rhmdky.adelphia.net) |
| 09:49.54 | orlock | so did the others |
| 09:49.54 | Zipper_32 | orlock: What's her #? |
| 09:49.56 | fourcheeze | hehe |
| 09:49.59 | orlock | except for one gamer guy |
| 09:50.01 | orlock | all goths, too |
| 09:50.05 | Zipper_32 | Bah! |
| 09:50.08 | Zipper_32 | Take em back! |
| 09:50.24 | orlock | hey! |
| 09:50.28 | orlock | i resemble that remark! |
| 09:50.31 | fourcheeze | Zipper_32: is it easy to get a * job in .ca ? |
| 09:50.38 | Altair256 | Hello everyone |
| 09:50.49 | orlock | canada has lots of cool stuff.. and not just the ice |
| 09:51.06 | orlock | i think its cos its easy to get to from america, but its not america, and it sure as hell isnt mexico |
| 09:51.09 | Zipper_32 | fourcheeze: I have no idea... I'm in Post Secondary doing this as a side gig for a small biz. |
| 09:51.17 | orlock | which is where all the other people who want our of the us go :) |
| 09:51.17 | fourcheeze | ahh |
| 09:51.33 | fourcheeze | I did the immigration quiz on the canadian government site |
| 09:51.40 | fourcheeze | and apparently I can come over any time I want ;-) |
| 09:51.41 | orlock | is pot legal over there? |
| 09:51.49 | Zipper_32 | It's decriminalized in small doses. |
| 09:51.50 | orlock | on the simpsons they keep saying it is |
| 09:51.50 | austinnichols101 | anyone using freepbx? I wanted to get a read on the effort from outside the project itself (too many fans in there to get a balanced opinion) |
| 09:52.00 | orlock | how small is small? |
| 09:52.03 | X-Rob | austinnichols101, join #freepbx |
| 09:52.04 | fourcheeze | austinnichols101: telling me |
| 09:52.11 | Zipper_32 | It's not legal, but you won't be thrown in jail for having a joint on you, or having a plant at home. |
| 09:52.20 | orlock | so no record, but a fine |
| 09:52.23 | orlock | like a parking ticket |
| 09:52.34 | Zipper_32 | You can grow your own for personal use, but the police have the right to stomp on it if they please. |
| 09:52.38 | fourcheeze | Zipper_32: sounds sensible |
| 09:52.40 | orlock | ok |
| 09:52.42 | austinnichols101 | x-rob: yeah, no kidding. that wasn't what I was asking |
| 09:52.42 | Zipper_32 | orlock: Correct, |
| 09:52.44 | orlock | i see why they all moved now |
| 09:52.51 | Altair256 | Here's a quick question... with AAH2.6 will genzaptelconf autoconfig TE110P cards? |
| 09:52.51 | fourcheeze | Zipper_32: what's the worst thing about Canada? |
| 09:52.55 | X-Rob | oh |
| 09:52.56 | X-Rob | sorry |
| 09:52.58 | orlock | i've been there before |
| 09:53.01 | X-Rob | didn't read the second sentance 8) |
| 09:53.04 | orlock | in 1988/89 |
| 09:53.07 | austinnichols101 | np |
| 09:53.20 | Altair256 | a quick search of the AAH forums say they will not, but it's an older post. Not sure if anything has changed |
| 09:53.21 | fourcheeze | Zipper_32: apart from all the Brits coming over |
| 09:53.23 | orlock | i remember it being -26 the day we went to the airport |
| 09:53.27 | austinnichols101 | I already KNOW what they think of it :) |
| 09:53.32 | X-Rob | austinnichols101, it's just AMP tho, it's been in Asterisk@home for years |
| 09:53.32 | orlock | so we left behind the gear w had borrowed |
| 09:53.34 | Zipper_32 | fourcheeze: The worst thing in Vancouver is the rain if you're not used to it, |
| 09:53.38 | orlock | those wonderfull wonderfull snow pants |
| 09:53.42 | orlock | and the beanie |
| 09:53.44 | orlock | etc etc |
| 09:53.50 | orlock | i remember my nose freezing |
| 09:53.53 | Zipper_32 | It's not cold on the coast either, which is nice. |
| 09:53.59 | Zipper_32 | orlock: Where did you visit? |
| 09:54.03 | austinnichols101 | altair256: didn't for me on 2.5 |
| 09:54.08 | orlock | um |
| 09:54.13 | orlock | montreal maybe? |
| 09:54.18 | rpm | w |
| 09:54.20 | Altair256 | ah, so you have a TE110P card then? and set it up with AAH? |
| 09:54.28 | Zipper_32 | The coldest it gets here on the coast of BC is -4, -5 at sea level. |
| 09:54.32 | X-Rob | Altair256, genzaptelconf is not part of AAH |
| 09:54.35 | orlock | saw a high school ice hockey game |
| 09:54.40 | X-Rob | and I really can't see it being one |
| 09:54.44 | X-Rob | uh |
| 09:54.44 | X-Rob | AMP |
| 09:54.46 | Zipper_32 | The interior can get -40.... so it's kind of ugly there. |
| 09:54.49 | rpm | Zipper_32: it snowed pretty good there tonight eh? |
| 09:54.49 | X-Rob | <-- AMP/FreePBX Developer |
| 09:54.50 | fourcheeze | Zipper_32: I quite like rain |
| 09:54.50 | Altair256 | austinnichols101: we don't have a PRI, just an FXS channel bank |
| 09:54.58 | fourcheeze | Zipper_32: are you in vancouver? |
| 09:54.59 | Zipper_32 | rpm: Still snowing a bit right now... |
| 09:55.07 | orlock | X-Rob: in vic? |
| 09:55.08 | X-Rob | Altair256, feel free to try to convince me otherwise though. |
| 09:55.08 | Zipper_32 | fourcheeze: 35 minutes south of. |
| 09:55.13 | X-Rob | orlock, yea, and still cold |
| 09:55.28 | Altair256 | X-Rob: thanks, it's hard to know which parts are AAH custom and which parts are from something else |
| 09:55.35 | Zipper_32 | rpm: I am going to hate my commute tomorrow morning... People don't know how to drive in the snow here... |
| 09:55.45 | X-Rob | Altair256, it's pretty easy - the web stuff is AMP. Everything else (including /maint) is AAH |
| 09:55.57 | austinnichols101 | altair256: yes - works great with 2.5 |
| 09:56.06 | austinnichols101 | err with AAH |
| 09:56.10 | Altair256 | ah, nice ^^ |
| 09:56.16 | fourcheeze | Zipper_32: how much of the year do you get snow? |
| 09:56.37 | Altair256 | austinnichols101 - What guide did you follow to set it up? |
| 09:56.48 | Altair256 | austinnichols101 - the one on voip-info.org? |
| 09:56.58 | Zipper_32 | fourcheeze: A week in November, a week in January, yesterday and today so far. |
| 09:57.06 | fourcheeze | sounds about right |
| 09:57.17 | fourcheeze | I can cope with that much |
| 09:57.27 | austinnichols101 | I actually paid someone to help me set it up the first time just so I could cut down the learning curve. He wallked me through the whole process, common problems, etc. Money well spent IMO. |
| 09:57.27 | fourcheeze | hoping to take the family over for a holiday |
| 09:57.35 | fourcheeze | don't know if we'll get as far as Vancouver though |
| 09:57.54 | Altair256 | a local company to you, or one of the guys listed on the voip-info.org site? |
| 09:57.58 | fourcheeze | which is a shame since I always wanted to go there since watching "Beachcombers" as a child |
| 09:58.12 | austinnichols101 | basically was just install the card and add a couple of parameters - not bad at all |
| 09:58.17 | Altair256 | I'm only doing a "single setup" (tm) |
| 09:58.37 | Altair256 | lol, happen to remember what those couple of parameters were? ;) |
| 09:58.51 | austinnichols101 | I can connect up to the office |
| 09:59.09 | Altair256 | realistically, once I learn to set this up at my company, I imagine i'll want to reuse the skill elsewhere as well... lol |
| 09:59.11 | Zipper_32 | fourcheeze: You really shouldn't skip out on Vancouver, and if possible, you should take a trek up to Whistler/Blackcomb. |
| 09:59.19 | austinnichols101 | fyi: I ended up working with Tom Vile at http://www.baldwintechsolutions.com/details.php?item_number=10 |
| 09:59.29 | X-Rob | Hey, what happened to the mad max convo? |
| 09:59.31 | X-Rob | I was enjoying that. |
| 09:59.46 | Altair256 | thanks for the link, austinnichols101 |
| 09:59.53 | *** join/#asterisk Possible (n=Babbel@23.255-136-217.adsl-fix.skynet.be) |
| 10:00.00 | Zipper_32 | X-Rob: Sorry. So anyway, Mel Gibson has really lost it since Mad Max 2, don't you think? |
| 10:00.12 | Zipper_32 | Honestly, what is with The Passion? |
| 10:00.15 | X-Rob | Heh |
| 10:00.17 | Altair256 | do they charge by the hour? |
| 10:00.19 | X-Rob | I haven't seen that |
| 10:00.27 | orlock | X-Rob: 4x4 hj kingswood panelvan, gunmetal grey |
| 10:00.32 | orlock | 31" wheels, raised 12" |
| 10:00.38 | austinnichols101 | basically - you paypal them $50 per half hour and then they have a ticket system |
| 10:00.40 | Zipper_32 | Gah, I have midterms in 6 hours... Good day to you all. |
| 10:00.42 | X-Rob | THAT's more like it |
| 10:00.50 | orlock | mine :) |
| 10:00.51 | X-Rob | Gotta love the shaggin' wagons. |
| 10:00.55 | Altair256 | ah, $50/30min not bad at all |
| 10:01.06 | X-Rob | patrol chassis? |
| 10:01.09 | austinnichols101 | there are several other people out there providing similar service and I picked tom because he was the most responsive |
| 10:01.09 | orlock | been off the road for a few years now though |
| 10:01.10 | orlock | nope |
| 10:01.15 | Altair256 | what part of the country (er.. which country) are they from? |
| 10:01.22 | orlock | all holden, landcruiser front steering/suspension |
| 10:01.23 | austinnichols101 | northeast us |
| 10:01.26 | Altair256 | bleh, New York |
| 10:01.37 | Altair256 | they'll speak a different language than me...lol.. South East |
| 10:01.38 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:01.49 | orlock | only a 202/4 speed though |
| 10:01.58 | orlock | 4 speed kept breaking, its from a jackeroo |
| 10:02.06 | austinnichols101 | I try to learn everything I can here in #asterisk but it's also that someone 'has my back' in case of a bigger problem |
| 10:02.10 | X-Rob | Hmm, ok. The SWB MQ Patrols were pretty much bolt-on for the h* era |
| 10:02.19 | orlock | heard that? |
| 10:02.22 | orlock | swb? you sure? |
| 10:02.25 | Altair256 | austinnichols101 did you say you had a list of what you had to do for the TE110P card setup somewhere? |
| 10:02.27 | austinnichols101 | and if you been around here for a while and you're using AAH you better get ready for some abuse from time-to-time |
| 10:02.30 | austinnichols101 | yes |
| 10:02.40 | X-Rob | plus you've got the advantage of being able to use a nissan engine in it, including a R32 GTR engine |
| 10:02.48 | austinnichols101 | zaptel.conf need to add span, bchan and dchan lines |
| 10:02.51 | orlock | plthhh |
| 10:02.52 | orlock | chev |
| 10:02.53 | Altair256 | as in, hoity-toity roll your own types? |
| 10:02.54 | X-Rob | We stuck an EH on a SWB patrol |
| 10:03.01 | orlock | niiice |
| 10:03.12 | X-Rob | Not engineered, and unsafe as buggery, but jeez it was fun |
| 10:03.37 | Altair256 | alrighty... I think that's where I nearly was when I quit |
| 10:03.46 | Altair256 | one thing that's weird... you can tell me if you've ever seen it before... |
| 10:03.57 | austinnichols101 | you're on aah, right? |
| 10:04.02 | X-Rob | no sway bars and had 900mm of flex on the back before it lifted the front, and about 700 on the front before it lifted a back wheel |
| 10:04.02 | Altair256 | I installed AAH2.6 first, then added the card.. |
| 10:04.16 | Altair256 | all sound "originating |
| 10:04.22 | austinnichols101 | big thing you need to do is ssh up to the box and go explore /usr/src/zaptel |
| 10:04.25 | Altair256 | ..." from the server is silent |
| 10:04.28 | austinnichols101 | zttool = your friend |
| 10:04.31 | Altair256 | I can still make SIp to SIP calls |
| 10:04.36 | X-Rob | we used to take it up to lithgow and go mental through the bleu mountains |
| 10:05.07 | X-Rob | blink |
| 10:05.08 | Altair256 | k, writing those locations down |
| 10:05.11 | X-Rob | how did I turn french there? |
| 10:05.13 | X-Rob | blue |
| 10:05.25 | austinnichols101 | second thing you need to do is add signalling, switchtype, group, context and channel to zapata.conf |
| 10:05.51 | Altair256 | k |
| 10:05.52 | *** join/#asterisk Skid (i=chris@unaffiliated/skid) |
| 10:05.59 | austinnichols101 | altair256: where are you located? |
| 10:06.06 | Altair256 | Knoxville, Tennessee |
| 10:06.08 | Altair256 | US |
| 10:06.10 | X-Rob | austinnichols101, re your first comment, what do you see as 'wrong' with FreePBX? |
| 10:06.15 | austinnichols101 | no shit. I grew up in knoxville |
| 10:06.21 | austinnichols101 | I'm in Miami now |
| 10:06.24 | Altair256 | awesome ^^ |
| 10:06.36 | austinnichols101 | go vols |
| 10:06.44 | Altair256 | w00t ^^ |
| 10:06.59 | austinnichols101 | central high school |
| 10:07.11 | Altair256 | I actually grew up in Maryville, went to Maryville High School |
| 10:07.17 | Altair256 | few years at Pellissippi >.> |
| 10:07.20 | austinnichols101 | Mom works at Maryville college |
| 10:07.30 | Altair256 | now I actually work at New Horizons here in Knoxville |
| 10:07.40 | austinnichols101 | so you know Paul Carney... |
| 10:07.43 | Altair256 | half an instructor, half network adminsitrator |
| 10:07.50 | Altair256 | yeah, I recognize the name |
| 10:07.55 | dpryo | Small world! |
| 10:08.00 | Altair256 | very... lol |
| 10:08.18 | austinnichols101 | did you fire up zttool yet? |
| 10:08.35 | Altair256 | not yet... lemme try to remotely connect in |
| 10:08.44 | Altair256 | can't remember if I shut the machine down before I left work >.> |
| 10:09.20 | Altair256 | wow |
| 10:09.28 | Altair256 | I can already tell zttool is going to be my new friend |
| 10:09.36 | austinnichols101 | http://www.napoftheamericas.net |
| 10:10.01 | austinnichols101 | check around - there are a few other goodies in that directory |
| 10:10.14 | austinnichols101 | zttest, ztspeed |
| 10:10.59 | Altair256 | can't wait to get back to work now and pull the T and plug it back into my box.. lol |
| 10:11.17 | austinnichols101 | in my case signaling is pri_cpe and switchtype = national |
| 10:11.18 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:11.29 | Altair256 | if I don't have a PRI, do I have to have dchan entry in zaptel.conf? |
| 10:11.30 | austinnichols101 | and asterisk doesn't really care if it's ni1 or ni2 |
| 10:11.46 | austinnichols101 | are you going to be talking t1 or pri? |
| 10:12.03 | Altair256 | t1 for now, since our old system currently uses FXS channel bank loopstart |
| 10:12.12 | Altair256 | was assuming I'd set it to FXS_KS |
| 10:12.19 | Altair256 | channels 1-13 for voice |
| 10:12.23 | austinnichols101 | I *think* you're right |
| 10:12.28 | austinnichols101 | something like |
| 10:12.35 | austinnichols101 | span=1,1,0,esf,b8zs |
| 10:12.41 | trnygaar | Any clue why timing of MOH is ok when i use the incoming trunks, but when i call internal it is really really slow? |
| 10:12.46 | austinnichols101 | bchan=1-13 |
| 10:12.46 | Altair256 | yeah, kewlstart is supposedly just loopstart with battery drop detection |
| 10:12.53 | trnygaar | incoming trunks are also sip |
| 10:12.53 | Altair256 | but I really have no clue what I'm talking about... lmao |
| 10:12.54 | austinnichols101 | #dchan= |
| 10:13.02 | X-Rob | nnight all |
| 10:13.16 | Altair256 | wha ttype of hardware, trnygaar? |
| 10:13.32 | Altair256 | alright, austinnichols101, I'm writing this down now... |
| 10:14.05 | Altair256 | what do 1,1,0,esf,b8zs do? |
| 10:14.09 | austinnichols101 | altair256: I'm doing most of this from memory as I have my notes at the office |
| 10:14.18 | propagandhi | can anyone give me an idea on how to eliminate a 2 second delay when asterisk picks up the call from PSTN |
| 10:14.30 | Altair256 | since I'm on a T1 instead of a PRI was wondering if it might be different |
| 10:14.34 | propagandhi | it doesnt do that when it comes in on ISDN |
| 10:15.05 | austinnichols101 | I know 1 is the span and I'm not sure on parms 2/3. You'll definitley need to set 4 and 5 to match the settings on the other side of the T |
| 10:15.17 | Altair256 | do you have SpanDSP (fax) turned on, propagandhi? |
| 10:15.22 | austinnichols101 | is the t coming from a carrier or from another switch? |
| 10:15.26 | propagandhi | Altair256: that I need to check |
| 10:15.39 | Fedoracore6 | some budy famliar with database mysql --> palse help me |
| 10:15.51 | Altair256 | not sure austinnichols101, having our local carrier come out tomorrow or Monday to "point to things and tell me what they are" |
| 10:16.01 | Altair256 | what's your question Fedoracore6? |
| 10:16.22 | austinnichols101 | k - piece of advice on that. You need to ask them "how the circuit is configured" |
| 10:16.32 | Altair256 | the T1 is coming from a carrier and they've confirmed them as FXS channel bank 1-13 voice loopstart |
| 10:16.51 | austinnichols101 | avoid questions like "are you using esf" and make them tell YOU what they're running |
| 10:16.55 | Altair256 | k, I can do that |
| 10:17.15 | austinnichols101 | I've seen techs that do actually do the lookup and go from memory and then you waste a bunch of time chasing down stuff that doesn't work |
| 10:17.36 | austinnichols101 | high percentage chance that the span settings I gave you will work |
| 10:17.45 | Altair256 | alright |
| 10:17.46 | *** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F28C2.dip0.t-ipconnect.de) |
| 10:17.53 | austinnichols101 | you should also be able to call the telco or look at your existing switch and find the settings too |
| 10:17.55 | ambriento | altair256, 1,1,0,esf,b8zs do: span 1, 1st source of sync, LBO, framing, coding |
| 10:18.12 | Altair256 | I know NOTHING about our existing system except how to move extensions |
| 10:18.20 | austinnichols101 | ambriento: tks |
| 10:18.32 | austinnichols101 | what's the system? |
| 10:19.00 | [ProB]CrazyMan | anybody tested asterisk 1.2.5 with bristuff-0.0.2 ? does that work togheter ? |
| 10:20.25 | orlock | oh my god |
| 10:20.29 | orlock | my mother just called me |
| 10:20.31 | orlock | i was pissed |
| 10:20.41 | orlock | i wasnt slurring, but i was talking for 15 minutes |
| 10:20.53 | austinnichols101 | orlock: nice! |
| 10:21.18 | orlock | yeah, i guess now she at least knows my job is goin ok, and we still own the same old car |
| 10:21.46 | austinnichols101 | took me a second to realize that you were drunk and not angry |
| 10:22.00 | orlock | hahaha |
| 10:22.13 | orlock | pissed/pissed off |
| 10:22.29 | austinnichols101 | yeah - quit calling me mom, I'm getting pissed |
| 10:23.27 | *** join/#asterisk Altair256 (n=icechat5@tn-greenback1a-12.rhmdky.adelphia.net) |
| 10:23.31 | Altair256 | grrr |
| 10:23.42 | Altair256 | that's the 3rd time I've been disconnected from clarke.freenode.net |
| 10:23.50 | Altair256 | I need to change servers |
| 10:24.02 | Altair256 | anyhow... last thing I saw was when I said "thanks ambriento" |
| 10:25.12 | austinnichols101 | you didn't miss that mutch |
| 10:25.21 | austinnichols101 | what switch do you have now? |
| 10:25.37 | Altair256 | switch? |
| 10:26.28 | ambriento | austinnichols101, do you mean, swithtype in zapata-channels.conf? |
| 10:26.36 | ambriento | oops, switch* |
| 10:26.42 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 10:27.06 | Altair256 | lemme look and see what I have in there now |
| 10:27.10 | Fedoracore6 | Altair256: i wanna do the some system registration use asterisk |
| 10:27.35 | Fedoracore6 | i already have database and ..done with conection but |
| 10:27.36 | austinnichols101 | there's your paste |
| 10:28.04 | ambriento | welll I'm on my way to work |
| 10:28.05 | ambriento | later guys |
| 10:28.31 | Altair256 | cya ambriento |
| 10:28.43 | Altair256 | thanks for paste, austinnichols101 |
| 10:29.04 | Fedoracore6 | when i do the student key in password the asterisk cannor read from my databases |
| 10:29.14 | *** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 10:29.14 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
| 10:30.03 | rpm | has anyone got iaxmodem working? |
| 10:30.43 | propagandhi | Altair256: what effect does the immediate parameter in zapata.conf have |
| 10:31.06 | Fedoracore6 | Altair256 |
| 10:31.18 | Fedoracore6 | http://pastebin.com/594148 |
| 10:32.05 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:32.30 | Enth | Guys, any idea why sip show peers gives me the following users , when they are not even connected? : 4545/4545 (Unspecified) D N 0 Unmonitored |
| 10:32.38 | Enth | 6666/6666 (Unspecified) D N 0 Unmonitored |
| 10:32.41 | Enth | :/ |
| 10:35.40 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 10:36.18 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 10:36.50 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 10:37.49 | puzzled | morning |
| 10:38.14 | Skid | Enth: their probably configured in your sip.conf file |
| 10:38.39 | Enth | 4545 isnt. |
| 10:38.46 | Skid | do a search for it |
| 10:38.52 | Skid | then, sip reload |
| 10:38.57 | Enth | it isnt there |
| 10:38.59 | Enth | I know :) |
| 10:39.08 | Skid | their still showing, after sip reload? |
| 10:39.13 | Skid | they're even |
| 10:43.18 | Enth | yes |
| 10:43.21 | Enth | :/ |
| 10:44.13 | Skid | are any files being included into sip.conf? |
| 10:46.26 | Enth | fn~Skid: yeah |
| 10:46.39 | Skid | :) psybnc? ;) |
| 10:47.04 | Skid | check to see if theres any of those 4545/6666 entries in there? sounds daft, but i dont see why else it'd be in there |
| 10:47.24 | Enth | ahh yes |
| 10:47.26 | Enth | ur right. |
| 10:47.29 | Skid | :) |
| 10:47.47 | Enth | but |
| 10:47.54 | Enth | only 4545 is there. |
| 10:48.04 | Enth | still doesnt solve why 6666 is still present. |
| 10:48.10 | Skid | any files being included in that one? :P |
| 10:48.14 | Skid | or contexts heh |
| 10:48.30 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:49.01 | Enth | let me see. |
| 10:49.53 | Enth | hrmmm |
| 10:49.57 | Enth | well it cleared 4545 |
| 10:51.19 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:51.23 | backblue | hi* |
| 10:51.39 | backblue | does anyone know, any comand to force iax2 trunks to come up? |
| 10:52.13 | Enth | but still shows 6666 |
| 10:53.00 | propagandhi | blackblue: iax2 reload |
| 10:56.20 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 10:56.20 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.5 released! (March 3, 2006) -=- Asterisk-addons 1.2.2 now available (March 6, 2006) |
| 10:56.25 | backblue | my trunks it's down now, and the machines have connectivity, i'm trying to understand why... |
| 10:57.35 | Enth | RoyK any idea why certain peers appear to be connected when they are not (sip show peers) |
| 10:57.46 | Enth | I've checked any #include sections and there are non there. |
| 10:58.15 | RoyK | no idea |
| 10:58.31 | Skid | hmm |
| 10:59.01 | Skid | for "open hours", is it better to use GotoIfTime, or just include => context|times|here|etc ? |
| 11:00.07 | RoyK | doesn't matter :) |
| 11:00.13 | RoyK | we use gotoiftime |
| 11:00.42 | backblue | iax2 show stats -> how this work? |
| 11:01.39 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 11:15.44 | Skid | what codec does playback use? |
| 11:15.53 | Skid | im gettin errors when only allowing ulaw and g729 |
| 11:16.10 | FuriousGeorge | licence for g729? |
| 11:16.34 | Skid | yeah |
| 11:16.52 | Skid | according to techie, he sorted it not me |
| 11:16.52 | FuriousGeorge | seems like ur not transcoding to me |
| 11:17.34 | FuriousGeorge | ~g729 |
| 11:17.36 | jbot | methinks g729 is It was in November 1995 that the G.729 standard, also referred to as CS-ACELP was adopted by the ITU, a United Nations organization. Similar, quality-wise, to 32 kbps ADPCM, G.729 offers toll quality speech. Furthermore, being only an 8 kbps codec, G.729 offers opportunities for significant increases in bandwidth utilization to existing telephony ... |
| 11:18.18 | FuriousGeorge | maybe you are in pass-through only mode |
| 11:18.23 | Skid | ah i see |
| 11:18.30 | Skid | i misspelt the sound file |
| 11:18.30 | FuriousGeorge | for lack of a properly installed licence |
| 11:18.31 | Skid | doh! |
| 11:18.33 | FuriousGeorge | lol |
| 11:18.39 | Skid | :) |
| 11:19.04 | Skid | 8Kbps, humph, i was seeing 60Kbps for two conversations :) |
| 11:19.44 | Skid | right, can't use bloody gotoiftime as im using macro's to route incoming calls |
| 11:19.45 | Skid | grr |
| 11:20.08 | FuriousGeorge | macroif? |
| 11:21.19 | Skid | or |
| 11:21.22 | Skid | i can do it an easier way |
| 11:21.27 | Skid | man |
| 11:21.27 | FuriousGeorge | ? |
| 11:21.29 | Skid | i need some spleep |
| 11:21.31 | Skid | sleep |
| 11:21.32 | Skid | argh!! |
| 11:21.40 | Skid | inside the macro, i can just use gotoiftime |
| 11:22.43 | FuriousGeorge | thats a nice solution. i wanna set mine up to pass the dialtimeout, and use only one macro for all inbound calls |
| 11:23.06 | Skid | yep, that's what im doing once I get my head around macros a bit more |
| 11:23.20 | Skid | it does it fine at the moment, but its not perfect |
| 11:23.21 | Skid | and it bugs me |
| 11:23.23 | Skid | :P |
| 11:24.46 | FuriousGeorge | mine works fine, the only thing that bugs me is that i know i could call one macro of just a few lines, based on the time, and make it shorter |
| 11:25.07 | FuriousGeorge | rather than have three different context for open closed and holiday |
| 11:25.46 | *** join/#asterisk sl16 (n=blah@tv.neterra.net) |
| 11:26.22 | Skid | i see |
| 11:26.35 | Skid | hmm, this means im going to have to crate another context |
| 11:26.43 | Skid | best have another think |
| 11:31.17 | backblue | i need a solution for the fromdomain on incoming calls, it uses allways the fromdomain from sip.conf |
| 11:32.20 | *** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F3923.dip0.t-ipconnect.de) |
| 11:34.31 | *** join/#asterisk themacuser (n=gm@ppp211-125.lns1.adl2.internode.on.net) |
| 11:34.48 | *** join/#asterisk robby (n=robby@host23-229.pool8252.interbusiness.it) |
| 11:36.55 | *** join/#asterisk starlein (i=star@fo0bar.de) |
| 11:37.46 | austinnichols101 | anyone familiar with the amp->freepbx migration? I've been watching over on #freepbx for a while but wanted to get an outside opinion of the project |
| 11:38.07 | austinnichols101 | (from the vim #asterisk crowd) |
| 11:41.19 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 11:43.17 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 11:43.44 | Skid | hmm, is there anyway to increase the timeout, sipgate's shitty service keeps timing out, and during the re-register phase I'm unable to call |
| 11:43.49 | Skid | or can it send "keep-alives" |
| 11:43.52 | Skid | ? |
| 11:45.02 | austinnichols101 | timeout of what |
| 11:45.23 | Skid | sipgate hae some timeout it looks like, if no traffic within x |
| 11:45.24 | Skid | then disconnect |
| 11:47.03 | austinnichols101 | what are you connecting with? |
| 11:47.31 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 11:47.38 | Skid | asterisk / sip ? |
| 11:47.40 | austinnichols101 | k |
| 11:47.41 | kardecallan | hi |
| 11:47.53 | Skid | no other way to connect to sipgate:) |
| 11:48.08 | austinnichols101 | I don't use sipgate so I wasn't sure |
| 11:48.17 | austinnichols101 | you using qualify=yes or qualify=nnnn? |
| 11:48.25 | Skid | using yes |
| 11:48.32 | Skid | just calling hteir moron support now |
| 11:48.42 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 11:48.47 | kardecallan | hi |
| 11:48.48 | austinnichols101 | qualify=yes should be pinging them for sip options every two seconds |
| 11:48.53 | *** part/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 11:48.54 | fourcheeze | Skid: try using something like qualify=10000 |
| 11:49.15 | fourcheeze | austinnichols101: does it say that sipgate is unreachable? |
| 11:49.35 | austinnichols101 | didn't catch that part. trying to help but not really familar with sipgate |
| 11:49.51 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 11:50.01 | kardecallan | Good Morning |
| 11:50.04 | *** part/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 11:50.16 | austinnichols101 | he was looking for a way to implement a keep alive because he can't make calls during the re-register process |
| 11:50.29 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 11:50.44 | Skid | I presume it's their end, if no data exchange or invites go across |
| 11:50.48 | sambal | strange, i see audio playing on my asterisk system but the line is quiet, anyone has a idea? there is nothing wrong with the audio files |
| 11:51.19 | austinnichols101 | you could probably turn on sip debug through the cli and see what's happening with the sip options messages |
| 11:51.41 | austinnichols101 | you should see the message go out and then an immediate response with no retries |
| 11:51.53 | kardecallan | Please!! You can help me? |
| 11:52.13 | austinnichols101 | kardecallan: just ask your question |
| 11:52.17 | kardecallan | I'm brazilian programer |
| 11:52.24 | austinnichols101 | save the begging for later :) |
| 11:53.16 | kardecallan | I have a little problem whith english |
| 11:53.42 | kardecallan | but, let me try explan my problem |
| 11:54.24 | kardecallan | I have asterisk server behind of firewall |
