| 00:00.26 | twanny796 | segmentation fault |
| 00:00.27 | twanny796 | ?? |
| 00:01.20 | drmessano | bsod? |
| 00:01.24 | drmessano | snow crash? |
| 00:01.35 | drmessano | illegal exception? |
| 00:02.34 | drmessano | kernel panic? |
| 00:03.13 | *** join/#asterisk joobie (~joobie@CPE-124-181-130-3.vic.bigpond.net.au) |
| 00:03.50 | beardy | General Failure and his squadron of badgers. |
| 00:27.23 | mmlj4 | o/~ I'm a little segfault o/~ |
| 01:19.12 | xheliox | Why would you want to box a candy bar? |
| 01:19.13 | *** join/#asterisk ruied (~ruied@89.214.112.67) |
| 01:28.06 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 01:31.39 | ChannelZ | Tamer1111: sorry no I don't use GUIs for Asterisk. |
| 01:36.36 | Gershwin | there are a few presence, pbx and user status clients out there like switchboard, hud and cic... |
| 01:36.53 | Gershwin | anyone know of any others, eitehr commercial or free? |
| 01:37.36 | doolittlework | elastix works great for the beginer, but don't stray away from asterisk conf files |
| 01:37.38 | Gershwin | i think i saw NEC and maybe a couple others have software clients |
| 01:39.02 | Gershwin | that do server status (queues, calls, history, etc.) along user presence info |
| 01:56.48 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-34-135.nycmny.east.verizon.net) |
| 01:57.29 | KingDavidNYC | hello everybody |
| 02:06.28 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-34-135.nycmny.east.verizon.net) |
| 02:06.43 | KingDavidNYC | anybody here? |
| 02:06.49 | xheliox | Sure. |
| 02:06.55 | KingDavidNYC | hi1 |
| 02:07.32 | doolittlework | yes |
| 02:07.35 | KingDavidNYC | can anybody please recommend me a good website for jobs?... it is tough |
| 02:07.36 | doolittlework | whats up? |
| 02:07.52 | doolittlework | job doing what? |
| 02:08.08 | KingDavidNYC | asterisk, .net developer |
| 02:08.13 | KingDavidNYC | voip |
| 02:08.23 | KingDavidNYC | telecom |
| 02:08.25 | doolittlework | are you qaulified? |
| 02:08.31 | KingDavidNYC | very much |
| 02:08.59 | doolittlework | check out freelance.com |
| 02:09.01 | Gershwin | do you have a website? |
| 02:09.20 | Gershwin | ...something that details your qualifications, experience and sample work? |
| 02:09.32 | KingDavidNYC | no.. freelance.com... isn't that the one that people pay $50 to $250? |
| 02:09.49 | Gershwin | i've done a few odds and ends from craigslist |
| 02:10.02 | Gershwin | but we're talking $200 here, $50 there and all local |
| 02:10.08 | KingDavidNYC | I correct myself, that is freelancer.com |
| 02:10.13 | Gershwin | last time i did that was 2 years ago |
| 02:10.54 | KingDavidNYC | Craiglist has been good, I've found work for the last couple of years, but nothing in 2 months |
| 02:11.03 | doolittlework | you from new york KingDavidNYC? |
| 02:11.06 | KingDavidNYC | yes |
| 02:11.31 | doolittlework | what legacy system experience do you have |
| 02:11.35 | KingDavidNYC | I believe that asterisk, in and by itself, is no great paying job skill |
| 02:12.06 | Gershwin | maybe you've got dms, 5ess, 7ess or some fancy ivr/cti exp. |
| 02:12.09 | KingDavidNYC | legacy system?...the only pbx I have worked with is asterisk, and ksus |
| 02:12.33 | Gershwin | i think he meant legacy telephony platforms |
| 02:12.53 | Gershwin | switching, billing, scripting, etc |
| 02:13.01 | KingDavidNYC | you mean like avaya's and nortels,? none of those |
| 02:13.17 | KingDavidNYC | asterisk for about 5 years |
| 02:13.18 | Gershwin | ya, lucent, mitel, you name it |
| 02:13.26 | doolittlework | looking for a german supplier for philips systems |
| 02:13.55 | Gershwin | how is your understanding of SIP, IP networking, BGP, SBCs, routing, etc? |
| 02:14.14 | KingDavidNYC | and when it comes to telecom, everybody wants cisco CCVP |
| 02:15.02 | Gershwin | or vendor-specific applications thereof... Cisco, AdTran, Hatteras, AudioCodes, Patton, Quintum... |
| 02:15.14 | Gershwin | well, not necessarily |
| 02:15.47 | KingDavidNYC | I can do sip troubleshooting, and I have written parts of sip softphones, and even front panel displays |
| 02:16.56 | Gershwin | are you really looking for full-time employment or more for piecemeal consulting jobs where you have some flexibility? |
| 02:16.57 | KingDavidNYC | I am very good at programming the AMI |
| 02:17.11 | KingDavidNYC | I am looking for anything :) |
| 02:17.37 | KingDavidNYC | very good writing telecom billing systems |
| 02:17.41 | Gershwin | well, i guess a resume is something that should be priority 1... tho i'm sure you've already got one or 2 of those |
| 02:18.08 | Gershwin | if you wanted #2 (consulting)... you gotta have one or more really nice brochures |
| 02:18.15 | Gershwin | and/or a website |
| 02:19.04 | Gershwin | brochures and a good 5 or 10 sentence sales pitch with a few bullet points are enough to get you going on CL |
| 02:19.12 | KingDavidNYC | where I am getting at is that I believe I have spend the last 10 years working in voip/asterisk, but unfortunately I did not focused on legacy systems |
| 02:19.27 | Gershwin | not that big a deal |
| 02:19.34 | Gershwin | you're in a metro area |
| 02:19.51 | Gershwin | i'm sure there are 10,000 smb's out there right now looking for a person like you |
| 02:19.56 | KingDavidNYC | to be concrete: I thought I was going to make a lot of money with asterisk, and I dont think that will ever happen |
| 02:20.26 | Gershwin | ...if only for perhaps a few 1 - 4 hour consultation, planning, installation, configuration and support sessions |
| 02:20.50 | Gershwin | nope, world is your oyster my friend |
| 02:21.02 | Gershwin | you know more about voip than 99.99% of the population out there |
| 02:21.51 | KingDavidNYC | Gershwin: any other web sites where people are looking for consultants? |
| 02:21.56 | Gershwin | and there are thousands that would feel great about spending small $ (say, $35 - $75 / hr) to have a professional guide them in their decision making and deployment |
| 02:22.12 | Gershwin | first thing that came to mind is CL |
| 02:22.56 | Gershwin | but i guess a super-active forum where you have a "VoIP Expert Consulting" tag wouldn't hurt |
| 02:23.32 | Gershwin | i would have said guru.com years ago, but it's deserted these days |
| 02:23.36 | KingDavidNYC | great tips, specially that one about having my web site |
| 02:23.43 | Gershwin | i've heard a bunch of ads for angies list on the radio, but i've never visited there |
| 02:23.46 | raden | What are we talking about ? |
| 02:23.59 | Gershwin | he's looking for work |
| 02:24.01 | KingDavidNYC | yes!, I found a job onece in guru.com, but nowadays is dry |
| 02:24.14 | raden | KingDavidNYC, what kinda work you looking for ? |
| 02:25.06 | *** join/#asterisk Corydon76-dig (pink@c-69-137-80-31.hsd1.tn.comcast.net) |
| 02:25.06 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 02:25.07 | KingDavidNYC | voip, integrating web with asteisk, asterisk development |
| 02:25.11 | Gershwin | hey, KingDavidNYC, have someone review your resume and any brochures you create.... someone w/some marketing savvy |
