00:12.14 | SlicerDicer | PSTN with SPA3102, I cannot get my callwaiting to work.. I call my PSTN, then have another call on PSTN I cannot flash to it? Anybody know why this would be? I have tried the double hook flash and it does not seem to work either? |
00:12.26 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
00:15.27 | [TK]D-Fender | SlicerDicer: might be no way of doing this |
00:19.21 | SlicerDicer | what would I need to make it work? |
00:20.49 | [TK]D-Fender | SlicerDicer: Not good at understanding "no" I see |
00:21.10 | SlicerDicer | "might" was my hope ;-) |
00:24.16 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
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00:43.46 | drmessano^ | Ok, everyone, I have some bad news |
00:43.52 | drmessano^ | They are recalling BACON |
00:43.56 | drmessano^ | DONT PANIC |
00:43.59 | drmessano^ | Wait |
00:44.02 | drmessano^ | EVERYBODY PANIC |
00:44.52 | denon | drmessano^: bacon? http://twitpic.com/zyoo |
00:46.32 | drmessano^ | The Patrick Cudahy firm, of Cudahy, Wisconsin is recalling more than 3,500 pounds of bacon bit products that may be contaiminated with Listeria monocytogenes. |
00:46.42 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
00:46.45 | drmessano^ | They're recalling teh bacons |
00:48.21 | jaytee | 3500 lbs? that's nothing |
00:48.35 | drmessano^ | 3500 of bacon BITS |
00:48.42 | drmessano^ | Thats like.. a weeks supply |
00:49.02 | *** join/#asterisk Borai (i=Bora@S0106001c109e98db.no.shawcable.net) |
00:49.03 | Borai | Hi |
00:49.03 | jaytee | I don't use bacon bits, I use fresh cooked bacon |
00:49.05 | drmessano^ | I wonder what that is in uncooked bacon |
00:49.23 | drmessano^ | What the hell am I supposed to put on my ice cream? |
00:49.25 | jaytee | yeah, the shrinkage factor must be pretty high |
00:49.33 | jaytee | chocolate jimmies? |
00:49.42 | drmessano^ | Do they have bacon in them? NO |
00:49.43 | drmessano^ | FAIL |
00:49.50 | jaytee | lol |
00:50.37 | jaytee | drmessano^, did you see the question above from SlicerDicer about using a 3102 with call waiting? |
00:50.57 | drmessano^ | Im eating a hot dog right now.. and as much as I love hot dogs, it feels like dating the hot chick at schools "sorta hot from the side" sister |
00:51.14 | jaytee | haaahhaaaaaaa |
00:51.22 | drmessano^ | jaytee: Trying to flash the call waiting on the PSTN? |
00:51.29 | jaytee | yeah |
00:52.02 | drmessano^ | I cant be bothered with such shenanigans |
00:52.13 | drmessano^ | Who the hell uses Call waiting on a PBX |
00:52.17 | drmessano^ | Seriously.. |
00:52.36 | denon | drmessano^: actually, Ive used SendFlash to do that beforfe |
00:52.49 | denon | not for call waiting .. to remote transfer to get calls off pstn trunks |
00:53.39 | drmessano^ | "What if my cat gets out and needs to call home. Mr Pimpernoodle will be stranded because I cant answer my beeps" |
00:53.45 | drmessano^ | Answer: Buy him a cell phone |
00:55.35 | Borai | 1.6.0.3-rc1 or 1.4.22 ? |
00:55.55 | [TK]D-Fender | Borai: YES |
00:55.55 | drmessano^ | Cheese or wine? |
00:56.02 | drmessano^ | [TK]D-Fender: WIN |
00:56.10 | jaytee | [TK]D-Fender, heheehee |
00:56.10 | [TK]D-Fender | \o/ |
00:56.18 | drmessano^ | "Would you like the steak or the fish?" "Yes" |
00:56.34 | drmessano^ | Someone had to |
00:56.37 | Borai | which one should I upgrade to? |
00:56.55 | [TK]D-Fender | Borai: From? |
00:57.02 | drmessano^ | Windows 98 |
00:57.07 | jaytee | Borai, I suppose that all depends on whether your a bottom or a top. If your a bottom I'd go with 1.6 |
00:57.29 | Borai | 1.4.20.1 |
00:57.46 | drmessano^ | jaytee: Fail. Bottoms use "trunk" |
00:57.48 | Borai | well I want to start from the beggining this time I dont want to make the mistakes I did in the past |
00:58.08 | drmessano^ | Borai: Bear us your soul |
00:58.11 | jaytee | drmessano^, I completely forgot. yep, that would be the "max pain" avenue |
00:58.32 | [TK]D-Fender | Borai: 1.6.0.3-rc1 then, becuase you clearly can't upgrade to soemthing with FEWER decimal version positions, can't you? |
00:58.37 | drmessano^ | I think Borat wants to reveal his innermost child to us |
00:59.25 | drmessano^ | Go with the Really Cool (RC) one.. you know you want to.. All the other kids are doing it.. |
00:59.25 | jaytee | I couldn't get 1.4.22 to compile with zaptel instead of dahdi. thought it was still an option, dropped down to 1.4.21 instead |
01:00.02 | drmessano^ | I'm holding out for 1.6.2 |
01:00.37 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
01:00.47 | drmessano^ | AKA the "GTFO Dahdi, you've mixed me up long enough" release |
01:00.53 | jaytee | think they'll finally have SIP TCP working right by 1.8.0? |
01:00.55 | NovceGuru | I wonder if asterisk will compile/run on a RMI 1250 RISC Processor |
01:01.34 | drmessano^ | jaytee: No, but i'm sure something useless like SIP IPX will be implemented |
01:01.41 | jaytee | some guy was trying to get it running on an ARM processor a few days ago with no success |
01:02.02 | NovceGuru | Somebody has ported it to the gumstix which is the intel pxa270 arm |
01:02.07 | jaytee | hahaha, IPX, wow that brings back memories...........bad ones |
01:02.14 | drmessano^ | SIPBeui |
01:02.17 | drmessano^ | There u go |
01:02.20 | jaytee | hahahaaha |
01:02.37 | NovceGuru | I'm using a http://www.chippc.com/thin-clients/linux/?p=linux somewhere where I could use the audio out for a PA system |
01:02.43 | jaytee | I'm sorry but you can't call that person, they're on another segment! |
01:03.19 | drmessano^ | You know, that is the ultimate in network security |
01:03.26 | drmessano^ | Who the hell would see you running NetBeui |
01:03.44 | jaytee | SIP SNA/LU6.2 over Token Ring |
01:04.03 | drmessano^ | grabs some frozen yellow hose and small mallet |
01:04.29 | drmessano^ | Speaking of "What new again" |
01:04.35 | drmessano^ | I saw the coolest thing today |
01:04.56 | drmessano^ | Netgear has an Ethernet over Coax adapter for using your home CATV cable for networking |
01:05.19 | drmessano^ | Has an in/out to so you can split your TV connection off it, and the RJ45 for the ethernet |
01:05.30 | drmessano^ | 270Mbps |
01:05.41 | NovceGuru | I'm waiting for hdmi/component baluns over a single coax |
01:05.47 | NovceGuru | THAT will be the day |
01:06.22 | drmessano^ | I was thinking for a 300 foot run at 270Mbps.. You can bury some RG-6 between two buildings |
01:06.29 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
01:06.34 | drmessano^ | Beats the crap out of fiber |
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01:07.24 | NovceGuru | direct burial cat5 isn't THAT bad |
01:07.42 | drmessano^ | Yuck |
01:07.43 | NovceGuru | ~$120 for 1000ft |
01:07.45 | drmessano^ | Its still cat5 |
01:07.50 | NovceGuru | gigabit! |
01:07.54 | drmessano^ | So? |
01:08.00 | NovceGuru | yeah |
01:08.19 | drmessano^ | I would want something less fragile buried |
01:08.27 | drmessano^ | Unless they encase it in a garden hose |
01:08.28 | jaytee | underground CAT5 for 120 bucks a 1000'? where you shopping? WalMart? |
01:08.43 | NovceGuru | monoprice, it's probably junk |
01:08.49 | drmessano^ | Standard Cat5 is 90 a roll |