| 11:55.20 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 11:56.17 | *** join/#asterisk stoffell (n=stoffell@d5153FC2C.access.telenet.be) |
| 11:58.17 | kardecallan | Babel Fish Translation Help |
| 11:58.17 | kardecallan | In English: |
| 11:58.17 | kardecallan | e when a external customer sip makes a call for an internal customer sip, does not have audio. |
| 11:59.27 | austinnichols101 | kardecallan: describe the setup (ie 7960 -> linksys (NAT) -> firewall (NAT) -> asterisk) |
| 12:00.42 | kardecallan | for these external users I placed in extern.conf the option of nat=route. |
| 12:01.18 | austinnichols101 | have you defined localnet and externip? |
| 12:01.24 | kardecallan | no |
| 12:01.53 | austinnichols101 | normally localnet will be something like 192.168.1.0 and externip = your public ip |
| 12:02.12 | kardecallan | I searched and read something on STUN. |
| 12:02.50 | *** join/#asterisk coppice (n=chatzill@239.192.17.210.dyn.pacific.net.hk) |
| 12:03.15 | austinnichols101 | stun is to help a remote device (like a phone) figure out what it's public ip address is |
| 12:03.52 | austinnichols101 | phone asks stun server 'what is my address' and stun server replies with the public address |
| 12:04.26 | abusenode | stop using nat: problem solved. |
| 12:04.42 | austinnichols101 | abusenode: yeah! |
| 12:06.32 | austinnichols101 | kardecallan: voice packets are not reaching the remote end because they're being routed to the wrong destination (because of nat) |
| 12:06.59 | kardecallan | when no need to use the STUN? Is this? |
| 12:07.18 | austinnichols101 | not in this case |
| 12:07.24 | coppice | this guy Nat must be employed by the telcos. he does a fantastic job of stopping VoIP from working well :-) |
| 12:07.40 | abusenode | i think the jews are involved |
| 12:07.42 | abusenode | one way or another. |
| 12:08.17 | coppice | they design all the really neat silicon, so most probably that is true |
| 12:08.20 | austinnichols101 | I felt really comfortable with nat until I started dealing with voip. nat + tcp = cake. nat + udp is just plain weird at times |
| 12:08.43 | [ProB]CrazyMan | coppice: I have an problem with useing txfax, when i try to send an fax to an AVM fritcard it fails. to an analog fax it works |
| 12:09.11 | abusenode | huhuuhuhhu |
| 12:10.37 | austinnichols101 | I'm tired of touching the router |
| 12:10.42 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:13.25 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 12:17.01 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 12:17.46 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 12:18.43 | octothorpe | np |
| 12:18.47 | Possible | bootx ? |
| 12:19.21 | Possible | I guess not |
| 12:20.10 | *** join/#asterisk Assid (n=assid@203.115.64.13) |
| 12:20.36 | kardecallan | austinnichols101 to make one test now, but doesn't have audio. |
| 12:22.47 | shiznatix | Hello, can anyone even read this? |
| 12:23.02 | kippi | shiznatix: yes |
| 12:23.34 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 12:23.42 | Winkie | freenode :( |
| 12:24.36 | kardecallan | I actived the localnet and externip, but doesn't have audio |
| 12:26.05 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 12:29.45 | *** join/#asterisk whirm (n=whirm@80.174.186.169) |
| 12:29.49 | whirm | hi |
| 12:30.09 | Winkie | sup? |
| 12:30.55 | whirm | is this the correct place for asking for help with compiling zaptel drivers? |
| 12:32.52 | *** join/#asterisk coppice (n=chatzill@73.199.17.210.dyn.pacific.net.hk) |
| 12:33.31 | coppice | [ProB]CrazyMan: try the version at http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps . It changes the buffering to make the timing more tolerant. |
| 12:34.39 | [ProB]CrazyMan | coppice: thx, i will try. other question, how do I know if the transmission was successfully ? |
| 12:36.31 | *** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au) |
| 12:36.39 | *** join/#asterisk fgomes (n=fgomes@201-13-79-92.dsl.telesp.net.br) |
| 12:37.01 | fgomes | hey guys |
| 12:37.22 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 12:38.41 | fgomes | I have one X100P card and when I receive a call from FWD I redirect it to ZAP/1. The problem is ZAP/1 immediately answer the call (that is still being connected by public telephony company) and then... i miss some seconds of FWD message. |
| 12:39.18 | Winkie | whirm: it's pretty dead in here so what's up? :) |
| 12:39.35 | fgomes | I have already immediate=no in zapata.conf . Any ideas? |
| 12:40.02 | Winkie | fgomes: that's an interesting problem |
| 12:40.09 | Winkie | how do you mean it immediately answers, what's on the end of ZAP/1? |
| 12:40.58 | fgomes | winkie: X100P is a fxo interface. When I receive calls from FWD I redirect to my mobile phone by using ZAP/1. |
| 12:41.15 | austinnichols101 | fgomes: put a delay step before answer and test |
| 12:41.55 | whirm | Winkie: http://pastebin.com/594296 <<-- I'm getting that trying to build the zaptel modules |
| 12:42.21 | Winkie | i think austinnichols101 is correct on this, why can't you just wait(4)? |
| 12:42.23 | whirm | i'm on a 32bit chroot on debian amd64 |
| 12:42.24 | Winkie | whirm: checking |
| 12:42.35 | austinnichols101 | kardecallan: http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html |
| 12:42.54 | whirm | i'm building the modules for a soekris embedded pbx |
| 12:43.05 | fgomes | whirm, austinick...: thanks... I will try. Testing now.... |
| 12:43.18 | Winkie | whirm: it looks like you don't have a 32 bit libm in your chroot? |
| 12:43.27 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:44.22 | coppice | you also have -m64 in your command line |
| 12:44.37 | whirm | Winkie: /lib/libm-2.3.6.so: ELF 32-bit LSB shared object, Intel 80386, version 1 (SYSV), stripped |
| 12:44.48 | Winkie | whirm: what's your chroot? |
| 12:45.02 | whirm | Winkie: debian SID |
| 12:45.19 | *** join/#asterisk danzig (n=chatzill@ruc-kj-013.ruc.dk) |
| 12:45.36 | Winkie | whirm: no i mean is this the /lib inside your chroot? |
| 12:45.59 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 12:46.02 | whirm | yup |
| 12:46.23 | Winkie | well it doesn't seem to be searching /lib anyway |
| 12:46.27 | Winkie | what's the ldconfig like in the chroot |
| 12:47.18 | whirm | http://pastebin.com/594299 |
| 12:47.29 | Winkie | the -m64 is a valid bit too |
| 12:47.40 | Winkie | by that i mean whyis it 64 |
| 12:47.42 | Winkie | not 32 |
| 12:47.44 | coppice | its looking for a 64 bit library, because you have -m64 on the command line |
| 12:47.57 | whirm | where the hell is getting the -m64 ?! it doesn't even appear in the makefile... :-? |
| 12:48.06 | Winkie | also looks like the ldconfig is a bit spares |
| 12:48.08 | Winkie | sparse* |
| 12:48.25 | danzig | EHLO * gurus :-) |
| 12:49.13 | austinnichols101 | danzig: nice nick |
| 12:49.24 | danzig | tnx |
| 12:49.53 | danzig | Does anyone know if there is an easy way to get an up-to-date Asterisk on debian, using apt-get, without upgrading the whole distro from stable to testing? |
| 12:50.24 | twisla | backporting it yourself |
| 12:51.32 | danzig | thats not easy ;-) I wouldn't mind backporting it once, but the whole point (for me) of using apt-get is that I get any important/security fixes without having to do a lot of thinking... |
| 12:53.15 | Winkie | i imagine you could screw with the .debs enough to get it to work |
| 12:53.26 | Winkie | but the point of debian is that it's slow to update if you're on stable |
| 12:53.42 | *** join/#asterisk Bambr (n=Bambr@213-35-233-22-dsl.end.estpak.ee) |
| 12:55.30 | Eitch | hau |
| 12:56.51 | whirm | wow! found it! DEB_HOST_GNU_TYPE=386 |
| 12:56.54 | whirm | thanks to all! |
| 12:57.58 | abusenode | uh what |
| 12:58.00 | abusenode | wtf is that shit? |
| 12:59.15 | [ProB]CrazyMan | coppice: rx/txfax does not compile (I'm using asterisk 1.0.10) |
| 12:59.55 | coppice | correct. that software will not compile with 1.0.10 |
| 13:01.22 | [ProB]CrazyMan | can't upgrade ti 1.2.5 because I do not know if bristuff ist working with that |
| 13:01.26 | *** join/#asterisk Seb7 (n=sebast@host217-34-0-168.in-addr.btopenworld.com) |
| 13:01.57 | coppice | OK, let me see if I can cook up a 1.0.x compatible version |
| 13:02.07 | stoffell | [ProB]CrazyMan, but you could upgrade to 1.2.4 (bristuff 0.3.0pre1k is pretty stable) |
| 13:02.18 | [ProB]CrazyMan | is it stable ? |
| 13:02.30 | [ProB]CrazyMan | so only dies once a day ? |
| 13:02.39 | stoffell | [ProB]CrazyMan, if you use k, not l ! (no, it doesn't die on me :) ) |
| 13:02.46 | stoffell | only if i want it to :D |
| 13:02.47 | fgomes | Winkie: Wait(10) doesnt work... It waits 10 secs before Dial(). ZAP/1 answers immediately instead of waiting for signaling from PSTN. |
| 13:02.48 | fgomes | <PROTECTED> |
| 13:02.48 | fgomes | <PROTECTED> |
| 13:02.48 | fgomes | <PROTECTED> |
| 13:02.48 | fgomes | <PROTECTED> |
| 13:03.24 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 13:03.43 | [ProB]CrazyMan | with 1.2.4 i have to change a lot in dialplan, there where some changes ... I see |
| 13:04.37 | stoffell | [ProB]CrazyMan, yeah, testing is prolly needed... but even if you'd use visdn, it's also based more and more on 1.2.X then 1.0.X |
| 13:05.30 | [ProB]CrazyMan | I realy want to update to 1.2.4 because I need some new functions ... like dynamical check external ip .. and so on |
| 13:05.36 | *** part/#asterisk whirm (n=whirm@80.174.186.169) |
| 13:05.49 | stoffell | [ProB]CrazyMan, time to get a test-machine out of the closet then! :) |
| 13:06.00 | Skid | hmph, can't connect to fwd |
| 13:06.10 | [ProB]CrazyMan | seems so |
| 13:06.22 | fgomes | Skid: how? IAX2? |
| 13:07.09 | austinnichols101 | connected iax2 to fwd from here |
| 13:07.24 | Skid | yeah |
| 13:07.28 | Skid | keep getting refused |
| 13:07.50 | Skid | do i have to email them? |
| 13:07.55 | Skid | i just read a post about emailing some "ed" |
| 13:10.26 | Skid | austinnichols101: could you pastbin your related configs minus the pass, etc please, so i can compare |
| 13:10.32 | Skid | i may be overlooking someething |
| 13:10.39 | Skid | NOTICE[11186]: chan_iax2.c:7410 socket_read: Registration of '754733' rejected: 'Registration Refused' from: '192.246.69.186' |
| 13:10.43 | Skid | is what im getting |
| 13:10.51 | austinnichols101 | sure |
| 13:11.05 | Skid | thanks |
| 13:11.11 | Skid | brb, just gonan get a sarnie |
| 13:12.11 | abusenode | fwd iax2 sucks |
| 13:12.15 | abusenode | well, it sucked a year ago |
| 13:12.20 | abusenode | probably still does |
| 13:12.22 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 13:12.53 | x86 | morning |
| 13:13.12 | Skid | yeah, its for my sister |
| 13:13.14 | Skid | she has it |
| 13:13.18 | Skid | and wants to be able to call me |
| 13:13.25 | Skid | and im sure as hell not using fwd ;P |
| 13:14.44 | austinnichols101 | http://pastebin.com/594328 |
| 13:15.11 | austinnichols101 | abusenode: still sucks |
| 13:15.17 | austinnichols101 | but it's connected to everything |
| 13:15.33 | x86 | FWD still sucks |
| 13:15.35 | fgomes | fwd iax2 is quite easy! I will copy/paste my config here... pls hold. |
| 13:15.48 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.adnc.com) |
| 13:16.17 | x86 | about 4 times a day i randomly look over at my asterisk server's monitor and see PEER FWD IS UNREACHABLE (1037ms) |
| 13:16.49 | x86 | fgomes: DO NOT PASTE HERE |
| 13:16.50 | FlyboySR22 | Good Morning Everyone |
| 13:16.56 | austinnichols101 | I normally see around 70ms turnaround, but I'm on a great connection too |
| 13:17.10 | Skid | thans |
| 13:17.11 | fgomes | register => 610743:password@iax.fwdnet.net |
| 13:17.11 | fgomes | [fwd-peer] |
| 13:17.11 | fgomes | type=peer |
| 13:17.11 | fgomes | auth=md5 |
| 13:17.11 | fgomes | username=610743 |
| 13:17.12 | fgomes | secret=password |
| 13:17.14 | fgomes | qualify=yes |
| 13:17.16 | fgomes | host=iax2.fwdnet.net |
| 13:17.18 | fgomes | disallow=all |
| 13:17.21 | x86 | GOD DAMMIT NO |
| 13:17.22 | fgomes | allow=ulaw |
| 13:17.24 | fgomes | callerid=Fernando Gomes<610743> |
| 13:17.30 | x86 | USE A FUCKING PASTEBIN YOU TIRD! |
| 13:17.34 | austinnichols101 | someone smack fgomes |
| 13:17.56 | sambal | playback command and background are not working, they play the files but there is no sound on the line, does anyone has a idea? |
| 13:17.59 | fgomes | [iaxfwd] |
| 13:17.59 | fgomes | type=user |
| 13:17.59 | fgomes | context=inbound-fwd |
| 13:17.59 | fgomes | auth=rsa |
| 13:17.59 | fgomes | inkeys=freeworlddialup |
| 13:18.00 | fgomes | disallow=all |
| 13:18.02 | fgomes | allow=ulaw |
| 13:18.04 | x86 | jesus christ |
| 13:18.06 | abusenode | pastebin sucks. |
| 13:18.08 | abusenode | use rafb.net/paste |
| 13:18.14 | x86 | where re the fucking bots when you need them |
| 13:18.15 | abusenode | pastebin is google ad filles commie garbage. |
| 13:18.16 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:18.20 | x86 | s/bots/ops ;) |
| 13:18.22 | abusenode | filles? filled. |
| 13:18.28 | x86 | [TK]D-Fender: anything yet man? :) |
| 13:18.36 | Winkie | oh hey x86 |
| 13:18.42 | x86 | abusenode: rafb is still a pastebin ;) |
| 13:18.43 | Winkie | i'm sorry i never got round to doing your voice prompts |
| 13:18.48 | Winkie | if you send me the stuff again i'll get on it tonight |
| 13:18.54 | x86 | Winkie: i got a canadian guy to do them hehe |
| 13:19.09 | Winkie | x86: haha, oh i've been beaten by the colonies :( |
| 13:19.19 | abusenode | uh |
| 13:19.22 | abusenode | no it fuckin isnt |
| 13:19.24 | x86 | canadia is a colony? |
| 13:19.26 | Winkie | i'm sorry it's just with everything going on i totally forgot about it till i was doing a mass /wind cl and found yours and hit myself :( |
| 13:19.32 | Winkie | it's a colony to the brits 8) |
| 13:19.50 | sambal | anyone a idea why playback / background plays no sound? does it use a external application to play the audio files? |
| 13:19.52 | abusenode | the whole "pastebin" shit is started by those faggot sites that auto-insert <? ?> between your crap so that it looks like noop php shit. |
| 13:19.55 | x86 | oh shit... i'm thinking some people are about to flail you now ;) |
| 13:20.00 | Enth | Guys, any idea why a hardware based IP Phone cannot hear the caller? It registers, rings fine but when answeredm both parties cannot hear eachother. Works fine for x-lite to x-lite. |
| 13:20.04 | abusenode | Winkie: it needs to have the sound to play. |
| 13:20.11 | abusenode | Enth: nat. |
| 13:20.19 | [TK]D-Fender | x86 : Geez... I'm not even caffeinated yet! |
| 13:20.26 | Winkie | abusenode: i'm sorry what? |
| 13:20.27 | x86 | [TK]D-Fender: haha |
| 13:20.33 | Enth | abusenode: What settings exactly? |
| 13:20.53 | Winkie | Enth: is it behind NAT? |
| 13:20.56 | Enth | yes |
| 13:21.15 | Enth | I've port forwared rtp 10000 to 20000 to that IP phone. |
| 13:21.18 | Winkie | that probably is your problem then, check the wiki for a rundown i believe |
| 13:21.22 | Enth | this is my sip.conf |
| 13:21.28 | Winkie | pastebin it |
| 13:21.31 | Enth | yup |
| 13:21.33 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 13:21.51 | austinnichols101 | enth: for nat start with http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html |
| 13:22.00 | Skid | austinnichols101: the top part of your pastebin, [fwd] - is that in iax.conf? |
| 13:22.06 | austinnichols101 | yup |
| 13:22.10 | austinnichols101 | all of it is |
| 13:22.16 | Skid | ah, i only have the register and iaxfwd |
| 13:22.26 | austinnichols101 | iaxfwd is the incoming context |
| 13:22.30 | abusenode | lool |
| 13:22.32 | abusenode | port forward. |
| 13:22.41 | abusenode | just stop using nat you fuckers |
| 13:22.49 | Enth | lol |
| 13:23.01 | fgomes | x86: sorry for stupidity... i seldom use irc! what a pastebin is? |
| 13:23.12 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:23.12 | Enth | Winkle: http://pastebin.ca/45127 |
| 13:23.13 | [TK]D-Fender | x86 : You have a lot of cross-context redundancy to fix.... |
| 13:23.30 | Skid | according to this though, http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76 I dont need the [fwd] part? |
| 13:23.43 | x86 | [TK]D-Fender: eh, i wasnt sure ;) |
| 13:23.54 | Skid | fs, still being rejected |
| 13:24.05 | x86 | [TK]D-Fender: friends couldnt call local, local couldnt call friends, all kinds of shit... put the includes in, worked like a champ ;) |
| 13:24.28 | x86 | [TK]D-Fender: doesnt work as blissfully with the IVR context, however ;) |
| 13:24.40 | [TK]D-Fender | x86 : friends & local seem to be the same.... |
| 13:24.47 | austinnichols101 | skid: the register should be enough on it's own to see it connect |
| 13:24.56 | fgomes | Skid: Is fwd working? |
| 13:24.56 | Skid | ya |
| 13:25.00 | Skid | fgomes: nope |
| 13:25.00 | [TK]D-Fender | x86 : is the version I have still current? |
| 13:25.09 | Skid | NOTICE[11236]: chan_iax2.c:7410 socket_read: Registration of '754733' rejected: 'Registration Refused' from: '192.246.69.186' |
| 13:25.12 | Skid | is what i see |
| 13:25.12 | abusenode | fgomes: rafb.net/paste is a pastebin |
| 13:25.17 | abusenode | Skid: YOU FAIL |
| 13:25.20 | fgomes | my config is working fine... and I |
| 13:25.33 | fgomes | <PROTECTED> |
| 13:25.47 | Enth | Winkle: any luck? |
| 13:25.48 | abusenode | dont expectr help from me |
| 13:25.57 | austinnichols101 | 192.246 isn't a private segment |
| 13:26.10 | abusenode | no shit |
| 13:26.13 | abusenode | thanks for the observation |
| 13:27.05 | austinnichols101 | abusenode: you workin on this problem or just hatin everyone? |
| 13:27.19 | austinnichols101 | get a drink and chill out |
| 13:28.54 | Skid | AH |
| 13:28.55 | Skid | ffs |
| 13:28.58 | Skid | you have to enable it in the account |
| 13:29.20 | Skid | thats nice of them to include that vital bit of info in the damn tutorial |
| 13:29.23 | Skid | idiots |
| 13:30.09 | fgomes | Skid: http://pastebin.ca/45128 |
| 13:31.04 | Enth | hrmmm |
| 13:31.07 | Skid | fwd-peer = for ourgoing stuff? |
| 13:31.11 | Enth | guess no one can help with the audio issue |
| 13:31.20 | real-dev | hi folks |
| 13:31.50 | real-dev | I have a six seconds break when originating a call with * |
| 13:32.10 | fgomes | Skid : You can choose whataver context name you want. OH... it seems you need a sample from my dialplan as well... pls hold on. |
| 13:32.11 | Skid | nah its cool |
| 13:32.17 | real-dev | either with the call-file or via originate |
| 13:32.26 | Skid | its the fwd settings on the account part i didnt have |
| 13:33.03 | real-dev | anybody here who uses call-file or originate? |
| 13:34.00 | fgomes | Skid: My dialplan is quite complicated... full of macros... I will give you just a debug line from CLI: |
| 13:34.03 | austinnichols101 | enth: check the url I sent you |
| 13:34.04 | fgomes | - Executing Dial("SIP/299101-995a", "IAX2/fwd-peer/612|20|Tt") in new stack |
| 13:34.49 | Enth | fn~austinnichols101: ok brb |
| 13:35.00 | Skid | fgomes: yeah; I had to login to FWD's site and enable IAX2 registering |
| 13:35.06 | Skid | just gotta wait for it to be live |
| 13:35.43 | Skid | :D registerd |
| 13:35.46 | Skid | right |
| 13:35.57 | *** join/#asterisk Lino` (n=Lino@i577BDBE1.versanet.de) |
| 13:36.04 | Lino` | ~seen Possible |
| 13:36.15 | jbot | possible <n=Babbel@23.255-136-217.adsl-fix.skynet.be> was last seen on IRC in channel #asterisk, 1h 16m 54s ago, saying: 'I guess not'. |
| 13:36.30 | Skid | sorted |
| 13:36.33 | Skid | thanks for the help |
| 13:36.41 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 13:37.03 | asteriskmonkey | whats new in the 1.2.2 addons? |
| 13:37.11 | fgomes | Skid: Oh...yes! It was a long time ago I've enabled IAX2. I ve already forgotten it. |
| 13:37.42 | Skid | :) |
| 13:37.46 | Skid | i can call my self fine |
| 13:37.50 | Skid | so, all is well |
| 13:37.53 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 13:38.06 | fgomes | Skid: great! :_) |
| 13:38.11 | Dr-Linux | simple question, |
| 13:38.23 | Dr-Linux | <PROTECTED> |
| 13:38.48 | Dr-Linux | what can i do for file "No response recieved" ? |
| 13:39.01 | asteriskmonkey | can anyone help me debug a spandsp issue? |
| 13:39.49 | Dr-Linux | :S |
| 13:39.55 | Dr-Linux | anybody active? |
| 13:40.05 | asteriskmonkey | nope |
| 13:40.18 | Dr-Linux | asteriskmonkey: do you understand my question? |
| 13:40.32 | asteriskmonkey | cam in late i didnt see you question what is the question |
| 13:40.58 | Dr-Linux | asteriskmonkey: i have an IVR |
| 13:41.00 | [TK]D-Fender | Dr-Linux : Look for a recording that says what you want. If you can't find one, you'll have to make it yourself |
| 13:41.12 | asteriskmonkey | yes |
| 13:41.18 | [TK]D-Fender | "pbx-invalid" is just a pre-made recoding in the sounds folder |
| 13:41.37 | Dr-Linux | [TK]D-Fender: sir i understand, i have all recordings .. but my question is something different |
| 13:41.43 | [TK]D-Fender | Allison didn't record every phrase imaginable so go an look what you have. |
| 13:42.00 | Dr-Linux | if the callers doesn't enter any digit >>> it should SAY >> no response recieved |
| 13:42.20 | Dr-Linux | if the caller enters wrong number >> it should SAY >> invalid.... |
| 13:42.35 | fgomes | Skid: notice that you are not allowed to change context [iaxfwd] bcos FWD assumes you have one in your aix.conf ! |
| 13:42.37 | Winkie | Dr-Linux: use the default extensions? |
| 13:43.08 | Dr-Linux | Winkie: what do you mean? |
| 13:43.24 | asteriskmonkey | rtm :) |
| 13:43.24 | Skid | ye |
| 13:43.28 | Skid | works all a treat |
| 13:43.30 | Dr-Linux | actually in my IVR, "i" and "t" options are aleady in use, |
| 13:43.40 | asteriskmonkey | i=invalid t=timeout |
| 13:43.44 | Dr-Linux | so i just wanna use something for >> no response recieved? |
| 13:43.57 | asteriskmonkey | timeout is where you set that |
| 13:44.07 | [TK]D-Fender | Dr-Linux : GO LOOK AT WHAT RECORDINGS ARE AVAILABLE. *NAOW* |
| 13:44.11 | Dr-Linux | what should i use, as "t" and "i" are alraedy busy for some other stuff |
| 13:44.11 | Dr-Linux | ? |
| 13:44.15 | Skid | do FWD accept g729 as a codec? |
| 13:44.19 | iGotNoTime | I am fairly new and don't want to make a mistake, could some look at a provider for me and tell me if it would work with * ? |
| 13:44.19 | Skid | or is it all ulaw shite? |
| 13:44.31 | asteriskmonkey | you need to know how to write an ivr properly then |
| 13:44.53 | Skid | what provider? |
| 13:44.57 | asteriskmonkey | iGotNoTime: we got no time to just ask them if they support asterisk |
| 13:44.58 | [TK]D-Fender | "t" = no response received |
| 13:45.05 | Dr-Linux | [TK]D-Fender: well, i have recoreded all the .gsm files .. thats not a problem |
| 13:45.11 | Dr-Linux | yess |
| 13:45.20 | Dr-Linux | [TK]D-Fender: i think you got my question |
| 13:45.23 | fgomes | Skid: not sure... the issue is g729 is proprietary. If you have a softphone you possibly do not have g729 enabled. ulaw is bad... but it is universal. |
| 13:45.32 | Winkie | ulaw's not that bad! |
| 13:45.36 | Winkie | also alaw over here 8) |
| 13:45.37 | iGotNoTime | asteriskmonkey, ok thanks :) |
| 13:45.38 | asteriskmonkey | Skid: ulaw is the best codec for sound quaily |
| 13:45.40 | Dr-Linux | [TK]D-Fender: but sir "t" is already being used for something else? |
| 13:45.40 | Skid | I run all cisco 7940/7960's :) |
| 13:45.44 | Dr-Linux | lemme show you |
| 13:45.46 | [TK]D-Fender | ULAW is great if you can afford the bandwidth. |
| 13:45.55 | Skid | b/w isnt so much as a problem |
| 13:46.01 | Dr-Linux | [TK]D-Fender: look |
| 13:46.02 | Dr-Linux | exten => t,1,Set(TRIES=$[${TRIES} + 1]) |
| 13:46.03 | Dr-Linux | exten => t,2,GotoIf($["${TRIES}" = "1"]?t,3:s,5) |
| 13:46.15 | Winkie | Dr-Linux: use pastebin, also that timeout is your problem |
| 13:46.25 | Dr-Linux | i'm already using "t" for this |
| 13:46.36 | fgomes | Winkie: ... in bandwidth terms... sound quality is pretty good! |
| 13:46.37 | [TK]D-Fender | Dr-Linux : So what? Thats what gets called when there is no response! Whats not clear about that? |
| 13:46.37 | *** join/#asterisk trelane` (n=trelane@208.64.32.51) |
| 13:46.46 | trelane` | is there any way besides creating a queue to ring two extensions at once? |
| 13:46.53 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 13:46.59 | [TK]D-Fender | Dr-Linux : You want a recording in there too? Just add it in front! |
| 13:47.07 | [TK]D-Fender | *sheesh* |
| 13:47.10 | *** join/#asterisk ToTo (n=ToTo@host214-134.pool872.interbusiness.it) |
| 13:47.23 | *** join/#asterisk Meaty-Wrk (n=cp_simbu@office.abi.ca) |
| 13:47.30 | Dr-Linux | [TK]D-Fender: so where can i put the file name "no-response-recieved" :S |
| 13:47.47 | asteriskmonkey | if anyone can answer this is would be cool, im trying to use spandsp and it appears to be working in the console but it dosnt write a file |
| 13:48.04 | Dr-Linux | as i don't wanna loose exten => t,1,Set(TRIES=$[${TRIES} + 1]) << functionality, |
| 13:48.06 | *** join/#asterisk blkremedy (n=ur3rdeye@142M28.oasis.mediatti.net) |
| 13:48.15 | [TK]D-Fender | You put it wherever you want and just play it back! Playback(/path/to/my/stupid/recording) |
| 13:48.20 | Dr-Linux | its mean use will listen 2 time this menu |
| 13:48.24 | [TK]D-Fender | Dr-Linux : ADD IT IN FRONT! |
| 13:48.32 | [TK]D-Fender | IN FRONT! |
| 13:48.45 | [TK]D-Fender | get it? |
| 13:48.51 | fgomes | Skid: g729 and g723 are proprietary (paid). g729 costs $10 / connection. Not expansive! |
| 13:48.52 | Enth | Winkie - any luck with that audio issue ? |
| 13:48.55 | [TK]D-Fender | renumber those lines |
| 13:49.10 | Skid | i know the chap who wrote g729 |
| 13:49.12 | Dr-Linux | [TK]D-Fender: sirry in that case my current "t" things will work too? right |
| 13:49.13 | Skid | :-) |
| 13:49.27 | Skid | he offered me a number, for our business name, cept it was 5,000 GBP |
| 13:49.30 | Skid | like, no thanks:) |
| 13:49.31 | [TK]D-Fender | Dr-Linux : its just a bunch of commands in a row! |
| 13:49.57 | Dr-Linux | [TK]D-Fender: wait let me try and the pastbin you |
| 13:50.32 | fgomes | Skid: you can licence from Digium. For my server (low number of users/connections) is is affordable. |
| 13:50.43 | Enth | hrmmm |
| 13:50.50 | *** join/#asterisk heison (n=heison@w3.somanetworks.com) |
| 13:51.23 | Skid | I have a license, thanks |
| 13:51.24 | Skid | =] |
| 13:51.39 | Enth | guys, pls take a look at this and tell me why a hardware IP phone does not receive audio (yes its behind the same NAT as the * server) http://pastebin.ca/45127 |
| 13:51.49 | iGotNoTime | this provider sells a sipura 1001 does that automatically mean they would work with * ? |
| 13:52.03 | Dr-Linux | [TK]D-Fender: sir look at this >> http://pastebin.com/594372 |
| 13:52.37 | Dr-Linux | i only add one line infront, thats what i need beside all other "t" functions |
| 13:52.57 | [TK]D-Fender | Enth : Describe the exact call. Can 2 phones INSIDE your LAN call each other properly? |
| 13:53.38 | Enth | [TK]D-Fender: two x-lite clients inside the LAN can call and talk etc. |
| 13:53.50 | Dr-Linux | [TK]D-Fender: lemme know if i'm wrong? |
| 13:53.57 | Enth | it's only when calling a hardware ip phone that there is no audio. |
| 13:54.37 | [TK]D-Fender | Enth : Where is the IP phone located? |
| 13:54.38 | Enth | everyone says that it's a NAT issue but all settings are correct. |
| 13:54.43 | Enth | inside the same LAN |
| 13:54.54 | fgomes | Enth: all is behind NAT but IP phones are not separeted from * server by nat. So nat=no in the IP phones config. |
| 13:54.59 | Enth | behind NAT as are the other clients and the * server |
| 13:55.16 | [TK]D-Fender | Enth : so all of your phones are local to your server? |
| 13:55.25 | Meaty-Wrk | Dr-Linux : You have a infinite loop in your context |
| 13:55.35 | Enth | fgomes: Done that, still doesnt work |
| 13:55.36 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:55.41 | [TK]D-Fender | Dr-Linux : fic your gotoif... the #'s changed... |
| 13:55.44 | [TK]D-Fender | fix* |
| 13:56.00 | Enth | [TK]D-Fender: yes. The ip phone rings and answers but no audio |
| 13:56.07 | iGotNoTime | I live in the USA can I use a provider than grants me a UK (or any other country) PSTN number, is it ok? |
| 13:56.27 | asteriskmonkey | Enth: you have a codec issue or dodgy build of asterisk |
| 13:56.46 | fgomes | Enth: every client presents the problem? even IP phones in the same LAN present one way audio? |
| 13:56.47 | Enth | fn~asteriskmonkey: then why do the x-lite clients work? |
| 13:56.59 | asteriskmonkey | clearly codec issue then |
| 13:57.05 | trelane` | is there any way besides creating a queue to ring two extensions at once? |