| 02:25.24 | Gershwin | i'm sure you can dig up someone |
| 02:25.33 | raden | KingDavidNYC, that a hard line to get into |
| 02:25.52 | Gershwin | it's like saying, opening a fast food restaurant is a hard line to get into |
| 02:25.57 | xheliox | Those that can, usually don't have to ask where. Just my trolling 2 cents. |
| 02:25.59 | Gershwin | anything is hard |
| 02:26.22 | Gershwin | esp. when you're not motivated or don't have any experience with it |
| 02:26.24 | KingDavidNYC | raden: I agree with you... most work available is either depelover in pure .net or php or cisco |
| 02:26.33 | Gershwin | i know opening a fast food restaurant would be hard for me |
| 02:26.43 | xheliox | And am I the only one revolted by putting .net and Asterisk in the same sentence? :) |
| 02:26.53 | Gershwin | but i'm sure there are ppl out there that have been managing a macdonalds for 10 years that know most what needs to be done |
| 02:27.20 | Gershwin | use mono in place of .net and it'll make you feel better xheliox |
| 02:27.24 | raden | Gershwin, food business a breeze |
| 02:27.34 | raden | I could open a restraunt in 6 weeks given the cash |
| 02:27.41 | xheliox | Gershwin: Only by the slightest of margins. :) |
| 02:27.48 | KingDavidNYC | you know, actually I thougth that because not many people wrote asterisk to .net systems, there would be a market for it |
| 02:27.48 | Gershwin | i'd end up killing my first customer :) |
| 02:28.04 | raden | xheliox, I am as well |
| 02:28.19 | KingDavidNYC | after all, windows bases voip systems used to suck |
| 02:28.32 | Gershwin | ya, tell that to interactive intelligence |
| 02:28.48 | KingDavidNYC | but I guess time is proving me wrong |
| 02:29.19 | Gershwin | tho their prices have come down significantly in the past 2 yrs |
| 02:29.47 | Gershwin | still, it's a top-notch and very stable product |
| 02:29.49 | raden | I just wish i worked somewhere where i could use all my skills and improve on them :( |
| 02:30.04 | KingDavidNYC | raden: what are your skills? |
| 02:30.12 | Gershwin | who knows, maybe u 2 are neighbords |
| 02:31.49 | KingDavidNYC | and then everybody now has a gui for asterisk...basically, any computer installer can nowadays deploy asterisk |
| 02:35.31 | raden | Asterisk, PHP, MySQL, Javascript, Cisco, graphics, electronics design, and many others but thats where i try to focus |
| 02:35.41 | *** join/#asterisk Corydon76-dig (nine@c-69-137-80-31.hsd1.tn.comcast.net) |
| 02:35.41 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 02:37.43 | raden | KingDavidNYC, you have projects ? |
| 02:39.59 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 02:41.16 | KingDavidNYC | raden: yes, 2, but I have been begging those jerks to hurry up and give me the go ahead, and I dont want to wait for them anymore |
| 02:41.47 | *** join/#asterisk resno (~bryan@cpe-075-177-155-247.nc.res.rr.com) |
| 02:42.00 | *** part/#asterisk resno (~bryan@cpe-075-177-155-247.nc.res.rr.com) |
| 02:42.31 | Gershwin | might be helpful if you set up a few fs and sipx boxes |
| 02:42.40 | Gershwin | just to familiarize yourself |
| 02:42.58 | Gershwin | broaden your exp |
| 02:43.14 | KingDavidNYC | I've been trying to sell pbxs for years and I dont find customers |
| 02:44.05 | KingDavidNYC | once, I paid 15 thousand dollars to verizon for a colored ad in their yellow pages and I only rececived 3 phone calls in the entired year |
| 02:44.49 | KingDavidNYC | I mean, I am not negative, I am just trying to find out what is it that other people do right |
| 02:45.53 | Gershwin | you'd do better door-to-door than 3 friggin phone calls/yr |
| 02:46.21 | KingDavidNYC | I've done a lot of cold calling |
| 02:46.42 | Gershwin | get out in the burbs into industrial/commercial parks |
| 02:46.53 | KingDavidNYC | mmm...good idea |
| 02:47.13 | KingDavidNYC | you really mean door to door :) |
| 02:47.15 | Gershwin | gotta have a good brochure tho... or intro letter |
| 02:47.27 | Gershwin | ya, most of the time you'll be dropping shit off w/the secretary |
| 02:47.37 | raden | <PROTECTED> |
| 02:47.46 | Gershwin | nice! |
| 02:48.04 | KingDavidNYC | you do raden? I like that |
| 02:48.05 | raden | what I normally do for businesses starting up cold is write letters to there CFO or CEO's |
| 02:48.22 | raden | then follow up a week later |
| 02:48.29 | raden | this way you have something to talk about |
| 02:48.51 | raden | hello, yes I sent your CEO MR BLA a letter last week and I would Like to speak to him to make sure he recieved it |
| 02:49.04 | raden | this will get you past the gatekeeper AKA sexatery |
| 02:52.15 | Gershwin | i'm of the opinion that if you get any hook, you're persistent |
| 02:52.37 | Gershwin | i mean, you get any sort of feedback that they're interested, you don't stop giving/getting info until you get a yes or a no |
| 02:53.45 | KingDavidNYC | thanks man |
| 02:54.12 | Gershwin | e-mail just as good as a phone call imo too... tho maybe not for some who don't like to type |
| 02:54.38 | Gershwin | in that case, obvoiusly, call or face time better |
| 03:02.00 | ChannelZ | and if none of that works, offer him hookers |
| 03:04.06 | KingDavidNYC | ChannelZ: undoubtedly for jobs over 100K |
| 03:08.39 | raden | Gershwin, but it difficult to get addys for email sometimes :( |
| 03:14.42 | Gershwin | yup, i normally don't give mine out |
| 03:15.19 | Gershwin | i was referring to followups |
| 03:16.20 | Gershwin | i had a guy get my contact info from a sales call w/a another vendor and he called and left me vm 6 times last week |
| 03:16.49 | Gershwin | i did end up e-mailing him after his 2nd e-mail |
| 03:17.12 | Gershwin | and I do want to hear back from him if another thing doesn't work out |
| 03:17.20 | Gershwin | but I give him an A for persistence |
| 03:18.20 | Gershwin | and I have no doubt i'll be getting an e-mail or 2 and perhaps a voicemail or 2 from him next week whether i contact him or not |
| 03:18.39 | Gershwin | that's the kind of salesperson i want working for me |
| 03:26.47 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 03:40.08 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 03:47.45 | *** join/#asterisk radic (~radic@178.2.209.244) |
| 03:49.30 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 03:57.11 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 03:58.39 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 04:08.51 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:09.48 | Kyosh | i think i just made linux crash by setting 100 .call files at once. could asterisk really cause that? |
| 05:11.07 | Kyosh | 200 .call files :-p |
| 05:12.34 | Kyosh | yea the machine died |
| 05:15.29 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 05:31.53 | ChannelZ | hmm. |
| 05:41.48 | raden | Interesting ... |
| 05:41.52 | carrar | get a faster box |
| 05:42.03 | carrar | stop using old 386 servers |
| 05:48.13 | Gershwin | anyone ever heard of a product that's RFID based |
| 05:48.41 | Gershwin | that causes a user's extension to transfer to the telephone that is physicially nearest to them in a building? |