01:08.50 | NovceGuru | lowes has some with a pretty thick jacket |
01:08.56 | drmessano^ | lol |
01:09.03 | jaytee | I can get standard for 79 bucks a 1000 |
01:09.05 | drmessano^ | "Pretty thick" does not make direct burial |
01:09.06 | NovceGuru | I get it for 65 at the electric store |
01:09.11 | drmessano^ | and I dont buy cable at LOWES |
01:09.14 | drmessano^ | Or Home Depot |
01:09.18 | drmessano^ | Or Rickels Lumber |
01:09.34 | drmessano^ | NovceGuru: Not Belden |
01:09.34 | NovceGuru | I do when I can't wait for shipping |
01:09.38 | jaytee | nope, neoprene jacket with foil shield for underground and that gets pricey. like 180 a 1000 |
01:09.39 | NovceGuru | they have the WORST cat5 |
01:10.08 | drmessano^ | Belden is worse than Lowes brand? |
01:10.19 | drmessano^ | Lay off the crack |
01:10.27 | NovceGuru | never used belden |
01:10.47 | NovceGuru | fortunately, it sounds like :) |
01:11.11 | NovceGuru | I did get some krap that was pre lubed and didn't have the nylon tension cord thingy in it |
01:11.15 | drmessano^ | If you buy a roll of Cat5 for $65 at todays prices, you can't possibly be getting decent cable |
01:11.31 | jaytee | I like Belden myself, I've never had a problem with it |
01:11.37 | NovceGuru | copper has came down, btw |
01:11.40 | coppice | Belden have been very good at getting crazy high prices out of defence customers :-) |
01:11.41 | drmessano^ | Belden is pretty much the best |
01:11.46 | drmessano^ | Not that much it hasnt |
01:11.54 | Borai | ok so I installed dahdi-linux-2.1.0 as well as dahdi-tools now configuring and installing (Really Cool)-1 |
01:12.10 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5446245918ccf1d7) |
01:13.48 | NovceGuru | http://www.kitcometals.com/charts/copper_historical_large.html#6months |
01:13.50 | drmessano^ | Armored Cat5 for the win |
01:14.13 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
01:14.18 | NovceGuru | http://www.pcconnection.com/IPA/Shop/Product/Detail.htm?sku=9136254&oext=1038A&ci_src=14110944&ci_sku=9136254 YES |
01:14.23 | NovceGuru | $800! cheap! |
01:16.28 | jaytee | 80 cents a foot isn't cheap |
01:17.15 | coppice | it is for shoes |
01:17.16 | drmessano^ | Cat5 will never be desirable for outdoor use unless its fully dipped |
01:17.25 | jaytee | lol |
01:17.51 | drmessano^ | Thats why RG6 is a great idea |
01:18.01 | jaytee | drmessano^, that was gel filled, foil shielded and polyethylene outside jacket. |
01:20.07 | jaytee | here ya go, 139 bucks for gel filled outdoor, just as good as that overpriced BlackBox cable at a way better price: http://www.jack2rack.com/index.php/cPath/97_73_197?gclid=CJz77Yrj-JcCFQ89awodtindEA |
01:20.11 | coppice | 2 layers are stainless steel armouring, and shark repellant |
01:20.42 | jaytee | yeah, those land sharks swimming underground love to bite into that cable |
01:20.48 | drmessano^ | Id be sadistic enough to use RG11 |
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01:21.12 | jaytee | moles, voles and friggin gophers |
01:21.18 | drmessano^ | HA |
01:21.22 | drmessano^ | Black box cable |
01:21.41 | jaytee | that was the 800 bucks for a 1000 feet link he posted |
01:22.11 | drmessano^ | It could be a run of Belden that didnt make the grade, or it could be a run of cable Kelloggs made in one of their offshore subsidiaries.. who knows |
01:22.22 | drmessano^ | I secretly suspect Callweaver is owned by Cisco |
01:22.26 | drmessano^ | But thats just me |
01:22.27 | coppice | black box are good for finding unusual things, but bad by any other measure |
01:23.34 | *** join/#asterisk sosoriri (n=chatzill@222.47.180.130) |
01:24.14 | sosoriri | hello, anybody have seen the soudn delay problem? |
01:25.15 | jaytee | [TK]D-Fender, your friend is back :-) |
01:25.46 | sosoriri | for example, when one user join in the conference, he must hear "Please enter your conference number followed by the pound key.", but he heard"...your conference number followed by the pound key." |
01:26.13 | sosoriri | but i can't catch it again. |
01:27.21 | sosoriri | it occurs one or two times in a month or not... |
01:28.24 | drmessano^ | Its moon phases |
01:28.25 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
01:28.33 | jaytee | gravity waves |
01:28.42 | [TK]D-Fender | drmessano^: Sheer lunacy I say... |
01:28.57 | jaytee | groans |
01:29.38 | drmessano^ | That was enlightening |
01:33.35 | Borai | ok |
01:34.10 | mchou | anyone have a spa-3000 or 3102? |
01:34.22 | Borai | success |
01:34.32 | Borai | clean insall of 1.6.0.3-rc1 |
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01:35.27 | Borai | now i got a polycom IP 550 unpacked sitting next to me |
01:36.11 | jaytee | nice phones |
01:36.14 | jaytee | I have a few |
01:36.44 | Borai | is there a guide on how to configure this sweet phone? |
01:37.46 | jaytee | there's a section in the book on how to setup FTP provisioning and if you use that, the Polycom whitepaper on provisioning and the SIP Admin guide for your version of the SIP firmware you should be able to do it. I did. |
01:38.38 | jaytee | or you could take the lame shortcut route and use the web interface |
01:38.55 | Borai | to access those guides |
01:39.01 | Borai | you have to have a polycom account right? |
01:39.05 | jaytee | nope |
01:39.10 | Borai | I couldnt register at their site their server has an error |
01:39.12 | Borai | ok |
01:39.15 | Borai | the book I have that |
01:40.05 | jaytee | http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip550.html |
01:40.51 | NovceGuru | their config files are crazy |
01:41.24 | jaytee | on that page you need to download the SIP Admin guide for YOUR phone's current version of the SIP firmware and under the section titled Other Technical Documents get the White Paper titled Configuration File Management on Soundpoint IP Phones. |
01:41.25 | Sargun_screen | jaytee: I have one of those, I like it. |
01:41.51 | jaytee | NovceGuru, with great power comes great responsibility |
01:41.55 | Borai | ok dont you need to download the firmware? |
01:42.08 | NovceGuru | jaytee: yeah |
01:42.34 | NovceGuru | I'm getting an IP670 in next week, color screen!(woopdeedoo?!?) |
01:42.45 | Borai | firmware is bootrom: in the settings of the phone right? |
01:42.47 | jaytee | yes, you can download the same version you have plus the bootrom files for what's on there now or you can use a new version. I prefer to start with whatever's on the phone or a minor version or two higher. |
01:43.41 | jaytee | bootrom boots the phone. Having a copy on the FTP server lets the phone boot from it and search there for it's configs to download. read the whitepaper, it'll all come clear to you. |
01:44.11 | bkw_ | has two 670's |
01:44.20 | bkw_ | two 650's, one 550, and two 330's |
01:44.38 | jaytee | I have 53 330's and 6 550's. |
01:44.47 | bkw_ | the 670 isn't that grand of an upgrade |
01:45.05 | jaytee | but it's got that larger screen and HD |
01:45.15 | bkw_ | so does the 650 and 550 |
01:45.25 | NovceGuru | I think the 650 and 550 are the same except for color screen and gige? |
01:45.33 | bkw_ | right |
01:45.41 | bkw_ | I'm also getting a Cisco 7975G |