| 13:57.07 | Dr-Linux | :S |
| 13:57.12 | Enth | fgomes: there is only one LAN. |
| 13:57.18 | [TK]D-Fender | Enth : then NONE of your phones should ne "nat=yes". |
| 13:57.19 | Enth | hrmm codec issue |
| 13:57.20 | Dr-Linux | [TK]D-Fender: whats wrong in this >> exten => t,3,GotoIf($["${TRIES}" = "1"]?t,3:s,5) |
| 13:57.52 | [TK]D-Fender | Dr-Linux : Read the damn line. if that is TRUE then it GOTO's Line 3. this IS line 3!!! infinite loop! |
| 13:57.58 | Enth | asteriskmonkey: what codecs would you suggest? |
| 13:58.20 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 13:58.22 | Stephnie | hi |
| 13:58.26 | Stephnie | Mar 10 19:11:46 WARNING[6667]: channel.c:784 channel_find_locked: Avoided initial deadlock for '0x8135b60', 10 retries! |
| 13:58.29 | [TK]D-Fender | trelane` : Dial(SIP/100&SIP/200&ZAP/1) |
| 13:58.31 | Stephnie | what kind of warning is this? |
| 13:58.35 | Dr-Linux | huh |
| 13:58.41 | trelane` | [TK]D-Fender, thanks |
| 13:58.57 | PakiPenguin | evening |
| 13:59.20 | asteriskmonkey | enth: use ulaw only to start |
| 13:59.25 | [TK]D-Fender | Dr-Linux : line t,3 has a goto t,3 in there! it means it will keep doing that commend FOREVER when it is true and never hang up! |
| 13:59.26 | Enth | hrmmm...any suggestions guys? |
| 13:59.37 | [TK]D-Fender | Enth : remove those NAT lines. |
| 13:59.38 | Dr-Linux | [TK]D-Fender: that's not my problem, i just type all to show you if this is a right way to use "no-response-recieved" line :( |
| 13:59.39 | Stephnie | is there anyone aware of this message? |
| 13:59.41 | Stephnie | Mar 10 19:11:46 WARNING[6667]: channel.c:784 channel_find_locked: Avoided initial deadlock for '0x8135b60', 10 retries! |
| 13:59.52 | Stephnie | ????? |
| 14:00.04 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-197.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 14:00.26 | Dr-Linux | [TK]D-Fender: ofcos i'll change the line 3 stuff |
| 14:00.49 | Dr-Linux | but all i want to know about damn "no-response-recieved" |
| 14:00.58 | [TK]D-Fender | Dr-Linux : Does the recording you are trying to play EXIST? |
| 14:01.12 | Enth | [TK]D-Fender: that did not fix the audio issue :( |
| 14:01.23 | Dr-Linux | [TK]D-Fender: yes i have all recordings |
| 14:01.28 | Enth | Could it be a codec issue for the ip phone? |
| 14:01.28 | [TK]D-Fender | Enth : maybe MIC is set wrong? |
| 14:01.47 | [TK]D-Fender | Dr-Linux : Did you find the EXACT FILE you are calling? How about TESTING it? |
| 14:02.11 | Dr-Linux | [av]bani: i'm going to bulit a very big production ivr with lot of AGI stuff, but this is a little problem for me |
| 14:02.12 | Enth | mic on where? the ip hardphone? at least i should be able to hear the person if the mic is not set |
| 14:02.15 | Dr-Linux | that's what i asking |
| 14:02.32 | [TK]D-Fender | Enth : You also have 2 context statements in [general] .... not good |
| 14:02.40 | heison | any SIP expert willing to offer help SIP 403 forbidden error? http://pastebin.ca/45130 |
| 14:02.59 | Enth | ok |
| 14:02.59 | Dr-Linux | [av]bani: sir, with testing maybe i'll get, but i wanna understand the logic, bcoz later i need to use it on may other palces .. |
| 14:03.08 | Dr-Linux | so i just wanna understand the logic |
| 14:03.32 | mikefoo | [TK]D-Fender: whats up |
| 14:03.36 | [TK]D-Fender | Enth : When you're done, pastebin your full new sip.conf |
| 14:03.47 | Dr-Linux | but there is aleady "t" in use ... so i want if someone takes long to enter digits, he/she should listen, "no-response-recieved" |
| 14:03.54 | [TK]D-Fender | mikefoo : Pulling my hair out on ridiculous question :) |
| 14:04.02 | Dr-Linux | and all other "t" stuff should also work |
| 14:04.12 | mikefoo | heh.. |
| 14:04.44 | Dr-Linux | [TK]D-Fender: i have more then 14 sub menu, so everything i have to use same, so all i need to understand, if i'm going rignt or wrong |
| 14:04.45 | [TK]D-Fender | Dr-Linux : IT works ok? Do you understand the basic principles of programming? It does t,1 then goes on to t,2 until something tells it to stop or go somewhere else! |
| 14:04.57 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 14:04.57 | [TK]D-Fender | its s friggen sequence! |
| 14:05.01 | [TK]D-Fender | just lines of code! |
| 14:06.05 | iGotNoTime | Guys I am sorry my questions are 'noob' questions, I am reading so much trying to grasp it all. This channel clearly is beyond me, maybe there is a * channel for people like me? |
| 14:06.07 | Dr-Linux | yes, i know that ... |
| 14:06.13 | Enth | [TK]D-Fender: http://pastebin.ca/45131 |
| 14:06.20 | Enth | done as requested. |
| 14:06.25 | Enth | and now... |
| 14:06.31 | Skid | like you? :o |
| 14:07.05 | willt | Hey gotnotime did you take your router back and get something else? |
| 14:07.38 | iGotNoTime | willt, Yes I returned it and bought a sipura online |
| 14:07.47 | iGotNoTime | should be here Tuesday |
| 14:08.03 | willt | thats cool they work pretty good |
| 14:08.06 | [TK]D-Fender | Enth : remove all of the "username" lines |
| 14:08.41 | Enth | [TK]D-Fender: you dont want me to add no usernames? |
| 14:08.44 | iGotNoTime | the thing is the provider I am looking for is a bit more intracate than what most are looking for :( |
| 14:08.51 | Enth | then how will it identify the extension? |
| 14:09.15 | [TK]D-Fender | heison :You can't put multiple codecs on an "allow" or "disallow line. make them seperate lines. |
| 14:09.40 | [TK]D-Fender | Enth : the name of your sections like [6666] *is* the username. |
| 14:09.55 | Enth | ok |
| 14:09.58 | Dr-Linux | [TK]D-Fender: how it looks for now you? >> http://pastebin.com/594415 |
| 14:10.34 | Enth | ok done. Now? sip reload ? |
| 14:11.09 | heison | [TK]D-Fender: sure, changed that... for some reason, the register line seems to be failing... i have even tried with the optional :authuser, same deal |
| 14:11.26 | [TK]D-Fender | Dr-Linux : Looks fine, but does that soudn file exist? |
| 14:11.33 | Enth | [TK]D-Fender: In doing so, my clients cannot login anymore. |
| 14:12.24 | Dr-Linux | [TK]D-Fender: thanks sir, i'll put that file, thats not a problem for |
| 14:12.33 | [TK]D-Fender | heison : Put "nat=yes" in [general] |
| 14:12.47 | Dr-Linux | [TK]D-Fender: sorry for my bad english, that i can't make you understand my question () |
| 14:12.52 | [TK]D-Fender | Enth : pastebin your attempt |
| 14:13.07 | Zipper_32 | iGotNoTime: Regarding your question if there are people like you out there, I was in your shoes last summer. We've all been there. |
| 14:13.23 | [TK]D-Fender | Dr-Linux : Learn to just TRY things and wath the CLI and see if it does what you expect. |
| 14:13.57 | iGotNoTime | Zipper_32, where did you go to get you answers? The Wiki on voip-info is not detailed enough for my questions :( |
| 14:13.57 | Winkie | iGotNoTime: a 941? |
| 14:14.04 | Enth | [TK]D-Fender: http://pastebin.ca/45135 |
| 14:14.27 | iGotNoTime | Winkie, sorry I don't understand? The ATA? The ATA is a 2001 I think |
| 14:14.36 | Zipper_32 | iGotNoTime: Where are you at right now regarding experience? |
| 14:14.37 | Winkie | 14:07.39 < iGotNoTime> willt, Yes I returned it and bought a sipura online |
| 14:14.40 | heison | [TK]D-Fender: nat=yes now in general, reload (as well as restart) still return 403 forbidden, |
| 14:14.43 | Winkie | i thought you meant a sipura spa 941 |
| 14:14.44 | Winkie | nice phones |
| 14:14.45 | Winkie | anyway brb |
| 14:15.08 | iGotNoTime | Zipper_32, I have * running, I ordered the ATA, and am shopping for a provider before the box arrives |
| 14:16.00 | Zipper_32 | iGotNoTime: One thing I used was just sittin in this channel and listening. Another was reading product descriptions on websites, and a third was the Oreilly asterisk book to get the very basics. |
| 14:16.02 | iGotNoTime | Zipper_32, the install was easy and have no technical questions, just provider questions :( |
| 14:16.20 | iGotNoTime | Zipper_32, how much was the book? |
| 14:16.40 | willt | i like the sipura ata adapters. I tried a few of their phones but the sound quality wasn't as good as the cisco 7960's i use |
| 14:16.44 | Zipper_32 | iGotNoTime: Free: http://voipspeak.net/index.php?/content/view/33/2/ |
| 14:16.59 | [TK]D-Fender | Enth : I said get rid of the "username=6666" type lines, NOT the [6666] context hearders! |
| 14:17.05 | Enth | ah ok |
| 14:17.08 | Enth | hang on |
| 14:17.11 | iGotNoTime | Zipper_32, LOL will download now :P |
| 14:17.18 | iGotNoTime | Zipper_32, thank you :) |
| 14:17.25 | *** join/#asterisk kpettit (n=keith@69.15.174.113) |
| 14:17.36 | Zipper_32 | It has a lot of basic information, but even if you already know some, you'll be more well-rounded. |
| 14:17.47 | Enth | [TK]D-Fender: ok done, sip reload? |
| 14:17.56 | [TK]D-Fender | yup |
| 14:18.10 | Enth | done |
| 14:18.12 | Enth | now? |
| 14:18.38 | iGotNoTime | Zipper_32, asterisk is capable of anything the provider permits correct? |
| 14:18.58 | [TK]D-Fender | x86 : done |
| 14:19.01 | iGotNoTime | Zipper_32, even local dial in number in several countries? |
| 14:19.05 | Enth | still no audio. |
| 14:19.07 | *** join/#asterisk Nand0 (n=Nando@unaffiliated/nand0) |
| 14:19.21 | Zipper_32 | iGotNoTime: Sure, just set up the dialplan appropriately. |
| 14:19.40 | iGotNoTime | Zipper_32, that was my only question :) Happy you knew the answer :D |
| 14:20.15 | Enth | hrmmm |
| 14:20.43 | Zipper_32 | Enth: Don't litter |
| 14:20.48 | [TK]D-Fender | Enth : Which? |
| 14:20.48 | Enth | its useless. good luck trying to get it to work |
| 14:21.05 | Enth | well's from ipchitchat dot com |
| 14:21.06 | [TK]D-Fender | Enth : and re-pastebin please |
| 14:21.10 | Enth | ok wait |
| 14:22.31 | Enth | [TK]D-Fender: http://pastebin.ca/45137 |
| 14:22.49 | Dr-Linux | [TK]D-Fender: sir i created an IVR in /etc/asterisk/something.conf , i want this file work with extension.conf file |
| 14:23.00 | Dr-Linux | should i just #include something.conf |
| 14:23.07 | Dr-Linux | in extensions.conf or what else i need ? :S |
| 14:23.26 | Dr-Linux | i never did this before, but i saw you things, you have done such things :) |
| 14:23.40 | Dr-Linux | s/you/your |
| 14:24.43 | [TK]D-Fender | Dr-Linux : #include looks fine |
| 14:25.22 | [TK]D-Fender | enth do any of the phones work fine between each other? |
| 14:25.42 | heison | [TK]D-Fender: i have tried a soft phone on the same network, it works fine... so, it's unlikely the firewall that I'm behind... |
| 14:25.42 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 14:26.10 | [TK]D-Fender | Enth : And you have no passwords on any of your accounts? |
| 14:26.15 | Enth | nope |
| 14:26.17 | Enth | not needed |
| 14:26.26 | Enth | x-lite clients work fine |
| 14:26.34 | [TK]D-Fender | Enth : Do it.. maybe there is another problem, and double check the user you put into the phone... |
| 14:26.35 | Enth | x-lite to x-lite = works fine |
| 14:26.46 | [TK]D-Fender | so x-lite -> phone = 1 way audio? |
| 14:26.53 | Enth | I have done that far too many time snow |
| 14:26.57 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 14:27.04 | Enth | x-lite - phone = no audio at all |
| 14:27.15 | heison | [TK]D-Fender: I basically followed these instructions...http://www.voip-info.org/wiki/view/asterisk+settings+HKBN+2b |
| 14:27.16 | Dr-Linux | [TK]D-Fender: so if i do in extensions.conf "#include" so will it's all context be for all, or i'll still need "include" option for diferent context? |
| 14:27.19 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 14:27.43 | Enth | phone rings fine, answers fine but after connected, cant hear the caller or he/she cant hear me |
| 14:27.48 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfimu.dialup.mindspring.com) |
| 14:28.08 | [TK]D-Fender | Dr-Linux : #include will include another FILE. not to be mistaken for "include => anothercontextsextens" |
| 14:28.23 | [TK]D-Fender | Enth : 1-way audio or NONE at all? |
| 14:28.30 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:28.31 | *** join/#asterisk robby (n=robby@host23-229.pool8252.interbusiness.it) |
| 14:28.35 | Enth | none at all |
| 14:28.43 | Dr-Linux | oo ic |
| 14:28.43 | [TK]D-Fender | and which # is the hardphone, they all say "x-lite"? |
| 14:28.44 | Dr-Linux | ok thanks |
| 14:28.54 | Enth | hardphone = 3333 |
| 14:29.12 | Enth | those are just comments i didnt remove :) |
| 14:29.43 | Enth | since x-lite work fine. |
| 14:29.54 | [TK]D-Fender | Enth : Ok well that setup is as basic as can be.... I don't see why... do "set verbose 10" in CLI and pastebin a call attempt. |
| 14:30.05 | Enth | ok wait |
| 14:30.18 | [TK]D-Fender | Enth : Yeah, double check the phone's config... |
| 14:30.29 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 14:31.15 | Enth | http://pastebin.ca/45139 |
| 14:31.18 | Enth | :) |
| 14:31.56 | x86 | [TK]D-Fender: done? where can i get it? :) |
| 14:32.03 | iDunno | morning |
| 14:32.14 | [TK]D-Fender | Enth : ok, while the call is in progress do "sip show channels" and pastebin it |
| 14:32.16 | PakiPenguin | hey x86 :) |
| 14:32.21 | x86 | yo |
| 14:32.24 | [TK]D-Fender | Katty: mew. |
| 14:32.26 | PakiPenguin | sup? |
| 14:32.28 | Katty | i hereby pronounce autozone as being dreamy. |
| 14:32.33 | x86 | i'm on my way out the door, but [TK]D-Fender owes me a config file ;) |
| 14:32.41 | x86 | [TK]D-Fender: so how about it? :) |
| 14:32.42 | Katty | they told me what my check engine light was trying to tell me |
| 14:32.44 | Katty | for free! |
| 14:32.56 | x86 | dreamy hahaha |
| 14:33.13 | x86 | hey... the 50's just SIP'd me, they want thier dialect back.... |
| 14:33.16 | [TK]D-Fender | x86 : http://pastebin.ca/45140 |
| 14:33.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:33.21 | x86 | [TK]D-Fender: danke |
| 14:33.36 | coppice | Katty: they only tell you that so you'll check it in for the far from free repairs :-) |
| 14:33.38 | Katty | [TK]D-Fender: mew. |
| 14:34.02 | Katty | coppice: it doesn't need repair. (= |
| 14:34.07 | Katty | coppice: just the gas cap. |
| 14:34.17 | heison | hi coppice |
| 14:34.26 | Enth | hrmmmm |
| 14:34.26 | Katty | coppice: it has happened 2 before.. |
| 14:34.36 | Enth | bah i give up with this fscking hardphone |
| 14:34.43 | Enth | what an absolute waste of time. |
| 14:34.48 | Enth | heh |
| 14:34.58 | Katty | i had to fsck my asterisk box this morning :< |
| 14:35.11 | Katty | we had storms last night and the power went out for awhile |
| 14:35.45 | *** join/#asterisk horvath (i=horvath@efnut.com) |
| 14:35.51 | X-Gen | Katty: sounds like your linux box has been sitting 2 close 2 a windows box |
| 14:35.59 | Katty | X-Gen: it's on a rack. |
| 14:36.07 | Katty | X-Gen: with several windows boxes (= |
| 14:36.15 | jsharp | And no UPS? |
| 14:36.34 | Katty | i see someone thinks i'm an idiot (= |
| 14:36.41 | Katty | that, or they're one. |
| 14:37.01 | Winkie | haha, the number of times the power's gone out here and i've had to do emergency fscks |
| 14:37.03 | Winkie | not for much longer! |
| 14:37.49 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfimu.dialup.mindspring.com) |
| 14:38.00 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 14:40.36 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 14:41.03 | Katty | X-Gen: and i'm an mcp, by the way :) |
| 14:41.06 | Katty | doomed :< |
| 14:41.41 | *** join/#asterisk octothorpe_away (n=octothor@198.60.73.171) |
| 14:42.20 | Katty | octothorpe: what's an octothorpe? |
| 14:42.33 | Corydon76-home | aka a hash mark or pound |
| 14:43.43 | octothorpe | thanks Corydon76 |
| 14:44.18 | coppice | hash mark? dope leaves special marks? :-\ |
| 14:44.25 | Corydon76-home | Although technically, an 8-pointed star would also be an octothorpe |
| 14:45.01 | MikeJ[Laptop] | coppice, only if you fall asleep with your face pressed againsy somthing@ |
| 14:45.27 | Enth | fn~[TK]D-Fender: Well I'll be damned. If I use DTMF, that is pressing the keys on the phone, I can hear it. |
| 14:45.38 | Enth | but not voice. |
| 14:46.19 | Enth | any ideas on why no voice but can hear dtmf ? |
| 14:47.00 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 14:47.04 | coppice | because DTMF is not being sent as audio, I assume |
| 14:47.05 | jsharp | Codec mismatch? |
| 14:47.13 | Enth | yeah |
| 14:47.16 | Enth | possibly |
| 14:47.23 | Enth | what's the best codec to use for a home LAN? |
| 14:47.24 | MattB2 | hi all |
| 14:47.32 | Enth | gsm or g711u ? |
| 14:47.33 | MattB2 | struggling with call limits on 1.2.5. Basically need to allow a user to make two outgoing calls (for attended transfer) but only accept one incoming |
| 14:47.41 | MattB2 | according to docs, incominglimit and outgoinglimit are no longer used, there's just call-limit |
| 14:47.50 | MattB2 | but call-limit=2 allows someone to call in when the user's already on phone, setting call-limit=1 means that user can't dial out for attended/conf calls |
| 14:47.56 | MattB2 | any suggestions? |
| 14:48.17 | jsharp | Home lan? G711. |
| 14:48.22 | jsharp | Plenty o' bandwidth. |
| 14:48.35 | Enth | ok |
| 14:48.48 | Enth | applying QoS will also help I guess. |
| 14:48.51 | Enth | :) |
| 14:49.33 | Corydon76-home | QoS is overkill for a home network, unless you have Windows machines that are prone to virus infestation |
| 14:49.42 | jsharp | If you need QoS on a local lan, you've got bigger problems. |
| 14:49.54 | horvath | MattB2: I would use the set group stuff to count it manualy |
| 14:50.16 | Enth | jsharp: Well, I just like to be perfect. |
| 14:50.23 | Enth | overkill I guess. |
| 14:50.34 | horvath | MattB2: Set(GROUP(${ACCOUNTCODE})=CHANNELS) |
| 14:50.57 | coppice | QoS doesn't generally function on LANs |
| 14:50.58 | horvath | MattB2: and then just a GotoIf or something |
| 14:51.18 | jsharp | Besides, unless you have a bunch of reallllllly smart layer 3 switches that can grok QoS, nothing would actually pay attention to the QoS bits. |
| 14:51.34 | MattB2 | horvath: thanks, i'll look into it. outgoinglimit and incominglimit used to work perfectly... feels like we're moving backwards not forwards! |
| 14:51.57 | horvath | MattB2: GotoIf($[${GROUP_COUNT(CHANNELS@${ARG2})} > 1]?s-BUSY|1) |
| 14:52.06 | horvath | MattB2: Yea I know :) |
| 14:52.41 | horvath | MattB2: ingore the arg2 thats just my own thing anyways yea just check the wiki |
| 14:53.01 | MattB2 | will do, thanks again. |
| 14:54.13 | blkremedy | does anyone here know how to apply a diff file? |
| 14:54.38 | Winkie | blkremedy: patch file-to-patch < patch.file ? |
| 14:54.54 | Winkie | (you might need -p) |
| 14:57.02 | blkremedy | is diff and patch the same? |
| 14:57.05 | Winkie | no |
| 14:57.29 | Winkie | diff is the utility that produces diffs, which are commonly referred to as 'patches', patch is the program which applies them |
| 14:57.55 | *** join/#asterisk octothorpe_ (n=octothor@198.60.73.170) |
| 14:58.18 | blkremedy | O...ok thanks for clearing that up. |
| 14:58.39 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 14:59.01 | Winkie | that's no problem, now tell me how to pump CDR data into a pgsql database |
| 14:59.01 | Winkie | ta |
| 14:59.45 | Kumbang | guys, what FXO voip gateway does support threeway calling and call waiting? |
| 15:01.21 | mphill | thats done at the station |
| 15:01.22 | *** part/#asterisk octothorpe[away] (n=octothor@198.60.73.170) |
| 15:01.36 | mphill | in short, yes |
| 15:01.53 | Fedoracore6 | hello all |
| 15:01.53 | mphill | just don't get a modem |
| 15:02.17 | mphill | most voice mmodems support caller ID and call waiting these, but the quality it God aweful |
| 15:02.24 | heison | I'm having trouble registering SIP with a provider... can anyone offer some help? I keep getting 403 Forbidden and if I use a softphone under Windows on the same network, everything works fine... http://pastebin.ca/45130 |
| 15:03.01 | *** join/#asterisk octothorpe_ (n=octothor@198.60.73.171) |
| 15:03.11 | mphill | heison: from the phone dial *60, see if the chick tells you the time |
| 15:03.20 | Fedoracore6 | spme budy expert ini asterisk realtime system ,, |
| 15:03.27 | Fedoracore6 | some budy * |
| 15:03.28 | [TK]D-Fender | Enth : Obvious test : is the handset plugged in right? :) |
| 15:03.37 | *** part/#asterisk octothorpe_ (n=octothor@198.60.73.171) |
| 15:03.58 | heison | mphill: you mean on the softphone under windows? |
| 15:04.54 | mphill | heison: i assume you are trying to get voip sip trunks setup, right? |
| 15:05.05 | mphill | with asterisk as the gateway |
| 15:06.14 | [TK]D-Fender | heison : You should put "nat=yes" and "canreinvite=no" into [general] |
| 15:06.15 | *** join/#asterisk octothorpe_ (n=octothor@198.60.73.170) |
| 15:06.20 | heison | mphill: yes |
| 15:06.22 | *** part/#asterisk octothorpe_ (n=octothor@198.60.73.170) |
| 15:06.26 | mphill | ok |
| 15:06.36 | [TK]D-Fender | heison : And make sure the appropriate ports are forwarded to * |
| 15:06.38 | mphill | so dial *60 and see if that works |
| 15:06.43 | fgomes | I'm facing a problem redirecting an incoming call to landline ZAP/1. ZAP/1 immediately "answers the call" instead of waiting for external party to answer. Any idea? |
| 15:06.49 | Enth | heh |
| 15:06.52 | Enth | brb |
| 15:07.09 | mphill | fgomes: check your incoming routes |
| 15:07.34 | mphill | Enth: its worth it, if you don't want echo land up in your ear |
| 15:07.46 | [TK]D-Fender | Enth : Polycom is a better deal..... |
| 15:07.55 | fgomes | mphill: the incoming routes are OK, I mean... a Call-Me from FWD is received and redirected to ZAP/1 (landline). |
| 15:08.10 | [TK]D-Fender | mphill : Echo? He doesn't even hear anything the FIRST time! |
| 15:08.13 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
| 15:08.33 | mphill | [TK]D-Fender: thats might be user error :\ |
| 15:08.35 | fgomes | ... but ZAP/1 channel immediately answers the call when I Dial( .... ) |
| 15:08.36 | heison | [TK]D-Fender: I have canreinvite=yes so that my SIP phones can communicate without Asterisk being in the middle after the call is setup... sure, i can remove that for now |
| 15:08.44 | gambolputty | Is it possible with the mysql command to select more than one field at a time? |
| 15:09.21 | heison | mphill: i'm on the windows softphone, calling *60 returns user not found... |
| 15:10.05 | fgomes | mphill: Seems to be something in zapata.conf. |
| 15:10.43 | mphill | i think your extensions are setup wrong |
| 15:11.01 | mphill | you should be able to talk to asterisk at the least, you know what i mean? |
| 15:11.11 | mphill | *60 is the time nazi on asterisk |
| 15:11.25 | [TK]D-Fender | fgomes : Analog Zap channels have no means of knowing if the other side answers so it is considered answered as soon as * is capable of dialing. |
| 15:12.03 | [TK]D-Fender | mphill : what are you talking about? *60 is someone elses invention, not *'s.... |
| 15:12.20 | heison | mphill: i have no problems communicating between my SIP phones and Asterisk, the problem is with Asterisk and the SIP provider |
| 15:12.33 | twisla | "DEBUG[13911]: Dropping voice to exceptionally long queue on IAX2/qz00@qz00/1" does anyone have seen something like that ? |
| 15:12.58 | Hmmhesays | anyone know the default login info for netopia routers? |
| 15:13.02 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 15:14.35 | heison | [TK]D-Fender: i have port 5060 fowarded to *, what else do i need? |
| 15:14.53 | [TK]D-Fender | heison : 10000-20000 UDP |
| 15:15.00 | [TK]D-Fender | 5060 is UDP as well |
| 15:16.02 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:16.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:17.59 | heison | i don't have RTP ports forwarded just yet... but that shouldn't cause the registration to fail |
| 15:18.12 | heison | yes, 5060 UDP is forwarded |
| 15:19.54 | fgomes | D-Fender: what do you suggest? I've changed some parameters and call remains up after I drop the phone (the extenal one, my mobile called by ZAP/1). Any idea? |
| 15:21.11 | *** join/#asterisk unixgeek (n=unixgeek@12.45.238.189) |
| 15:21.16 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
| 15:22.40 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 15:25.33 | *** join/#asterisk aah_user (n=octothor@198.60.73.230) |
| 15:25.39 | *** part/#asterisk aah_user (n=octothor@198.60.73.230) |
| 15:25.57 | fgomes | [TK]D-Fender, mphill: Pls see http://rafb.net/paste/results/dqs70K21.html |
| 15:30.02 | *** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net) |
| 15:30.11 | [TK]D-Fender | fgomes : Disconnect detectiong on analog is also a PITA... get rid of your signalling line in there and change to fxsks. |
| 15:32.19 | *** join/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 15:32.54 | *** part/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 15:34.24 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 15:35.06 | *** join/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 15:35.15 | *** part/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 15:37.29 | *** join/#asterisk pycsusz (i=root@IP-183-69.TvNetWork.Hu) |
| 15:37.40 | pycsusz | Hi Everybody! |
| 15:37.46 | *** join/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 15:38.04 | Hmmhesays | haha i found a website where you can type in lyrics and it guesses which song you are looking for |
| 15:38.05 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 15:38.08 | SpaceBass | morning |
| 15:38.20 | SpaceBass | so...who knew spoofing caller ID could really make some people mad! |
| 15:38.24 | pycsusz | I would like to know, how to use the extensions.conf and extensions.ael togother |
| 15:38.30 | octothorpe | Hmmhesays: sweet, where is it |
| 15:38.46 | pycsusz | If somebody cab help me, then do it please! |
| 15:39.01 | Hmmhesays | www.findmeatune.com |
| 15:39.20 | octothorpe | SpaceBass: Yeah, my mom hates it when I call with CallerID of her own number |
| 15:39.34 | [TK]D-Fender | Hmmhesays : Sounds like google ;) |
| 15:39.35 | PakiPenguin | can anyone suggest some service , with unlimited calling to the US and one incoming number? |
| 15:39.55 | SpaceBass | octothorpe, I had one friend worked up enough to call verizon...apparently (he claims) they traced it back to me |
| 15:40.09 | Hmmhesays | PakiPenguin: kind of |
| 15:40.15 | SpaceBass | PakiPenguin, broadvoice |
| 15:40.17 | octothorpe | PakiPenguin: Telasip |
| 15:40.25 | SpaceBass | jsharp, only works in cingulair, right? |
| 15:40.33 | PakiPenguin | SpaceBass, broadvoice works okay? |
| 15:40.40 | SpaceBass | i like it |
| 15:41.16 | octothorpe | Broadvoice has "issues" with international calls though I hear |
| 15:41.23 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 15:41.34 | PakiPenguin | i just need us48 calling |
| 15:41.36 | SpaceBass | i use BV to call france and ireland sometimes...seems to work fine |
| 15:41.55 | Hmmhesays | SpaceBass: try calling your X girlfriends new bf from her number |
| 15:41.56 | Hmmhesays | hahah |
| 15:42.05 | SpaceBass | lol |
| 15:42.11 | Hmmhesays | you want some drunken fun, there it is |
| 15:42.13 | Hmmhesays | lol |
| 15:42.27 | SpaceBass | don't mention drinking....im so bloody hung over right now i may pule |
| 15:42.30 | SpaceBass | puke |
| 15:42.38 | Hmmhesays | you too huh? it was ladies night last night, sweet geebus |
| 15:42.57 | SpaceBass | nope....not coming to me |
| 15:43.13 | Hmmhesays | i talked to some hotties, had a salad passed out |
| 15:43.20 | Hmmhesays | woke up already an hour late for work |
| 15:43.20 | SpaceBass | salad sounds good |
| 15:43.31 | SpaceBass | i have to catch a plane to New York in a few....going to do it all again |
| 15:43.37 | Hmmhesays | lovely |
| 15:43.39 | SpaceBass | this weekend is going to hurt my poor liver |
| 15:43.42 | Hmmhesays | and i'm out of smokes |
| 15:43.54 | Hmmhesays | I got band practice this weekend so i'm not going to drink *much* tonight |
| 15:44.02 | jsharp | SpaceBass: No, I can do it on my Sprint phone too....if I have it set for "no pin if calling from my cell phone" |
| 15:44.12 | *** join/#asterisk tooms (n=hype@203.57.131.22) |
| 15:44.26 | SpaceBass | jsharp, I have that setting turned on on my sprint phone too...and when I call I get asked for a password |
| 15:44.38 | SpaceBass | that's how this all started...someone bet me I couldn't check their voicemail |
| 15:45.10 | jsharp | I haven't checked it in a while. I'll have to test it in a few minutes. |
| 15:46.08 | Hmmhesays | SpaceBass LOL |
| 15:46.19 | Hmmhesays | thats why i have my pin enabled ALL the time |
| 15:46.43 | SpaceBass | i thought about that...but then realized I dont get important messages |