| 05:49.29 | WIMPy | What about giving them phones instead? |
| 05:50.05 | Gershwin | got me man... just thought it was cool when i saw it |
| 05:50.12 | drmessano | I know there's been some proximity stuff done with bluetooth headsets. But bluetooth makes me stabby sometimes |
| 05:50.46 | Kyosh | carrar: its a dual core xeon with 8gb ram. although i have to say linux is running in a vm :p |
| 05:50.57 | Kyosh | its prolly cause im running an old version of asterisk |
| 05:51.23 | drmessano | How old? |
| 05:51.30 | carrar | then it's not really what you describe |
| 05:51.34 | drmessano | and is the VM running in Windows 98? |
| 05:51.41 | Kyosh | hehe |
| 05:51.42 | Kyosh | no |
| 05:51.43 | carrar | probably have a missconfigured vm |
| 05:51.44 | Kyosh | esx |
| 05:51.51 | WIMPy | BT won't do handovers, I'm sure. |
| 05:51.54 | Kyosh | carrar:? not misconfigured |
| 05:52.00 | carrar | obviously |
| 05:52.08 | Kyosh | if you wanna think so, go ahead |
| 05:52.22 | Kyosh | its been working fine for about a year. this is the first time ive used .call files with it tho |
| 05:52.34 | drmessano | Running Asterisk 1.08 in VMware under Windows 95 says a lot more than "Asterisk crashes with 100 call files on a Dual Core Xeon with 8GB RAM" |
| 05:52.47 | carrar | heh |
| 05:53.02 | Gershwin | 3.11 |
| 05:53.15 | drmessano | Don't make the Asterisk After Dark crew pwn you like [TK]D-Fender on a GOOD day |
| 05:53.17 | Kyosh | i prolly should give the vm more memory |
| 05:53.24 | drmessano | Yes, maybe you should |
| 05:53.30 | drmessano | CAPS LOCK |
| 05:53.46 | carrar | should just dedicate the box to asterisk and dump the vm crap |
| 05:53.51 | Kyosh | is there a way to limit the number of concurrent calls the spooler will handle while using .call files? |
| 05:53.54 | carrar | then come back |
| 05:54.14 | drmessano | Kyosh: Yeah, drop the memory back until the spooler just crashes when you put in X number of files |
| 05:54.21 | Kyosh | i have 9 vm's running on that machine. cant run ast by itself on there |
| 05:54.25 | drmessano | We can call that a "limit" |
| 05:54.29 | drmessano | 9 VMs? |
| 05:54.36 | Kyosh | yup |
| 05:54.47 | carrar | 8vm's with a box that has 8gb of ram? |
| 05:54.49 | carrar | err 9 |
| 05:54.53 | drmessano | Dual Core Xeon and 8GB RAM and you have 9VMs? |
| 05:54.56 | drmessano | FAIL |
| 05:55.09 | Kyosh | not in the least, not if you do it right :) |
| 05:55.09 | carrar | all my vm servers have 32GB |
| 05:55.17 | drmessano | Obviously you havent |
| 05:55.26 | drmessano | Game, set, match |
| 05:55.36 | Kyosh | carrar:? this is only a dev server, sorry the boss wont give me a production level machine for development |
| 05:56.46 | Kyosh | i would like if asterisk didnt try to launch all 200 calls at the same time tho. maybe schedule them. hmm |
| 05:57.06 | carrar | thats entirely your call |
| 05:57.18 | drmessano | carrar: No, that's 100 calls |
| 05:57.22 | Kyosh | 200 calls |
| 05:57.23 | drmessano | Can you count? |
| 05:57.37 | carrar | You have 200 concurrent SIP channels avail? |
| 05:57.42 | drmessano | Now you're just being silly |
| 05:57.51 | drmessano | Do you even know 200 people? |
| 05:57.54 | carrar | haha |
| 05:58.08 | Kyosh | im test calling different hotels worldwide |
| 05:58.15 | Kyosh | and yes, i have 200 sip chans |
| 05:58.28 | drmessano | Are you an admin for Priceline.com? |
| 05:58.32 | drmessano | Do you know Shatner? |
| 05:58.38 | carrar | ENGAGE!! |
| 05:58.49 | carrar | err no thats captin picard |
| 05:59.38 | carrar | Who's providing you 200 channels of concurrent SIPness |
| 05:59.39 | drmessano | Are you the priceline administrator? |
| 05:59.44 | carrar | Just curious |
| 05:59.51 | drmessano | Do you know the priceline negotiator? |
| 06:00.29 | *** join/#asterisk DogBoy (~john@unaffiliated/dogboy) |
| 06:00.30 | drmessano | OMG, did you wire up Shatners blue phone? |
| 06:02.07 | carrar | anyways |
| 06:02.15 | carrar | that was a fun conversation |
| 06:02.27 | drmessano | Not really. I hated it. |
| 06:02.41 | carrar | You didn't read between the lines |
| 06:03.17 | Kyosh | <PROTECTED> |
| 06:03.19 | Kyosh | my company |
| 06:03.29 | Kyosh | apparently the scheduling works a bit better |
| 06:03.34 | Kyosh | still not what i was looking for |
| 06:03.35 | carrar | nice |
| 06:03.45 | carrar | DS3 of PSTN? |
| 06:04.04 | Kyosh | neither |
| 06:04.30 | carrar | Whats your gateway? |
| 06:04.39 | Kyosh | gateway? |
| 06:04.46 | Kyosh | what kind of provider? |
| 06:04.48 | carrar | SIP gateway |
| 06:04.50 | Kyosh | is that what you mean |
| 06:04.51 | Kyosh | ahh |
| 06:04.54 | Kyosh | umm |
| 06:04.57 | Kyosh | asterisk? |
| 06:05.01 | Kyosh | i dont follow you |
| 06:05.06 | rlankfo | heh |
| 06:05.08 | carrar | You said you are your own provider |
| 06:05.08 | Kyosh | you mean sip provider? |
| 06:05.13 | Kyosh | i did? |
| 06:05.25 | carrar | <Kyosh> <carrar> Who's providing you 200 channels of concurrent SIPness |
| 06:05.25 | carrar | <Kyosh> my company |
| 06:05.54 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 06:06.12 | carrar | Where do your calls go once they leave asterisk? |
| 06:06.17 | Kyosh | i never said my company is a provider |
| 06:06.17 | Kyosh | you asked who is giving me those channels |
| 06:06.18 | Kyosh | my company is |
| 06:06.21 | carrar | if are calling someone in PSTN LAND |
| 06:06.21 | drmessano | He's got unlimited SIP channels |
| 06:06.23 | Kyosh | they get it from our provider |
| 06:06.37 | drmessano | He never said "...that connect to the PSTN in any way" |
| 06:06.47 | drmessano | I have 1 billion IAX2 channels on my home system |
| 06:06.52 | drmessano | 0 concurrent |
| 06:07.34 | carrar | so back to my original question |
| 06:07.43 | carrar | <carrar> Who's providing you 200 channels of concurrent SIPness |
| 06:09.05 | carrar | I need to get a unlimited SIP Provider |
| 06:10.00 | carrar | sticks his finger in his mouth, 1 MILLION SIP CHANNELS |
| 06:10.08 | Kyosh | our sip provider is also our customer. we design their systems so we get a few hundred test channels |
| 06:12.13 | Kyosh | gawd these test calls are screwing my irc |
| 06:12.51 | carrar | How man BW you got? |
| 06:13.03 | Gershwin | prob on fiber |
| 06:13.09 | carrar | FIOS! |
| 06:13.23 | carrar | 200 calls is about 18 mbps |
| 06:13.25 | Gershwin | ya right, like they've got fiber in garden city |
| 06:13.26 | drmessano | Digital Subscriber Line |
| 06:13.35 | drmessano | DIGITAL |
| 06:13.42 | drmessano | 5x faster than dialup |
| 06:13.48 | carrar | 3G!! |
| 06:14.01 | Gershwin | you could stream video w/that |
| 06:14.18 | Kyosh | gershwin, actually fios is all over nassau |
| 06:14.39 | Gershwin | you could watch the youtubes with teh wifize |
| 06:14.56 | Kyosh | except south nassau. but i am not in garden city, sorry if the ip tracker is wrong |
| 06:15.03 | Gershwin | ya, i got at&t getting me a quote, tho i think it's gonna be too high |
| 06:15.13 | Kyosh | for dsl? |
| 06:15.20 | Kyosh | :p |
| 06:15.20 | carrar | AT&T is always too high |
| 06:15.33 | Gershwin | comcrap came in at 1200/mo |