01:45.47 | jaytee | yeah, 650 has color and gig |
01:45.49 | NovceGuru | I dont understand the oogling over gige....as is networks wont be backward compatable for the next 10 years? |
01:45.52 | NovceGuru | as if* |
01:46.00 | bkw_ | NovceGuru: never know |
01:46.20 | NovceGuru | and let me know when sip needs > 100mbit !! |
01:46.24 | NovceGuru | heh :P |
01:46.47 | bkw_ | it does if you do more than 1200 calls with ulaw |
01:46.50 | jaytee | for when your phone also has HDTV videoconferencing built in I imagine |
01:46.54 | bkw_ | NovceGuru: its not for the phone |
01:46.58 | bkw_ | its for the PC sitting on the otherside |
01:47.02 | bkw_ | on the switch port s |
01:47.08 | NovceGuru | no it's obviously clear |
01:47.18 | NovceGuru | now* |
01:47.37 | jaytee | gigabit on the backbone or on a segment's one thing, nowadays on a station or phone uplink to the switch it's a waste of bandwidth |
01:47.38 | bkw_ | I can't wait till the 550, 650 and 670 support G722.1 and G722.1C |
01:48.10 | jaytee | why, cuz you wrote it? |
01:48.21 | bkw_ | No I wrote the module for FreeSWITCH :P |
01:48.33 | bkw_ | but they'll have it sometime in 09 |
01:48.38 | bkw_ | for the other phones i'm told |
01:48.49 | bkw_ | lower bandwidth usage |
01:48.58 | jaytee | well, last time I looked at my calendar it was ...... hmmmm, 09! |
01:49.04 | bkw_ | try Q3 09 |
01:49.30 | jaytee | I'm patient. I can wait till Q3.......of 2011 or 2012 |
01:49.46 | NovceGuru | I never did follow asterisk and it's g722 issues |
01:49.53 | BBHoss | bkw_: how is g722 different from .1? |
01:49.58 | NovceGuru | even with ulaw the polycoms sounds great |
01:50.08 | bkw_ | G722.1 can run better on non-dsp hardware ie PC's |
01:50.11 | jaytee | BBHoss, a period and an extra number? |
01:50.16 | NovceGuru | 2-3x bandwidth |
01:50.21 | bkw_ | NovceGuru: Nope |
01:50.25 | bkw_ | G722 runs 64k just like ulaw |
01:50.28 | BBHoss | here we go again |
01:50.43 | NovceGuru | bkw_: I mean then vs .1/c? |
01:51.00 | jaytee | yep, down the old "yadda, yadda, yadda, 64k, yadda, DSP, yadda." yawn |
01:51.19 | BBHoss | bkw_: so it takes less time to transcode or w/e, lower translation time? |
01:51.56 | bkw_ | BBHoss: it could.. I have never really calculated that |
01:52.13 | BBHoss | so its just easier to implement then |
01:52.14 | bkw_ | G722.1 can run at 16 and 32kbit, while G722.1C can run at 24, 32 and 48kbit |
01:52.20 | bkw_ | BBHoss: resource wise it is |
01:52.50 | BBHoss | not patent encumbered either, correct? |
01:53.05 | bkw_ | you still have to get a royalty free license from polycom to use it commercially |
01:53.12 | bkw_ | but its free |
01:53.26 | jaytee | if it wasn't for the fact that computerized engine analyzers cost a fortune most of today's geeks would be motorheads instead. |
01:53.26 | BBHoss | you mean to like put it in your own product? |
01:53.34 | bkw_ | <PROTECTED> |
01:53.34 | bkw_ | <PROTECTED> |
01:53.35 | bkw_ | right |
01:54.32 | jaytee | so it's already available then |
01:54.49 | BBHoss | where does it talk about royalties? |
01:55.16 | bkw_ | BBHoss: you have to contact polycom for the contract |
01:55.23 | BBHoss | ahh |
01:55.45 | BBHoss | they should really make it 100% free if they want widespread adoption |
01:55.51 | bkw_ | thats what they are working on |
01:56.00 | bkw_ | thats why they let us put it all in FreeSWITCH codec lib and all |
01:56.31 | BBHoss | oh that was nice of them |
01:56.59 | BBHoss | now they just need to get it in all of the media gateways |
01:57.43 | bkw_ | pstn isn't wideband |
01:57.45 | bkw_ | so why bother |
01:57.49 | bkw_ | its more for sip to sip usage |
01:58.38 | BBHoss | well if every itsp would put thier numbers in enum or dundi or similar |
01:58.45 | bkw_ | true |
01:58.52 | BBHoss | or if you called someone on the same service |
01:59.20 | bkw_ | thats where it will shine |
01:59.24 | BBHoss | its almost as low-rate as g729, higher quality, and cheaper/free |
01:59.38 | bkw_ | I think G729E is wideband |
01:59.48 | BBHoss | yeah but who supports it |
02:00.00 | bkw_ | G722.2 is good also... AMR-WB if I remember my numbers correctly |
02:00.25 | bkw_ | G719 is Siren22 |
02:00.28 | bkw_ | Stereo baby |
02:00.32 | BBHoss | bkw_: does freeswitch have any kind of dundi type network, or just enum? |
02:00.37 | bkw_ | just enum |
02:00.40 | bkw_ | and ISN |
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02:01.20 | BBHoss | isn? |
02:01.24 | bkw_ | www.freenum.org |
02:01.36 | bkw_ | the stuff John Todd did |
02:01.52 | NovceGuru | ahhh so much krap I know nothing about |
02:02.43 | bkw_ | BBHoss: FreeSWITCH 1.0.2 is the first to include any of the Siren codec stuff... |
02:02.55 | BBHoss | bkw_: i run trunk anyways :) |
02:03.01 | bkw_ | hehe |
02:03.07 | jaytee | wow, let's all dump * and run that then :-) |
02:03.25 | bkw_ | jaytee: if Asterisk fits your needs you don't really have a reason to switch |
02:03.48 | jaytee | i know, i was being facetious |
02:04.03 | bkw_ | I think BBHoss runs both don't you? |
02:04.09 | BBHoss | yeah |
02:04.14 | NovceGuru | I used to toy with sipxecs |
02:04.22 | bkw_ | PAIN |
02:04.35 | jaytee | i still run sipX as a upd/tcp proxy to Exchange UM |
02:04.47 | bkw_ | jaytee: people use FreeSWITCH for that too |
02:04.59 | jaytee | until I can get 1.6 working right with TCP |
02:05.15 | BBHoss | i want to use freeswitch for my application, but it seems the guy who wrote the Telegraph library for it is MIA, and the Asterisk Telegraph library is a good deal better |
02:05.21 | NovceGuru | I like the sipx endpoint manager |
02:05.25 | bkw_ | BBHoss: I know him |
02:05.34 | bkw_ | BBHoss: you could use liverpie |
02:05.43 | bkw_ | http://www.liverpie.com/ |
02:05.46 | bkw_ | no clue where they got the name |
02:05.47 | Borai | so i need to have an ftp account that has the 2345-12500-001.bootrom.ld and 2345-12500-001.sip.ld, as well as a sip.cfg? |
02:05.59 | bkw_ | Borai: yes |
02:06.10 | BBHoss | bkw_: oh yeah i remember seeing that |
02:06.21 | Borai | anything else |
02:06.22 | bkw_ | they started with Telegraph and did that if I recall |
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02:06.36 | bkw_ | Borai: I just unzip the whole file from polycom into the dir |
02:06.41 | bkw_ | and leave it at that |
02:06.53 | Borai | ok |
02:07.02 | BBHoss | you should see the fun shit i wrote today for a voice broadcasting system |
02:07.04 | jaytee | leave it? that'll fubar it |
02:07.21 | jaytee | you need to more than just unzip it. read the damn whitepaper |
02:07.21 | bkw_ | what? |
02:07.24 | BBHoss | they should put it on github |
02:07.38 | bkw_ | jaytee: I always unzip it and reboot the phone |
02:07.43 | bkw_ | never once have i had problem |
02:07.47 | bkw_ | I started at 2.0 |
02:08.20 | jaytee | without creating a master config file to match the mac address of the phone? or a phone config file with the unique parameters for that phone? |
02:08.57 | BBHoss | wow, looks like its being semi-actively developed |
02:08.58 | bkw_ | jaytee: thats already done |
02:09.15 | bkw_ | BBHoss: also check out esl.. if you haven't already |
02:09.16 | jaytee | bkw_, how? |