| 15:46.57 | Hmmhesays | i the carrier has the option for pinless access and its enabled you can check anyones voicemail |
| 15:47.14 | SpaceBass | im fairly sure sprint and verizon smartened up |
| 15:47.31 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 15:48.02 | Hmmhesays | ok i'm looking for a song it goes "something something, dirty little secret, dirty little secret" |
| 15:48.12 | Hmmhesays | punkish song |
| 15:48.41 | Hmmhesays | "i'll show you my dirty little secret, dirty little secret" something like that |
| 15:48.47 | SpaceBass | all american rejects |
| 15:49.09 | Hmmhesays | thanks |
| 15:49.57 | iGotNoTime | remember yesterday that it was said the TOS of some provider prohibits * ? |
| 15:49.58 | SpaceBass | np |
| 15:50.07 | Hmmhesays | they have any other decent songs? |
| 15:50.16 | *** join/#asterisk loveas (n=lab0rize@port927.ds1-fa.adsl.cybercity.dk) |
| 15:50.17 | viperdude | hi guys |
| 15:50.18 | iGotNoTime | will they really have any way of knowing? |
| 15:50.30 | jsharp | They can look at the SIP headers. |
| 15:50.34 | Hmmhesays | so change it |
| 15:50.35 | viperdude | anyone using externnotify option in voicemail.conf? |
| 15:51.44 | SpaceBass | viperdude, I use it...albeit via AMP |
| 15:51.55 | iGotNoTime | jsharp, that reply was to me? |
| 15:52.08 | Hmmhesays | SpaceBass you checked out freepbx? |
| 15:52.15 | SpaceBass | no |
| 15:52.20 | Hmmhesays | amp 2.0 beta |
| 15:52.24 | jsharp | Yeah. |
| 15:52.25 | Hmmhesays | pretty sweet |
| 15:52.29 | SpaceBass | checking now |
| 15:52.31 | iGotNoTime | jsharp, k thx |
| 15:52.35 | SpaceBass | hummm |
| 15:52.45 | Hmmhesays | much more pretty than the current amp incarnation |
| 15:52.52 | SpaceBass | ohhh |
| 15:52.59 | SpaceBass | sounds like a good replacement for my AAH box |
| 15:53.23 | Hmmhesays | you can upgrade aah's amp to freepbx |
| 15:53.24 | SpaceBass | spacey need pictures |
| 15:53.28 | SpaceBass | really?!?!?! |
| 15:53.31 | Hmmhesays | yeah |
| 15:54.03 | octothorpe | I upgraded my AAH to FreePBX |
| 15:54.16 | trelane` | iGotNoTime, why use them? I mean if they don't want you as a customer why support htem? |
| 15:54.23 | trelane` | iGotNoTime, vote with your money and use a different provider |
| 15:54.26 | Hmmhesays | check that link I pm'd you |
| 15:54.31 | octothorpe | it is just an update for the AMP that is included with AAH |
| 15:54.33 | viperdude | SpaceBass: I am trying to differentiate between a new voicemail and the voicemail app being called. Both of which fire externnotify |
| 15:55.34 | iGotNoTime | trelane`, no I just am looking for the one's that are ok with * :) |
| 15:55.55 | iGotNoTime | trelane`, I feel the same way, if they are not letting me use the system the way I want, then I don't need them |
| 15:56.10 | trelane` | iGotNoTime, make sure you call them and tell them that. |
| 15:56.11 | iGotNoTime | trelane`, that is why I cancelled Vonage 48 hours ago |
| 15:56.27 | trelane` | that was a brilliant move |
| 15:56.30 | trelane` | go grab a beer! |
| 15:56.46 | trelane` | but if they don't, screw 'em |
| 15:57.02 | iGotNoTime | LOL |
| 15:58.18 | trelane` | in this channel there are 312 voip users. who here uses vonage? |
| 15:58.34 | trelane` | (or voip in a busines setting?) |
| 15:58.36 | iGotNoTime | trelane`, now you make fun of me LOL |
| 15:58.44 | trelane` | so if you give a shit about voip... you wouldn't use vonage |
| 15:58.57 | trelane` | absolutely not, making the smart decision in this case absolutely makes up for past mistakes |
| 15:59.00 | trelane` | it's no issue ;) |
| 15:59.05 | *** join/#asterisk Hmm-work (i=PJirc@66.173.103.100) |
| 15:59.07 | iGotNoTime | trelane`, Skypebay too ;) |
| 15:59.16 | Hmm-work | coming to you from freepbx's irc client |
| 15:59.16 | trelane` | iGotNoTime, I've been considering setting up skype |
| 15:59.30 | iGotNoTime | for SIP-like only |
| 15:59.39 | iGotNoTime | trelane`, not for their paid systems |
| 15:59.42 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:00.22 | trelane` | iGotNoTime, right, they're not significantly different for what I'd do than say FWD |
| 16:00.37 | iGotNoTime | it would be kinda cool to have their 2 line phone setup with * though |
| 16:00.48 | iGotNoTime | trelane`, those 2 line cordless phones are sweet |
| 16:01.42 | trelane` | iGotNoTime, umm get a cisco or rca or etc 2 line or multi line business sip cordless? |
| 16:01.43 | iGotNoTime | well if you have Skype contacts that is |
| 16:02.07 | iGotNoTime | trelane`, I have a wifi sip now, but it doesn't support Skype contacts :P |
| 16:03.04 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 16:03.43 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 16:03.51 | trelane` | iGotNoTime, they'll probably add the functionality to asterisk |
| 16:04.13 | iGotNoTime | can Skype be routed through *? I thought it was proprietary. |
| 16:04.20 | *** join/#asterisk nagl (n=nagl@137.208.4.181) |
| 16:05.11 | PakiPenguin | SpaceBass, can 2 people call with broadvoice business plan at the same time ? |
| 16:05.31 | PakiPenguin | or for 2 people simultaneously calling i'd need 2 different connections? |
| 16:05.41 | Hmmhesays | this theory of a deadman album is seriously kickass |
| 16:06.06 | kippi | Hi |
| 16:06.22 | kippi | Just tried installing the new asterisk and getting this error http://pastebin.ca/45144 can anyone help? |
| 16:06.34 | *** part/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 16:06.49 | kippi | is it because I havn't install asterisk-addons? |
| 16:08.09 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 16:08.13 | nettie | Hi guys, I was wondering what's the best way to define sip phones extensions in extensions.conf ? just a variable pointing to Dial(SIP/username,20) ? |
| 16:09.07 | Skid | i use usernames |
| 16:09.11 | viperdude | nettie: use a macro |
| 16:09.11 | blkremedy | is there a solution for TDM400P cards not hanging up on FXO line? |
| 16:10.09 | [TK]D-Fender | kippi : Do you need ODBC for anything in *? |
| 16:10.25 | [TK]D-Fender | kippi : because its not finding the .so where it expects it. |
| 16:10.27 | *** join/#asterisk justnulling2 (n=justnull@ool-18bc017c.dyn.optonline.net) |
| 16:10.32 | jsharp | Heee. I just called one of my satellite vendors and the phone system answered with very familiar asterisk voice and prompts. |
| 16:10.50 | *** join/#asterisk justnulling2 (n=justnull@ool-18bc017c.dyn.optonline.net) |
| 16:11.15 | Fedoracore6 | hai all ... now i use grandstream phone ,but when the system have choice like press '1" to add and '2" drop ... |
| 16:11.27 | [TK]D-Fender | kippi : if not then just put "noload => res_odbc.so" into modules.conf |
| 16:11.30 | SpaceBass | ok.... |
| 16:11.35 | Fedoracore6 | why when i press the button ,,... the system cannot work hemm |
| 16:11.50 | [TK]D-Fender | Fedoracore6 : Make sure your DTMF mode matches. |
| 16:12.09 | Fedoracore6 | any configuration ..should i do to ... grandstream |
| 16:12.11 | SpaceBass | so the callerID spoofing to check voicemail ...not really a good idea... my buddy had a messaged on his cell from Verizon that there were 4 unauthorized attempts to check messages |
| 16:12.18 | SpaceBass | this isnt going to end well for me |
| 16:12.26 | [ProB]CrazyMan | I read now threw the dial() app, in 1.2.0 the priorityjump changed, so how does it there work with busy and not availible? |
| 16:12.41 | Fedoracore6 | oic [TK] how i can know that DTMF mode matches |
| 16:13.24 | *** join/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 16:13.26 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 16:13.30 | [TK]D-Fender | Fedoracore6 : Look at your config on the phone and what you set inSIP.CONF |
| 16:13.58 | nettie | viperdude thanx, where shall I read about that please? I also need to clear my mind regarding voicemail setup.. at the moment I defined an extensions followed by a Dial command pointing to the SIP phone, defined a timeout and the correct voicemail number. If you have a suggestion for a more scalable solution please let me know. |
| 16:15.05 | viperdude | nettie: http://www.voip-info.org/wiki/view/Stdexten+macro |
| 16:15.07 | wunderkin | wow this "voip xpress" guy just doesn't get it (lagged) |
| 16:15.39 | viperdude | using a macro you have a one line per extension for dialling, then any changes to macro are replicated across all extensions using that macro |
| 16:15.52 | *** part/#asterisk octothorpe (n=octothor@198.60.73.230) |
| 16:16.23 | nettie | viperdude seems perfect, thanx a lot |
| 16:16.42 | nettie | what's exaclty CONGESTION? |
| 16:17.14 | fourcheeze | when you can't breath properly |
| 16:17.20 | RoyK | :) |
| 16:17.20 | nettie | eheh |
| 16:17.29 | RoyK | nettie: no more available channels |
| 16:18.01 | nettie | RoyK that's chanunavailable I think |
| 16:18.23 | viperdude | CONGESTION is what I had on the M6 yesterday |
| 16:18.44 | nettie | I thoguht you had a doge viper :) |
| 16:18.54 | nettie | dodge |
| 16:18.55 | [TK]D-Fender | nettie : Here's a good sample for you : http://pastebin.ca/45147 |
| 16:19.05 | nettie | thanx guys |
| 16:19.16 | viperdude | nettie: I wish, all women on the 'net think that, I hate to shattter the illusion :-) |
| 16:19.51 | *** join/#asterisk justnulling2 (n=justnull@ool-18bc017c.dyn.optonline.net) |
| 16:20.28 | kippi | could someone have a look at http://pastebin.ca/45144 and point me in the right dir |
| 16:20.36 | *** join/#asterisk octothorpe (n=octothor@unaffiliated/octothorpe) |
| 16:20.37 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:23.21 | *** join/#asterisk justnulling2 (n=justnull@ool-18bc017c.dyn.optonline.net) |
| 16:23.23 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 16:23.28 | Hmmhesays | looks like you don't have that lib |
| 16:23.35 | Hmmhesays | or its not in your path |
| 16:24.38 | kippi | its looking for libodbc.so.1 ? |
| 16:25.23 | Hmmhesays | looks like it |
| 16:25.29 | Juggie | kippi, did you install * from source |
| 16:25.31 | Juggie | or from rpm |
| 16:25.34 | kippi | source |
| 16:25.52 | Juggie | the problem is exactally as it appears :) |
| 16:25.52 | kippi | but i am using my old config |
| 16:26.04 | Juggie | libodbc is missing |
| 16:26.09 | Juggie | what distro? |
| 16:26.14 | kippi | redhat |
| 16:27.26 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 16:27.30 | Juggie | 'yum install unixODBC unixODBC-devel' |
| 16:28.15 | kippi | is that a new one for the lastest version of asterisk |
| 16:28.26 | Juggie | umm no? |
| 16:28.34 | kippi | hmm |
| 16:28.42 | kippi | woo |
| 16:28.43 | Juggie | just making sure you have odbc installed on your box |
| 16:28.54 | Hmmhesays | so cal is where my mind states, but its not my state of mine |
| 16:29.04 | Juggie | that worked? |
| 16:29.15 | kippi | asterisk has started |
| 16:29.27 | jsharp | The altered state of california. |
| 16:29.40 | Juggie | kippi, so installing unixODBC solved the problem? |
| 16:30.07 | Juggie | i'm curious as to why res_odbc would even be compiled if unixodbc wasnt on the system |
| 16:30.23 | kippi | yeah, well asterisk has started, can't make calls but i'll look at the mnow |
| 16:30.50 | Juggie | alright, do you want my email to send payment? :) |
| 16:30.59 | Juggie | i'm 50$ an hour, 3 hour minimum :) |
| 16:31.34 | Hmmhesays | I'm $75 2 hour min |
| 16:32.09 | trelane` | you flat out cant afford me. |
| 16:32.17 | trelane` | I win |
| 16:32.51 | Katty | Hmmhesays: cheap bastard ;) |
| 16:32.54 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 16:32.56 | jsharp | I'm free, but you have to buy me dinner first. |
| 16:33.10 | Juggie | haha |
| 16:33.24 | Katty | hey now! |
| 16:33.32 | Katty | you didn't tell me about expenses up front! |
| 16:33.42 | Juggie | almost every linux question about * can be solved with yum |
| 16:33.49 | Juggie | it wont start, or it wont compile |
| 16:33.52 | Juggie | blah blah |
| 16:33.58 | Juggie | yum is sex :) |
| 16:33.59 | coppice | i'm $10 an hour, but I bill a minimum of 1000 hours, and never take on anything complex |
| 16:34.03 | Qwell | sex? |
| 16:34.09 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 16:34.12 | Juggie | yes |
| 16:34.15 | Juggie | yum is like sex |
| 16:34.28 | Qwell | quick and painful the first time you use it? |
| 16:34.30 | jsharp | yum fix asterisk |
| 16:34.30 | coppice | you get endless new partners? |
| 16:34.34 | Juggie | being the only package manager to correctally support multi arch |
| 16:34.42 | Hmmhesays | lol |
| 16:34.46 | Hmmhesays | you get that tune yet? |
| 16:34.56 | Qwell | Juggie: That's like...the only thing I like about yum |
| 16:35.02 | Qwell | the multiarch... |
| 16:35.05 | Juggie | i used to hate yum too |
| 16:35.06 | Qwell | yum can't even resume like apt :p |
| 16:35.08 | Juggie | because it was slow as shit |
| 16:35.18 | Juggie | but newer versions are much better |
| 16:35.26 | Juggie | or maybe i'm just using faster machines |
| 16:35.34 | [TK]D-Fender | kippi : I already gave you the answer earlier... |
| 16:35.39 | [TK]D-Fender | kippi : if not then just put "noload => res_odbc.so" into modules.conf |
| 16:36.02 | Juggie | why does res_odbc even compile if unixODBC isnt installed? |
| 16:36.04 | Juggie | that seems odd. |
| 16:36.05 | coppice | yum just gets on with it, without much hassle. only big pain is when there is a broken RPM in the repo |
| 16:36.22 | [TK]D-Fender | [11:31] <Hmmhesays> I'm $75 2 hour min <- yeah... and we're sure you'll last 3 minutes tops ;) |
| 16:36.48 | kippi | [TK]D-Fender sorry I didn't see that, now geting chan_zap errors |
| 16:37.12 | Juggie | have fun :) |
| 16:37.38 | [TK]D-Fender | kippi : care to SHOW us the errors and your config files? |
| 16:38.18 | kippi | just trying a reboot and loading modprobe zaptel and modprobe wcte11xp |
| 16:38.37 | Winkie | wcte11xp etc should autoload zaptel 8) |
| 16:39.00 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 16:41.30 | *** join/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m) |
| 16:44.38 | kippi | http://pastebin.ca/45150 is the first error |
| 16:44.59 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:47.03 | _foxfire_ | kippi : are you using udev ? |
| 16:47.49 | kippi | IF i DO zttool it can see the card |
| 16:48.18 | *** join/#asterisk Navman (n=Navman@62.108.206.82) |
| 16:48.36 | kippi | hmm |
| 16:48.40 | kippi | got it working some how |
| 16:48.59 | kippi | runing zttool and clicking loopback |
| 16:50.17 | kippi | i'll try rebooting and see what happends |
| 16:51.35 | brettnem | wheee rebooting |
| 16:51.48 | shido6 | you do not need to reboot |
| 16:53.01 | backblue | reboot? are you running on windows? |
| 16:53.07 | kippi | nope |
| 16:53.37 | backblue | have you updated kernel stuff? |
| 16:53.44 | kippi | yep |
| 16:53.52 | backblue | modules? |
| 16:53.58 | *** join/#asterisk Gertrude (n=gert@chickenbones.bflony.adelphia.net) |
| 16:55.03 | kippi | if I run wcte11xp and then run zttool and hit loop that seems to fix it |
| 16:55.17 | *** join/#asterisk Mw3 (n=mw3@national.t-error.hu) |
| 16:55.21 | kippi | if I run wcte11xp and then run zttool and hit loop that seems to fix it |
| 16:59.50 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:59.55 | *** join/#asterisk nagl (n=nagl@137.208.4.163) |
| 17:02.42 | *** part/#asterisk Gertrude (n=gert@chickenbones.bflony.adelphia.net) |
| 17:03.15 | *** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com) |
| 17:03.47 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 17:03.54 | Seldon1975 | can someone remind me of the url for Polycom binaries; www.freedomphones.com or something? |
| 17:04.47 | tehdely | Seldon1975: ---> pm |
| 17:05.17 | stack_ | I'm looking at building a system that would house 12 users, 6 of which will be on the phone all day. We also will be growing, so the system should handle about double that. What sort of specs should I look for in a system (i.e. Processor, RAM) |
| 17:06.08 | jsharp | A current model P4 with 512MB o' ram should cover it. |
| 17:06.39 | stack_ | jsharp: really?! that's it? I imagined much much more |
| 17:06.57 | Fedoracore6 | hemm i wanna use u all .. i wanna build one data bases name asterisk , connection to databases success but why i try my codeing to put data ,asterisk say cannot insert into databases , when i use databases cdr then asterisk can use the data bases |
| 17:06.58 | stack_ | jsharp: at least on RAM |
| 17:07.21 | jsharp | Nah. Asterisk doesn't use much ram. You'll run out of CPU horsepower before you run out of ram. |
| 17:07.46 | tehdely | and that's a really light load you've described |
| 17:07.57 | stack_ | awesome, thanks guys |
| 17:08.08 | stack_ | I was specing a dual core with 2 GB of ram |
| 17:08.10 | Fedoracore6 | http://pastebin.com/594740 |
| 17:08.26 | Fedoracore6 | plase some budy check that link |
| 17:08.42 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:08.49 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
| 17:08.52 | brettnem | stack_: you could probably do that on a junky box.. really.. unless you are doing some fancy codcin |
| 17:08.54 | brettnem | g |
| 17:08.57 | jsharp | That's a serious machine. You could handle a small telco with that. |
| 17:09.24 | stack_ | wow, I will be doing a lot of queueing and special routing |
| 17:09.53 | jsharp | stack_: You don't start burning CPU cycles until you start transcoding between codecs. |
| 17:10.01 | jsharp | queueing and routing are very minimal impacts. |
| 17:10.12 | stack_ | k... thanks a lot. My boss will be happy :) |
| 17:10.15 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 17:10.58 | eric_hill | Can someone help me with a PRI configuration issue? |
| 17:11.05 | jsharp | Sure. |
| 17:11.24 | eric_hill | I'm attaching an Asterisk box to our corporate phone system. |
| 17:11.36 | stack_ | one more question, is there any benefit to going 64-bit? |
| 17:11.42 | eric_hill | I have a Wildcard TE110p on the asterisk box and a T1 card on the phone system. |
| 17:11.45 | jsharp | No, not really. |
| 17:12.10 | jsharp | So far so good. |
| 17:12.21 | eric_hill | Both configured for PRI q.SIG |
| 17:12.56 | *** part/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 17:12.59 | eric_hill | Talking with kippi for a sec... |
| 17:14.10 | eric_hill | My TE110p has a green light on it. |
| 17:14.30 | eric_hill | pri show span 1 on the asterisk box gives Status: Provisioned, Down, Active |
| 17:14.39 | *** join/#asterisk ToTo (n=ToTo@host214-134.pool872.interbusiness.it) |
| 17:14.42 | eric_hill | The primary D channel is on slot 24. |
| 17:15.01 | eric_hill | If I unplug the cable, the asterisk box sees an alarm. |
| 17:15.12 | eric_hill | So I'm not quite sure why the two boxes aren't talking to each other. |
| 17:15.22 | eric_hill | Thoughts? |
| 17:15.25 | kippi | mine says Status: Provisioned, Up, Active |
| 17:16.04 | eric_hill | Is there any way to find out more details on the connection? Like "D channel ok"? |
| 17:16.15 | jsharp | You can do PRI intense debugging |
| 17:17.32 | *** join/#asterisk tuxinator_linux (n=tuxinato@166.173.10.112) |
| 17:17.47 | jsharp | Do you have signalling set for pri_net or pri_cpe? |
| 17:17.59 | eric_hill | CPE |
| 17:18.06 | eric_hill | The phone system is set to "NETWORK_SIDE" |
| 17:18.29 | jsharp | Which side is providing circuit clocking? |
| 17:18.30 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-133-198-176.prod-infinitum.com.mx) |
| 17:18.39 | gongoputch | How! what a lot of config files. Is there a front end for configuring *? |
| 17:18.46 | eric_hill | I would assume the phone system (i.e. network side). |
| 17:19.03 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 17:20.04 | eric_hill | Intense debugging shows me some info, but I'm not quite sure what I'm looking at. |
| 17:20.11 | jsharp | That's network side at ISDN level. It may not be providing clocking. |
| 17:20.22 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 17:20.46 | jsharp | Paste the debug to pastebin.com |
| 17:21.36 | eric_hill | Pased. |
| 17:21.42 | eric_hill | Er, pasted. |
| 17:21.55 | eric_hill | I get that over and over and over. |
| 17:22.50 | jsharp | Thats asterisk trying to bring the circuit up, but not getting any answer. |
| 17:23.06 | jsharp | in your zaptel.conf, what do you have set for your span line? |
| 17:23.28 | eric_hill | span=1,1,0,esf,b8zs |
| 17:23.29 | *** join/#asterisk apardo (n=apardo@87.218.45.124) |
| 17:24.23 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 17:26.04 | jsharp | Try setting it to span=1,0,0,esf,b8z8. |
| 17:26.13 | jsharp | stop asterisk and rerun ztcfg, then restart asterisk. |
| 17:26.51 | jsharp | That's the first thing I can think of. |
| 17:27.36 | eric_hill | Changing the span to 1,0,0... didn't seem to cure the problem. :( |
| 17:28.33 | eric_hill | One more ambeguity: I have the choice between ERB_ESF_ROBBED_B8ZS and ECB_ESF_CLEAR_B8ZS on the phone system. Which one should I use? |
| 17:28.34 | jsharp | hmmm. |
| 17:28.50 | jsharp | ECB_ESF_CLEAR_B8ZS |
| 17:29.24 | jsharp | That should be equal to b8zs line coding, with extending superframe framing. |
| 17:29.38 | jsharp | extended, rather. |
| 17:30.29 | eric_hill | Switching the framing on the phone system side, I see asterisk show the line go down then come back up, but still not in service. |
| 17:30.43 | jsharp | What kinda phone system? |
| 17:30.53 | eric_hill | Cortelco Millenium |
| 17:31.10 | eric_hill | I'm very good on it - just learning Asterisk though. |
| 17:31.42 | jsharp | You don't see any alarms on the phone switch, do you? |
| 17:32.12 | iGotNoTime | What is everyone's favorite softphone? |
| 17:32.17 | *** join/#asterisk steveaj (n=steve@82-71-15-37.dsl.in-addr.zen.co.uk) |
| 17:32.35 | [TK]D-Fender | iGotNoTime : eyeBeam, but it costs |
| 17:32.35 | jsharp | Did you not have it set to CLEAR_B8ZS earlier? If not, put the timing back to span = 1,1,0,b8zs,esf |
| 17:33.02 | iGotNoTime | [TK]D-Fender, support multiple lines? |
| 17:33.46 | fourcheeze | I'm trying to use the * during voicemail to drop out to an operator. I want to know what the original extension dialled was - however ${EXTEN} is 'a' once the * is pressed |
| 17:33.47 | eric_hill | I had it set to ROBBED earlier, not CLEAR. Changing it now... |
| 17:34.07 | fourcheeze | can I get to that old value? |
| 17:34.10 | *** join/#asterisk jdbecker1968 (n=jbecker@S0106000f1fa407a0.cg.shawcable.net) |
| 17:34.14 | iGotNoTime | that's X-lite? |
| 17:34.46 | eric_hill | Nuts - still down. |
| 17:35.03 | jdbecker1968 | howdy, anyone else having issues compiling zaptel 1.2.4 on latest CentOS kernel (2.6.9-34.EL-i686)? |
| 17:35.34 | backblue | where the hell are the demo sound files located? |
| 17:36.04 | jdbecker1968 | http://pastebin.ca/45155 |
| 17:36.12 | eric_hill | They're in /var/lib/asterisk/sounds |
| 17:37.11 | [TK]D-Fender | iGotNoTime : Yup, 6 |
| 17:37.25 | iGotNoTime | sexy |
| 17:37.26 | backblue | eric_hill: hoo, i miss it! tks. |
| 17:37.43 | [TK]D-Fender | fourcheeze : set a temp variable to hold it. |
| 17:38.00 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 17:38.01 | Hmmhesays | this avenged sevenfold song just rocks |
| 17:38.28 | *** join/#asterisk Assid (n=assid@203.115.64.13) |
| 17:38.47 | Assid | heya |
| 17:38.50 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 17:39.13 | Assid | anyone know a good provider which i can port a toll free number to |
| 17:39.22 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 17:40.07 | backblue | it's there a "welcome" somewhere in the asterisk sound files? |
| 17:40.54 | Qwell[] | like welcome.gsm? |
| 17:40.55 | blkremedy | If I did a make samples, how do I un make samples? anyone |
| 17:41.11 | Qwell[] | blkremedy: why would you remove them? |
| 17:41.36 | [TK]D-Fender | blkremedy : restore from a backup.... your old configs are TOAST. |
| 17:41.47 | Qwell[] | only if you run `make samples` twice |
| 17:41.55 | Qwell[] | They're still there as .old or something |
| 17:41.56 | Hmmhesays | indeed |
| 17:41.58 | blkremedy | I did a make samples and now all of my original files are ending with .old |
| 17:42.04 | Hmmhesays | your old configs are now *.old |
| 17:42.17 | Hmmhesays | but don't try that again |
| 17:42.32 | blkremedy | do I have to go back and edit each file? |
| 17:42.40 | jdbecker1968 | latest trunk zaptel also won't compile on latest CentOS kernel |
| 17:42.47 | Hmmhesays | or write a script to do it |
| 17:42.47 | jdbecker1968 | you heard it hear first! |
| 17:43.14 | Hmmhesays | I should go take that dead hooker out of my trunk |
| 17:43.23 | Qwell[] | Hmmhesays: why? |
| 17:43.28 | backblue | Qwell[]: its there a welcome.gsm? |
| 17:43.36 | Qwell[] | backblue: sure |
| 17:43.39 | backblue | where? |
| 17:43.49 | Qwell[] | asterisk-sounds maybe |
| 17:43.52 | Hmmhesays | Qwell[], its nice out she's going to start to stink |
| 17:44.01 | jsharp | Hmmhesays: Leave her back there for a few more weeks. She'll leak out eventually. |
| 17:44.01 | Assid | heya tkd , qwell |
| 17:44.04 | backblue | Qwell[]: tjs |
| 17:44.07 | backblue | tks |
| 17:44.24 | Hmmhesays | true |
| 17:44.41 | kardecallan | Where material meeting on STUN server? I have firewall in my net and my server asterisk is behind of it, and when I receive a external call I do not obtain to have audio. |
| 17:44.44 | *** part/#asterisk jdbecker1968 (n=jbecker@S0106000f1fa407a0.cg.shawcable.net) |
| 17:45.07 | Assid | does anyone know a good provider which i can port a toll free number? |
| 17:45.14 | Qwell[] | Assid: asterlink |
| 17:45.16 | Hmmhesays | you set your localnet and externip |
| 17:45.57 | kardecallan | excuse! I'm Brazilian. |
| 17:46.08 | kardecallan | My english is so-so |
| 17:46.23 | Hmmhesays | kardecallan: set your localnet and externip in sip.conf |
| 17:46.34 | kardecallan | no |
| 17:46.39 | Hmmhesays | and send me pictures of those pretty brazilian women |
| 17:46.43 | kardecallan | I active nat for yes |
| 17:46.47 | Hmmhesays | like the ones I see on tv |
| 17:47.43 | kardecallan | ehehe, it's true |
| 17:47.54 | kardecallan | women brazilian is very beautiful |
| 17:47.57 | *** join/#asterisk nagl (n=nagl@137.208.4.184) |
| 17:48.06 | *** join/#asterisk arosen (n=arosen@modemcable229.135-82-70.mc.videotron.ca) |
| 17:48.15 | Hmmhesays | women from north dakota are corn feed |
| 17:48.27 | *** part/#asterisk arosen (n=arosen@modemcable229.135-82-70.mc.videotron.ca) |
| 17:48.29 | Hmmhesays | there's a few hotties though |
| 17:48.34 | Qwell[] | that was random |
| 17:48.57 | *** join/#asterisk sloeber (n=mirc@157-180.245.81.adsl.skynet.be) |
| 17:49.03 | Hmmhesays | I have a pretty wicked hangover, my brain is pretty random today |
| 17:49.38 | sloeber | hehe... got the problem also |
| 17:50.10 | sloeber | is anyone using h323 with asterisk? |
| 17:52.46 | *** join/#asterisk RoyK (n=roy@ti211310a080-16128.bb.online.no) |
| 17:54.13 | kardecallan | Hmmhesays: thanks, I need to install the server stund? |
| 17:55.12 | kardecallan | I am using the sip protocol |
| 17:55.21 | Hmmhesays | i think you need to set the settings I told you in sip.conf |
| 17:56.09 | kardecallan | localnet and externip? |
| 17:56.49 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 17:58.05 | sloeber | sip proto is working fine... just as iax... |
| 17:59.24 | RoyK | methinks this memory leak with asterisk is wierd |
| 17:59.37 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.16.10.telnor.net) |
| 17:59.40 | kardecallan | ip address lan stund server is 10.75.2.30 and my ip address wan (Firewall) is 201.45.22.140, I need set localnet=10.75.2.30 and externip=201.45.22.140. It is this? |
| 17:59.53 | RoyK | er |
| 17:59.56 | RoyK | LAN stun server? |
| 18:00.38 | backblue | anyone knows if there is a .wav files from asterisk-sounds? |
| 18:00.42 | RoyK | Isn't the whole reason with STUN to be on an official address??? |
| 18:00.54 | RoyK | backblue: just do a 'sox file.gsm file.wav' :) |
| 18:00.58 | RoyK | sox is nice |
| 18:01.12 | backblue | yes, but .gsm to wav i will lost quality. |
| 18:01.17 | RoyK | no |
| 18:01.19 | backblue | lose |
| 18:01.20 | kardecallan | sorry! I have difficulty with the writing in English. |
| 18:01.20 | RoyK | it will not |
| 18:01.24 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 18:01.25 | backblue | RoyK: why not? |
| 18:01.43 | RoyK | because it is only decoded from gsm and saved as raw/wav |
| 18:02.03 | RoyK | the quality was lost when it was compressed to gsm |
| 18:02.27 | RoyK | you may be able to add some warmth lost with gsm by using audacity or something |