| 06:15.34 | Kyosh | AT&T has nothing good |
| 06:15.44 | Kyosh | what bw are you looking for? |
| 06:15.50 | Gershwin | just 10meg |
| 06:15.54 | Kyosh | what area? |
| 06:15.57 | Gershwin | chicago |
| 06:16.12 | Gershwin | i already signed w/XO just a few days ago |
| 06:16.20 | Kyosh | comcrap's price is good, but its still comcrap |
| 06:16.37 | Gershwin | 10 meg on 7 bonded T's for 620/mo... nobody else could touch it |
| 06:16.42 | carrar | I have nothing good to say about comcast either |
| 06:16.48 | Kyosh | umm |
| 06:16.50 | Gershwin | same |
| 06:16.56 | Kyosh | 7xT1 for $620??? |
| 06:17.10 | Gershwin | yup... i couldn't believe it either...way undercut everyone else |
| 06:17.19 | Kyosh | that should be double the price |
| 06:17.25 | Gershwin | told my boss... it's a no brainer, sign it |
| 06:17.32 | Kyosh | i cant see how they are doing under $100 per T |
| 06:17.45 | carrar | same CO? |
| 06:17.52 | Kyosh | 0 mile? |
| 06:17.54 | Gershwin | ya, lowest anyone else could get me was over double |
| 06:18.01 | Kyosh | im quite sure |
| 06:18.12 | Gershwin | no, but loop cost is just to next town |
| 06:18.29 | Gershwin | i think their loop cost is $90 per pair |
| 06:18.37 | Gershwin | but i'm not sure, i'm not in that business |
| 06:19.03 | Kyosh | is it $620 for the T's including bw and loops or just bw? |
| 06:19.21 | Gershwin | just bandwidth... but we're not paying loop costs |
| 06:19.26 | Kyosh | doh |
| 06:19.33 | Kyosh | why no loop costs? |
| 06:19.34 | Gershwin | 3 year deal too |
| 06:19.53 | carrar | They doing a trade? |
| 06:19.55 | Gershwin | i'm guessing because it's 3 year, dunno |
| 06:19.59 | Kyosh | nopes |
| 06:20.01 | Kyosh | always a loop cost |
| 06:20.08 | carrar | servicing other people with your location? |
| 06:20.14 | Gershwin | nope |
| 06:20.21 | Gershwin | just us |
| 06:20.29 | Kyosh | so its $62/Mbps which is not really good, but since there's no loop cost, its amazing! |
| 06:20.48 | Kyosh | the loop cost usually starts at $75/Mbps |
| 06:20.50 | Gershwin | i agree |
| 06:20.57 | Kyosh | so i wouldnt bitch |
| 06:21.08 | Kyosh | id just make sure the contract doesnt have a snag in there for loop costs |
| 06:21.11 | Gershwin | how much do they pay at&t for each loop? is $90 close? |
| 06:21.26 | Kyosh | who? XO? |
| 06:21.35 | Gershwin | yes, or any clec/ilec |
| 06:21.47 | Kyosh | gimme a nearby address and i'll get you a price in 90 secs |
| 06:22.05 | Kyosh | company writes management software for isp's |
| 06:22.16 | Gershwin | i know their POP is on Jorie Boulevard in Oakbrook, IL |
| 06:22.37 | drmessano | I use A2Billing on Trixbox for my ITSP and it works great |
| 06:22.40 | Gershwin | hmmm, let me find a nearby address because my addy is a brand new building |
| 06:22.47 | Gershwin | that's not going to be in your database |
| 06:23.05 | Gershwin | might be Jorie lane, i forget the name of that street |
| 06:23.16 | Kyosh | even a place around the corner would be fine |
| 06:25.29 | Gershwin | here we go... macdonalds nearby |
| 06:25.31 | Gershwin | Route 83 & Mockngbird Ln |
| 06:25.31 | Gershwin | Willowbrook, IL 60527 |
| 06:25.45 | carrar | quarter pounder pls |
| 06:26.00 | Gershwin | here's a GM link: http://maps.google.com/maps?f=q&source=s_q&hl=en&q=macdonalds,&sll=41.718991,-87.948303&sspn=0.044269,0.071068&ie=UTF8&t=h&radius=2.2&split=1&rq=1&ev=zi&hq=macdonalds,&hnear=&ll=41.718159,-87.948303&spn=0.04427,0.071068&z=14 |
| 06:26.05 | Kyosh | no bldg address? |
| 06:26.17 | Gershwin | i dunno... let me look for some real address |
| 06:27.23 | carrar | nice pond |
| 06:28.13 | Gershwin | here we go: 16W135 Honeysuckle Rose Ln, Hinsdale, IL |
| 06:29.29 | Gershwin | here's another address... decent restaurant too: |
| 06:29.31 | Gershwin | 10S656 Kingery Hwy |
| 06:29.31 | Gershwin | Willowbrook, IL 60527-0730 |
| 06:29.51 | Kyosh | 16W135 Honeysuckle Rose Ln, Hinsdale, IL - not on any existing network |
| 06:30.25 | Gershwin | ya, it's prob. got to do w/those wierd direction-embedded addresses |
| 06:31.21 | Kyosh | 10S656 Kingery Hwy |
| 06:31.27 | Kyosh | $1200/mo 10M |
| 06:31.29 | carrar | How about |
| 06:31.30 | carrar | 8802 Illinois 83, Willowbrook, IL 60527 |
| 06:31.56 | carrar | or 8806 |
| 06:32.07 | Kyosh | but T1's are $89 loop carrier cost |
| 06:32.33 | Kyosh | carrar: T1 or 10M? |
| 06:32.43 | Gershwin | what's their recurring monthly cost for the loop? like $12? |
| 06:32.47 | carrar | no thats just a building |
| 06:32.49 | Kyosh | fuk you guys aint got EoC over there |
| 06:32.51 | Kyosh | sad |
| 06:33.09 | Gershwin | yup, called around to everyone and their mother to see if I could get EoC |
| 06:33.23 | Gershwin | the ONLY ones who said they might be able to do it is |
| 06:33.24 | Kyosh | ATT sucks |
| 06:33.28 | Gershwin | Paetec and XO |
| 06:33.36 | Kyosh | nopes,t hey cant |
| 06:33.36 | Gershwin | and the ATT guy never got back to me on it |
| 06:33.53 | Kyosh | ATT controls the lines and paetec/xo has to go thru them |
| 06:33.57 | Gershwin | and XO said they couldn't because their Hatteras boxes wont' do the distance |
| 06:34.24 | Kyosh | its all based on the ilec and what they have available |
| 06:34.29 | Gershwin | we're something like 25K from the CO and their Hatteras boxes go max to something like 12 or 13 or 15--can't remember |
| 06:34.45 | Kyosh | <carrar> 8802 Illinois 83, Willowbrook, IL 60527 |
| 06:34.45 | Kyosh | <PROTECTED> |
| 06:35.01 | Kyosh | hatts go 12k |
| 06:35.01 | Gershwin | well, the SE I had in the office said the circuit was the same... a standard alarm circuit--unconditioned bare copper... 5 pairs |
| 06:35.07 | carrar | What, just finding another address near him |
| 06:35.12 | Kyosh | oo |
| 06:35.29 | Gershwin | i've never worked with 'em, so ya go me |
| 06:35.33 | Kyosh | yea ATT is robbing clecs for interconnects |
| 06:35.41 | Gershwin | there is a lit building less than 1/2 mile away from our new location |
| 06:35.43 | Kyosh | att is nutorious for that |
| 06:35.52 | Gershwin | but that's right next to a national laboratory |
| 06:36.03 | Gershwin | so i'm thinking there's an att node right there |
| 06:36.18 | carrar | They can shoot some microwave over to ya |
| 06:36.24 | Gershwin | nope |
| 06:36.39 | Gershwin | we're in this goofy ass miniscule little valley |
| 06:36.55 | Kyosh | you can set up some private LOS wireless |
| 06:36.56 | Gershwin | prob 1/4 mi wide by 1/4 mi long... can't see nothin' :( |
| 06:37.11 | carrar | Nothing a 500' tower can't fix!! |
| 06:37.17 | Gershwin | heh |
| 06:37.40 | Gershwin | i'm sure the residential development next door to our commercial development would love that |
| 06:37.59 | Gershwin | "oh look honey, what a lovely tower" |
| 06:38.10 | Kyosh | put it on top of a building |
| 06:38.19 | Kyosh | roof rights rocks |
| 06:38.33 | Gershwin | i asked our current wisp if there was anythign he could do |
| 06:38.40 | Gershwin | and they did a survey.... |
| 06:38.42 | Kyosh | get a 100mbp line from xo, set up some ubiquity repeaters for los then sell it off |
| 06:38.43 | Gershwin | NO dice |