02:09.25 | bkw_ | jaytee: I never modify the .cfg files from the tarball |
02:09.33 | bkw_ | I just include the sip.cfg and phone.cfg |
02:09.38 | jaytee | I use them as templates |
02:09.39 | bkw_ | and modify them locally in my per phone config |
02:10.02 | BBHoss | i was looking for some tips on rails and freeswitch, so i googled "rails freeswitch" and MY name was the first to come up :) |
02:10.09 | bkw_ | BBHoss: hehe |
02:10.20 | Borai | ok i just have 1 phone at this point, only one polycom) and i am downloading both zip files on to the server and extracting them |
02:10.22 | bkw_ | I'm not sold on the rails stuff for voip yet |
02:10.31 | Borai | and yes the whitepaper tells me i have to create a config file |
02:10.50 | BBHoss | i'm just integrating it into my web app, not using rails for 100% voice |
02:11.02 | bkw_ | BBHoss: thats kewl |
02:11.23 | BBHoss | right now its just a voice broadcasting system for emas |
02:11.26 | BBHoss | like r-911 |
02:11.46 | Borai | but the default phone1.cfg that comes in the zip is totally confusing |
02:11.46 | BBHoss | but eventually i want to do TTS with it, so they dont have to record a message |
02:13.08 | *** join/#asterisk sdaniels (n=chatzill@cpe-72-190-76-209.tx.res.rr.com) |
02:14.36 | sdaniels | how do i remove a digit from a dialed number? I understand how to add one ex: exten => _NXXNXXXXXX,n,Dial(SIP/+1${EXTEN}@whatever) but what if i want to have _81NXXNXXXXXX and remove the 8? |
02:14.51 | bkw_ | sdaniels: ${EXTEN:X: |
02:14.54 | bkw_ | er |
02:14.59 | bkw_ | ${EXTEN:1} |
02:15.12 | bkw_ | that'll remove the 8 |
02:15.33 | bkw_ | exten => _8NXXNXXXXXX,n,Dial(SIP/+1${EXTEN:1}@whatever) |
02:16.14 | Borai | ok now both zips are in my ftp |
02:16.28 | bkw_ | unzip them both |
02:16.38 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-154-243.dsl.stlsmo.sbcglobal.net) |
02:16.52 | *** join/#asterisk Hanif08 (n=bucoo77@netop.jaring.my) |
02:17.00 | LemensTS | will stop gracefully kick me out of the cli when its done? |
02:17.14 | Borai | both unzipped |
02:17.30 | Borai | oh wait they have to be in the main directory right? |
02:17.35 | Borai | so I cant have it under a sub directory |
02:17.51 | bkw_ | Borai: you can if you feed it the full path ot the files in the phone on the ftp |
02:17.56 | bkw_ | I use http://ip/polycom/ |
02:18.35 | Borai | ok |
02:18.43 | LemensTS | Seems like it is sitting here a while, and i tried show channels and that didnt do anything... |
02:18.52 | Borai | thats what i did too ip/polycom/ |
02:18.54 | bkw_ | LemensTS: sounds like something locked up |
02:19.07 | Borai | but in the phone's settings menu i didnt see a path |
02:19.21 | bkw_ | you set it at the end of the IP |
02:19.24 | bkw_ | once you set the type to FTP |
02:19.25 | bkw_ | or HTTP |
02:20.20 | Borai | ok im under server menu i changed server type to http |
02:20.24 | jaytee | better to setup DHCP to pass Option 66 |
02:20.31 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
02:20.47 | bkw_ | he has one phone |
02:20.53 | bkw_ | option 66 is over kill right now |
02:21.01 | jaytee | oh, ok. agreed |
02:21.11 | Borai | well dhcp i cant |
02:21.21 | Borai | i dont have dhcp at this place and the asterisk server is on a remote location |
02:21.33 | LemensTS | bkw_: exited out of the cli and went back in and it lets me do cmds now. |
02:21.41 | [TK]D-Fender | Borai: How do you NOT have DHCP? |
02:21.57 | bkw_ | LemensTS: you did stop gracefully |
02:21.57 | jaytee | hell, unless I wanted to configure special features I'd have just used the web interface if all I wanted was to get the phone working. |
02:21.58 | Borai | I dont use DHCP |
02:21.58 | bkw_ | if I recall correctly asterisk shouldn't be running at all once that command finishes |
02:22.04 | [TK]D-Fender | Borai: In a local LAN? |
02:22.10 | Borai | yes |
02:22.25 | Borai | static ip's manual entered on all the pcs |
02:22.38 | Borai | have 3 pc's here |
02:22.43 | [TK]D-Fender | Borai: Fine, just start the phone, go into the bootrom and set the parms manually as well as the server for provisioning |
02:22.58 | [TK]D-Fender | Borai: And for internet? |
02:23.14 | LemensTS | bkw_: there are 2 active calls still |
02:23.17 | Borai | ok cable modem connected to a linksys wrt54gl |
02:23.22 | Borai | that is connected to 550 |
02:23.27 | Borai | a server |
02:23.29 | [TK]D-Fender | Borai: and that doesn't ser DHCP? |
02:23.32 | Borai | and to another switch |
02:23.32 | LemensTS | bkw_: think it has to wait till those are done |
02:23.40 | Borai | it does but I dont have dhcp enabled |
02:23.42 | [TK]D-Fender | serve* |
02:23.50 | [TK]D-Fender | Borai: CRAZY |
02:23.53 | Borai | it does I just dont use dhcp |
02:23.56 | Borai | why? |
02:24.06 | jaytee | LemensTS, until all calls are complete it won't exit |
02:24.10 | [TK]D-Fender | Borai: so you don't have to do the boring stuff... |
02:24.26 | Borai | boring stuff? |
02:24.39 | [TK]D-Fender | Borai: assigning IP's to devices |
02:24.43 | LemensTS | it wouldnt hurt to do a stop now after you did a stop gracefully eh? |
02:24.49 | bkw_ | you guy see this phone http://blog.voipsupply.com/new-products/first-look-cisco-spa-525g-desktop-ip-phone-with-wifi-bluetooth-and-more |
02:24.56 | bkw_ | I think they failed cuz they didn't include wideband |
02:25.53 | NovceGuru | do ciscos still suck ass for SIP? |
02:25.54 | jaytee | anyone see the video of the kid getting shot in the head at the Fruitvale BART station while two other cops were holding him down? |
02:26.07 | NovceGuru | I remember wrestling with some 7940s < then a year ago...pita |
02:26.14 | Borai | so server address should be myftpip/foldername right? |
02:26.20 | NovceGuru | granted 7940 is a bit older then this spa525g |
02:26.56 | bkw_ | jaytee: yes I seen it |
02:27.17 | jaytee | disgusting |
02:27.27 | bkw_ | someone better go away for that one |
02:27.33 | bkw_ | or hell will break loose |
02:27.35 | jaytee | indeed |
02:27.46 | bkw_ | is happy he lives in Oklahoma |
02:27.47 | jaytee | hell's already broken loose |
02:27.54 | *** part/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
02:27.56 | jaytee | an Okie? OKC? |
02:28.03 | bkw_ | McAlester |
02:28.12 | bkw_ | southeastern oklahoma |
02:28.19 | jaytee | ah, I was stationed at Tinker during the late 70's |
02:28.29 | bkw_ | I was born in the late 70's |
02:28.30 | bkw_ | :P |
02:28.40 | bkw_ | I like it here.. its calm |
02:28.41 | jaytee | I was around when they invented dirt |
02:28.42 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
02:28.42 | bkw_ | and I can carry a gun |
02:29.02 | jaytee | I liked the people but the landscape's a bit dreary |
02:30.04 | NovceGuru | ahha that cisco has a "Kensington security slot support" |
02:31.04 | *** join/#asterisk jtodd (n=jtodd@117.sub-70-214-1.myvzw.com) |
02:31.32 | Borai | erm |
02:32.20 | bkw_ | the snom 820 has wifi too... I have one on the way |
02:33.13 | BBHoss | bkw_: what hardware did you use to test CELT? |
02:33.21 | bkw_ | BBHoss: my Mac |
02:33.23 | bkw_ | and FS |
02:33.30 | bkw_ | but Ekiga has support for CELT now in the latest builds |
02:33.35 | bkw_ | stkn added support for it |
02:33.36 | BBHoss | oh ok cool |