| 18:02.31 | coppice | RoyK: but your wave file might contain LPC10 |
| 18:02.56 | RoyK | coppice: ? |
| 18:03.01 | backblue | RoyK: wav its better quality, i want wav to -> alaw or something, i dont want gsm -> wav so wav will only have gsm quality, never more. |
| 18:03.07 | kardecallan | RoyK: Address IP of my net lan is 10.75.2.30 |
| 18:03.37 | coppice | All you can do by playing with the decoded data from GSM is make things worse. the GSM decoder is reasonably well optimised to decode the best it can |
| 18:03.39 | RoyK | backblue: true, but once the quality is lost, it's lost |
| 18:03.56 | coppice | wave is a file format. almost anything might be in the wave file, including GSM |
| 18:04.32 | RoyK | sox docs says |
| 18:04.33 | RoyK | <PROTECTED> |
| 18:04.36 | backblue | RoyK: that's why i'm asking for the wav files. |
| 18:04.46 | RoyK | backblue: then contact digium :P |
| 18:04.56 | RoyK | you might have to pay for them |
| 18:05.01 | RoyK | s/might/may/ |
| 18:05.07 | Qwell[] | pay for what? |
| 18:05.24 | RoyK | asterisk sounds in wav format... i don't know |
| 18:05.32 | Qwell[] | didn't Kristian do asterisk-sounds too? |
| 18:06.03 | sloeber | you can download better sound files on : http://www.astlinux.org/index.php?option=com_docman&task=cat_view&gid=36&Itemid=36 |
| 18:07.08 | *** part/#asterisk steveaj (n=steve@82-71-15-37.dsl.in-addr.zen.co.uk) |
| 18:07.51 | freat | sloeber: with 1.2 having "native" music on hold... you know if anyone has provided the music in this native format? |
| 18:08.05 | backblue | sloeber: i dont want native sounds, i want extra sounds. |
| 18:08.08 | freat | not native... but the raw format whatever that is... |
| 18:08.38 | sloeber | freat: i don't know sorry |
| 18:08.44 | freat | sloeber: thx |
| 18:09.12 | sloeber | backblue: u can use 'seak' enguines on the internet... or do the talking yourself... we are doing it to for specific languages |
| 18:09.46 | sloeber|eat | seak = speak sorry |
| 18:09.49 | backblue | sloeber|eat: we are doing that too, but i dont want. |
| 18:09.57 | sloeber|eat | hehe lazy ;) |
| 18:10.05 | sloeber|eat | have t eat |
| 18:10.07 | sloeber|eat | sorry |
| 18:10.52 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 18:11.02 | freat | sloeber|eat: @home has the calm-river etc converted to wav format |
| 18:14.54 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 18:16.01 | kink0 | a question, when Asterisk get i.e. an ISDN CAUSE 34, then generates a SIP 503, right ? but just send to the peer the 503 and never the g931 causes ? right ? |
| 18:16.15 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
| 18:16.26 | *** join/#asterisk PMantis (n=pmantis@66.251.89.34) |
| 18:16.32 | Qwell[] | oh file... |
| 18:16.40 | Qwell[] | nevermind, no singing today |
| 18:17.04 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 18:17.05 | PMantis | Is there a link on voip-info.org to correctly size a box to support * with a # of simultaneous calls, etc? |
| 18:17.15 | file | Qwell[]: hiiiiiiii |
| 18:17.42 | asteriskmonkey | PMantis: how many uses do you need |
| 18:17.45 | Qwell[] | PMantis: "correctly" no, but there are "guidelines". do a google search for "asterisk dimensioning site:voip-info.org" |
| 18:17.55 | kink0 | PMantis, that are subject to codec used also. |
| 18:18.08 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 18:18.32 | *** join/#asterisk fabsoft (n=fabsoft@host200-85.pool8254.interbusiness.it) |
| 18:19.04 | PMantis | Right now, 10 users 23 channels inbound. Mostly hard wired inbound, low omplression SIP to a channel gateway |
| 18:19.14 | Qwell[] | PMantis: easy |
| 18:19.25 | Qwell[] | ghz at best |
| 18:19.46 | [TK]D-Fender | PMantis : Subject to weather, APR financing approval, transportation and prep, insurances extra, please see your dealer for more details.... |
| 18:19.53 | PMantis | lol |
| 18:19.54 | Qwell[] | oac |
| 18:20.11 | [TK]D-Fender | OAC... never forget that... |
| 18:20.12 | Seldon1975 | all your base are belong to me |
| 18:20.20 | [TK]D-Fender | Make your time!!! |
| 18:20.49 | PMantis | I'm recommending a Dual Xeon 2 Ghz w/ 2Gb RAM.. room for expansion, huh? :-) |
| 18:21.11 | Qwell[] | PMantis: yeah...easily :p |
| 18:21.16 | PMantis | Need queuing, some AGI scripting, MOH |
| 18:22.09 | [TK]D-Fender | 2GHZ? buying used hardware? |
| 18:22.27 | PMantis | wellng, bigger is better, of course |
| 18:22.44 | [TK]D-Fender | PMantis : How is it you're going to have more channels than users? |
| 18:22.58 | PMantis | Inbound call center with queuing |
| 18:23.12 | [TK]D-Fender | PMantis : figured.... |
| 18:23.20 | PMantis | Allow for conferencing, faxing, etc |
| 18:23.34 | [TK]D-Fender | PMantis : Yeah, jsut get a decent xeon server or something..... |
| 18:23.34 | PMantis | Right now, only 11 channels active |
| 18:23.46 | PMantis | ok, cool |
| 18:24.00 | [TK]D-Fender | nothing special at all.. |
| 18:24.13 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 18:24.19 | PMantis | I'll still recommend something strong... for longevity |
| 18:24.25 | [TK]D-Fender | Since its internal, use G711 for your phones and the load will be petty. |
| 18:24.39 | diclophis | do they make 8 pri span cards? |
| 18:24.49 | Qwell[] | diclophis: two cards |
| 18:24.59 | [TK]D-Fender | PMantis : Not too strong, its diminishing returns. Remember that Astersik scales better with multiple servers than it does with simply BIGGER servers. |
| 18:25.02 | PMantis | [TK]D-Fender, They're plannig to use a 4 port T1 card, just in case of expansion |
| 18:25.05 | diclophis | is that ok to have two cards on one pci bus? |
| 18:25.15 | Qwell[] | diclophis: it isn't recommended...at all |
| 18:25.28 | diclophis | interesting |
| 18:25.33 | [TK]D-Fender | diclophis : can work fine depending |
| 18:25.40 | PMantis | [TK]D-Fender, Ahh, that's true. Internal private DUNDi, with multiple servers. hmmm |
| 18:25.41 | *** join/#asterisk AlexCeli (n=Alex@200.89.15.171) |
| 18:25.49 | kink0 | a question, when Asterisk get i.e. an ISDN CAUSE 34, then generates a SIP 503, right ? but just send to the peer the 503 and never the g931 causes ? right ? |
| 18:26.12 | [TK]D-Fender | PMantis : Your company would have to more than double before you'd even give second thoughts to upgrading anything... |
| 18:27.02 | blkremedy | it was a lot of work but, I have removed all of the *.old files and things are back in order. |
| 18:27.19 | [av]bani | ... |
| 18:27.23 | Qwell[] | .. |
| 18:28.25 | diclophis | [TK]D-Fender depending on what? |
| 18:28.26 | PMantis | [TK]D-Fender, Ok, so a dual isn't necessary at all? |
| 18:28.57 | heison | [TK]D-Fender: i'm still battling with SIP 403 for bidden messages, could you help? |
| 18:29.05 | *** join/#asterisk joeqread (n=joe@207.40.150.15) |
| 18:29.13 | joeqread | hey, anyone awake? |
| 18:29.22 | Qwell[] | zzz |
| 18:29.39 | hardwire | faker |
| 18:29.45 | hardwire | say is asterisk 1.2.9 out yet? |
| 18:29.50 | Qwell[] | soon |
| 18:30.01 | hardwire | I was really hoping to not have to update to 1.5.6 until it was really stable |
| 18:30.14 | joeqread | you guys know if asterisk has any sort of voice detection available to (E)AGI scripts? |
| 18:30.25 | hardwire | I don't think I was one of the guys that was banging on the walls for more asterisk releases :) |
| 18:30.54 | joeqread | not speech recognition, just basic vox |
| 18:31.00 | blkremedy | does anyone have a work around for a tdm400p FXO module that does not recognize hangup |
| 18:31.05 | diclophis | if its like, 200% better to just have two machines, with 1 card each, than it is to try and hack 2 cards into one machine that doesnt seem unreasonable |
| 18:31.23 | hardwire | its 200% more expensive |
| 18:31.24 | Qwell[] | blkremedy: tried changing the signalling? I know some do better than others |
| 18:31.27 | hardwire | so thats 200% more better |
| 18:31.48 | Qwell[] | such as ls vs ks |
| 18:32.11 | diclophis | mm |
| 18:32.35 | Qwell[] | diclophis: All I can say is "try it", and if it doesn't work...you know what to do |
| 18:33.05 | Qwell[] | but really, do you want 100% of your 190+ calls going through one box? |
| 18:33.12 | Qwell[] | erm |
| 18:33.22 | Qwell[] | double that |
| 18:33.23 | PMantis | Thanks for all the info guys!! |
| 18:33.25 | PMantis | GTG |
| 18:33.26 | jaiger | diclophis, if you can afford 2 servers, then for a business I'd recommend you go that way so you have decent redundancy |
| 18:33.27 | Qwell[] | no, wait |
| 18:33.29 | Qwell[] | meh :p |
| 18:34.13 | joeqread | anyone? even a decent URL on the matter would do... |
| 18:35.03 | joeqread | also wondering if anyone has set up anything so that each FXO port can be listened to real-time through a shoutcast server, anyone set anything like that up? |
| 18:35.21 | asteriskmonkey | blkremedy: try softhangup |
| 18:35.54 | kink0 | my peer appears to have problems managing SIP 503 ( congestion ) messages, he ask me if we send a ISDN CAUSE n , but as I see , SIP never sends the q931 causes, right ? |
| 18:36.40 | salviadud | more better? |
| 18:36.45 | salviadud | just better |
| 18:36.52 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 18:36.52 | *** join/#asterisk [vmwarez]dotcom (n=jjones@216.147.224.254) |
| 18:38.27 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 18:38.51 | *** join/#asterisk knobo (n=knobo@liberalitas.freecode.no) |
| 18:43.48 | UdontKnow | oi |
| 18:43.52 | UdontKnow | WARNING[25423]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x775e30 (len 395) to 213.91.9.213:5060 returned -1: Bad file descriptor |
| 18:43.54 | UdontKnow | whats that? |
| 18:44.06 | tzanger | UdontKnow: looks like hte call went away just as it went to transmit |
| 18:44.24 | diclophis | thanks |
| 18:44.25 | UdontKnow | when I boot asterisk it gives me that |
| 18:44.40 | |omni| | yay.. TAPI dialing |
| 18:44.44 | UdontKnow | I have no firewall rules active, and its a plain blank asterisk config |
| 18:44.55 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 18:45.58 | UdontKnow | I just added 2 things to asterisk: 1) softphone, 2) wengo service |
| 18:46.28 | austinnichols101 | Enth: did you get your nat issue solved? |
| 18:46.32 | joeqread | is that IP to your softphone? |
| 18:46.39 | UdontKnow | joeqread: no, to wengo |
| 18:47.12 | UdontKnow | my softphone runs on 172.16/12 ;) |
| 18:47.27 | joeqread | hmm.. over my head then, but Bad file descriptor usually means it's trying to read/write to a file/socket that doesn't exist yet, in this case it may not be connected to wengo yet and trying to initiate a stream... check timeouts maybe? |
| 18:47.56 | hardwire | joeqread: talkdetect.so ? |
| 18:48.19 | joeqread | no idea, sorry.. |
| 18:48.33 | hardwire | heh |
| 18:48.36 | joeqread | oh, hardwire, you talking about my voice detection? |
| 18:48.43 | hardwire | in the agi do a background w/ talk detect |
| 18:48.51 | hardwire | tada.. the function you wanted is now there |
| 18:48.58 | hardwire | yeh |
| 18:49.02 | blkremedy | •asteriskmonkey• what's softhangup? |
| 18:49.11 | hardwire | BackgroundDetect() I thought |
| 18:49.16 | joeqread | is that part of the asterisk distro or is it 3rd-party? |
| 18:49.48 | UdontKnow | joeqread: I found it |
| 18:49.51 | caio1982 | UdontKnow: dont you have to resolve wengo' server name? |
| 18:49.55 | UdontKnow | its the register => line |
| 18:50.04 | joeqread | what was it? |
| 18:50.10 | UdontKnow | didnt solve it |
| 18:50.17 | UdontKnow | but commenting register => |
| 18:50.21 | UdontKnow | stops the error |
| 18:50.30 | UdontKnow | caio1982: http://voip-info.org/wiki/view/Asterisk+settings+for+Wengo |
| 18:50.48 | joeqread | but then you're not connecting to wengo, right? |
| 18:50.50 | UdontKnow | I made wengo work with kphone first |
| 18:50.58 | caio1982 | googling for your error msg pointed this out, ok then |
| 18:51.02 | UdontKnow | joeqread: well, I am not registering with it to receive calls, yes |
| 18:51.39 | UdontKnow | caio1982: url? |
| 18:52.19 | caio1982 | http://www.google.com/search?hl=en&lr=&q=sip_xmit+bad+file+descriptor&btnG=Search |
| 18:53.21 | hardwire | joeqread: 1.2.x and up definatly.. for that function |
| 18:53.45 | joeqread | k, thx hardwire, found docs on that, appreciate the help |
| 18:53.55 | joeqread | any idea on streaming channels to a shoutcast server? |
| 18:54.22 | joeqread | seems somthing could be rigged up polling the monitor every x seconds, looking for new sessions, then listening in and streaming the output |
| 18:54.36 | joeqread | but not sure where to go about getting audio out of the monitor |
| 18:55.17 | *** join/#asterisk tuxinator_linux (n=tuxinato@adsl-69-235-144-138.dsl.irvnca.pacbell.net) |
| 18:55.48 | *** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe) |
| 18:56.38 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:57.53 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 18:58.36 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 18:58.36 | *** join/#asterisk revar (i=[U2FsdGV@216.127.82.54) |
| 18:58.50 | revar | n |
| 18:59.02 | revar | hi all |
| 18:59.09 | revar | any voicetronix users out there? |
| 18:59.10 | caio1982 | UdontKnow: then? |
| 18:59.48 | revar | or anyody who might be able to help me get started with a non zap board and asterisk@home? |
| 19:00.19 | joeqread | revar: what's the prob? |
| 19:00.42 | joeqread | I got voicetronix with ctserver going, might be able to help if it's just a basic driver issue |
| 19:00.54 | revar | vpb is fine |
| 19:01.05 | revar | I have no idea how to get it working with asterisk@home |
| 19:01.25 | revar | ie configure asterisk@home to use it as a trunk and incoming lines |
| 19:01.49 | revar | not sure if I have to abandon AMP or not and go to config files |
| 19:01.57 | UdontKnow | caio1982: nah, it needed the bind address |
| 19:02.10 | austinnichols101 | revar: if you get it going so that asterisk sees it then all AAH is doing is layering AMP on top |
| 19:02.24 | joeqread | OpenLine 4 card? |
| 19:02.28 | revar | yes |
| 19:02.39 | revar | I have my vpb.conf setup properly |
| 19:02.55 | joeqread | I would set it up as best you can through the web interface, then check the config files closely and see if anything's messed up |
| 19:03.29 | austinnichols101 | how does that board appear from the zaptel / zapata level - as a trunk group? |
| 19:03.53 | UdontKnow | Incoming call: Got SIP response 405 "invalid method" back from 213.91.9.213 |
| 19:03.53 | revar | doesnt show up at all automatically |
| 19:03.55 | UdontKnow | hmmmm |
| 19:04.12 | joeqread | you check this yet? http://www.voip-info.org/wiki/view/Voicetronix |
| 19:04.39 | revar | yeah, but it wan't very helpful as to exactly how to get it talking to asterisk |
| 19:04.51 | revar | just the basic config file setup |
| 19:05.12 | austinnichols101 | All you should need from the AAH/AMP side is to add a trunk from the GUI and that group is going to have an 'zap identifier (trunk name) of 'g0' or whatever group number it's listed as in zaptel/zapata |
| 19:05.19 | joeqread | k, so you do an lsmod and you see the driver running, you already got vpb.conf set up, but you can't call in or do anything with it? |
| 19:05.52 | revar | exactly |
| 19:06.03 | stack_ | Anyone have a problem faxing through an ATA box? |
| 19:06.23 | Qwell[] | stack_: daily |
| 19:06.26 | revar | ive seen some examples where people have the vpb in thier dial plan |
| 19:06.35 | austinnichols101 | revar: what are your zapata.conf entries for that board? |
| 19:06.42 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 19:06.43 | stack_ | Qwell[], so it doesn't work well at all? |
| 19:06.49 | Qwell[] | not really, no |
| 19:06.56 | revar | like Dial(vpb/1-9/${VBP}XXXXXXXXXX |
| 19:07.10 | Damin | Anyone care to put on their SIP Goggles for a second and tell me why I am an idiot? |
| 19:07.14 | stack_ | well that sucks... |
| 19:07.15 | revar | but I was hoping to not edit all of my extensions and use AMP |
| 19:07.25 | austinnichols101 | revar: no - zapata.conf |
| 19:08.06 | revar | I should be looking at settings in zapata.conf? |
| 19:08.31 | austinnichols101 | I'm just tracing it back from there (probably someone more experienced in here could do it in a better way) |
| 19:08.56 | UdontKnow | hmmmm |
| 19:09.06 | austinnichols101 | specifically need to know what group it's a part of |
| 19:09.14 | austinnichols101 | then all you do is set up the corresponding group in amp |
| 19:09.26 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:09.28 | austinnichols101 | only AMP calls it 'trunk' |
| 19:09.49 | UdontKnow | http://rafb.net/paste/results/A47AD988.html -- ideas? |
| 19:10.11 | austinnichols101 | IOW, you should be able to just go in and create a zap trunk called vpb and then start using it in your dialplan |
| 19:10.27 | austinnichols101 | if that dial string you gave earlier actually works |
| 19:10.34 | Hmmhesays | c |
| 19:10.39 | Qwell[] | d |
| 19:10.44 | octothorpe_ | Damin: happy to call you an idiot anytime (jk) |
| 19:10.50 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 19:11.13 | revar | create a zap trunk via the zapata.conf? |
| 19:11.46 | austinnichols101 | try this - go into AMP and select 'setup/trunks/add a zap trunk |
| 19:11.53 | revar | ok cool |
| 19:11.59 | Qwell[] | it isn't zap though, is it? |
| 19:12.08 | austinnichols101 | the last entry on the page is zap identifier and stick in vpb |
| 19:12.42 | austinnichols101 | that's what I don't know for sure, but I thought all of the physically installed boards end up with the zap driver layer on top of them |
| 19:13.20 | austinnichols101 | leave everything else blank |
| 19:13.39 | revar | ok |
| 19:13.47 | revar | now have ZAP/vpb |
| 19:14.00 | austinnichols101 | then go to setup/outbound routing and create a route with dial pattern 8|. that points to your trunk |
| 19:14.16 | austinnichols101 | then dial 8 and the number and see what happens. |
| 19:14.31 | austinnichols101 | that should put all of the basics you need into the dialplan so that you can troubleshoot from there |
| 19:16.17 | *** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com) |
| 19:16.44 | pifiu | hey everyone |
| 19:16.49 | pifiu | qwell my favorite person here lol |
| 19:16.56 | Qwell[] | shh, I'm sleeping |
| 19:16.59 | pifiu | LOL |
| 19:17.19 | pifiu | let me ask you guys something |
| 19:17.27 | Qwell[] | sure, but it might cost ya |
| 19:17.32 | pifiu | in just general nothing detailed |
| 19:17.41 | pifiu | but could asterisk be setup to call a customer to tell them their order is ready? |
| 19:17.46 | Qwell[] | sure |
| 19:17.56 | Qwell[] | easily |
| 19:17.56 | UdontKnow | <PROTECTED> |
| 19:17.58 | UdontKnow | yay |
| 19:17.58 | pifiu | are there plugins someone has developed for them? |
| 19:17.59 | UdontKnow | :/ |
| 19:18.04 | Qwell[] | no plugins needed |
| 19:18.10 | pifiu | oh yeah how? |
| 19:18.12 | Qwell[] | just a .call file, or the manager interface |
| 19:18.17 | pifiu | just very broadly without going into details though |
| 19:18.26 | pifiu | oh but thats with a@h? |
| 19:18.31 | Qwell[] | no |
| 19:18.39 | pifiu | what is the "manager interface"? |
| 19:18.51 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 19:18.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 19:18.59 | Qwell[] | something to manage * with |
| 19:19.01 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
| 19:19.09 | austinnichols101 | something to search google with :-) |
| 19:19.32 | pifiu | oh how come ive never heard of it? |
| 19:19.36 | pifiu | is ti new? |
| 19:19.37 | Qwell[] | no |
| 19:19.42 | pifiu | wtf lol cant be |
| 19:19.59 | pifiu | oh but its a plugin |
| 19:20.05 | pifiu | its not part of * |
| 19:20.18 | Qwell[] | yes it is |
| 19:20.28 | pifiu | the manager interface or .call? |
| 19:20.31 | blitzrage | both |
| 19:20.31 | Qwell[] | both |
| 19:20.36 | Qwell[] | jynx! |
| 19:20.38 | pifiu | wow i must be ignorant |
| 19:20.39 | pifiu | lol |
| 19:20.40 | blitzrage | :) |
| 19:20.46 | pifiu | let me take a look at that |
| 19:20.52 | Qwell[] | blitzrage: we need to go drinking at von |
| 19:20.52 | pifiu | but its basically a web gui control panel? |
| 19:20.53 | blitzrage | stop using a@h and learn * for real! :) |
| 19:20.59 | blitzrage | Qwell[]: actually, I quit drinking |
| 19:21.00 | pifiu | im not using a@h |
| 19:21.10 | blitzrage | Qwell[]: believe it or not |
| 19:21.14 | pifiu | thats why i asked if that was a a@h plugin or something |
| 19:21.30 | Qwell[] | well then |
| 19:21.34 | pifiu | ok so let me google this |
| 19:21.39 | pifiu | asterisk manager interface |
| 19:22.00 | blitzrage | Qwell[]: and I quit smoking too |
| 19:22.03 | revar | austinnichols - I only am setup for incoming calls, can you help me test that? |
| 19:22.06 | Qwell[] | good... |
| 19:22.09 | Qwell[] | smoking sucks :P |
| 19:22.16 | austinnichols101 | pifu: http://www.voip-info.org/wiki-Asterisk+manager+API |
| 19:22.24 | blitzrage | Qwell[]: yah, luckily I wasn't smoking cigarettes anymore -- I had already quit those a few months ago |
| 19:22.28 | Damin | blitzrage: You quit drinking??????? |
| 19:22.29 | Qwell[] | heh |
| 19:22.32 | Damin | blitzrage: WOW! |
| 19:22.34 | blitzrage | Damin: yah... :) |
| 19:22.34 | pifiu | thanks qwell, let me check it out |
| 19:22.37 | pifiu | so its pretty basic then |
| 19:22.38 | austinnichols101 | revar: can you explain further? |
| 19:22.42 | Qwell[] | Damin: yeah...now who am I gonna go drink with?! |
| 19:22.43 | Damin | blitzrage: Is this the same as when I said I stopped drinking? |
| 19:22.53 | blitzrage | Damin: no -- this is for real :) |
| 19:23.07 | revar | so I've got two lines on my openline 4 attached with analog lines and no extensions |
| 19:23.15 | blitzrage | Damin: I had 2 beers last weekend... other than that, I haven't really drank in like... 3 weeks |
| 19:23.17 | russellb | blitzrage: w00t |
| 19:23.24 | trelane` | has anyone considered fixing the Snom360 SIP Register issue? |
| 19:23.26 | revar | I guess I want to try to get the openline to answer theline |
| 19:23.28 | blitzrage | russellb: today is day 1 for not smoking :) |
| 19:23.43 | russellb | nice man |
| 19:23.45 | revar | when I call one of the two lines attached |
| 19:23.52 | blitzrage | can't wait to get rid of this 'fog' |
| 19:24.06 | austinnichols101 | revar: what happens now? |
| 19:24.14 | revar | no answer |
| 19:24.16 | russellb | blitzrage: you'll be a new man |
| 19:24.22 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:24.29 | blitzrage | russellb: aye -- hopefully I don't go insane :D |
| 19:24.34 | austinnichols101 | connect up to the cli and see what it shows |
| 19:24.40 | Damin | blitzrage: Next, you'll need to get off the heroin! |
| 19:24.43 | Qwell[] | ooo, that kind... |
| 19:24.47 | Qwell[] | yeah, quitting smoking is no fun :p |
| 19:24.50 | blitzrage | Damin: lol |
| 19:25.16 | austinnichols101 | qwell: I quit chewing last october and had that hung over feeling for almost two weeks |
| 19:25.31 | revar | cli the flash panel? |
| 19:25.56 | austinnichols101 | revar: no. on your asterisk box, type asterisk -r and you can see the debug stuff about the calls |
| 19:25.58 | blitzrage | oh yah ... and I shaved my head last night as my "new beginning" |
| 19:26.05 | revar | ah |
| 19:26.06 | revar | ok |
| 19:26.19 | blitzrage | I look wierd with no hair :D |
| 19:26.28 | justnulling2 | any 7960 gurus here? |
| 19:26.33 | Qwell[] | justnulling2: sccp? |
| 19:27.09 | revar | no activity on the cli when I dial in |
| 19:27.31 | revar | an when I look at flash panel trunks is empty |
| 19:27.36 | revar | should it show vpb? |
| 19:28.09 | revar | bbl thanks for your help austin!! |
| 19:28.14 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 19:28.14 | austinnichols101 | this is where I'm lost with that particular board. I've never used one and everything we've done so far assumes that the board is exposed to asterisk via zap |
| 19:28.27 | austinnichols101 | it's ok - thanks for bearing with me as I'm learning myself |
| 19:28.35 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-143-33.dsl.irvnca.pacbell.net) |
| 19:28.48 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
| 19:29.05 | justnulling2 | qwell[]: no idea what it has now trying to flash it for sip but get application error and all bottons are locked |
| 19:29.15 | Qwell[] | fun |
| 19:29.24 | austinnichols101 | revar: were you actually able to use a staqnd-alone dial statement to call? |
| 19:29.48 | austinnichols101 | revar: here it is: http://www.voip-info.org/wiki/view/Asterisk+vpb+channels |
| 19:29.52 | kink0 | my peer appears to have problems managing SIP 503 ( congestion ) messages, he ask me if we send a ISDN CAUSE n , but as I see , SIP never sends the q931 causes, right ? |
| 19:29.57 | MGSsancho | i cant decide if phonesecks is more fun over voip or pstn |
| 19:30.02 | austinnichols101 | so instead of a zap trunk we need to set it up as a custom trunk in AMP |
| 19:30.10 | Qwell[] | MGSsancho: in person |
| 19:30.16 | Qwell[] | or, over voip, with video |
| 19:30.24 | MGSsancho | oh baby dont teawse my iptables hehehehe |
| 19:30.40 | MGSsancho | like convention vs voip over phone of course |
| 19:31.25 | MGSsancho | one time talking to my gf on the phone and it was getting dirty talk. then i said i wanted sip in my zaptel.conf |
| 19:31.33 | *** join/#asterisk razu_ (n=razu@213-35-173-39-dsl.prn.estpak.ee) |
| 19:31.36 | MGSsancho | yeah one way to ruin a moment |
| 19:31.50 | MGSsancho | but i'll stop now cuz ur having a serious convo |
| 19:32.05 | jsharp | Ohyeahbaby....twiddle my bits with your clock. |
| 19:32.15 | Qwell[] | mmm, zap timing |
| 19:32.33 | justnulling2 | qwell[]: i get protocol application invalid, and it is not asking any files on tftp, any ideas how to flash it? |
| 19:32.50 | Qwell[] | justnulling2: You could try doing a factory reset |
| 19:33.03 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
| 19:33.07 | justnulling2 | qwell[]: how do i do that? |
| 19:33.14 | Qwell[] | umm |
| 19:33.26 | Qwell[] | there are instructions on cisco.com - google them up |
| 19:33.35 | MGSsancho | hahaha jsharp |
| 19:34.03 | jsharp | Hold down # as it powers up, then dial 123456789*0# |
| 19:34.11 | jsharp | That'll factory reset it. |
| 19:34.26 | justnulling2 | jsharp: thanks |
| 19:34.32 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:34.47 | Qwell[] | http://www.sellsbrothers.com/fun/exam.gif |
| 19:34.50 | gambolputty | In *, is there a way to get the number of rows of a result set in advance from a mysql select statement? |
| 19:35.07 | justnulling2 | qwell[] google said to go to settings then press 3 then 3 or something but my keyboard is locked so that didn't work let me try jsharp method |
| 19:35.12 | Qwell[] | gambolputty: select count(*) fr.. |
| 19:35.14 | _Sam-- | how would you know the number of rows before you select? |
| 19:35.23 | Qwell[] | jsharp's method is right |
| 19:35.38 | *** join/#asterisk rsaf (n=rsaf@pgw.paskov.cz) |
| 19:36.24 | justnulling2 | cool it is reset :) |
| 19:36.29 | justnulling2 | thanks guys |
| 19:36.43 | *** join/#asterisk veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 19:37.01 | *** join/#asterisk digime (n=digime@ip68-101-196-93.sd.sd.cox.net) |
| 19:37.14 | *** join/#asterisk jsharp (n=jsharp@65.88.255.245) |
| 19:37.18 | jsharp | Dammit. |
| 19:38.02 | *** join/#asterisk innosent (n=jmd@65.169.18.1) |
| 19:39.33 | innosent | having a problem with FreePBX/asterisk when calling extensions or using a queue. Queues are dropping calls after 5 seconds, and dialparties.agi is returning with no extensions to call when you call an extension that is available |
| 19:39.36 | innosent | any help? |
| 19:40.36 | innosent | This is a fresh install on CentOS 4.2 amd64 |
| 19:42.00 | stack_ | how reliable is the iaxmodem with hylafax? Anyone using this? |
| 19:43.09 | rsaf | Hello. i've problem with Asterisk. I'm registered to SIP provider, when someone call from outside, inside phone rings and can speak... but when inside phone hungup it rings again and can speak with caller again ! |
| 19:43.50 | rsaf | the first call dones't hungup correctly |
| 19:47.34 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 19:48.02 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 19:48.06 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 19:48.47 | nettie | Hey guys, anyone know how to start the voicemail without any voice prompt please? I need just the beep.. |
| 19:49.40 | eric_hill | FYI, I got the PRI span set up by using ERB_ESF_ROBBED_B8ZS instead of ECB_ESF_CLEAR_B8ZS and forcing the span to remote-clock. |