| 06:38.54 | Kyosh | whos the local wisp? |
| 06:39.03 | Gershwin | local company... BoB |
| 06:39.10 | Kyosh | bah |
| 06:39.11 | Kyosh | of course |
| 06:39.14 | Kyosh | BoB |
| 06:39.17 | Gershwin | called 2 otehrs... same answer, no los in that goofy valley |
| 06:39.27 | carrar | better move |
| 06:39.32 | Kyosh | unless you centralize your own center point |
| 06:39.36 | Gershwin | bob used to be just across teh street from our headquarters |
| 06:39.47 | Gershwin | they moved a few towns away tho |
| 06:40.14 | Kyosh | whats the mileage in radius of the valley? |
| 06:40.28 | Gershwin | it's tiny, i dunno |
| 06:40.52 | Kyosh | 50 miles from center to edge? |
| 06:40.55 | carrar | Looks like hughes satellite service is your only choice :) |
| 06:42.05 | Gershwin | we're going to do standard business cable w/comcast |
| 06:42.12 | Gershwin | for our backup |
| 06:42.19 | Gershwin | be nice to have 3 tho |
| 06:42.37 | Gershwin | comcast reliablility is the worst of all our isps |
| 06:43.14 | Gershwin | tho it's only down 4 or 5 times a year, some of those outages can be 12 hrs |
| 06:43.30 | Kyosh | ack thats bad |
| 06:44.12 | Gershwin | ya, we only need to service < 50 trunks, so we should be ok |
| 06:44.20 | Gershwin | for now |
| 06:52.00 | Gershwin | I see hatteras has repeaters that can go over 25 mi on copper |
| 06:52.31 | Gershwin | http://www.hatterasnetworks.com/default.aspx?pagename=Repeater_and_Line_Power_Module |
| 06:53.16 | Gershwin | shoulda looked that up before... tho for the price we got, guess it doesn't matter |
| 07:07.10 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 07:10.54 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 07:12.34 | Gershwin | hmmm, AdTran has a device that'll do 720 Mbps EoC |
| 07:12.50 | Gershwin | 16 pairs tho |
| 07:13.15 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 07:14.36 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:21.27 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 07:42.50 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 08:05.12 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 08:09.31 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 08:11.16 | *** join/#asterisk mifadir (~mifadir@dynamic.casap1-184-30-137-41.wanamaroc.com) |
| 08:11.31 | mifadir | how to make video conference with asterisk |
| 08:19.22 | mifadir | any answer?? |
| 08:22.01 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 08:41.22 | *** join/#asterisk mpe_ (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 08:45.34 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 09:07.36 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 09:37.57 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 09:49.08 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 09:57.05 | *** join/#asterisk Abd4llA (~Abd4llA@196.219.96.141) |
| 09:57.44 | Abd4llA | hi everyone, I'm trying to find the required table structre for ARA "asterisk reatime" in static mode |
| 09:58.07 | Abd4llA | I can't find doc/extconfig on 1.6 |
| 10:04.18 | tzafrir | Abd4llA, the same as for the non-static case, IIRC |
| 10:06.30 | *** join/#asterisk BANSAL (~bansal@117.199.113.67) |
| 10:38.36 | BlackBishop | ca .. arde .. 40 de grade'i febra .. stam batuti de soare .. ca niste rechini de stepa .. si .. arde uite cum arde .. arestat o vodca .. am condamnat un joint la moarte ! \:D/ |
| 10:39.22 | BlackBishop | woups |
| 10:39.24 | BlackBishop | wrong window |
| 10:45.48 | *** join/#asterisk Abd4llA (~Abd4llA@196.219.96.141) |
| 10:46.03 | Abd4llA | tzafrir: yea idd, so any idea where to get that info ? |
| 11:06.45 | *** join/#asterisk bmg505 (~leon@196-209-120-126.dynamic.isadsl.co.za) |
| 11:07.40 | bmg505 | sorry for the spam yesterday, south african internet isps :( |
| 11:13.42 | Abd4llA | Anyone knows where can I find the required table structure for configure asterisk realtime in static mode? |
| 11:21.28 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 11:21.53 | tzafrir | Abd4llA, contrib/realtime/ ? |
| 11:31.16 | RypPn | ~book |
| 11:31.17 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 11:31.26 | *** join/#asterisk sylar (~sylarrrr@bzq-79-177-48-232.red.bezeqint.net) |
| 11:55.54 | Abd4llA | zane93: are those for static config ? don't think so |
| 11:56.02 | Abd4llA | if so they're not complete |
| 11:56.36 | Abd4llA | at least the mysql |
| 12:09.08 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 12:11.36 | *** join/#asterisk ruied (~ruied@62.28.165.222) |
| 12:15.08 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 12:52.54 | *** join/#asterisk sgimeno (~chatzilla@95.122.11.99) |
| 13:19.59 | *** join/#asterisk Skten (Nick@118.11.233.220.static.exetel.com.au) |
| 13:20.30 | Skten | Evening all. Anyone familiar with the radius cdr module? |
| 13:34.32 | *** join/#asterisk imox1234 (~imox1234@p4FC5C303.dip0.t-ipconnect.de) |
| 13:41.59 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:42.21 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:45.24 | *** join/#asterisk ruied (~ruied@62.28.165.222) |
| 13:57.01 | *** join/#asterisk riddlebox (~james@75-132-216-43.dhcp.stls.mo.charter.com) |
| 14:13.12 | Skten | I realise that Asterisk is and always will be a PABX, or more so a platform to develop telephony applications. Subsequently the focus is always going to be on the dialplan logic, the core SIP stack etc. But is it just me or is there an increasing amount of crap that either doesn't work, is buggy or just isnt complete? I also realise its pretty dead in here but I need to vent :-) |
| 14:17.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 14:23.55 | pabelanger-lap | Skten: such as? |
| 14:26.30 | Skten | Logged a bug the other day for TOS support in chan_sip, need to log a bug after doing a bunch of testing / packet captures regarding music on hold handling + invites being parsed incorrectly where NAT is involved, I can't find a straight answer on how the CDR Radius config is supposed to look as the documentation is all over the shop, to do things like disable transfers period the source needs to be modified |
| 14:27.51 | Skten | I realise these are not things that would be common to the typical Asterisk user - but Im getting frustrated that things are being included in builds presumably on the basis that a specific user requires them, and then they are screwing up, however no one except the original contributor really knows what is going on |
| 14:29.00 | russellb | you sure are high and mighty this morning |
| 14:31.16 | Skten | I know, I need to get my knickers out of a knot. I've just been left with a sour taste in my mouth after finally being able to move a bunch of stuff from OpenSIPs to Asterisk, and now Im running into a whole new set of issues. Its not a go at the dev's listening, sorry if thats how its come across. |
| 14:31.43 | *** join/#asterisk bintut (~bintut@cm11.kappa11.maxonline.com.sg) |
| 14:31.47 | bintut | waves |
| 14:32.04 | bintut | i am now getting this sip message: SIP/2.0 488 Not Acceptable Here |