02:33.51 | bkw_ | I have an XM radio celt stream up for people to call if they want |
02:34.02 | BBHoss | where |
02:34.09 | bkw_ | sip:886@taz.bkw.org:5080 |
02:34.13 | Borai | ok |
02:34.28 | Borai | it took like 5 minutes on the initializing screen but now its downloading the new bootrom |
02:34.54 | BBHoss | bkw_: so whats the easiest way to test it on a mac? |
02:34.55 | bkw_ | Borai: give it time |
02:35.01 | bkw_ | BBHoss: portaudio + freeswitch |
02:35.05 | bkw_ | BBHoss: thats hwo I did it |
02:35.23 | Borai | the download was fast tho |
02:35.25 | BBHoss | guess i'll do it another day then :) |
02:35.35 | bkw_ | BBHoss: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-December/009746.html |
02:35.43 | Borai | formatting fs please wait. |
02:35.55 | bkw_ | Borai: just chillax you'll be ok |
02:36.14 | BBHoss | its fun when its says ERROR 10 or w/e |
02:36.23 | Borai | downloading bootrom again |
02:36.30 | bkw_ | haha |
02:36.36 | Borai | im excited what a phone can do |
02:36.42 | Borai | its interesting |
02:36.54 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
02:37.45 | bkw_ | Borai: you have caught the voip bug I take it? |
02:38.03 | Borai | voip bug? |
02:38.14 | Borai | formatting again |
02:38.26 | Borai | endless loop? |
02:38.48 | Borai | oh no nevermind now its downloading new application |
02:38.53 | BBHoss | Borai: no it has to download all of them up to the latest version i think |
02:38.54 | bkw_ | see told you to chillax |
02:39.31 | Borai | oh wait so you cant just go from ex. 1.1 to 1.7 you have to do 1.1, 1.2, 1.3, ...? |
02:40.29 | BBHoss | Borai: dunno, but its grabs the file like 10 times |
02:40.50 | BBHoss | you can tail -f the tftp log or the log it uploads and see what its doing |
02:44.00 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
02:45.15 | Borai | ok booted |
02:45.50 | *** join/#asterisk fatnasty1 (n=fatnasty@cpe-72-190-76-209.tx.res.rr.com) |
02:48.22 | fatnasty1 | Hello? |
02:49.14 | Borai | goodbye? |
02:49.18 | fatnasty1 | Cool it works |
02:49.36 | jaytee | no it doesn't |
02:51.28 | Borai | lol |
02:52.02 | Borai | ok now i changed to ftp so that the phone can save the config when i change something |
02:52.07 | Borai | what about tag sn to ua? |
02:59.16 | Borai | is there an easy way to create a config file for the polycom |
03:00.25 | jaytee | i use a bash script to copy template files and edit them with sed |
03:00.57 | jaytee | so i just type ./prepphone.sh macaddress XXXX and i'm done |
03:01.17 | Borai | ok nice but the problem is |
03:01.23 | jaytee | or in the case of multiple lines on a 550 I have another script to do that. |
03:01.31 | Borai | a) I have never configured a polycom |
03:01.47 | carrar | you get to learn! |
03:01.50 | Borai | before, b) I dont understand anything in the example file phone1.cfg |
03:02.13 | carrar | helps to understand XML |
03:02.31 | carrar | and helps to understand how the phone works |
03:02.43 | carrar | so need to learn all that 1st |
03:02.47 | jaytee | the configuration parameters are documented in the SIP Admin guide |
03:02.50 | carrar | start with the polycom admin guide |
03:03.40 | carrar | then you can write scripts to make the configs for you |
03:03.46 | carrar | once you know what you need to change |
03:04.30 | Borai | I dont need to create a script I am not that lazy I can hand edit it but |
03:04.36 | carrar | haha |
03:04.48 | carrar | even I wouln't want to hand edit 5 polycom phones |
03:04.59 | Borai | :) |
03:05.03 | [TK]D-Fender | carrar: I do mine by hand all the time |
03:05.07 | carrar | when spending that time to write the script can make life so simple |
03:05.08 | Borai | they are documented in the admin guide? |
03:05.13 | carrar | hahha |
03:05.14 | jaytee | what's lazy about using a script? want to spend 3 hours or 10 minutes configuring 30 phones? |
03:05.17 | jaytee | duh! |
03:05.35 | Borai | well i am not going to configure 30 phones |
03:05.42 | Borai | if i was i would write a script or even hire someone |
03:05.42 | jaytee | ever? |
03:05.43 | carrar | who wants to edit dozens of phone polycom config sby hand |
03:05.44 | Borai | that would do it |
03:05.54 | [TK]D-Fender | jaytee: 10 minutes? You saw how much time I've wasted to toto retards here? 10 minutes is a FIELD DAY! |
03:06.01 | [TK]D-Fender | total* |
03:06.22 | carrar | 1 phone is easy by hand once you know what to change |
03:06.48 | jaytee | [TK]D-Fender, what about my 10 minutes? |
03:07.06 | jaytee | ~IWMWB |
03:07.07 | jbot | I WANT MY WEEKEND BACK! |
03:07.16 | carrar | I took that 10 mins and wrote a script, so now going forward it takes 3 seconds |
03:08.10 | *** join/#asterisk fatnasty1 (n=fatnasty@cpe-72-190-76-209.tx.res.rr.com) |
03:08.45 | jaytee | yeah, I exaggerated. If I write a list of all the macs for the phones I can crank out 30 configs in about 4 minutes probably but I'm a fast typist. |
03:09.22 | carrar | list of mac's and a script, 4 seconds |
03:09.22 | jaytee | toughest part was learning to use SED |
03:10.14 | jaytee | that's cuz you're reading in the list into the script to batch process it. I hadn't taken it to the next level.......yet |
03:10.25 | carrar | I use perl |
03:10.32 | jaytee | I'm just learning perl |
03:10.42 | carrar | best way to learn, needing to write something |
03:10.43 | jaytee | got my llama book right here :-) |
03:11.44 | carrar | that same script can also generate your sip.conf config for the phone as well |
03:12.00 | carrar | making your time faster to complete that phone |
03:13.10 | carrar | or anything else uniqe to that extension/phone |
03:14.02 | *** join/#asterisk Sargun (n=Sargun@75-101-13-24.dsl.static.sonic.net) |
03:14.03 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
03:14.40 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
03:14.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
03:23.30 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-221-124.phlapa.east.verizon.net) |
03:23.33 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.133) |
03:23.45 | Daejeo | greetings :) |
03:24.50 | Daejeo | anyone knows path to documentroot on switchvox ? |
03:25.10 | [TK]D-Fender | Daejeo: I'm sure www.digium.com does |
03:25.19 | Daejeo | "/var/www/html" |
03:25.26 | carrar | documentroot? |
03:25.52 | carrar | You are not suppose to have access to the switchvox OS :) |
03:26.19 | Borai | rfc2543 hold yes or no? |
03:26.31 | [TK]D-Fender | Borai: leave alone |
03:27.31 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
03:28.07 | Daejeo | [TK]D-Fender: did you play with switchvox ? |
03:28.22 | [TK]D-Fender | Daejeo: No |
03:29.30 | carrar | Daejeo, it's all javascript anyways |
03:29.41 | carrar | if you know that you will find it easy |
03:34.22 | Borai | how can you save the config file in the phone to the boot server? |
03:34.53 | jaytee | can't |
03:34.56 | Bad_Robot- | boot your live linux cd ;) |
03:35.22 | jaytee | it'll upload log files but not configs |
03:35.31 | carrar | it will uplaod changes |
03:35.35 | Borai | ok |
03:36.03 | carrar | if you have the 'overrides' directory setup |
03:36.30 | carrar | 'limited changes' |