| 19:49.44 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 19:49.52 | [TK]D-Fender | nettie : Voicemail(s[box]) |
| 19:50.00 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
| 19:50.21 | nettie | GREAT! thanx a lot TK |
| 19:50.37 | [TK]D-Fender | np |
| 19:50.38 | DaveCanoe | I have a tough problem. call comes into asterisk #1 (on IAX). It answers it, plays a sound, and the Dials another asterisk server IAX-wise. Asterisk #2, 'answers' the call and calls an AGI that does a fancy IVR. AGI calls the get digit function and it seems to never hear the DTMF. I've verified that RFC DTMF is passed from #1 to #2 in the packets (and I even tried inband DTMF, but it didn't work either). The same script works if it is dialed di |
| 19:50.38 | DaveCanoe | rectly. |
| 19:51.17 | DaveCanoe | This behaviour holds true wether notransfer is yes _or_ no. |
| 19:51.30 | Hmmhesays | Chuck Norris has the greatest Poker-Face of all time. He won the 1983 World Series of Poker, despite holding only a Joker, a Get out of Jail Free Monopoloy card, a 2 of clubs, 7 of spades and a green #4 card from the game UNO. |
| 19:51.54 | [TK]D-Fender | ... |
| 19:51.56 | tzanger | Hmmhesays: you're reading the chuck norris page... awesome |
| 19:52.07 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
| 19:52.10 | tzanger | DaveCanoe: what codec? |
| 19:52.14 | DaveCanoe | g.711 |
| 19:52.14 | tzanger | (between #1 and #2) |
| 19:52.16 | tzanger | ok |
| 19:52.18 | DaveCanoe | all g.711 |
| 19:52.30 | nettie | TK, works great :) |
| 19:52.32 | Hmmhesays | i haven't been to it in awhile |
| 19:52.33 | Hmmhesays | lol |
| 19:52.46 | tzanger | DaveCanoe: what version of asterisk? |
| 19:52.47 | Hmmhesays | Chuck Norris and Mr. T walked into a bar. The bar was instantly destroyed, as that level of awesome cannot be contained in one building. |
| 19:53.07 | Qwell[] | link? |
| 19:53.15 | DaveCanoe | full media path is Granstream 2000 (sip) -> Asterisk #0 IAX -> asterisk #1 (answers, plays sound, then DIAL IAX) -> asterisk #2 (answers, calls agi) |
| 19:53.23 | DaveCanoe | 1.2.3 |
| 19:53.24 | Hmmhesays | http://www.chucknorrisfacts.com |
| 19:53.36 | DaveCanoe | actually #4 is 1.2.4 |
| 19:53.36 | Qwell[] | lynx friendly? |
| 19:53.45 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:53.51 | DaveCanoe | #0, rather, can't type |
| 19:53.57 | DaveCanoe | #1 and #2 are 1.2.3 |
| 19:54.22 | DaveCanoe | intended media path replaces grandstream with cisco 5300 |
| 19:55.18 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 19:56.06 | *** join/#asterisk pcm (n=pcm@user-69-73-0-22.knology.net) |
| 19:57.22 | Hmmhesays | LOL, i just found a video of chuck norris reading off those facts |
| 19:58.42 | tzanger | hahaha |
| 19:58.44 | tzanger | Chuck Norris can win a game of Connect Four in only three moves. |
| 19:59.23 | Hmmhesays | Chuck Norris once walked into a burger king and ordered a big mac... He got it |
| 20:00.09 | *** part/#asterisk pcm (n=pcm@user-69-73-0-22.knology.net) |
| 20:00.12 | tzanger | haha |
| 20:00.14 | tzanger | Chuck Norris once bet NASA he could survive re-entry without a spacesuit. On July 19th, 1999, a naked Chuck Norris re-entered the earth's atmosphere, streaking over 14 states and reaching a temperature of 3000 degrees. An embarrassed NASA publically claimed it was a meteor, and still owes him a beer. |
| 20:00.21 | `Sauron | That's old |
| 20:00.24 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp73-adsl-204.ath.forthnet.gr) |
| 20:00.48 | Hmmhesays | lol |
| 20:00.53 | *** join/#asterisk redondos (n=redondos@190.48.46.251) |
| 20:00.54 | tzanger | Chuck Norris can hit you so hard that he can actually alter your DNA. Decades from now your descendants will occasionally clutch their heads and yell "What The Hell was That?" |
| 20:00.55 | jsharp | Someone replaced "Chuck Norris" with "Jack Bauer" for a bunch of those. |
| 20:01.04 | cthompson | Each night, before he goes to bed, the Boogeyman checks under his bed for Chuck Norris |
| 20:01.14 | Hmmhesays | chuck Norris CAN slam a revolving door |
| 20:01.20 | tzanger | yeah I like that one |
| 20:01.25 | Hmmhesays | http://www.i-am-bored.com/bored_link.cfm?link_id=16294 |
| 20:01.31 | redondos | Should the zaptel kernel module create the /dev/zap node? |
| 20:01.44 | cthompson | Chuck Norris' tears can cure cancer. Unfortunately, Chuck Norris never cries. |
| 20:02.52 | Alric | What is with this? I didn't hear any of this for years, and then in the last two weeks everyone I talk to seems to be spouting Chuck Norris facts. |
| 20:03.07 | mog_work | its a mob thing AlexCeli |
| 20:03.10 | mog_work | err Alric |
| 20:03.17 | mog_work | it will fade in a few months |
| 20:03.20 | mog_work | but it is funny |
| 20:03.35 | Alric | Its only funny if you haven't heard the cancer one 1.53x10^235 times :) |
| 20:03.54 | I-MOD | Chuck Norris is all funny all the time :) |
| 20:03.59 | Juggie | you cant make chuck norris dead, you can only make him angry |
| 20:04.20 | cthompson | We're all individuals. |
| 20:05.22 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 20:05.23 | Hmmhesays | According to Einstein's theory of relativity, Chuck Norris can actually roundhouse kick you yesterday. |
| 20:05.41 | *** join/#asterisk Possible (n=Babbel@23.255-136-217.adsl-fix.skynet.be) |
| 20:05.41 | tzanger | I saw the site a couple months back but it wasn't popular yet |
| 20:05.42 | *** part/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 20:06.18 | DaveCanoe | tzanger: any other thoughts? |
| 20:06.34 | tzanger | no... I remember hearing something about RTP DTMF problems but you said you went inband too |
| 20:07.07 | DaveCanoe | was that version specific? |
| 20:07.45 | Damin | All your base are belong to us! |
| 20:07.58 | tzanger | heh that's still funny IMO |
| 20:08.06 | tzanger | DaveCanoe: I don't recall; I never had the issues |
| 20:08.14 | *** join/#asterisk pryk (n=tmalkut@fw.orasoft.net.pl) |
| 20:09.25 | asteriskmonkey | anyone had issue with rx spilling into tx and vice versa? |
| 20:09.54 | hardwire | weird |
| 20:10.02 | hardwire | how do bits spill |
| 20:10.21 | Assid | just curious.. if i have box 1 on IAX and box2 on IAX and box3 on IAX.. and if a person calls box 3 VIA box 2 from box1, will the ip of box3 show on box1 as connecting? and will the CDR on box2 show the actual talk time between box1 and box2 ? |
| 20:10.28 | asteriskmonkey | dont know i run zt monitor and millitwatt or a call to the co test line and it spills |
| 20:10.40 | jsharp | Mine leak out of the ethernet cable all the time. |
| 20:11.35 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
| 20:12.11 | Assid | anyone know? |
| 20:12.20 | jsharp | It should show the actual talk time. |
| 20:13.03 | Assid | but normally in iax.. calls would interconnect box1-box3 |
| 20:13.10 | *** part/#asterisk Yashy (n=yashy@mail.yashy.com) |
| 20:13.10 | Assid | and then box2 would disapppear from the picture |
| 20:13.39 | jsharp | You'd still get call signalling to box 2. |
| 20:13.46 | jsharp | Call setup and teardown messages. |
| 20:14.25 | Assid | okay |
| 20:14.27 | Assid | what about the ip |
| 20:14.35 | Assid | will box1 know box3 is connecting to it? |
| 20:14.41 | *** join/#asterisk darby_t (i=darby_t@dkc125.neoplus.adsl.tpnet.pl) |
| 20:15.10 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 20:15.40 | joeqread | anyone know if you can have the same Meet-me conference room on multiple servers, each server taking a limited number of calls for that room then sharing that room's audio between each server? |
| 20:15.46 | *** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
| 20:16.19 | lathos42 | Howdy |
| 20:16.19 | mog_work | yes you can joeqread |
| 20:17.31 | jsharp | Assid: I don't know. |
| 20:17.46 | joeqread | sweet, can you sum very generally how or maybe send me a URL to instructions? |
| 20:18.04 | joeqread | I don't need specifics, just what I need in order to do it, evaluating a project requirement only at the moment |
| 20:18.11 | *** join/#asterisk SuPrSluG (n=chatzill@pool-71-243-164-226.bflony.east.verizon.net) |
| 20:18.25 | SuPrSluG | hello |
| 20:18.43 | [TK]D-Fender | joeqread : you can have someone call both conferences and confrence them on the phone.... thats a cheap way... |
| 20:19.22 | joeqread | no, need somthing automatic and scalable |
| 20:19.41 | joeqread | don't wanna train someone to bridge two servers together, then chance they'll mess up if I drop in a third |
| 20:19.42 | *** join/#asterisk X-Rob (n=rob@dsl-220-235-230-122.vic.westnet.com.au) |
| 20:19.51 | SuPrSluG | i'm having a voicemail issue. when 7 is pressed to delete the message it moves the message to the Old folder. Any ideas? |
| 20:20.52 | SuPrSluG | why is the message is not deleted? |
| 20:21.02 | jsharp | joeqread: You can have Asterisk autocall the other meet-me room through a .call file that gets automagically created. |
| 20:21.45 | joeqread | jsharp: that's prone to failure too if the other meet-me room is already full |
| 20:23.41 | fgomes | [TK]D-Fender: You sent me an answer hours ago... but I have to urgently connect to a customer site. |
| 20:23.57 | fgomes | I had, I mean. |
| 20:24.20 | jsharp | A full meetme room? |
| 20:24.27 | [TK]D-Fender | God I hate that word ..."automagically" ... a word used mostly by people who still think computers require witchcraft to understand.... |
| 20:24.30 | fgomes | [TK]D-Fender: It was about zapata.conf: Disconnect detectiong on analog is also a PITA... get rid of your signalling line in there and change to fxsks. |
| 20:25.30 | joeqread | jsharp: yes, don't you specify a room limit? |
| 20:25.53 | Hmmhesays | The pie scene in "American Pie" is based on a dare Chuck Norris took when he was younger. However, in Chuck Norris' case, the "pie" was the molten crater of an active volcano. |
| 20:26.43 | [TK]D-Fender | fgomes : ok, so go do it now :) |
| 20:26.52 | jsharp | I never have...but I suppose you could, but if you specify a room limit, I'm sure you could make exceptions. |
| 20:28.06 | kink0 | [TK]D-Fender, can you give me some help ? my peer gets 503 SIP, but they see then 63 ISDN CAUSE, while they with other peer when get 503 then see a 34 ISDN cause |
| 20:28.40 | kink0 | myAsterisk -> SIP 503 -> they see SIP 503, q931 63 |
| 20:28.41 | joeqread | jsharp: how do you make exceptions to room limits? Meetme's config seemed a tad lacking in that respect |
| 20:28.52 | nettie | [TK]D-Fender just got my ivr messages recorded :) do you have any hints on IVR, welcome auto-operator please? if I start from scratch it works but I really like to see what other people do to improve my setup and explore deeply.. any idea pls? thanx |
| 20:28.55 | fgomes | [TK]D-Fender: I was using fxsks before without good results... I'm in Brazil... not sure about telco standards. Can you quickly exapalin |
| 20:28.57 | kink0 | otherpeer -> SIP 503 -> they see SIP 503, 1931 34 |
| 20:29.17 | fgomes | explain |
| 20:30.15 | jsharp | joeqread: Just looking at the meetme stuff, you'd probably end up doing the conference limiting in the dialplan, not in the room configuration...so you could set up a special extension to let you dial into the conference without worrying about whether it was full or not. |
| 20:31.20 | *** join/#asterisk Seldon1975 (n=someone@toronto-HSE-ppp4239807.sympatico.ca) |
| 20:32.13 | heison | [TK]D-Fender: found the problem with my SIP provider... |
| 20:32.26 | [TK]D-Fender | fgomes : You'll have to check what kind of disconnect signalling your area and telco support. I suggest you also look up "disconnect detection" on the WIKI |
| 20:32.34 | [TK]D-Fender | heison : what was it? |
| 20:32.37 | kpettit | anybody point me to a good OpenVPN/Astersik tutorial? |
| 20:32.53 | Hmmhesays | why do you need a tutorial for openvpn |
| 20:33.04 | heison | their "version" of the softphone, secretly append a string to the end of the username |
| 20:33.06 | kpettit | I have some remote phones that i want to go through OpenVPN on a Linksys router to a OpenVPN/Asterisk machine |
| 20:33.29 | jsharp | Set up openvpn first, get packets flowing, then make the asterisk part work. |
| 20:33.42 | *** join/#asterisk apardo (n=apardo@62-14-56-136.inversas.jazztel.es) |
| 20:33.54 | kpettit | the LInksys has 1 IP and I have 5 phones that need to go through that to get to my PBX. I'm sure NAT will screw it up so VPN sounds like the best way to go. But I new to VPN's on linux |
| 20:34.13 | heison | [TK]D-Fender: http://www.voip-info.org/wiki/view/asterisk+settings+HKBN+2b |
| 20:34.46 | kpettit | OpenVPN is already installed on my */Gentoo box , and the OpenVPN client is setup on my Linksys router. It's just doing the configs and such Ineed help with |
| 20:35.59 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-197.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 20:36.54 | *** join/#asterisk X-Rob (n=rob@dsl-220-235-230-122.vic.westnet.com.au) |
| 20:37.25 | asteriskmonkey | got asterisk on my linksys router :) |
| 20:38.10 | cpm | Ewww! |
| 20:38.39 | *** part/#asterisk AlexCeli (n=Alex@200.89.15.171) |
| 20:38.45 | mog_work | heh your a light weight asteriskmonkey i have asterisk running on my dead badger |
| 20:39.21 | fugitivo | on this www.nokia.com/770 |
| 20:39.27 | [TK]D-Fender | asteriskmonkey : You'd be running if you were on a dead badger too! |
| 20:39.49 | cpm | I got some asterisk on my pants, now I have to go wash up |
| 20:41.10 | redondos | I'm trying to configure asterisk@home with an x100p card, using X-lite I connect to it and dial something that matches a route that I've se tup. I always get a "all circuits are busy" message. |
| 20:41.16 | redondos | What might be happening? |
| 20:42.08 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 20:42.24 | [TK]D-Fender | redondos : pastebin the full CLI of your call attempt with "set verbose 10" |
| 20:43.15 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 20:43.32 | asteriskmonkey | yes well i have asterisk running on my lego controller |
| 20:43.39 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 20:43.40 | CunningPike | Is anyone having trouble compiling zaptel on the latest RHEL kernel? |
| 20:43.41 | twisted[asteria] | uh |
| 20:44.07 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 20:44.17 | twisted[asteria] | hmm.. drawing a blank |
| 20:44.21 | mog_work | asteriskmonkey, you call it astleg? |
| 20:44.22 | redondos | [TK]D-Fender: Forgive my stupidit,I'm really new at this. Are you talking about asterisk's output or x-lite's? |
| 20:44.27 | asteriskmonkey | lol |
| 20:44.32 | DaveCanoe | I have a tough problem. call comes into asterisk #1 (on IAX). It answers it, plays a sound, and the Dials another asterisk server IAX-wise. Asterisk #2, 'answers' the call and calls an AGI that does a fancy IVR. AGI calls the get digit function and it seems to never hear the DTMF. I've verified that RFC DTMF is passed from #1 to #2 in the packets (and I even tried inband DTMF, but it didn't work either). The same script works if it is dialed di |
| 20:44.32 | DaveCanoe | rectly. <---- adding more strangeness to this report. Everything works if I send a digit within the first second of audio. If I don't send a digit in the first second, it will never recognise digits. |
| 20:44.43 | asteriskmonkey | i call my asterisk update script getass.sh :D |
| 20:45.06 | twisted[asteria] | and that's why you never get any :) |
| 20:45.17 | twisted[asteria] | DaveCanoe, IAX's dtmf is all out of band |
| 20:45.23 | asteriskmonkey | twisted: you ever heard of rx spilling into tx when doing a call from the test line at a co? |
| 20:45.23 | DaveCanoe | sure. |
| 20:45.25 | DaveCanoe | I know that. |
| 20:45.52 | DaveCanoe | I see it in the packets. the #2 asterisk just doesn't seem to pass it to the script (now ... I discover, unless there is a tone in the 1st second of the call) |
| 20:45.53 | twisted[asteria] | asteriskmonkey, you mean like crosstalk? |
| 20:46.00 | *** join/#asterisk cypromis (n=michael@asterisk.pl) |
| 20:46.31 | twisted[asteria] | DaveCanoe, are you natively transferring the iax call to the 2nd box so that it takes box A out of the loop? |
| 20:47.03 | asteriskmonkey | twisted: same happens with milliwatt just vise versa.. dont know if its cross talk im using ztmonitor -vv mode :P |
| 20:47.16 | fgomes | [TK]D-Fender: yes. I was having this problem on the beginning: line stays connected after the other party drops. Thanks! |
| 20:47.20 | redondos | [TK]D-Fender: Really,m please tell me what output you were expecting. |
| 20:47.34 | twisted[asteria] | asteriskmonkey, analog lines, right? |
| 20:47.36 | justinu | woohoo... new laptop on it's way! |
| 20:47.39 | DaveCanoe | twisted: yes |
| 20:47.47 | twisted[asteria] | sounds like crosstalk to me |
| 20:47.47 | asteriskmonkey | twisted : sorry no a pri |
| 20:47.52 | DaveCanoe | well... I've tried both notransfer=yes and notransfer=no |
| 20:47.53 | twisted[asteria] | oh |
| 20:47.55 | asteriskmonkey | PRI on a te406 |
| 20:48.14 | twisted[asteria] | DaveCanoe, try it without allowing the reinvited stream |
| 20:48.25 | twisted[asteria] | DaveCanoe, well, not reinvited, but native transfer |
| 20:48.37 | Lino` | lol |
| 20:48.37 | twisted[asteria] | DaveCanoe, the time in whcih the dtmf will pass is about the amount of time it takes to hand off the call |
| 20:48.43 | *** join/#asterisk X-Rob (n=rob@dsl-220-235-230-122.vic.westnet.com.au) |
| 20:48.53 | Lino` | is there any possibility to use nortel meridian phones with asterisk? |
| 20:49.08 | [TK]D-Fender | redondos : pastebin the * CLI output of an attempted call. |
| 20:49.09 | asteriskmonkey | twisted[asteria]: can you get cross talk on a pri? |
| 20:49.12 | redondos | [TK]D-Fender: Embarrassing... here's the output: http://pastebin.com/595135 |
| 20:49.33 | octothorpe | Lino': I've heard of it |
| 20:49.38 | twisted[asteria] | asteriskmonkey, no, but if you're converting it to analog at any point, it could be getting it |
| 20:49.40 | Alric | Lino`: Seems like I heard someone saying something about nortel meridian integration, a long, long time ago. I think it was a personal project though. |
| 20:49.53 | Lino` | ok that must be hard. i don't want it then. |
| 20:49.54 | Lino` | :D |
| 20:49.59 | [TK]D-Fender | redondos : Line # 43 is BAD! You defined your trunk improperly. |
| 20:50.05 | jsharp | There's a gateway that lets you convert 24 Meridian phones to SIP, but its $3000. |
| 20:50.08 | [TK]D-Fender | oops, #42 |
| 20:50.09 | octothorpe | lino: check out google |
| 20:50.13 | octothorpe | ~google |
| 20:50.15 | jbot | it has been said that google is a search engine found at http://www.google.com/ |
| 20:50.21 | twisted[asteria] | jsharp, got the link to that? i've had a customer looking for one of those |
| 20:50.26 | [TK]D-Fender | # |
| 20:50.26 | [TK]D-Fender | <PROTECTED> |
| 20:50.29 | Lino` | yeah |
| 20:50.31 | jsharp | http://www.voiphardware.com/shop/item.asp?itemid=438 |
| 20:50.32 | Lino` | i know google |
| 20:50.35 | Lino` | i already googled |
| 20:50.38 | asteriskmonkey | twisted[asteria]: co (less than 100ft away) => pri => te406 => asterisk => sip/iax devices i dont see anywhere in that path i could be getting crosstalk |
| 20:50.48 | Lino` | yeah i know that stuff |
| 20:50.48 | redondos | [TK]D-Fender: What could I have done wrong? |
| 20:50.50 | Lino` | i don't like it |
| 20:50.58 | [TK]D-Fender | jsharp : its $2332 at atacomm |
| 20:51.04 | twisted[asteria] | jsharp, many thanks... i couldn't remember what it was called, i'd seen it before, and told customer about it, but forgot what it was.. |
| 20:51.10 | DaveCanoe | twisted: now it never accepts a digit |
| 20:51.24 | octothorpe | lino': not to easy of a project it seems |
| 20:51.30 | twisted[asteria] | asteriskmonkey, hmm... yeah, that's strange. |
| 20:51.31 | Lino` | :D |
| 20:51.36 | Lino` | yeah thats why i dont like it |
| 20:51.38 | redondos | [TK]D-Fender: What's wrong about that line you pasted? |
| 20:51.40 | [TK]D-Fender | redondos : You are the one who typed in "Telephonica" somewhere... find out where... is that a VoIP provider? Because it looks like you tried setting it up as TDM. |
| 20:51.41 | twisted[asteria] | asteriskmonkey, all 4 ports exhibit this behavior? |
| 20:51.48 | Lino` | i'll rather recommend buying a stack of new cisco sccp phones or something like that |
| 20:51.57 | asteriskmonkey | twisted[asteria]: yes ive tried a te110p aswell |
| 20:52.07 | DaveCanoe | Seems like this is easy to replicate: asterisk #1 "answers" and plays a sound calls asterisk #2 and plays a sound. I'm not even using the ivr. It doesn't go to the '1' exention even though I'm pressing '1' |
| 20:52.12 | Alric | Perhaps this weekend... |
| 20:52.15 | twisted[asteria] | asteriskmonkey, if you've tried multiple hardware, i'd be willing to bet the CO is doing something funky... |
| 20:52.32 | redondos | [TK]D-Fender: BTW, this isn't VoIP, I'm trying to use an FX0 card, so I can I can use POTS. |
| 20:52.46 | Lino` | sccp is good |
| 20:52.49 | *** join/#asterisk fu3 (n=kaa@234-200-29-134.hcc.mnscu.edu) |
| 20:52.51 | Lino` | :D |
| 20:52.52 | fu3 | hello |
| 20:52.54 | twisted[asteria] | DaveCanoe, you're using Background() to play that sound outside the ivr, right? |
| 20:53.04 | octothorpe | Alric: sccp is not too hard |
| 20:53.12 | [TK]D-Fender | redondos : Well go find out where you put "Telephonica" INTO amp, AND THATS THE SECTION YOU NEED TO FIX. |
| 20:53.14 | twisted[asteria] | sccp is fairly easy |
| 20:53.24 | asteriskmonkey | twisted[asteria]: ive had the co test for echo and they found nothing they said is there something else i should ask them to check? also once my rx/tx are set to 14500 (not gains but measurements dtmf stops working properly) |
| 20:53.29 | Qwell[] | sccp is fun |
| 20:53.36 | redondos | [TK]D-Fender: I know where I put it, but what's wrong about it? |
| 20:53.37 | [av]bani | sccp kicks your dog |
| 20:53.39 | octothorpe | ~sccp |
| 20:53.40 | jbot | hmm... sccp is Proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. Also supported by some other vendors. Also Signaling Connection Control Part (SCCP), a routing protocol in SS7 protocol suite in layer 4, provides end-to-end routing for TCAP messages to their proper database. |
| 20:54.00 | [TK]D-Fender | redondos thats not a valid way to name a channel interface. |
| 20:54.06 | octothorpe | Alric: http://chan-sccp.berlios.de/ |
| 20:54.25 | redondos | [TK]D-Fender: Oh, I see. |
| 20:54.28 | *** join/#asterisk cypromis (n=michael@asterisk.pl) |
| 20:54.32 | redondos | [TK]D-Fender: It should be numeric, shouldn't it? |
| 20:54.35 | [av]bani | sccp will abduct your children and put them in cages for display in a traveling circuis |
| 20:54.35 | DaveCanoe | background, yes |
| 20:54.38 | [av]bani | circus |
| 20:54.39 | twisted[asteria] | asteriskmonkey, i'm all shrugs at this point.. you have strangeness. |
| 20:54.42 | twisted[asteria] | DaveCanoe, k, just checking ;) |
| 20:54.54 | redondos | [TK]D-Fender: All right, it worked. Thank sa bunch. |
| 20:54.56 | Alric | Its not a lack of the chan_sccp channel, its a lack of manhours :) |
| 20:55.01 | twisted[asteria] | crap |
| 20:55.14 | twisted[asteria] | good luck guys |
| 20:55.48 | DaveCanoe | I've developed about 4000 lines of python for a complex dialaround application for a client. This is the crucial payment bit. The end problem is to interrupt the call and send it to the ivr. The ivr is running on a different server because asterisk completely refuses to play a sound after a call has hit it's timout limit. |
| 20:55.52 | [TK]D-Fender | redondos : What # were you trying to dial for your test? |
| 20:55.59 | octothorpe | alric: manhours? how many phones are we talking about? |
| 20:56.01 | DaveCanoe | ... but it will dial another thing. |
| 20:56.09 | Alric | Its testing, not using. |
| 20:56.21 | octothorpe | alric: I got sccp up it about 15 minutes |
| 20:56.49 | DaveCanoe | so I have it dail the backup asterisk box (right now) and send the call to it. This works --- sound plays, but the DTMF is screwed somehow --- which is curious because I can see the DTMF going across in the IAX packets. |
| 20:58.30 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 20:59.58 | prongils | anyone with experience getting dial-up credit card processing machines working on VOIP using Sipura 2100 ATAs? |
| 21:00.40 | jsharp | Modems + voip are hit or miss. |
| 21:00.53 | prongils | just like fax then :( |
| 21:00.58 | prongils | well T.38 works nice |
| 21:01.04 | jsharp | fax == modem |
| 21:01.27 | stack_ | I'm attempting to use 'iaxmodem', but I get "Rejected connect attempt from 127.0.0.1, requested/capability 0x40/0x4c incompatible with our capability 0xff03." Anyone know what this means? |
| 21:01.40 | prongils | hmmm maybe I'll play with the settings on the Sipura 2100 then :( |
| 21:02.34 | jsharp | Try different codecs. |
| 21:02.40 | jsharp | That's what will eat you. |
| 21:02.45 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 21:02.45 | *** join/#asterisk cypromis (n=michael@asterisk.pl) |
| 21:02.48 | redondos | [TK]D-Fender: The number was 110. |
| 21:02.48 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 21:02.53 | redondos | [TK]D-Fender: It works now, so thank you. |
| 21:03.00 | prongils | jsharp: k thanks |
| 21:03.08 | redondos | [TK]D-Fender: I am ysing X-Lite to connect to asterisk, is there anything better? |
| 21:03.33 | octothorpe | redondos, yes |
| 21:03.59 | redondos | What, octothorpe ? |
| 21:04.02 | justinu | a real phone |
| 21:04.13 | redondos | heh... |
| 21:04.21 | octothorpe | redondos, windows, mac, or linux? |
| 21:04.21 | mog_work | octothorpe, is the pound key on a phone |
| 21:04.21 | redondos | Like.. any phone? |
| 21:04.25 | mog_work | octothorpe, is the real name |
| 21:04.28 | justinu | softphones tend to make a bad impression |
| 21:04.28 | redondos | octothorpe: Windows and Linux |
| 21:04.43 | mog_work | oops i misread that as what is octothorpe |
| 21:04.44 | justinu | redondos: one of the polycom IP series |
| 21:04.50 | octothorpe | redondos, for windows I prefer idefisk |
| 21:05.08 | GerbilWrk | Anyone familiar with a Lucent TNT connecting to an Asterisk box mind explaining how they physically connect? |
| 21:05.11 | redondos | Awesome, dling now. |
| 21:05.41 | Lino` | hmmm |
| 21:05.49 | octothorpe | redondos, for linux, I am not sure (no softphone on linux for me) |
| 21:05.54 | Lino` | what is the default extension for agent login logout? (a@h) |
| 21:06.10 | Lino` | giving a queue number 100 |
| 21:06.31 | *** join/#asterisk windowsrefund (n=windowsr@vtb2.fxserver.com) |
| 21:06.38 | windowsrefund | hello |
| 21:06.42 | octothorpe | redondos, idefisk: http://www.asteriskguru.com/tools/idefisk_beta.php |
| 21:06.47 | *** join/#asterisk jets (i=jetsnoc@216.83.66.202) |
| 21:06.53 | octothorpe | ~idefisk |
| 21:06.56 | fu3 | hi |
| 21:07.07 | windowsrefund | is it possible to use asterisk and completely bypass a phone provider? |
| 21:07.24 | Qwell[] | windowsrefund: only if you call other people who don't have phones |
| 21:07.33 | windowsrefund | damn |
| 21:07.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:07.56 | windowsrefund | those maggots |
| 21:08.39 | fu3 | yeah |
| 21:08.40 | fu3 | imagine that |
| 21:08.41 | octothorpe | windowsrefund: asterisk to asterisk is cool, but to reach PSTN you must use a provider |
| 21:08.45 | fu3 | providing a service in exchange for money |
| 21:08.47 | fu3 | fuck that |
| 21:09.16 | fu3 | we *could* |
| 21:09.22 | windowsrefund | the providers don't provide a service, they provide misery |
| 21:09.24 | fu3 | if everyone would simply say "i no longer value currency" |
| 21:09.29 | fu3 | but EVERYONE has to do that |
| 21:09.37 | fu3 | and then we have to agree not to simply overindulge ourselves |
| 21:09.55 | octothorpe | fu3: not gonna happen methinks |
| 21:10.02 | fu3 | not in this life time. |
| 21:10.05 | fu3 | unfortunately |
| 21:10.27 | fu3 | heh.. its crazy. all these wars, and fighting.. its all for control over the same oppressive systems. |
| 21:10.30 | fu3 | </rant> |
| 21:10.36 | fu3 | i need to work on this T1.. not rant |
| 21:10.43 | fu3 | so.. sorry :) |
| 21:10.56 | octothorpe | Have fun with your T1 |
| 21:11.00 | fu3 | i am! |
| 21:11.07 | fu3 | i just got my sangoma a104d |
| 21:11.13 | octothorpe | nice card |
| 21:11.16 | fu3 | i just got my DID numbers.. |
| 21:11.22 | jsharp | party time! |
| 21:11.23 | fu3 | now ive got to tackle asterisk |