| 14:32.28 | Skten | I am documeting these problems as I go though, and specifically intend to fix the voip-wiki inconceptions Im seeing once I have some time on my hands. |
| 14:32.49 | bintut | i'm running Asterisk 1.4.29.1 |
| 14:33.02 | Skten | *misconceptions |
| 14:34.48 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 14:34.54 | mattwj2002 | hi guys |
| 14:34.54 | Skten | Need more info bintut - when are you seeing that? |
| 14:35.12 | bintut | anyone's familiar on this message and on how i can fix this? the sip user is from the remote and is registered as per "sip show peers" |
| 14:35.24 | mattwj2002 | I am sorry first off for an offtopic question but I was hoping you guys could help me out with something |
| 14:35.27 | [TK]D-Fender | bitCodec mismatch <- |
| 14:35.34 | [TK]D-Fender | bintut: Codec mismatch <- |
| 14:35.49 | bintut | Skten: just now |
| 14:36.01 | mattwj2002 | anyone know of a good place online for phone cards.....my dad is in the hospital :( |
| 14:36.02 | Skten | *face palm* |
| 14:36.13 | bintut | [TK]D-Fender: meaning, the remote sip user is not using the same codec as on what the asterisk is expecting? |
| 14:38.07 | [TK]D-Fender | bitUnable to agree on anything |
| 14:38.14 | Skten | bintut: Meaning neither side is listing support for a codec that the other side supports. |
| 14:38.19 | [TK]D-Fender | bintut: Start by actually looking at the SIP debug <- |
| 14:39.03 | bintut | [TK]D-Fender: i did a debug already. the remote sip user is using x-lite |
| 14:39.10 | bintut | Skten: checking.. |
| 14:40.00 | *** join/#asterisk DogBoy_ (~john@unaffiliated/dogboy) |
| 14:47.49 | bintut | this is my sip debug output => http://www.pastie.org/1069663 |
| 14:47.55 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 14:54.51 | Skten | bintut: What Codecs do you have enabled in xLite? Your INVITE says you support ULAW + GSM only. Do you have one of these enabled in Xlite? (From memory I didn't think Xlite supported anything other than g711u) |
| 14:55.23 | bintut | Skten: isn't it ulaw == g.711u ? |
| 14:55.25 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 14:55.38 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 14:55.39 | Skten | bintut: Yes. |
| 14:56.10 | bintut | Skten: x-lite supports ulaw so it should be able to work |
| 14:57.26 | Skten | bintut: Yes but do you have it enabled in xlite? What happens when you make a call from xlite into the asterisk box - what codecs show as supported in the invite? |
| 14:57.51 | bintut | let me check |
| 15:04.55 | bintut | hhmmm |
| 15:05.10 | bintut | Skten: i'm sorry, the user who was testing remotely left already.. |
| 15:05.30 | bintut | it's already 23:05H here.. :( |
| 15:05.51 | bintut | thank you for your help, Skten and [TK]D-Fender |
| 15:08.11 | *** join/#asterisk fofware (fabian@190.137.125.220) |
| 15:09.28 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 15:19.09 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 15:20.04 | *** join/#asterisk modsaid (~modsaid@196.218.219.102) |
| 15:20.20 | modsaid | Greetings Everyone |
| 15:20.52 | modsaid | I'm new to asterisk, setting it up now to try it as a pure VoIP server. I assume i do not need any specific hardware.. right ? |
| 15:21.34 | pabelanger-lap | modsaid: Unless you want to interface to a PSTN (IE: Bell) |
| 15:21.42 | *** join/#asterisk lirakis (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 15:21.46 | pabelanger-lap | But yes, pure VoIP no hardware needed |
| 15:21.47 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 15:22.12 | modsaid | thanks guys |
| 15:22.17 | *** join/#asterisk lirakis (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 15:24.27 | Gugge | pabelanger: i would think you need at least one computer to run asterisk, and somewhere to run a couple of soft phones |
| 15:25.55 | pabelanger-lap | Gugge: Of course, but that is not specific hardware to asterisk |
| 15:27.45 | modsaid | :) |
| 15:27.49 | modsaid | thanks Cugge |
| 15:28.07 | modsaid | My plan is, installing asterisk on a linux machine (probably debian or ubuntu) |
| 15:28.26 | modsaid | and using it for VoIP conversation with VoIP client from 2 other computers |
| 15:38.36 | mmlj4 | ~book |
| 15:38.37 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:38.44 | mmlj4 | modsaid: go read that |
| 15:39.58 | modsaid | the pdf book |
| 15:40.11 | modsaid | mmlj4: I got that and going through it now... Thanks |
| 15:42.21 | *** join/#asterisk Shaaan (~Un1x@CPE000024cccb7c-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 15:48.40 | *** join/#asterisk Alagar (~Administr@122.164.36.94) |
| 16:23.38 | *** join/#asterisk Tim_Toady (~moi@178.128.17.211.dsl.dyn.forthnet.gr) |
| 16:44.04 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 16:54.29 | *** join/#asterisk dustybin (dustybin@wizbox.org) |
| 16:54.47 | dustybin | anybody know how to create a suitable .wav file for asterisk with the newer version of sox? |
| 16:55.00 | dustybin | i keep on getting this error msg |
| 16:55.00 | dustybin | sox WARN sox: effect `resample' is deprecated; see sox(1) for an alternative |
| 16:55.25 | dustybin | could this be the work of a virus / malware / spyware ? |
| 16:56.05 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 17:07.04 | tzafrir | dustybin, what command do you run? |
| 17:09.25 | dustybin | i found some info about it http://corelatus.wordpress.com/2009/02/19/sox-parameters-for-downsampling-to-8khz-alaw/ |
| 17:09.46 | dustybin | those commands dont seem to work either |
| 17:10.27 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 17:21.36 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:24.28 | ChannelZ | sox inputfile.ext -b 16 -c 1 -r 8000 outputfile.wav |
| 17:25.39 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:25.47 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 17:31.55 | dustybin | ChannelZ: THAT WORKS!!!!!!!!!!!!!!!!!!!!!!!!!!!! THANKS :D |
| 17:32.06 | dustybin | feels really excited |
| 17:34.01 | ChannelZ | not sure what it didn't like about those samples on the web page, they looked fine to me.. |
| 18:07.41 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:10.48 | *** join/#asterisk Tim_Toady (~moi@178.128.17.211.dsl.dyn.forthnet.gr) |
| 18:16.46 | *** join/#asterisk MiserySoft (~Lee@nat67.mia.three.co.uk) |
| 18:23.16 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 18:31.53 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 18:32.37 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 18:55.49 | *** join/#asterisk sylar (~sylarrrr@bzq-79-177-48-232.red.bezeqint.net) |
| 18:59.24 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 19:09.28 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 19:14.23 | *** join/#asterisk philipp64|laptop (~chatzilla@64.134.128.79) |
| 19:25.02 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 19:25.16 | *** join/#asterisk Z_God (~julius@wlan237116.mobiel.utwente.nl) |
| 19:30.04 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 19:30.10 | *** part/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 19:47.30 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 19:47.54 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 20:32.14 | *** part/#asterisk BlackBishop (dexter@d3xt3r01.tk) |