03:37.21 | Borai | ok |
03:37.38 | Borai | well that means i will have to create a config file |
03:37.43 | carrar | yes |
03:37.50 | Borai | i just configured the phone using the menu on it |
03:37.59 | [TK]D-Fender | In Soviet Russia, phone configures YOU |
03:38.21 | [TK]D-Fender | Borai: Only thing you should program on the phone itself is IP parms & the IP & type of boot server |
03:38.44 | carrar | not even that!! use DHCP to do that for you |
03:38.59 | carrar | must be lazier! |
03:40.35 | [TK]D-Fender | carrar: No, you seem to have missed that... he doesn't even HAVE DHCP |
03:40.41 | carrar | oh |
03:40.45 | carrar | that has to suck |
03:41.08 | carrar | put that on your switchvox box |
03:41.10 | carrar | :) |
03:41.28 | carrar | I'm assuming he's running switchvox |
03:41.55 | carrar | wrong person |
03:42.00 | carrar | man |
03:42.07 | carrar | I'm just not paying enough attention |
03:42.28 | jaytee | no, it's that Daejeo guy that keeps asking questions about hacking switchvox |
03:43.18 | carrar | Borai, any reason why you can't run a DHCP service? |
03:43.35 | jaytee | he has ALS |
03:43.44 | carrar | ALS? |
03:43.52 | jaytee | Acute Laziness Syndrome |
03:43.55 | carrar | haha |
03:44.14 | carrar | I think DHCP is easier then configuring a handfull of phones via the menus |
03:44.22 | jaytee | ya think? |
03:44.23 | [TK]D-Fender | jaytee: My mom told her co-workers she hard SARS (Serios Anal Retentive Secretary) |
03:44.31 | [TK]D-Fender | had* |
03:44.32 | jaytee | lol |
03:44.39 | carrar | heh |
03:45.57 | carrar | I could even be talked into writting his dhcp.conf |
03:46.03 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-7b6abd2a67c06443) |
03:46.38 | [TK]D-Fender | carrar: I mean NO DHCP. He's running a Linksys router and disabled it. Its not for LACK of it... |
03:47.07 | carrar | Whats Asterisk running on then? |
03:47.19 | carrar | some remote hop someplace else? |
03:47.35 | carrar | must be |
04:08.51 | *** join/#asterisk Lord_Drachenblut (n=drachenb@12.197.74.66) |
04:10.59 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
04:12.02 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
04:15.14 | SlicerDicer | anybody know about Aastra 480i models? |
04:15.19 | SlicerDicer | and thoughts perhaps? |
04:15.34 | *** part/#asterisk Lord_Drachenblut (n=drachenb@12.197.74.66) |
04:17.50 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
04:18.07 | [TK]D-Fender | SlicerDicer: Largely decent, but rare to advise |
04:18.28 | SlicerDicer | what there is better models or what? |
04:19.20 | [TK]D-Fender | Polycom > All |
04:19.38 | SlicerDicer | uglier than all too |
04:20.13 | SlicerDicer | if I were to vomit.. thats what a polycom would look like.. but in all seriousness what features makes them better? |
04:20.15 | [TK]D-Fender | SlicerDicer: Far from |
04:20.48 | [TK]D-Fender | SlicerDicer: Solid phones, highly configurable, superior call handling, physical & audio quality |
04:21.20 | SlicerDicer | features please... |
04:21.47 | SlicerDicer | those are just thoughts of why they are better IMO |
04:21.53 | SlicerDicer | feature for feature what makes them better |
04:22.42 | [TK]D-Fender | SlicerDicer: And a piece of shit tah echoes on all your calls and feels like a flimsy piece of shit with an unreadable display must be AWESOME |
04:22.57 | SlicerDicer | that makes no sense |
04:23.04 | SlicerDicer | I just asked feature for feature what is better |
04:23.23 | SlicerDicer | you come back with some garbage.. |
04:23.31 | maddog01 | what phones are you comparing |
04:23.44 | SlicerDicer | obviously you are biased with polycom if you wont answer a simple question |
04:23.48 | [TK]D-Fender | SlicerDicer: wXML microbrowsers, call join/ split, drop-conferencing, etc |
04:23.52 | SlicerDicer | maddog01: I am asking about the Aastra 480i |
04:24.02 | maddog01 | good phones |
04:24.06 | maddog01 | i use them |
04:24.09 | SlicerDicer | good deal :) |
04:24.22 | SlicerDicer | I have the Aastra 480 analogs and such :) thats why I am considering them for sip |
04:24.27 | maddog01 | but i would go for the new 57i |
04:24.35 | [TK]D-Fender | SlicerDicer: What was that about BIAS? |
04:24.56 | SlicerDicer | [TK]D-Fender: if you are aprehensive and get so angry obviously there is something in your line of thinking |
04:24.58 | [TK]D-Fender | and I had a 57i CT... made me wish for my old bedside Polycom IP 301 <- |
04:24.59 | SlicerDicer | just sayin |
04:25.27 | [TK]D-Fender | SlicerDicer: Aastra took a bad turn on the 5i seriesYes... |
04:25.34 | SlicerDicer | 57i yeah I was looking at that one |
04:25.50 | maddog01 | i bought five different phones when i converted to voip |
04:25.52 | SlicerDicer | however I am able to snag some 480i for 75$ |
04:26.12 | SlicerDicer | 480i CT for 135 |
04:26.18 | SlicerDicer | so thats why I am considering :) |
04:26.22 | [TK]D-Fender | SlicerDicer: that is a great deal... |
04:27.39 | maddog01 | good price i perfered the look of the 57i |
04:27.54 | SlicerDicer | understood :) |
04:28.06 | [TK]D-Fender | SlicerDicer: 480i was better than the 5i series... their production standard went crappy... |
04:28.55 | SlicerDicer | what happened that made them go crappy? |
04:29.29 | maddog01 | i don't think so 57i supports alot more feature though |
04:29.44 | [TK]D-Fender | Quick summary of 57i CT = Base & handset have NO wieght, shifted on the table... rubber feet = BLEH. Backlit LCD... with NO VIEWING ANGLE! pixel disply with retarded char-martix DRIVERS (DIE!). Crashed somewhat regularly. shit-for-all rubbery buttons. |
04:29.47 | maddog01 | newer firmware and hardware |
04:29.52 | [TK]D-Fender | ^^^^^^^^ |
04:30.10 | [TK]D-Fender | 5i series buttons = STAB |
04:30.22 | maddog01 | the viewing angles sucks. hes right |
04:30.26 | [TK]D-Fender | grabs a big knife |
04:30.33 | stintel | agrees with the viewing angle ツ |
04:30.36 | SlicerDicer | :/ |
04:30.57 | maddog01 | the buttons are okay |
04:31.04 | [TK]D-Fender | H8 |
04:31.14 | SlicerDicer | maddog01: is it lightweight? |
04:31.16 | maddog01 | they actually feel better then the 3com |
04:31.18 | SlicerDicer | cause I mean my 480's are bricks |
04:31.19 | SlicerDicer | lol |
04:31.31 | maddog01 | i used at another place |
04:31.51 | [TK]D-Fender | 480 is a very decent phone and at 75$ its a no-brainer. |
04:31.51 | SlicerDicer | ahh |
04:32.03 | SlicerDicer | [TK]D-Fender: thats what I was thinking |
04:32.04 | maddog01 | well they are light but that dosn't mean anything |
04:32.17 | [TK]D-Fender | SlicerDicer: now, NEW would be another matter, but would I suggest you spend more and pass up that deal? Hell no. |
04:32.28 | maddog01 | i havent had anyone complain about the weight |
04:32.35 | SlicerDicer | :) |
04:33.01 | [TK]D-Fender | maddog01: Mine dragged across the desk and the PITA of resetting it after calls because of the SHIT FOV... forget it |
04:33.14 | SlicerDicer | [TK]D-Fender: I was thinking of getting 2 of them heh |
04:33.37 | maddog01 | [TK]D-Fender: get a longer cord |
04:33.41 | [TK]D-Fender | SlicerDicer: I'm not sure, but they might be PoE only... be warned |
04:33.49 | maddog01 | no |
04:33.52 | maddog01 | there both |