| 21:11.23 | octothorpe | with EC I assume |
| 21:11.26 | fu3 | yes.. |
| 21:11.34 | fu3 | $2345.64 :) |
| 21:11.38 | octothorpe | nothing but the best for you |
| 21:11.52 | fu3 | its not for me.. its for my end users. |
| 21:12.07 | octothorpe | they are lucky to get such good equipment |
| 21:12.16 | fu3 | am I picking up on sarcasm? :) |
| 21:12.25 | octothorpe | not at all |
| 21:12.40 | fu3 | oh.. well thanks :) I'm glad to hear the card is well liked. |
| 21:13.33 | octothorpe | enjoy, it will save you headaches later |
| 21:14.15 | fu3 | im counting on that |
| 21:15.27 | FuriousGeorge | a school has me looking into a paging system |
| 21:16.04 | octothorpe | FuriousGeorge: that sound fun |
| 21:16.31 | FuriousGeorge | the only thing i cant figure out is how am i gonna replace physical switches for logic in the dialplan |
| 21:16.37 | FuriousGeorge | page a and C but not B or D |
| 21:16.44 | octothorpe | contexts |
| 21:17.04 | octothorpe | or maybe a variation on ring-groups |
| 21:17.14 | FuriousGeorge | yeah |
| 21:17.21 | justinu | fu3: your card cost more than my laptop :P |
| 21:17.31 | justinu | big spender |
| 21:17.46 | fu3 | haha |
| 21:17.47 | FuriousGeorge | i can also have an extension that asks for a list of extension to page |
| 21:18.02 | [TK]D-Fender | I couold buy 4 lapotops for that price.... |
| 21:18.03 | fu3 | thank the taxpayers of the state of minnesota |
| 21:18.17 | octothorpe | FuriousGeorge: explain more please |
| 21:18.21 | justinu | [TK]D-Fender: you must buy shitty laptops :P |
| 21:19.08 | [TK]D-Fender | justinu : I said I *could* buy that many, not that I'd CHOOSE those :) |
| 21:19.11 | justinu | heh |
| 21:19.17 | Qwell[] | :D |
| 21:19.26 | justinu | i just ordered a thinkpad x60s |
| 21:19.29 | Qwell[] | eww |
| 21:19.31 | justinu | 2gig ram, 100gig drive |
| 21:19.38 | [TK]D-Fender | justinu : I'd need the next up model with DVDrw and double the ram which would then be comfortable... |
| 21:19.39 | octothorpe | Qwell never misses a beat |
| 21:19.39 | FuriousGeorge | octothorpe: *9 is page all *100 is page first floor, *200 is page second floor, *XXX is page only that extension, and *99 will prompt the user for a list of extensions where he can put in *100, *201, *109 |
| 21:19.50 | [TK]D-Fender | So 3 decent laptops then :) |
| 21:19.56 | Qwell[] | octothorpe: That's what happens when you don't ever sleep |
| 21:19.59 | octothorpe | cool, that should work |
| 21:20.06 | fu3 | hey |
| 21:20.14 | fu3 | in zaptel.conf i dont understand this LBO option |
| 21:20.15 | justinu | dual laver DVDrw |
| 21:20.22 | Zodiacal | anyone know why xlite's transfer button is disabled? |
| 21:20.26 | fu3 | is that the distance between where my T1 enters teh building, to the PBX? |
| 21:20.30 | justinu | because they want to you to buy eyebeam |
| 21:20.30 | Qwell[] | Zodiacal: It's in the pay version |
| 21:20.32 | fu3 | or from the PBX to the Channel Bank? |
| 21:20.37 | Zodiacal | qwell ahh |
| 21:20.45 | jsharp | fu3: Both. |
| 21:20.49 | fu3 | ahh |
| 21:20.57 | Zodiacal | qwell so i can't even press like #ext to transfer either? |
| 21:21.02 | Qwell[] | nope :p |
| 21:21.10 | jsharp | Set the LBO according to your cable length. |
| 21:21.10 | Zodiacal | but its free |
| 21:21.14 | Zodiacal | qwell thanks! |
| 21:21.23 | fu3 | cable length between what though? |
| 21:21.29 | FuriousGeorge | ~seen schmaltz |
| 21:21.33 | jbot | i haven't seen 'schmaltz', FuriousGeorge |
| 21:21.49 | jsharp | Between your T1 card and the channel bank and between your T1 card and your provider's smartjack. |
| 21:22.11 | fu3 | ahhhh ok! |
| 21:22.15 | fu3 | yeah.. that makes sense. |
| 21:22.23 | fu3 | I dont have a CB |
| 21:22.29 | fu3 | but I do know what I think is the smartjack. |
| 21:22.38 | FuriousGeorge | anyone know of ANALOG phones with autanswer mode AND hands free |
| 21:22.47 | FuriousGeorge | speakerphone |
| 21:22.58 | jsharp | smartjack == wherever the telco hands your your circuit. |
| 21:23.11 | FuriousGeorge | ~seen shmaltz |
| 21:23.13 | jbot | shmaltz <n=mybox@mail.dmaven.com> was last seen on IRC in channel #asterisk, 3d 5h 35m 16s ago, saying: 'rene-, try it and tell us'. |
| 21:23.27 | fu3 | yep |
| 21:23.36 | fu3 | ok.. I think thats way longer than 655 feet :/ |
| 21:23.43 | fu3 | well maybe not. |
| 21:24.06 | fu3 | haha |
| 21:24.15 | fu3 | fudge.. i have no real idea how far that is. |
| 21:24.40 | jsharp | its that far? |
| 21:24.48 | fu3 | I didnt work here when they installed the conduit that goes from my server room to the head end. |
| 21:25.21 | fu3 | well i'm going to specify option 3, and experiment from there. |
| 21:25.24 | jsharp | Oh. |
| 21:25.25 | jsharp | Heh. |
| 21:25.32 | *** join/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer) |
| 21:25.34 | *** part/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer) |
| 21:28.10 | [ProB]CrazyMan | question ... what should tell me this ... app_rxfax.c:335 rxfax_exec: Unable to restore write format |
| 21:28.12 | Qwell[] | so, anybody sell their soul to go to VON next week? |
| 21:28.29 | Hmmhesays | No but I think i'll add to my dead hooker collection |
| 21:28.32 | mog_work | you are selling out for skinny Qwell that is if you want a place to sleep |
| 21:28.39 | Qwell[] | ack! |
| 21:28.42 | jsharp | Not after getting 40 or 50 spams from Pulver about it. |
| 21:29.02 | tzanger | dead hooker collection? are you Robert Pickton? |
| 21:29.23 | Hmmhesays | i wish I got that reference cause it bet it would have been funny |
| 21:30.24 | tzanger | google for him |
| 21:30.28 | Hmmhesays | yep, google confirmed it |
| 21:30.29 | tzanger | I'm sure he's all over google |
| 21:30.30 | Hmmhesays | it was funny |
| 21:30.44 | *** join/#asterisk exonic (n=exonic@209.172.11.54) |
| 21:31.00 | Hmmhesays | freaking drop C tunings |
| 21:31.03 | Hmmhesays | ARGH |
| 21:31.04 | exonic | Does everyone tend to configure seperate user and peer sections for iax connections? I can't get it to cleaning work using type=friend |
| 21:31.18 | tzanger | Hmmhesays: hahaha |
| 21:31.19 | tzanger | drop C? |
| 21:31.22 | tzanger | that's a low drop |
| 21:31.30 | Hmmhesays | you play tzanger? |
| 21:31.48 | tzanger | yeah |
| 21:31.53 | tzanger | not particuarly well but yes |
| 21:32.20 | Hmmhesays | figured you out by nickelback and say goodbye by theory of a deadman are both D-tuned and drop C |
| 21:33.47 | Hmmhesays | one thing I can figure out, no surprise by theory of a deadman is supposedly in drop D, but then never play a low D in the entire song |
| 21:34.09 | fu3 | damnit |
| 21:34.18 | fu3 | i get "ZT_SPANCONFIG device not configured" error. |
| 21:34.20 | tzanger | I all but refuse to play songs that are in fucked up tunings |
| 21:34.26 | Hmmhesays | drop D is ok |
| 21:34.28 | fu3 | ztcfg -vvvvv shows everything as good. |
| 21:34.33 | fu3 | until that error |
| 21:34.34 | Hmmhesays | in fact i kind of like it |
| 21:34.48 | Hmmhesays | i don't play heavy enough strings to drop C |
| 21:34.51 | jsharp | modules loaded and everything? |
| 21:35.13 | fu3 | yes |
| 21:35.21 | _Sam-- | [av]bani : did you try the new teliax gw? |
| 21:35.46 | jsharp | dmesg shows the hardware showing up when you load the modules? |
| 21:35.48 | [av]bani | Hmmhesays: http://bani.anime.net/nickelbackRecycled.swf |
| 21:36.28 | fu3 | yes |
| 21:36.44 | fu3 | wait :) |
| 21:37.21 | Zodiacal | anyone know of a free softphone that can do transfers? |
| 21:37.29 | fugitivo | xlite? |
| 21:37.32 | Zodiacal | nope |
| 21:37.42 | Hmmhesays | [av]bani thats pretty funny, but nickelback still rocks |
| 21:37.44 | fugitivo | it can't transfer??? |
| 21:37.48 | Zodiacal | not the free one |
| 21:37.52 | fu3 | yeah |
| 21:37.55 | Hmmhesays | their face melting live shows kick ass |
| 21:37.56 | fu3 | dmesg shows my sangoma card |
| 21:37.57 | fugitivo | that sucks |
| 21:37.59 | [av]bani | Hmmhesays: they rock the same song over and over :) |
| 21:38.07 | Zodiacal | fugitivo yes in deed it surly does |
| 21:38.09 | Hmmhesays | so what, ac/dc has been doing it for years |
| 21:38.19 | [av]bani | ac/dc does it better :) |
| 21:38.24 | Hmmhesays | they just do it different |
| 21:38.29 | Lino` | hmmm |
| 21:38.29 | Hmmhesays | ever seen nickelback play live? |
| 21:38.31 | jsharp | zaptel.conf is span=1,1,3,b8zs,esf or whatever your framing is? |
| 21:38.31 | [av]bani | they do it down under |
| 21:38.40 | Lino` | until now i never noticed but i got an email |
| 21:38.52 | Lino` | somebody claims that asterisk uses american dialtones |
| 21:38.57 | Lino` | or tones at all |
| 21:38.57 | fu3 | fuck |
| 21:39.04 | Hmmhesays | nickelback puts on one hell of a live show |
| 21:39.07 | fu3 | i have span=1,1,3,esf,b8zs |
| 21:39.12 | Lino` | like when you pick up the phone |
| 21:39.14 | Lino` | beeep |
| 21:39.21 | Hmmhesays | not like their cd's at all |
| 21:39.29 | Lino` | he wants german tones. beep beep beep *pause* beep beep beep *pause* beep beep beep *pause* |
| 21:39.31 | Hmmhesays | dialtones are configure on the endpoint itself |
| 21:39.36 | Lino` | hmm |
| 21:39.47 | jsharp | No, your right. I got the b8zs & esf backwards. |
| 21:39.50 | Lino` | so how can he set the dialtones using cisco 7960 SIP |
| 21:40.10 | fu3 | ok |
| 21:40.13 | fu3 | yeah |
| 21:40.26 | jsharp | dialtones on the 7960 are generated on the phone, not by asterisk. |
| 21:40.26 | [hC] | anyone have an idea on the power consumption of a polycom ip501 phone, for volts/amps? |
| 21:40.35 | fu3 | then below that i have fxsls=1-8 and then fxols=9-24 |
| 21:40.44 | Lino` | yeah, but is there a way to set german dialtones *on* the phone? |
| 21:40.48 | Hmmhesays | probably |
| 21:40.52 | Lino` | :D |
| 21:41.35 | [hC] | maybe someone who has a phone on PoE or something.. |
| 21:41.57 | jsharp | fu3: What does cat /proc/zaptel show you? |
| 21:42.07 | fu3 | haha here it comes |
| 21:42.09 | fu3 | "I use FreeBSD" |
| 21:42.15 | jsharp | Doh. |
| 21:42.16 | fu3 | and now "i dont support that" |
| 21:42.17 | jsharp | Ohyeah. |
| 21:42.19 | fu3 | and "cya" |
| 21:42.20 | fu3 | :) |
| 21:42.31 | jsharp | I'd support it if I knew it. |
| 21:42.35 | fu3 | i appreciate that |
| 21:42.43 | fu3 | like i said.. i'll go to linux if I cant get this to work. |
| 21:43.37 | fu3 | there is virtually nothing to be found through google on "ZT_SPANCONFIG failed on span 1: Device not configured" |
| 21:43.55 | [av]bani | fu3: linux sux |
| 21:43.59 | fu3 | although just before that, it says "24 channels configured" |
| 21:44.00 | Hmmhesays | power requirements should be on the spec sheet |
| 21:44.13 | fu3 | [av]bani.. i agree, but dont want to start THIS conversation all over again :) |
| 21:44.20 | [av]bani | fu3: opensores sux |
| 21:44.39 | *** join/#asterisk palomiux (n=lecaus@200.30.160.186) |
| 21:44.48 | palomiux | Hi there |
| 21:44.49 | fu3 | hi |
| 21:44.55 | palomiux | can you help me_ |
| 21:44.56 | palomiux | ? |
| 21:45.09 | palomiux | i have a few questions about asterisk required hardware |
| 21:45.15 | octothorpe | palomix: ask away |
| 21:45.35 | Hmmhesays | no you can't get 5 million users on a p133 |
| 21:45.47 | Hmmhesays | unless of course you're chuck norris |
| 21:45.48 | fu3 | hahah |
| 21:45.55 | octothorpe | ahh, why not? (jk) |
| 21:46.00 | fu3 | who saw Walker, Texas Ranger: Trial By Fire? |
| 21:46.08 | Hmmhesays | was that any good? |
| 21:46.12 | fu3 | no idea |
| 21:46.15 | fu3 | thats why im asking :) |
| 21:46.19 | Hmmhesays | i'll have to download thta |
| 21:46.21 | Hmmhesays | *that |
| 21:46.22 | fu3 | but its got the Norris in it |
| 21:46.28 | Hmmhesays | yeah reprising his walker role |
| 21:46.38 | fu3 | sudbury sucks |
| 21:46.42 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
| 21:46.50 | Seldon1975 | in any room there are at least 10 objects Chuck Norris could kill you with, including the room itself |
| 21:46.58 | Hmmhesays | I actually kind of liked walker texas ranger. it was a feel good show, the bad guy always got a chuck norris style ass whooping and the good guys won |
| 21:47.12 | fu3 | yeah.. it was entertaining. |
| 21:47.24 | fu3 | and Chuck actually KNOWS martial arts.. he isnt just acting. |
| 21:47.29 | [av]bani | Hmmhesays: black and white morality tales have wide appeal |
| 21:47.35 | fugitivo | isn't that the movie show where all chapters ended with a chuck norris kick to the bad boy? |
| 21:47.46 | Qwell[] | ROUNDHOUSE kick |
| 21:47.48 | Qwell[] | yes |
| 21:47.53 | fugitivo | yes, that one |
| 21:47.59 | Seldon1975 | Chuck Norris does not sleep. He waits. |
| 21:48.11 | Seldon1975 | Chuck Norris defines love as the reluctance to murder. If you’re still alive, it’s because Chuck Norris loves you. |
| 21:48.13 | _Sam-- | Chuck Norris counted to infinity. Twice. |
| 21:48.24 | Qwell[] | okay, okay :P |
| 21:48.31 | Seldon1975 | Chuck Norris can divide by zero. |
| 21:48.32 | Seldon1975 | soz |
| 21:48.35 | Seldon1975 | ill stop now |
| 21:48.38 | fugitivo | I like steaven seagal style |
| 21:48.39 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 21:48.41 | fugitivo | steven |
| 21:48.43 | fu3 | hahahahah _Sam--.. that was good |
| 21:48.54 | Seldon1975 | fugitivo: pity he continued to make movies waaay past his prime |
| 21:48.57 | fu3 | Steven is fake though |
| 21:48.59 | fu3 | Chuck is REAL |
| 21:49.04 | Seldon1975 | fu3: errrr, no |
| 21:49.06 | fugitivo | why fake? |
| 21:49.08 | fu3 | i saw a documentary on segal.. they film shit all slow motion and then speed it up. |
| 21:49.11 | fugitivo | aikido is not fake |
| 21:49.17 | fugitivo | really? |
| 21:49.20 | Seldon1975 | fu3: he's kick YOUR ass |
| 21:49.20 | fu3 | yes |
| 21:49.26 | fu3 | well.. he'd kick my ass in the kitchesn |
| 21:49.32 | fugitivo | oh man, now i'm depressed |
| 21:49.32 | fu3 | i mean.. he IS america's favorite chef. |
| 21:49.40 | fu3 | yeah.. segal IS A FRAUD! |
| 21:49.46 | fu3 | Although, entertaining. |
| 21:49.51 | fu3 | I enjoyed his ass kickings. |
| 21:50.08 | fu3 | taking on the entire soviet army and MAYBE getting a scratch on the arm, or forehead. |
| 21:50.13 | fu3 | classic :) |
| 21:50.31 | Seldon1975 | Steven Seagal is a 7th Dan black belt in Aikido |
| 21:50.59 | _Sam-- | Chuck Norris is the reason why Waldo is hiding. |
| 21:51.00 | fugitivo | the techniques he uses are not 100% aikido |
| 21:51.03 | fu3 | well.. I suppose we can blame it on the director. He didnt want Steven to kick everyone ass for real, so he made him slow it down. |
| 21:51.03 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
| 21:51.28 | Seldon1975 | "If you look at him run, he runs like a woman," John Connolly, who wrote a scathing profile of Seagal for Penthouse, tells THS. |
| 21:52.06 | fu3 | yeah well he runs like a woman who could fuck you up :) |
| 21:52.08 | palomiux | guys, can you tell me which hardware is best to build my asterisk server? |
| 21:52.25 | Seldon1975 | get a Dell Poweredge server |
| 21:52.29 | jsharp | Hardware that runs pretty much sums it up. |
| 21:52.29 | _Sam-- | [av]bani : did you try voip-co4 teliax yet? |
| 21:52.37 | [av]bani | no |
| 21:52.37 | fugitivo | palomiux: what are you going to run on it? |
| 21:52.49 | Nodren | would anyone recommend running asterisk on CentOS? |
| 21:53.05 | _Sam-- | Nodren : probably people that know CentOs would recommend that. |
| 21:53.06 | Seldon1975 | Nodren: thats how asterisk@home is set up |
| 21:53.27 | Seldon1975 | that's what I'm using, seems to work |
| 21:53.43 | Nodren | should i go with asterisk@home or just asterisk? |
| 21:53.46 | Seldon1975 | if I built it from scratch I'd probably use Debian |
| 21:53.49 | brad_mssw | i personally wouldn't use a redhat-based distro, but hey, that's just me |
| 21:53.51 | jsharp | Depends on your application. |
| 21:53.54 | fugitivo | Nodren: asterisk |
| 21:53.56 | FuriousGeorge | i really love AMD chips as an enthusiast, but seems like all the * pros prefer intel. that safe to say? |
| 21:54.01 | Seldon1975 | Nodren: ppl have said @home has sucked in the past |
| 21:54.08 | fugitivo | Nodren: you'll not get support if you use a@h |
| 21:54.11 | Nodren | thank you |
| 21:54.11 | Seldon1975 | Nodren: but the latest 2.6 version may be ok |
| 21:54.12 | brad_mssw | FuriousGeorge: we run it on AMD, no issues |
| 21:54.17 | _Sam-- | i have heard good things about the dual core AMDs and * |
| 21:54.29 | jsharp | Every * server I've built for customers has been with AMD, without a problem. |
| 21:54.30 | FuriousGeorge | brad_mssw: AMD64? |
| 21:54.33 | Nodren | ok |
| 21:54.34 | FuriousGeorge | i use amd in most places too |
| 21:54.51 | Nodren | i run a few centos servers on AMD64, they run great |
| 21:54.54 | brad_mssw | FuriousGeorge: actually, our primary box is on Athlon XP ... testing some AMD64 stuff now, probably will end up replacing that box |
| 21:54.56 | Nodren | assuming you use a 32bit distro |
| 21:55.13 | Seldon1975 | Nodren: sounds like Centos on AMD is your best bet then |
| 21:55.25 | Seldon1975 | Nodren: since you're most familiar with it |
| 21:55.34 | Seldon1975 | Nodren: and there are no known issues with that setup |
| 21:55.43 | FuriousGeorge | why would i want to use a 32bit distro? |
| 21:55.59 | FuriousGeorge | so * can run? |
| 21:56.01 | Nodren | because 64bit distros arnt fully developed enough |
| 21:56.03 | Nodren | to be stable |
| 21:56.11 | FuriousGeorge | fair enough |
| 21:56.17 | Nodren | even if you have a 64 bit processor, i've personally seen only trouble with 64 bit os |
| 21:56.19 | palomiux | fugitivo: Asterisk |
| 21:56.20 | brad_mssw | Nodren: eh, i wouldn't claim that |
| 21:56.22 | *** join/#asterisk Smi|k (n=smilk@netblock-72-25-103-142.dslextreme.com) |
| 21:56.27 | palomiux | maybe Asterisk@home |
| 21:56.32 | Nodren | infact, i had a linux server crash and lose data, even using ext3, when on 64bit |
| 21:56.40 | Nodren | i switched to 32bit and have over 100 days uptime, no errors |
| 21:56.44 | Smi|k | anyone know the best way for a voip service provider that is larger in size to offer incoming phone service for businesses? |
| 21:56.49 | brad_mssw | Nodren: running a 2.4 kernel or something ? |
| 21:56.58 | Nodren | 2.6 |
| 21:57.03 | Smi|k | pots lines are expensive, and its hard to be competitive even with PRI T1's |
| 21:57.19 | _Sam-- | Smi|k : get a DS3 to your VOIP provider |
| 21:57.20 | _Sam-- | and speak SIP |
| 21:57.29 | brad_mssw | Smi|k: yeah, T3, sangoma has a card |
| 21:57.33 | Smi|k | how many lines will a DS3 handle for incoming? |
| 21:57.53 | mog_work | doesnt work in asterisk brad_mssw |
| 21:57.55 | _Sam-- | depends what codec you use etc |
| 21:57.57 | iGotNoTime | is Teliax ok with using *? |
| 21:58.03 | Smi|k | and speak sip to what kind of priver? stuff like fonality and zaltus? |
| 21:58.04 | mog_work | its just unchanilized |
| 21:58.04 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 21:58.05 | brad_mssw | mog_work: really? never used it |
| 21:58.17 | mog_work | thats why its so cheap |
| 21:58.20 | brad_mssw | iGotNoTime: yeah, if you're close enough to them |
| 21:58.34 | iGotNoTime | brad_mssw, USA? |
| 21:58.39 | brad_mssw | iGotNoTime: yeah |
| 21:58.41 | _Sam-- | Smi|k : for real business enterprise service, you probably wont want to speak SIp over a remote gateway, at least not one that is too far away. |
| 21:58.42 | iGotNoTime | ok :D |
| 21:58.46 | CunningPike | Does anyone know why this would always give a result of APPERROR, even if the file exists? |
| 21:58.47 | brad_mssw | iGotNoTime: latency can be bad though |
| 21:58.49 | CunningPike | System(ls /var/lib/asterisk/sounds/call-processors/${PHRASEID}.*) |
| 21:58.58 | CunningPike | It used to work :( |
| 21:59.02 | iGotNoTime | brad_mssw, noticable? |
| 21:59.35 | brad_mssw | iGotNoTime: yeah, lots of packetloss too ... usually 70+ms latency |
| 21:59.41 | brad_mssw | iGotNoTime: they're in Denver, I'm in Florida |
| 21:59.51 | _Sam-- | brad_mssw : things are better for me there. |
| 22:00.01 | _Sam-- | plus now they have another colorado server in testing |
| 22:00.10 | brad_mssw | iGotNoTime: seems like the data tends to transfer over savvis or cogent, I'm on UUNET |
| 22:00.14 | iGotNoTime | brad_mssw, I'll check out out I am in Ohio might be good for me :) |
| 22:00.20 | _Sam-- | my SIP to teliax are fine, never any latency that is their fault. |
| 22:00.33 | _Sam-- | if there is latency / packet loss, its been all internet related |
| 22:00.39 | _Sam-- | and my calls are sounding good again |
| 22:00.40 | brad_mssw | _Sam--: yeah, i ran some latency checks to the new one, it's about the same for me ... different provider though which is nice |
| 22:00.54 | brad_mssw | _Sam--: changed to SIP, hasn't really helped |
| 22:01.23 | brad_mssw | _Sam--: waiting on their east-coast server |
| 22:01.26 | iGotNoTime | while I await my sipura I can use a softphone with my Teliax account right? |
| 22:01.48 | brad_mssw | iGotNoTime: yeah |
| 22:01.51 | _Sam-- | alls i know is that my teliax stuff sounds perfect...so i do not believe the problems people are having are coming from teliax. |
| 22:01.56 | _Sam-- | i mean, i think its not teliax's fault. |
| 22:02.11 | brad_mssw | _Sam--: well, it's their ISPs fault for using sub-par backbone providers |
| 22:02.21 | brad_mssw | _Sam--: so indirectly it's teliax's fault for using that ISP |
| 22:02.23 | _Sam-- | i come in over the same routes as anyone else |
| 22:02.42 | _Sam-- | i can set you up an sip account on my * and you can make some calls |
| 22:02.46 | _Sam-- | and see if its any better |
| 22:02.48 | *** join/#asterisk Axel69 (n=alexlsf@200.62.38.91) |
| 22:02.53 | brad_mssw | _Sam--: yeah, but it depends on what provider you start on, what does your route look like |
| 22:03.17 | _Sam-- | i come in to teliax over cogent . |
| 22:03.26 | jsharp | The joys of the suckiness of the intardweb. |
| 22:03.38 | Smi|k | so if I have a group of customers who in total need 500 incoming phone lines, in various locations etc..etc.. |
| 22:03.47 | brad_mssw | _Sam--: UUNET is one of the best, and always has been, and rockynet does have a UUNET link, but they have their low-cost links favored over it ... so I'm getting screwed at the peering points, and the sub-par networks |
| 22:03.52 | Axel69 | Hi guys, i have a problem with h323... i installed the addon but when i type channeltypes it doesn't appers the h323 |
| 22:03.59 | brad_mssw | _Sam--: cogent is great if you're on cogent networks, but their peering points suck |
| 22:04.11 | Smi|k | is there any provider that can give great service and charge me say .005c/minute incoming and then I resell it out to my customers for .01c/minute? |
| 22:04.22 | Qwell[] | .01c? |
| 22:04.27 | Qwell[] | good luck with that |
| 22:04.27 | fu3 | Come on, man. I had a rough night and I hate the fuckin' Eagles, man |
| 22:04.33 | brad_mssw | _Sam--: 9 hops after leaving UUNET to reach teliax |
| 22:04.38 | _Sam-- | brad_mssw : i also noticed a difference when i upgraded my *....are you on 1.2.5? |
| 22:04.46 | brad_mssw | _Sam--: yes, 1.2.5 |
| 22:04.53 | _Sam-- | im 15 hops to teliax, 80ms, and my calls are mint |
| 22:05.03 | brad_mssw | wow, 80ms is kind of high |
| 22:05.09 | brad_mssw | i'm sitting at 15 hops too |
| 22:05.13 | Smi|k | any idea what the best option is for that? |
| 22:05.20 | _Sam-- | i did have problems, but i dont any longer. |
| 22:05.22 | brad_mssw | if you count my internal router it sits behind |
| 22:05.31 | Qwell[] | Smi|k: probably $500k to level3 |
| 22:05.32 | _Sam-- | and the 15 people who were bitching daily, are all off my back. |
| 22:05.37 | Qwell[] | per month |
| 22:05.38 | brad_mssw | _Sam--: i think my main problem comes from packetloss on savvis |
| 22:05.42 | Smi|k | i.e. how can I hook right into telecom so I can provision my own "phone lines" that are not really lines |
| 22:06.16 | _Sam-- | Smi|k : if you used an ISP like mine you would be good to go :) |
| 22:06.28 | _Sam-- | Smi|k : you need to find a small, adept, CLEC to connect to |
| 22:06.34 | *** join/#asterisk revar (i=[U2FsdGV@216.127.82.54) |
| 22:06.37 | Smi|k | how can I find that for my ISP? |
| 22:06.49 | _Sam-- | unless your ISP is a CLEC as well, you cant. |
| 22:07.01 | revar | hi all, I'm trying to get an openline working with asterisk@home |
| 22:07.10 | brad_mssw | _Sam--: though it's only 13 hops to the voip-co4.teliax.com |
| 22:07.21 | Smi|k | how does an ISP become a CLEC? |
| 22:07.31 | revar | I've got a vpb trunk setup and the vpb drivers going but no answer when I call |
| 22:07.39 | revar | any ideas how I can diagnose this? |
| 22:07.42 | Smi|k | I've been complaining to my ISP who offers their own VOIP service so much that they are coming here for a meeting for me to explain their options to them |
| 22:07.48 | _Sam-- | Smi|k : from what i gather, they hire an attorney and file some stuff with the FCC and/or PUC? |
| 22:07.49 | brad_mssw | _Sam--: hmm, no packetloss yet though to the voip-co4.teliax.com |
| 22:07.51 | Smi|k | they need to become a CLEC to make money with voip right? |
| 22:08.15 | _Sam-- | Smi|k : i would say the NEED to |
| 22:08.30 | _Sam-- | there are a zillion ways to make money working with VOIP...selling per minute based services is just one of them. |
| 22:08.38 | _Sam-- | er i WOULDNT say they need to |
| 22:08.39 | _Sam-- | they COULD |
| 22:08.41 | kink0 | curiosity... what is a CLE ? |
| 22:08.45 | _Sam-- | clec |
| 22:08.45 | kink0 | CLEC ? |
| 22:08.55 | _Sam-- | competitive local exchange carrier |
| 22:09.00 | _Sam-- | http://isp.webopedia.com/TERM/C/CLEC.html |
| 22:09.26 | kink0 | ah ok, that concept is no managed here (Spain) |
| 22:11.28 | palomiux | you know where to compare Asterisk with @home_ |
| 22:11.29 | palomiux | ? |
| 22:12.25 | palomiux | anyone? |
| 22:12.35 | austinnichols101 | palomuix: sure asterisk is asterisk and @home is asterisk + a centos build + AMP + several other pieces |
| 22:12.41 | brad_mssw | _Sam--: ok, just set up my primary outgoing as teliax voip-co4 ... see how that is |
| 22:12.55 | palomiux | Austin: AMP? |
| 22:13.06 | _Sam-- | brad_mssw: what codec and what are your SIP devices? |
| 22:13.27 | fu3 | WOOHOO!! |
| 22:13.29 | austinnichols101 | palomuix: Asterisk Management Portal (#amportal). Everything is managed through AMP which is a subject of contention among many. |
| 22:13.31 | fu3 | justinu.. it works. |
| 22:13.36 | brad_mssw | _Sam--: g711u, going to a Zap FXS channel locally |
| 22:13.39 | fu3 | it WASNT loading the modules properly |
| 22:13.43 | fu3 | but it is now ;) |
| 22:14.15 | FuriousGeorge | what kind of processor would be able to handle 10-20 sip clients in a meetme |
| 22:14.26 | FuriousGeorge | maybe even 18 sip and 2 zap |
| 22:14.30 | austinnichols101 | palomiux: concensus (with which I agree) is that if you were building your own box with asterisk and all of the add-on pieces that you would probably do it in a different way and could improve upon what was done. OTOH, it's all there for you in one place with the AAH ISO. |
| 22:15.37 | austinnichols101 | palomiux: big issue is that many people don't dig much deeper under the covers once they install AAH which can lead to a lot of stupid questions later on because they don't have a fundamental understand of what asterisk is doing. |
| 22:15.44 | *** join/#asterisk Tikola (n=nix@222-154-13-78.jetstream.xtra.co.nz) |
| 22:15.54 | Tikola | hi pplz |
| 22:15.54 | Tikola | anyone got the AstTAPI thing working? |
| 22:15.59 | palomiux | I see |
| 22:16.01 | Tikola | it says my manager account has logged in, but never dials. no commands are issued |
| 22:16.19 | _Sam-- | i got ASTtapi working, i use it with outlook |
| 22:16.20 | austinnichols101 | palomiux: http://www.mundy.org/blog is a good place to start reading. Look for their soup-to-nuts on AAH and read through to understand the differences. |
| 22:16.23 | _Sam-- | i think thats what i use |
| 22:16.28 | palomiux | Austin: what is AMP? what AMP stands for? |
| 22:16.47 | Tikola | configured it under modems/advanced. i think its configured correct. i told it to use the context everything is setup under. with the dialplans |
| 22:16.48 | austinnichols101 | Asterisk Management Portal. |
| 22:17.02 | palomiux | hmmm, it comes with both versions? |
| 22:17.08 | austinnichols101 | and it's in the process of being reworked and rebranded as freePBX |
| 22:17.16 | Tikola | sam; yes, this is with outlook |
| 22:17.16 | austinnichols101 | nope |
| 22:17.25 | _Sam-- | ya i use asttapi fine, works. |
| 22:17.26 | _Sam-- | i like it |
| 22:17.39 | Tikola | did you configure it to use a channel, or context |