| 20:42.09 | *** join/#asterisk aurix (~aurelio@81.174.13.196) |
| 20:46.39 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 20:50.38 | *** join/#asterisk _zoom_ (~user@196.29.186.102) |
| 20:51.02 | _zoom_ | hi, how to specify rtp ports for each sip channel? |
| 20:52.43 | pabelanger-lap | rtp.conf |
| 20:54.11 | _zoom_ | pabelanger-lap: per sip channel? |
| 20:54.43 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 20:57.20 | drmessano | Why would you do that? |
| 21:06.32 | carrar | Because |
| 21:07.06 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 21:07.32 | *** join/#asterisk chuckz (~lechuck@93-40-114-140.ip38.fastwebnet.it) |
| 21:08.30 | carrar | _zoom_, just hack up the source to add that feature |
| 21:09.10 | carrar | #DEFINE_UNIQUE_RTP_PORTS_PER_SIP_CHANNEL=ENABLE |
| 21:09.12 | *** part/#asterisk dustybin (dustybin@wizbox.org) |
| 21:09.18 | tzafrir | Those channels are allocated dynamically. Managing them on your own won't be fun |
| 21:11.24 | carrar | then in rtpports_to_ip_match.conf, just add the ports to use per matching destination ip range :) |
| 21:14.08 | tzafrir | carrar, again, why do you need to set the port numbers? |
| 21:14.17 | carrar | <_zoom_> hi, how to specify rtp ports for each sip channel? |
| 21:14.21 | carrar | I don't |
| 21:14.48 | tzafrir | oops, sorry :-) |
| 21:15.03 | carrar | I was trying to be humorus |
| 21:15.08 | carrar | FAIL |
| 21:15.10 | carrar | heh |
| 21:15.31 | tzafrir | let's hope he'll zoom in again |
| 21:27.16 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 21:27.40 | boodu | hello |
| 21:28.46 | *** join/#asterisk DelphiWorld (~Delphi@41.200.16.77) |
| 21:28.47 | DelphiWorld | hi |
| 21:28.52 | DelphiWorld | any knowne open stun server? |
| 22:03.35 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 22:04.33 | xSmurf | hey all, fooling around with AGI, how do I select the language for audio streaming? agi_language doesn't seem to reflect the calling extension's setting... |
| 22:07.23 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 22:07.59 | kerframil | DelphiWorld: counterpath have one |
| 22:08.52 | nextime | hello. Asterisk 1.6.9. I have a sip DID with a local voip provider. If i try to do a Record over an incoming call, it ever drop after 45 sec of recording, i have rtpkeepalive at 20 seconds, but this doesn't solve the problem. Any idea on what can cause the calldrop even with the rtpkeepalive? |
| 22:15.29 | kc8pxy | do i need to set nat for any sip channels if some sip clients are on the lan, and some on teh internet, and the asterisk server is on the gateway system? |
| 22:18.21 | DelphiWorld | kerframil: open? |
| 22:18.58 | kerframil | DelphiWorld: as far as I'm aware. its hostname is stun.counterpath.net. |
| 22:20.15 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 22:21.20 | DelphiWorld | pyite: ;) |
| 22:23.33 | kc8pxy | i don't need to add nat to a sip channel definition if all the phones can get to teh server /wo snat/dnat rules on the firewall, yes? |
| 22:28.54 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 22:29.03 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 22:31.04 | *** join/#asterisk mutineer612 (~mutineer6@68-117-44-195.dhcp.roch.mn.charter.com) |
| 22:31.44 | mutineer612 | Anyone running Switchvox in a virtual environment? |
| 22:31.53 | mutineer612 | VMWare ESXi or XEN? |
| 22:32.29 | nextime | nothing to do, even if i set rtpkeepalive to 1 second, itdrop the call exactly after 45 seconds from when i start Record() |
| 22:32.41 | *** join/#asterisk _structz (~structz@201-42-156-238.dsl.telesp.net.br) |
| 22:32.53 | _structz | Hello All, can somebody give me a help |
| 22:33.49 | _structz | i have asterisk 1.6.2.9 working with mysql realtime(sippeers,sipusers and voicemail) sip.conf ras rtcache = yes autoclear=no adn rtupdate=yes, but somehow from time to time rtcache keeps gettin flushed and some fones go to voicemail :( |
| 22:36.40 | *** join/#asterisk saisoma (4aa7881e@gateway/web/freenode/ip.74.167.136.30) |
| 22:38.03 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 22:39.27 | saisoma | Hey guys, question about queues. Customers are complaining that they are in position #2 and then back to #5. Any ideas? Here is the config: http://pastebin.com/mgB25atG |
| 22:40.19 | saisoma | We are using fop2 for monitoring and are NOT using agent logins, so unmanned phones are being rung and not answered. Does that then shift the caller back to the end of the line? |
| 22:42.26 | _structz | by the way that happens when i sip reload |
| 22:44.05 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 22:44.24 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:45.39 | nextime | ok |
| 22:45.55 | nextime | setting transmit_silence_during_record=yes in asterisk.conf [options] solved my issue |
| 22:48.27 | _structz | anyone ? |
| 22:51.36 | mutineer612 | Sorry _structz, I have no idea. This room is pretty quite. Not a very helpful room... I took a quick look, but did not find anything helpful for you. |
| 22:51.49 | _structz | yes |
| 22:51.51 | _structz | its strange |
| 22:52.07 | _structz | the phones registers and works just fine after a while the phones go to voicemail |
| 22:52.28 | _structz | like if the phone unregistered but no messages at asterisk console |
| 22:52.34 | *** part/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 22:53.16 | Gugge | _structz: sip reload flushes the cache |
| 22:53.44 | _structz | the flushing the cache causes this event that i'm having? |
| 22:53.58 | _structz | phones going to voicemail cuze there aren't in the cache table |
| 22:54.01 | Gugge | no |
| 22:54.16 | Gugge | asterisk should load it from the realtime db whenever it needs it |
| 22:54.26 | Gugge | check of the right info is in the db when it happens |
| 22:56.34 | _structz | it ain`t |
| 22:56.46 | _structz | that info keeps getting flushed i guess because the phone says its registered |
| 22:56.52 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 22:57.59 | Gugge | so something changes regseconds in the db? |
| 22:58.44 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:59.07 | _structz | name 6919 |
| 22:59.08 | _structz | <PROTECTED> |
| 22:59.08 | _structz | <PROTECTED> |
| 22:59.08 | _structz | <PROTECTED> |
| 22:59.08 | _structz | <PROTECTED> |
| 22:59.08 | _structz | <PROTECTED> |
| 22:59.10 | _structz | <PROTECTED> |
| 22:59.12 | _structz | <PROTECTED> |
| 22:59.14 | _structz | <PROTECTED> |
| 22:59.16 | _structz | <PROTECTED> |
| 22:59.18 | _structz | <PROTECTED> |
| 22:59.20 | _structz | <PROTECTED> |
| 22:59.22 | _structz | <PROTECTED> |
| 22:59.24 | _structz | <PROTECTED> |
| 22:59.26 | _structz | name 6919 |
| 22:59.28 | _structz | <PROTECTED> |
| 22:59.30 | _structz | <PROTECTED> |
| 22:59.32 | _structz | <PROTECTED> |
| 22:59.34 | _structz | <PROTECTED> |
| 22:59.38 | _structz | <PROTECTED> |
| 22:59.40 | _structz | <PROTECTED> |
| 22:59.42 | _structz | <PROTECTED> |
| 22:59.44 | _structz | <PROTECTED> |
| 22:59.46 | _structz | <PROTECTED> |