04:34.05 | [TK]D-Fender | maddog01: 280i's? You sure his have bricks? |
04:34.09 | [TK]D-Fender | 480i* |
04:34.30 | maddog01 | ? |
04:34.43 | [TK]D-Fender | maddog01: Both what? |
04:34.51 | SlicerDicer | [TK]D-Fender: PoE? |
04:34.53 | maddog01 | oh poe |
04:35.04 | [TK]D-Fender | maddog01: Yes.. welcome to the conversation.. |
04:35.07 | [TK]D-Fender | ~poe |
04:35.08 | jbot | methinks poe is Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt. Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet |
04:35.09 | [TK]D-Fender | ^^^ |
04:35.11 | maddog01 | power pack or poe |
04:35.40 | maddog01 | the 480i don't have PoE |
04:35.55 | [TK]D-Fender | SlicerDicer: Most decent SIP phones support PoE for power and some that do don't have wall-plugs. |
04:35.57 | SlicerDicer | I dont think any of the 480i have PoE thats why I was wodnering... |
04:36.01 | [TK]D-Fender | maddog01: Yes they do... |
04:36.16 | SlicerDicer | well fricken A |
04:36.21 | SlicerDicer | thay have it? |
04:36.27 | SlicerDicer | but also all of them have power adapter |
04:36.37 | [TK]D-Fender | SlicerDicer: If they do then you're covered |
04:36.39 | maddog01 | built in or they have a clip on spliter |
04:36.47 | maddog01 | ??? |
04:38.02 | jaytee | and be careful what you pick to use PoE. Cisco's poe switches and injectors only work with their phones, not with Polycoms or any other 802.3af compliant phones |
04:38.24 | SlicerDicer | I will make sure they are not PoE |
04:38.29 | SlicerDicer | that should solve that eh? ;-) |
04:38.46 | jaytee | long as you have an outlet nearby |
04:39.16 | [TK]D-Fender | SlicerDicer: they ARE Poe. That is a FACT. I'm not sure if they are Poe *ONLY* |
04:39.18 | drmessano^ | I love 802.11ruok |
04:39.25 | SlicerDicer | yeah thats what I mean fender |
04:39.44 | [TK]D-Fender | SlicerDicer: If they aren't and you have bricks for them, great. If they come with injectors, equally great. |
04:39.53 | SlicerDicer | jaytee: there is Aastra phones there now currently :) |
04:39.55 | jaytee | only downside to that is power failures. If you have a 802.3af poe switch or midspan hub plugged into a UPS to power phones they'll stay up when the power blips. If you don't then the phone has to reboot |
04:40.00 | maddog01 | i just checked aastras site |
04:40.15 | SlicerDicer | jaytee: they are just analog not voip |
04:40.26 | maddog01 | they don't say anything about PoE |
04:40.43 | [TK]D-Fender | SlicerDicer: 280i no voip? |
04:40.49 | [TK]D-Fender | 480* |
04:40.50 | jaytee | maddog01, what phone? |
04:40.54 | [TK]D-Fender | WTF!? |
04:41.18 | [TK]D-Fender | [23:25]<SlicerDicer>however I am able to snag some 480i for 75$ |
04:41.26 | [TK]D-Fender | SliWTF are are you talking ANALOG for here? |
04:41.55 | SlicerDicer | yep [TK]D-Fender |
04:42.03 | jaytee | fuck sip phones, everyone who's anyone knows that FXS is the new black! |
04:42.08 | SlicerDicer | they are 480 non voip they are old |
04:42.38 | [TK]D-Fender | SlicerDicer: You said 480i there. Multiple times |
04:42.39 | SlicerDicer | remember earlier talking about fxs module stuff ADSI etc etc? |
04:42.46 | drmessano^ | I had a 480i analog |
04:42.47 | [TK]D-Fender | SlicerDicer: 480i = SIP <--- |
04:42.57 | drmessano^ | I gave it to goodwill |
04:42.59 | SlicerDicer | I said there is aastra phones there |
04:43.08 | [TK]D-Fender | SlicerDicer: http://www.aastra.com/cps/rde/xchg/SID-3D8CCB6A-07999B78/04/hs.xsl/19493.htm |
04:43.08 | SlicerDicer | they are analog no voip |
04:43.12 | [TK]D-Fender | Sli480e = analog |
04:43.23 | [TK]D-Fender | SlicerDicer: COMPLETELY different |
04:43.32 | SlicerDicer | I know |
04:43.39 | SlicerDicer | but they look almost identical thats what I was driving at |
04:43.43 | SlicerDicer | same footprint etc etc |
04:43.48 | [TK]D-Fender | SlicerDicer: And you would be a complete retard to pay $75 for a &#^$%ing analog phone |
04:43.51 | SlicerDicer | thus no problem delete/replace |
04:43.52 | drmessano^ | LOL |
04:43.55 | drmessano^ | Yeah |
04:43.59 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
04:44.01 | SlicerDicer | [TK]D-Fender: JESUS H CHRIST!!! |
04:44.14 | maddog01 | sorry im back |
04:44.31 | drmessano^ | Is it "Let me do the worst thing I can do with VoIP" day? |
04:44.32 | SlicerDicer | I have analog phones NOW!!! I am going to buy VOIP!!! readzor please |
04:44.32 | [TK]D-Fender | SlicerDicer: You want to compare FEATURES?! You can't hold a friggen call on those! No MoH! |
04:44.37 | drmessano^ | I thought that was last week? |
04:44.40 | drmessano^ | Shit.. I am off |
04:44.52 | [TK]D-Fender | SlicerDicer: You just said you're goign to go buy the analog 480's |
04:45.05 | SlicerDicer | [TK]D-Fender: as in I am taking your advice and doing away with my analog phones that you said earlier |
04:45.11 | drmessano^ | HES CONFUSED BY THE "PHONE" PART |
04:45.16 | drmessano^ | LEAVEM A LOAN |
04:45.21 | maddog01 | the 480i» Compatible with IEEE 802.3af inline power device |
04:45.21 | maddog01 | » Optional Power Over Ethernet (POE) injector available, but not provider |
04:45.38 | maddog01 | provided |
04:45.45 | SlicerDicer | [TK]D-Fender: where di I say I was going to buy "analog" |
04:45.48 | drmessano^ | How does 802.3af work with analog? |
04:45.51 | maddog01 | straight from aastra |
04:45.53 | drmessano^ | Ping 10 and 14? |
04:45.57 | drmessano^ | Pin |
04:46.02 | maddog01 | the 57i is built in |
04:46.17 | SlicerDicer | [21:40:42] <SlicerDicer> jaytee: there is Aastra phones there now currently :) |
04:46.17 | SlicerDicer | SlicerDicer> jaytee: they are just analog not voip |
04:46.27 | SlicerDicer | [TK]D-Fender: does that clear it up? |
04:46.42 | SlicerDicer | I said nothing about buying analog phones |
04:46.49 | [TK]D-Fender | SlicerDicer: I hope to hell not. |
04:47.10 | [TK]D-Fender | SlicerDicer: But your broken prasing wasn't helping |
04:47.14 | [TK]D-Fender | phrasing* |
04:47.20 | SlicerDicer | lol |
04:47.22 | [TK]D-Fender | dang, typing skills fading fast... |
04:47.32 | SlicerDicer | sorry my brain works |
04:47.34 | SlicerDicer | in clusters! |
04:47.36 | SlicerDicer | ;-) |
04:47.50 | SlicerDicer | I will try to complete my thoughts better :) |
04:47.50 | maddog01 | SlicerDicer: i would buy 5xi series they have alot more feature and faster hardware but if your looking from the cheapes price go with the 480i |
04:48.27 | SlicerDicer | maddog01: basicly what I wanted to do was get my house fitted out with voip phones to start with |
04:48.36 | SlicerDicer | once that transition is complete I can work on going "nicer" |
04:48.48 | SlicerDicer | then pawn the other phones down the food chain to family and what not maybe :) |
04:48.56 | SlicerDicer | pawn = give |
04:49.29 | maddog01 | lol |
04:50.08 | maddog01 | you don't need aastra phones for your house |
04:50.20 | maddog01 | get some thing cheap |
04:50.25 | maddog01 | and wireless |
04:50.47 | maddog01 | vtech or panasonic sip phones with dect |
04:51.00 | SlicerDicer | hmm |
04:51.14 | SlicerDicer | I dont like "cheap" cause that leads to shit quality |
04:51.15 | maddog01 | these are business phones |
04:51.21 | SlicerDicer | least I know the aastra had good quality :) |