| 22:17.39 | Tikola | ? |
| 22:17.46 | austinnichols101 | asterisk is 'just asterisk' by itself. Think of AMP as an add-on that puts a web gui on managing your config files |
| 22:18.06 | austinnichols101 | asterisk by itself and vim can do the same thing and is definitely less 'restrictive' |
| 22:18.07 | palomiux | Can it be used with Asterisk and AAH? |
| 22:18.17 | austinnichols101 | AMP is part of the AAH distro |
| 22:18.17 | _Sam-- | it uses a user channel, and an outgoing channel |
| 22:18.27 | palomiux | Thanks |
| 22:18.29 | _Sam-- | user channel == your sip/iax account |
| 22:18.35 | CunningPike | I am trying to test for the existence of a sound file from the dialplan so I can decide whether to play it or not. System (ls /file) used to work (1.0.9) but no longer works in 1.2 - any ideas? |
| 22:18.40 | _Sam-- | i use the dial outgoing channel part, not dial by context. |
| 22:18.46 | _Sam-- | and i specify the outgoing channel |
| 22:19.25 | _Sam-- | works fine for me, i click to dial, my phone rings, i answer, then it connects me up |
| 22:19.37 | Tikola | what have you got in the outgoing channel box? |
| 22:19.59 | _Sam-- | my outgoing channel: SIP/teliax/ |
| 22:20.02 | Zodiacal | anyone know how parked calls are suposed to notify me what # its parked at? cuz im using eyebeam softphone and transfering a call to 70 (which is my park number) and then it just ends my softphone call. but in the cli i see that it parked it to say71, but how is a normal use to see that? |
| 22:20.06 | Zodiacal | is this a limit of my softphone? |
| 22:20.41 | Tikola | can i have Zap/3/ |
| 22:20.54 | _Sam-- | sorry i dont have time to be your tutor on it... |
| 22:20.56 | _Sam-- | but you can try it |
| 22:21.00 | _Sam-- | i have to get some work done |
| 22:21.06 | brad_mssw | Zodiacal: it usually tells you where it parked it (audio) when you park a call |
| 22:21.09 | Tikola | ok cheers anyway, ill try somethings |
| 22:21.15 | Zodiacal | brand_mssw any ideas why its not? |
| 22:21.23 | brad_mssw | Zodiacal: are you doing a blind transfer ? |
| 22:21.29 | palomiux | what can you tell me about the best hardware cards for Asterisk? |
| 22:21.40 | Zodiacal | brad_mssw ahh maybe eyebeam softphone is doing blind transfer |
| 22:21.45 | Zodiacal | im just pushing the transfer button |
| 22:22.01 | brad_mssw | Zodiacal: yeah, see if you can figure out how to do an attended transfer ... it should tell you there |
| 22:22.05 | *** part/#asterisk windowsrefund (n=windowsr@vtb2.fxserver.com) |
| 22:22.05 | Zodiacal | can i transfer manualy by pressing #ext i forget |
| 22:22.14 | austinnichols101 | palomiux: check voip-info.org as there are extensive writeups on asterisk hardware there |
| 22:22.24 | austinnichols101 | palomiux: it's already been discussed to death |
| 22:22.25 | Hmmhesays | but only if you can read |
| 22:22.36 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 22:22.38 | Tikola | yes! working :) |
| 22:22.54 | palomiux | Austin: I want to know what do you think based on your experience |
| 22:22.59 | Zodiacal | brad_mssw worked! Thank You! |
| 22:23.11 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 22:23.43 | austinnichols101 | palomiux: my only experience so far has been with the Digium cards and they've worked just fine. However, I'm just like you in that I'm new to the game so I'm definitely not the one to rely on in that area. |
| 22:24.21 | _Sam-- | the sangoma cards with echo cans have been getting good reviews here |
| 22:24.29 | _Sam-- | the A200* |
| 22:24.33 | austinnichols101 | palomiux: voip-info will show you who the major players are and you'll usually find two camps that disagree over which one is best. |
| 22:24.36 | _Sam-- | but i dont have one |
| 22:24.38 | fu3 | so far, im loving my a104d |
| 22:24.45 | Hmmhesays | i only use external gateways |
| 22:24.54 | austinnichols101 | _sam: yeah - I want to replace my TE110P with one of those |
| 22:24.56 | fu3 | those lead to harder drugs |
| 22:25.05 | FuriousGeorge | let me ask this way. is a 32bit amd barton (512 cache 400mhz fsb, ~2.GHZ) handle a meetme conference with 20 people speaking? 18 sip two zap? im trying to think of the worst possible scenario |
| 22:25.16 | Hmmhesays | you can't snort lines of coke off a pci card |
| 22:25.24 | fu3 | cant you? |
| 22:25.25 | fu3 | :) |
| 22:25.25 | FuriousGeorge | hmmmm? |
| 22:25.32 | Hmmhesays | not without cutting your nose up |
| 22:25.33 | _Sam-- | FuriousGeorge : if there is a lot of transcoding im not sure a barton could keep up |
| 22:25.36 | _Sam-- | im not positive though |
| 22:25.39 | FuriousGeorge | what about a bump off the northbridge |
| 22:25.43 | fu3 | how would you know that Hmmhesays? |
| 22:25.44 | austinnichols101 | ha |
| 22:25.46 | fu3 | :) |
| 22:25.51 | palomiux | Austin: no problem, nice to know it works fine |
| 22:26.00 | Hmmhesays | FuriousGeorge: that might work, better than cooking it up in a spoon with a lighter |
| 22:26.24 | revar | exit |
| 22:26.25 | revar | exit |
| 22:26.26 | FuriousGeorge | Hmmhesays: provided there is no heatsink on the northbridge, of course :) |
| 22:26.26 | revar | quit |
| 22:26.30 | Hmmhesays | fu3: because not only do I collect dead hookers, i'm a drug addict |
| 22:26.30 | revar | do |
| 22:26.31 | revar | sorry |
| 22:26.31 | austinnichols101 | palomiux: general rule of thumb I heard years ago. You have 1) Good, 2) Fast and 3) Cheap. Pick any two you want. |
| 22:26.34 | Hmmhesays | FuriousGeorge of course |
| 22:26.40 | FuriousGeorge | _Sam--: all ulaw |
| 22:26.50 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 22:27.03 | _Sam-- | FuriousGeorge : what is the timing device? ztdummy? |
| 22:27.03 | Primer | so what's the deal with having to 'license' the software on a cisco IP phone? Surely that doesn't apply if you flash it to use SIP...? |
| 22:27.18 | _Sam-- | Primer : of course it does, why wouldnt it |
| 22:27.23 | Hmmhesays | that's like the rules about trying to hook up with women from the internet. Hot, Single, Emotionally stable. pick two |
| 22:27.29 | Primer | that just seems...wrong |
| 22:27.34 | FuriousGeorge | _Sam--: good question, knowing what i know about HW EC now, i'd probably want to get a sangoma |
| 22:27.42 | FuriousGeorge | vs. a tdm400p |
| 22:27.44 | Primer | I mean, if I buy the phone, why must I continue to pay? |
| 22:27.55 | jsharp | Hmmhesays: That's not just women from the internet. That's women in general. |
| 22:27.56 | FuriousGeorge | do they have a timer (the sangoma's) |
| 22:27.58 | *** join/#asterisk pauldy (n=pauldy@m090e36d0.tmodns.net) |
| 22:28.04 | Hmmhesays | yeah who in here actually paid for their crisco firmware |
| 22:28.05 | Primer | like, if I buy a linksys router, I don't have to pay for firmware updates, or some stupid license |
| 22:28.10 | mog_work | yes FuriousGeorge |
| 22:28.16 | _Sam-- | if there is no transcoding and you have a hardware device for timing i bet you would be fine |
| 22:28.20 | mog_work | but why go that way when you can get the bohemeth |
| 22:28.22 | Hmmhesays | jsharp, you're right |
| 22:28.22 | mog_work | the 2400p |
| 22:28.23 | _Sam-- | but i just dont know |
| 22:28.40 | Primer | well, we're planning on deploying a bunch of IP phone in a business |
| 22:28.47 | Primer | and we don't want risk any exposure |
| 22:29.33 | Hmmhesays | last time i risked exposure the cop said I shouldn't be pissing on his bumper |
| 22:29.34 | FuriousGeorge | mog_work: they only have two pots lines comming in |
| 22:29.39 | _Sam-- | lol |
| 22:29.45 | palomiux | how do I get 2? |
| 22:29.47 | mog_work | and??? |
| 22:29.49 | palomiux | Good and cheap? |
| 22:29.50 | mog_work | ^_^ |
| 22:30.03 | FuriousGeorge | mog_work: does the 2400p have HW EC? |
| 22:30.07 | mog_work | yes |
| 22:30.12 | mog_work | as an option |
| 22:30.30 | FuriousGeorge | what about the price for the board and the modules, isnt that overkill? |
| 22:30.32 | fabsoft | hi, anyone know how to use app_txfax in asterisk ? |
| 22:30.36 | FuriousGeorge | for two fxo? |
| 22:30.47 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 22:30.52 | FuriousGeorge | option=more money? |
| 22:30.54 | mog_work | err sorry |
| 22:31.03 | FuriousGeorge | ill look into it :) |
| 22:31.06 | mog_work | probably more money than sangoma at 2 ports |
| 22:31.45 | mog_work | but much more room to grow |
| 22:31.49 | FuriousGeorge | well, its scalable, but they are a school, so i dont know how scalable they need to be |
| 22:32.08 | mog_work | and you only need the one echo can for 24 |
| 22:32.22 | stack_ | has anyone used iaxmodem & hylafax reliably for a long period of time. This could be an ideal solution, I just want to make sure it's stable |
| 22:32.22 | mog_work | so if you ever bought another sangoma a200 you need yet another ec |
| 22:33.17 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:33.39 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 22:33.52 | FuriousGeorge | mog_work: lemme check it out, i may just get regular tdm400s if the echocan is a lot more on the sangoma anyway. i didnt realize it was an option |
| 22:34.12 | mog_work | yeah |
| 22:34.17 | mog_work | hw ec is expensive |
| 22:34.23 | palomiux | Austin: what kind of cards are you using? |
| 22:34.25 | mog_work | and personally mg2 kicks more ass in most cases |
| 22:34.36 | mog_work | hw ec is only better in taking load off in my opinion |
| 22:34.47 | mog_work | at least with whats out today |
| 22:36.59 | Hmmhesays | is that you i hear screamin' my name? i'm a roller i'm a rider i'm #1 mutherfarkin survivor |
| 22:37.02 | FuriousGeorge | yeah, i just realized that the sangoma WITH ec is actually a lot more than a comparable tdm w/o it :) |
| 22:37.04 | stack_ | Can someone explain how a T1 works as far as attaching phone numbers to it? Ideally we would get a T1 (23 channel + 1 PRI) and would like to split it in 3. Some lines for one part of the company, some for another and the rest for our call center. How exactly does a voice T1 work? |
| 22:37.13 | FuriousGeorge | tdm400 it is |
| 22:37.15 | palomiux | Austin are you there? |
| 22:37.40 | brad_mssw | stack_: you probably don't want a voice T1, you probably want a PRI |
| 22:37.40 | mog_work | tdm2400p starts to make more sense finacially after 8-12 lines i think |
| 22:37.58 | brad_mssw | stack_: you don't get features like callerid on a T1 |
| 22:38.14 | stack_ | brad_mssw: isn't PRI the caller id stuff? |
| 22:38.15 | mog_work | pri is better |
| 22:38.18 | mog_work | its all pretty |
| 22:38.28 | brad_mssw | stack_: no, pri isn't an addon to a T1 |
| 22:38.33 | FuriousGeorge | mog_work: thing is, im learning that chan_sip and chan_iax are so much cooler than chan_zap, and cheaper |
| 22:38.39 | brad_mssw | stack_: different technology/delivery |
| 22:38.58 | brad_mssw | stack_: as far as how you split the usage, etc ... that's all up to your dialplan/asterisk config |
| 22:39.02 | stack_ | thanks, now I'm all confused :) |
| 22:39.16 | mog_work | can be FuriousGeorge |
| 22:39.31 | Hmmhesays | to fall in love and fall in debt, to alcohol and cigarettes |
| 22:39.35 | stack_ | brad_mssw: we were looking at the T1 for all the channels and have three phone numbers attached to it. Each number would be assigned to a group of channels |
| 22:39.36 | brad_mssw | stack_: search google for like PRI vs T1 |
| 22:39.38 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 22:39.50 | diclophis | is it possible to calculate .wav file duration from the filesize? |
| 22:39.50 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 22:40.34 | joeqread | except chan_sip and chan_iax2 won't work in a MeetMe room without a zaptel device. :( |
| 22:40.36 | kpettit | ?? cpm wassup? |
| 22:40.37 | brad_mssw | stack_: well, with a PRI it doesn't really work like that |
| 22:40.49 | joeqread | and there's no ztdummy driver for FreeBSD |
| 22:40.57 | mog_work | USE LINUX! |
| 22:41.01 | brad_mssw | stack_: you can assign an unlimited number of numbers to it, it delivers to your PBX what number the call is coming in on, and you can route from there |
| 22:41.22 | joeqread | naw, I like fbsd better |
| 22:41.24 | brad_mssw | stack_: and you can choose to reject the call if it exceeds the number of consecutive calls for a certain number or similar |
| 22:41.28 | asterboy | Does the Digium TE110P T1 card terminate a PRI line directly? |
| 22:41.37 | mog_work | yes it can |
| 22:41.39 | austinnichols101 | yup |
| 22:41.44 | stack_ | that was my next question |
| 22:41.47 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
| 22:41.53 | asterboy | so all 24 channels can be lit? |
| 22:42.06 | jsharp | 23 voice + the D control channel. |
| 22:42.11 | asterboy | right |
| 22:42.36 | asterboy | Anyone have a good T1 provider for Alberta, Canada? |
| 22:42.42 | masonf | is there any way to make my telephony card answer calls right when they come in and not wait a ring before it picks up? |
| 22:42.46 | asterboy | Allstream wants $680/month |
| 22:42.52 | brad_mssw | asterboy: Level3 in canada ? |
| 22:43.00 | stack_ | I see where my info was getting mixed up. The phone company said that a T1 with callerid was 23 channel with PRI, I assume they meant to say just PRI |
| 22:43.03 | asterboy | ahhhh...good ol' level3 |
| 22:43.34 | brad_mssw | stack_: yeah, pri can be delivered over T1 ... but the terms can't be used interchangably |
| 22:43.45 | stack_ | gotcha |
| 22:44.18 | stack_ | so the te110p can accept the pri over t1 then? |
| 22:44.22 | jsharp | Yes. |
| 22:44.24 | mog_work | yes |
| 22:44.50 | brad_mssw | stack_: yeah, but I'd use sangoma cards, personally |
| 22:45.08 | asterboy | cheaper too |
| 22:45.10 | stack_ | awesome, you guys had me scared for a bit... I'm new to all this and im trying to set up a server for a call center, couldn't ask for a harder situation :) |
| 22:45.12 | joeqread | mog: any particular reason to use linux over fbsd, besides personal taste and presence of a ztdummy driver? |
| 22:45.19 | asterboy | ebay has a quad for about $500 |
| 22:45.22 | FuriousGeorge | mog_work: cheaper in the long run, usually |
| 22:45.28 | mog_work | well i have used everything under the sun |
| 22:45.42 | mog_work | and i really dont care if im on a bsd box or a linux box |
| 22:45.49 | Qwell[] | even a sun? |
| 22:45.52 | diclophis | so.... no .wav calculation wizards in today? |
| 22:45.53 | mog_work | but i would never run asterisk on a non-linux box |
| 22:45.54 | stack_ | are the docs for sangoma as good as the ones available for zaptel stuff? |
| 22:45.57 | mog_work | i have a sun box Qwell |
| 22:45.58 | joeqread | so performance wise with * they're the same? |
| 22:46.02 | Qwell[] | ooo |
| 22:46.05 | Qwell[] | What kind? |
| 22:46.08 | mog_work | cobalt |
| 22:46.11 | Qwell[] | fun |
| 22:46.11 | mog_work | and a sparc 2 |
| 22:46.16 | Qwell[] | I've got an ss5, heh |
| 22:46.22 | *** part/#asterisk [vmwarez]dotcom (n=jjones@216.147.224.254) |
| 22:46.22 | mog_work | very similar joeqread |
| 22:46.23 | jsharp | Asterisk on an SS2. |
| 22:46.24 | Qwell[] | I love that slow POS |
| 22:46.29 | mog_work | but everything just works on linux |
| 22:46.34 | mog_work | i mean its a pbx |
| 22:46.38 | jsharp | As long as you've got a Weitek powerup. |
| 22:46.38 | mog_work | who cares what os it runs on |
| 22:46.40 | brad_mssw | stack_: some people say sangoma is slightly harder to set up, but their echo cancellation is 10x better |
| 22:46.41 | mog_work | just use what works |
| 22:46.49 | mog_work | 10x?!?! |
| 22:46.54 | stack_ | Lol, I've got an SGI O2 running OpenBSD, think it would run asterisk well? |
| 22:46.58 | mog_work | 99% of people use software echocancel |
| 22:47.00 | mog_work | it will be the same |
| 22:47.06 | mog_work | maybe stack_ |
| 22:47.18 | joeqread | k, just making sure there was no major issues with FBSD... I hate linux's dependency hell, like just installing perl requires about 30-generations of random, pointless libraries to get installed |
| 22:47.31 | Qwell[] | dependency hell? |
| 22:47.33 | brad_mssw | stack_: sangoma wiki here: http://sangoma.editme.com/ |
| 22:47.33 | mog_work | use debian and asterisk |
| 22:47.34 | Qwell[] | use a real distro... |
| 22:47.36 | mog_work | very little hell |
| 22:47.41 | asterboy | Is there a level 3 in Canada? |
| 22:47.42 | mog_work | much more fun |
| 22:47.46 | Qwell[] | gentoo ;] |
| 22:47.52 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 22:48.00 | joeqread | plus I've been out of practice with linux since pre-y2k |
| 22:48.02 | mog_work | gentoo isnt bad if you have time |
| 22:48.10 | mog_work | its not as scary as its made out to be |
| 22:48.11 | Qwell[] | well, if you're running fbsd, and use ports... |
| 22:48.11 | jsharp | If you use gentoo, though, you have to put the paint-can exhaust tips on the powersupply fan. |
| 22:48.12 | jets | lots of time |
| 22:48.29 | joeqread | yeah, always use ports for software I'm not comfortable tweaking myself |
| 22:48.39 | Qwell[] | well then, gentoo would be an easy jump |
| 22:48.43 | [av]bani | http://funroll-loops.org/ |
| 22:48.50 | joeqread | I'll install mysql myself, to get certain threading libraries working right, but otherwise ports work great |
| 22:50.14 | FuriousGeorge | we're gonna do a word association, i start, you guys just shout out whatever comes to mind... read... |
| 22:50.16 | *** join/#asterisk ToTo (n=ToTo@host214-134.pool872.interbusiness.it) |
| 22:50.17 | joeqread | is ztdummy a kernel module? |
| 22:50.25 | FuriousGeorge | ip-phone-cheaper-than-snom-320-with-PoE |
| 22:50.31 | asterboy | Level 3 is in Colorado, so I don't think they service Canada. |
| 22:50.38 | FuriousGeorge | joeqread: yes |
| 22:51.07 | joeqread | hmm, wonder if I run linux binary emulation if I can load it in freebsd |
| 22:51.14 | joeqread | does it need a specific kernel version? |
| 22:51.23 | asterboy | FuriousGeorge...an ebay snom-320? |
| 22:51.29 | asterboy | :P |
| 22:51.35 | FuriousGeorge | lol asterboy |
| 22:52.05 | FuriousGeorge | what about w/o poe. price to performance that sipura phone is pretty good |
| 22:52.06 | cpm | man, i love it when my folks put octothorpes in their mail folder names. |
| 22:52.07 | asterboy | It's the PoE part that is hard to beat. |
| 22:52.14 | FuriousGeorge | but it needs to have autoanswer, i cant recall if it does |
| 22:52.37 | joeqread | octothorpe? |
| 22:52.45 | asterboy | ~octothorpe |
| 22:52.47 | jbot | # |
| 22:53.04 | cpm | no wonder my scripts are breaking :) |
| 22:53.10 | masonf | do eveyone's zaptel cards ring once before they get to asterisk |
| 22:53.17 | joeqread | is that what a pound sign is called? |
| 22:53.30 | cpm | yup |
| 22:53.38 | joeqread | hah, never knew that |
| 22:53.39 | jsharp | masonf: If you're using an analog card, yes. |
| 22:53.41 | asterboy | in technical circles |
| 22:53.51 | joeqread | er, doesn't that break IMAP specs? |
| 22:54.06 | cpm | joeqread, no, it doesn't it's legal. |
| 22:54.34 | groogs | masonf: it does that to pick up caller id |
| 22:54.56 | joeqread | ah, you can't begin a folder name with it though I don't think |
| 22:55.38 | masonf | thanks all |
| 22:55.51 | asterboy | what are you guys paying for a pri line per month? |
| 22:56.04 | cpm | $500 |
| 22:56.10 | asterboy | USD? |
| 22:56.14 | joeqread | jesus |
| 22:56.16 | cpm | yup |
| 22:56.29 | asterboy | about the same here in Canada then |
| 22:56.30 | jsharp | $350, but we've got 24 of em. |
| 22:56.37 | asterboy | holy |
| 22:56.50 | asterboy | that's routing a lot of calls |
| 22:56.50 | kink0 | asterboy, 280 Euro/E1 |
| 22:56.51 | joeqread | we got 4, I think we only pay 240 |
| 22:57.07 | asterboy | Ya I have heard that Euro coms are far cheaper. |
| 22:57.11 | joeqread | no DID or ANI or anything though |
| 22:57.25 | jsharp | No, not routing a lot of calls. On a busy day, we use maybe 15 channels out of 1 PRI. |
| 22:57.44 | joeqread | so why do you need 24 pri's? |
| 22:57.46 | asterboy | man thats hardly scratching the surface |
| 22:58.04 | jsharp | But one of our customers paid big $$$ for us to guarantee 500 simultaneous calls. |
| 22:58.08 | Axel69 | those pri are for unlimited calls? |
| 22:58.32 | asterboy | usually just local |
| 22:58.38 | asterboy | unlimited local |
| 22:58.45 | cpm | mine are unlimited local. |
| 22:58.50 | cpm | but my local is pretty big |
| 22:58.57 | Axel69 | for an specific code or codes |
| 22:58.58 | jsharp | Yeah. We've got unlimited local into the Atlanta LATA. |
| 22:58.58 | cpm | njoy! |
| 22:59.06 | jsharp | plus LD through MCI. |
| 22:59.38 | joeqread | is there any way to hook asterisk into vonage? |
| 22:59.41 | Axel69 | and how much charge for all country calls |
| 22:59.43 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 22:59.46 | cpm | my lada is greater DC, baltimore, annapolis, down to just north of Richmond, Va. |
| 22:59.55 | Qwell[] | lata? |
| 23:00.08 | Zodiacal | anyone know of a way to disable the speaker on my phone when i use my sound card to page? |
| 23:00.27 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 23:01.24 | Axel69 | which hardwaare are u using for all the pri's? |
| 23:01.34 | Qwell[] | Axel69: digium |
| 23:01.41 | Qwell[] | oh, them, nm |
| 23:01.49 | Axel69 | all with asterisk? |
| 23:02.43 | jsharp | I've got all 24 pris coming into a Quintum CMS960, which then passes the calls to * |
| 23:03.05 | Zodiacal | anyone know if its posible for FOP to show parked calls? |
| 23:03.08 | Axel69 | none with cisco |
| 23:03.25 | jsharp | Some of the calls go from * to Cisco VG224s. |
| 23:04.41 | groogs | Zodiacal: yes |
| 23:05.08 | Zodiacal | groogs coolness |
| 23:05.15 | nextime | kink0 : 280 E in spain? |
| 23:05.21 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 23:05.23 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 23:05.57 | groogs | Zodiacal: you have to just add them all as extension=70 or whatever your parking ext is |
| 23:06.14 | Axel69 | great bussiness the voip |
| 23:06.20 | Axel69 | lol |
| 23:06.37 | Zodiacal | groogs so if i have a parking lot of 71-89 then i would add 18 exts at ext 70? |
| 23:06.45 | groogs | yep |
| 23:06.50 | Zodiacal | okie thank you! |
| 23:07.37 | Axel69 | how much charge if you call to another state from your pri's? |
| 23:08.37 | nextime | Axel69 : depend on the telco that give you the priu |
| 23:08.39 | nextime | pri |
| 23:08.55 | Axel69 | cost average? |
| 23:10.16 | kink0 | nextime: si |
| 23:10.21 | nextime | Axel69 : in italy i have a 0.01 for european destination with COLT and 0.008 for italian proper with FastWeb and 0.017 for european destination, but for international destination voip termination are cheaper |
| 23:10.24 | kink0 | nextime: yes |
| 23:10.34 | nextime | kink0 : si is ok for me :) |
| 23:10.46 | nextime | kink0 : same price in italy ( more or less ) |
| 23:10.57 | Axel69 | great |
| 23:11.02 | nextime | kink0 : and we have E1 pri, not T1, so 30 channels, not 24 :) |
| 23:11.30 | kink0 | nextime, here we are working around 0.005 for propper, and 0.065 for mobile |
| 23:11.44 | nextime | kink0 : with wich telco? |
| 23:11.45 | Axel69 | i have Guatemala VoIP for $0.11 if anyone need |
| 23:11.59 | nextime | Axel69 : guatemala for 0.11 is very expensive |
| 23:12.03 | palomiux | Axel? |
| 23:12.11 | Axel69 | for end users |
| 23:12.12 | palomiux | are you in Guatemala? |
| 23:12.19 | Axel69 | yes, i'm here |
| 23:12.30 | Axel69 | lol |
| 23:12.33 | Axel69 | yes, in guatemala |
| 23:12.59 | kink0 | nextime, France Telecom |
| 23:13.26 | nextime | Axel69 : i have a target of 0.05$ for guatemala... |
| 23:13.44 | Axel69 | can you sell to me? |
| 23:13.48 | De_Mon | asterisk is cutting off the first, ohh 200ms of audio when it answers a sip call. (Voicemail n stuff), what's the best way to correct this? |
| 23:13.58 | Qwell[] | De_Mon: polycom? |
| 23:14.07 | Qwell[] | seen a similar issue |
| 23:14.08 | De_Mon | sip phones? no xten |
| 23:14.36 | De_Mon | how'd you fix it on the polycom? |
| 23:14.51 | Qwell[] | try doing Answer(), then Playback(silence/1), then Voicemail() |
| 23:14.52 | palomiux | Axel, what company do you work for? |
| 23:15.09 | nextime | Axel69 : 0.05$ is my target, not the price that i can sell |
| 23:15.12 | Qwell[] | or make a 200ms silence file |
| 23:15.28 | nextime | Axel69 : anyeay, if you have a decent minutage we can discuss a good rate |
| 23:15.50 | De_Mon | ooh, a silence file, cool! |
| 23:16.11 | Axel69 | i work in my own company |
| 23:16.18 | nextime | kink0 : for dialout i use voip anyway, i use pri only for dialin, with premium or green numbers |
| 23:16.31 | nextime | ( green in italy is == tool free ) |
| 23:16.55 | Axel69 | i'm trying to do some end user mins...more proffit lol but i'm doing some Cell and proper |
| 23:17.11 | kink0 | nextime, we are ussing PRI also to voIP termination here, there actually only two PRI, one to proper and the other one for mobile. |
| 23:19.19 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 23:19.49 | nextime | kink0 : can i query you just a minute? |
| 23:19.56 | kink0 | nextime, sure |
| 23:19.58 | Qwell[] | kink0: yeah, me too... |
| 23:20.08 | Qwell[] | I need business cards. What's the turnaround time? :P |
| 23:20.17 | FuriousGeorge | hey, setting sip header on the spa841 works better for paging than the autoanswer on the snom 360. i dont theink the 360 supports it? |
| 23:20.48 | kink0 | Qwell[], ok |
| 23:21.25 | asterboy | ~ani |
| 23:21.30 | jbot | it has been said that ani is Automatic Number Identification Systems |
| 23:21.39 | Qwell[] | kink0: I was joking, of course |
| 23:24.14 | FuriousGeorge | anyone know if the snoms support this sip header paging? i cant see where they do but a firmare upgrade could fix that no? |
| 23:24.21 | Zodiacal | groogs its not showing the parked call, i have the parked extention there but it never changes when i park to it |
| 23:24.32 | fourcheeze | FuriousGeorge: they do |
| 23:24.41 | fourcheeze | I keep meaning to try it but haven't yet |
| 23:24.58 | fourcheeze | I think there's a page=on setting somewhere |
| 23:25.00 | FuriousGeorge | fourcheeze: thanks |
| 23:25.14 | fourcheeze | and you have to send your own sip header |
| 23:25.46 | groogs | Zodiacal: hm, i don't remember changing anything else. one sec, i think i did a patch for it so i'll look |
| 23:26.26 | fourcheeze | fourcheeze: I always support@snom.com quite help |
| 23:26.35 | fourcheeze | FuriousGeorge: ^ |
| 23:27.39 | fourcheeze | found out about putting address book onto ldap like that |
| 23:28.03 | groogs | Zodiacal: hm, yeah it looks like that's all i changed |
| 23:28.20 | Zodiacal | hrmm |
| 23:28.24 | Zodiacal | i'll keep playing with it |
| 23:28.33 | groogs | http://sourceforge.net/tracker/index.php?func=detail&aid=1173341&group_id=121515&atid=690574 |
| 23:32.06 | *** join/#asterisk BlacKNasH (n=BlacKNas@201.102.10.81) |
| 23:37.06 | *** part/#asterisk BlacKNasH (n=BlacKNas@201.102.10.81) |
| 23:38.34 | *** join/#asterisk tuxsoul (n=tuxsoul@dsl-201-129-224-248.prod-infinitum.com.mx) |
| 23:38.51 | tuxsoul | hello |
| 23:38.57 | tuxsoul | :-) |
| 23:40.24 | palomiux | hi tux |
| 23:40.48 | tuxsoul | i'm from méxico, sorry my english is poor :-P, i'm try buy in mexico hardware for asterisk, but not finding nothing, somebody know about reselles in méxico, |
| 23:40.54 | tuxsoul | palomiux: hi |
| 23:43.22 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 23:43.43 | _MartinCabrera_ | tuxsoul: yo compre en ifax.com, obtuve buenos precios y todo me ha llegado bien- |
| 23:43.51 | Juggie | tuxsoul http://www.voip-info.org/wiki-Asterisk+consultants+Mexico |
| 23:44.11 | Axel69 | Buy it in ebay and send it to mexico |
| 23:45.48 | tuxsoul | ok, muchas gracias, no sabia que hablan español :-D, thank's, ahorita mismo checo, pero no hay costos extras en los envios a mexico ? |
| 23:46.05 | tuxsoul | Juggie: thank's, i will check now ;-) |
| 23:47.11 | *** join/#asterisk Gamercjm (n=Gamercjm@pool-71-254-164-89.lsanca.fios.verizon.net) |
| 23:47.57 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:51.22 | tuxsoul | _MartinCabrera_: ok, muchas gracias, no sabia que hablan español :-D, thank's, ahorita mismo checo, pero no hay costos extras en los envios a mexico ? |
| 23:52.28 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 23:52.44 | *** join/#asterisk alephcom (n=darren@host75.net14.mcsnet.ca) |
| 23:55.02 | alephcom | hello everyone. |
| 23:55.12 | robin_sz | hello |
| 23:55.26 | tuxsoul | alephcom: hi |
| 23:59.45 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