| 22:59.48 | _structz | <PROTECTED> |
| 22:59.50 | _structz | <PROTECTED> |
| 22:59.52 | _structz | <PROTECTED> |
| 22:59.54 | _structz | <PROTECTED> |
| 22:59.56 | _structz | just makin a test, set the phone to reboot and register |
| 22:59.58 | _structz | its in the cache (sip show peers) |
| 23:00.00 | _structz | going to make a sip reload |
| 23:00.12 | Gugge | _dont_ paste all that stuff in here |
| 23:00.17 | _structz | sorry |
| 23:00.29 | cosmicwombat | use pastebin |
| 23:00.32 | _structz | i past it twice wasn`t suppost to do too much lines |
| 23:00.35 | _structz | sorry |
| 23:02.20 | Gugge | _structz: check if regseconds is updated in the db when you register a phone |
| 23:02.24 | *** join/#asterisk philipp64|laptop (~chatzilla@63.81.41.227) |
| 23:02.34 | _structz | it does |
| 23:02.41 | _structz | as you can see in the past |
| 23:02.46 | _structz | let me put in pastbin hold up |
| 23:03.13 | saisoma | _structz ~pastebin |
| 23:03.17 | _structz | http://pastebin.com/QXfpwVmS |
| 23:03.25 | saisoma | grr. nm. that doesn't work for me apparently |
| 23:04.22 | _structz | Gugge, this sip user, i just reboot the phone for register, and the information set to the database is that on pastebin |
| 23:04.56 | _structz | right now is everything fine, phone receiving and making calls |
| 23:05.02 | Gugge | _structz: the a sip reload should flush it from the cache, and when its needed it should be reloaded, because regseconds is > now |
| 23:05.31 | _structz | the sip show peers cache ain`t showing this phone right now |
| 23:05.46 | _structz | but information in database is ok, and the phone is working properly |
| 23:06.18 | _structz | just about know the phone got cache |
| 23:07.45 | _structz | http://pastebin.com/qAkcWWWe |
| 23:09.34 | Gugge | _structz: so what is not working? |
| 23:09.57 | _structz | so for now its just fine |
| 23:10.16 | _structz | but at some point someone try to call to the phone and it goes to voicemail |
| 23:10.38 | _structz | its been 11 minutes that is ok, i`ll perform a sip reload to see if the database will loose the information |
| 23:11.10 | Gugge | next time it happens, turn on verbose and check the cli |
| 23:11.23 | _structz | its already on |
| 23:11.36 | Gugge | do you have a dump from a failed call? |
| 23:12.17 | _structz | its a normal dump, when the dialstatus is differente than answer goes to voicemail |
| 23:12.50 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 23:12.58 | Gugge | does it try to call the phone |
| 23:13.03 | _structz | yes |
| 23:13.05 | Gugge | what does sip debug say |
| 23:13.40 | _structz | haven't deugged yet, doing that now |
| 23:13.45 | _structz | this is my steps |
| 23:13.53 | _structz | 1. reboot the phone to register again |
| 23:14.06 | _structz | 2. check realtime info and sip show peer info = OK |
| 23:14.15 | Gugge | is the phone on some firewalled / natted network ? |
| 23:14.19 | _structz | 3. sip reloaded (phone still working after that) |
| 23:14.38 | _structz | 4. now debug on and cheking info to see if gets flushed sometime and get sip info) |
| 23:14.41 | _structz | nope |
| 23:14.45 | _structz | public ip btw |
| 23:15.36 | saisoma | Hey guys, question about queues. Customers are complaining that they are in position #2 and then back to #5. Any ideas? Here is the config: http://pastebin.com/mgB25atG |
| 23:15.41 | saisoma | We are using fop2 for monitoring and are NOT using agent logins, so unmanned phones are being rung and not answered. Does that then shift the caller back to the end of the line? |
| 23:18.02 | _structz | Gugge, |
| 23:18.05 | _structz | there you go |
| 23:18.05 | _structz | http://pastebin.com/xLmd0qik |
| 23:18.08 | _structz | information just flushed |
| 23:19.42 | _structz | but in sip show peers its cached |
| 23:19.55 | _structz | and not messages on sip just |
| 23:20.00 | _structz | sip read from udp ... |
| 23:20.32 | _structz | but the phone is receiving calls |
| 23:20.48 | Gugge | as long as its cached, it wont read from the db |
| 23:21.05 | _structz | and now if a sip reload it won't find the phone because the db info was flushed |
| 23:21.17 | Gugge | do you have more than one asterisk instance using the same tables? |
| 23:21.33 | _structz | no |
| 23:21.59 | _structz | just sip reloaded |
| 23:22.03 | _structz | cached info flushed |
| 23:22.16 | _structz | and the db info flushed somehow |
| 23:22.20 | _structz | and the phone went to voicemail |
| 23:22.22 | Gugge | your problem is not that the cached info is flushing |
| 23:22.29 | _structz | the db info is flushed |
| 23:22.40 | _structz | i guess it shouln't |
| 23:24.06 | Gugge | you could set ignoreregexpire=yes, then realtime will never remove the info from db |
| 23:24.50 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 23:25.25 | Gugge | but i dont know why asterisk would remove it from the db, unless the phone unregisters, or times out |
| 23:25.34 | _structz | so that could cause a problem, if the phone is really turned off then won't ever go to voicemail right? |
| 23:25.52 | Gugge | it will, after failing the call |
| 23:26.45 | _structz | i see |
| 23:27.04 | _structz | my rtupdate was set to yes |
| 23:27.12 | _structz | i just setted to no to see if will do any changes |
| 23:27.29 | Gugge | it will |
| 23:27.35 | Gugge | if it is set to no it will never work after a sip reload :P |
| 23:27.45 | Gugge | so i guess you dont want that |
| 23:27.49 | _structz | yes |
| 23:27.55 | _structz | the information will never be updated on db |
| 23:28.31 | _structz | rtcachefriends=yes , rtupdate=yes, rtautoclear=no others rt options is default |
| 23:28.49 | _structz | let me pastebin my sip.conf |
| 23:33.15 | _structz | Gugge, take a look http://pastebin.com/2ZQHcY6K |
| 23:34.14 | Gugge | i would set ignoreregexpire=yes and rtautoclear=yes |
| 23:34.28 | Gugge | then it should still clear the cache, and not clear the db |
| 23:35.46 | _structz | i see |
| 23:49.04 | Lantizia | Hey you know RJ11 cables that headsets use? do you know if they use all of the 4 wires? |
| 23:50.48 | WIMPy | yes |
| 23:51.12 | *** join/#asterisk aurix (~aurelio@81.174.13.196) |
| 23:54.56 | TJNII | 2 for mic, 2 for earphone |
| 23:55.08 | Lantizia | oooh course! lol |
| 23:55.14 | Lantizia | duh |
| 23:55.47 | TJNII | I've seen one phone that used 3, but the company that made it ceased to exist on the 20s. |
| 23:56.02 | Lantizia | I don't get how our phone lines can use just 2 |
| 23:56.07 | Lantizia | and it provides it's own power |
| 23:56.18 | coppice | its not an RJ11. its a smaller plug, often called RJ9 or RJ22 |
| 23:56.26 | TJNII | Look up "hybrid network" |
| 23:56.53 | TJNII | And a analog phone is a current based device, not a voltage based one. |
| 23:57.48 | TJNII | The mic varies the resistance of the phone, which varies the apparent resistance of the phone. This effects both the local earpiece (sidetone) and the remote one. |
| 23:58.00 | TJNII | To oversimplify..... |
| 23:58.12 | Lantizia | right |
| 23:58.56 | TJNII | The two phones are in series, along with a power source from the telco. |