04:52.18 | drmessano^ | I've used ATA's and cheap phones at home.. Now I have a mix of things |
04:53.20 | drmessano^ | VoIP phones in the office, next to the bed.. ATA's with $25 5.8GHZ cordless on them for roaming around.. Then a couple $5 phones on ATA's that are wired into the house wiring for filling in the gaps |
04:53.28 | SlicerDicer | http://idisk.mac.com/slicerdicer/Public/phone.JPG blurry yes but thats what I currently use maddog01 |
04:53.59 | SlicerDicer | I actually have 3 of them all together |
04:54.10 | drmessano^ | I've been told the $18 vtech 5.8GHZ at walmart is an awesome phone |
04:54.27 | SlicerDicer | I gave my 5.8ghz vtech away |
04:54.39 | denon | but does it have a sepaeate flash button! |
04:54.44 | denon | separate |
04:54.47 | denon | that's the question |
04:54.52 | maddog01 | slicerdicer: check this out http://www.canadianvoipstore.com/product_info.php?products_id=3708 |
04:54.54 | drmessano^ | lol |
04:55.14 | maddog01 | thats more then enought |
04:55.23 | SlicerDicer | CAD? |
04:55.30 | SlicerDicer | thats like what 20$ USD? ;-) |
04:55.37 | drmessano^ | I'm done listening to SlicerDicer anyway.. $75 analog Aastras.. WTF |
04:55.50 | SlicerDicer | drmessano^: errrm... damn it |
04:56.01 | SlicerDicer | drmessano^: they are voip aastras... |
04:56.04 | maddog01 | SlicerDicer: shit wrong phone and whats wrong with canada |
04:56.07 | SlicerDicer | the one I linked the picture of is what I have |
04:56.12 | maddog01 | one csec |
04:56.14 | maddog01 | sec |
04:56.20 | SlicerDicer | maddog01: nothing is wrong with canada :) |
04:56.21 | drmessano^ | maddog01: Damn canucks live there |
04:56.40 | maddog01 | lol |
04:56.40 | drmessano^ | maddog01: Three words "Canadian bacon, eh" |
04:56.56 | maddog01 | i'm having some right now. lmao |
04:56.58 | SlicerDicer | lol |
04:56.59 | drmessano^ | and it's "ABOUT", not "ABOOT" |
04:57.14 | SlicerDicer | ABOOT is a car trunk |
04:57.27 | SlicerDicer | or for ABOOT up your ass ;-) |
04:57.48 | drmessano^ | jaytee: Did you LOL? |
04:57.51 | maddog01 | now now lets not start a international insident |
04:58.18 | jaytee | LOL |
04:58.32 | SlicerDicer | lol |
04:58.41 | drmessano^ | Go play some HACKEY, TOOK-face! |
04:58.45 | SlicerDicer | maddog01: ahh I am far enough away |
04:58.56 | SlicerDicer | I can flee to mexico before canada makes it to New Mexico ;-) |
04:59.12 | maddog01 | lol |
04:59.25 | drmessano^ | BYE,EH is NOT A SIP HEADER |
04:59.30 | maddog01 | but yet you want to buy a canadian made phone |
04:59.50 | SlicerDicer | maddog01: your argument is invalid as I like canada :) |
04:59.56 | drmessano^ | maddog01: An analog one, to BOOT |
04:59.59 | SlicerDicer | I just have some fun from time to time :) |
05:00.08 | maddog01 | aastra the phones you all love and hate |
05:00.21 | SlicerDicer | I mean hell I am in New Mexico.. think of all the shit you could come up with for me... LOL |
05:00.22 | drmessano^ | Oh god, he puts smileys at the ends of sentences to convey non-threatingness |
05:00.41 | drmessano^ | Oh god, he puts smileys at the ends of sentences to convey non-threatingness :) |
05:00.45 | SlicerDicer | lol |
05:00.58 | drmessano^ | Next it'll be "hah lol j/k" |
05:00.59 | SlicerDicer | only thing bad about canada? |
05:01.05 | SlicerDicer | fucking cold... |
05:01.10 | SlicerDicer | nuff said on that |
05:01.11 | SlicerDicer | lol |
05:01.24 | SlicerDicer | hay guise! does that work drfreeze |
05:01.29 | SlicerDicer | bah tab completion ftl |
05:01.31 | drmessano^ | maddog01: I do have one nice thing to say about canada |
05:01.56 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
05:02.04 | drmessano^ | maddog01: I have proof that contrary to popular comedic belief, all the assholes are NOT in canada |
05:02.15 | drmessano^ | maddog01: They migrate south |
05:02.17 | jaytee | speaking of Canadian, a friend of mine in another chat channel was just talking about Mafia Wars on Facebook, she said: "When I'm feeling silly I just go to the hit list and sucker punch someone at random, heh" |
05:02.18 | SlicerDicer | lol... |
05:02.24 | *** join/#asterisk rob0 (n=rob0@cardinal.lizella.net) |
05:02.33 | maddog01 | yah there just south of canada |
05:02.37 | maddog01 | lmao |
05:02.53 | drmessano^ | maddog01: Some even closer to mexico |
05:03.00 | jaytee | I'm a Masshole by birth but I live in Indiana now |
05:03.11 | maddog01 | lol |
05:03.12 | SlicerDicer | hey now |
05:03.14 | SlicerDicer | I resemble that remark |
05:03.17 | drmessano^ | maddog01: Where $75 analog phones roam free like el chupacabra |
05:03.24 | jaytee | why, you from Mass? |
05:03.38 | maddog01 | lmao |
05:03.41 | SlicerDicer | I am from New Mexico... thats close to Mexico |
05:03.46 | jaytee | oh |
05:04.10 | drmessano^ | maddog01: Apparently the legend of El Cheapacabra is true |
05:04.13 | jaytee | sad about Richardson having to bow out |
05:04.19 | SlicerDicer | but we dont use Pesos here just to clear that up |
05:04.22 | SlicerDicer | yeah jaytee |
05:04.29 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
05:04.31 | SlicerDicer | I hope he has done nothing wrong for the sake of new mexico.. |
05:04.33 | maddog01 | drmessano^: is on a roll |
05:04.35 | rob0 | It's newer than Mexico, obviously! |
05:04.36 | SlicerDicer | we dont need another scandal... |
05:05.06 | jaytee | Land of Enchantment my ass! how enchanting can several hundred thousand square miles of desert be anyways? |
05:05.17 | SlicerDicer | jaytee: well it keeps the unbelievers out |
05:05.23 | SlicerDicer | they rot at the borders |
05:05.27 | rob0 | I think they're talking about upgrading the name ... New and Improved Mexico |
05:05.31 | jaytee | leaving just you tin foil hat folk |
05:05.35 | *** join/#asterisk Zippoman (n=bobperry@cpe-76-95-113-203.socal.res.rr.com) |
05:05.49 | maddog01 | newer Mexico |
05:05.52 | maddog01 | lmao |
05:05.54 | SlicerDicer | lol |
05:06.20 | jaytee | how about changing the motto to "Yeah, we got Roswell and aliens" |
05:06.40 | SlicerDicer | not kosher with Groome Lake jaytee |
05:06.47 | Zippoman | hey can someone help me out with some asterisk issues? |
05:06.55 | drmessano^ | ~elcheapacabra |
05:06.55 | jbot | Watch out, EL CHEAPACABRA will suck your blood, leaving a lifeless corpse, all for the thrill of putting cheap analog phones on barely working Trixbox PII-300s with less RAM than an iPhone. BEWARE! |
05:07.11 | jaytee | what does Groome Lake have to do with Hassidic dietary laws? |
05:07.19 | maddog01 | slicedicer: okay i found a good site with a bunch of phones: http://www.cordless-phones.uk.com/voip-phones/voip-dect-phones/ |
05:07.30 | drmessano^ | VoIP DECT PHONES |
05:07.33 | drmessano^ | WHY O WHY |
05:08.02 | SlicerDicer | yeah but thats UK? |
05:08.02 | jaytee | because we can! |
05:08.07 | drmessano^ | ATA + Some Analog Deal of the Weak DECT/non-Dect phone |
05:08.12 | drmessano^ | oh lord |
05:08.18 | mchou | hell no to VoIP DECT |
05:08.18 | maddog01 | cordless voip phones for the house |
05:08.22 | Zippoman | what does it mean if i call my number and i get a long silence |
05:08.30 | mchou | agrees with drmessano^ |
05:08.39 | drmessano^ | maddog01: STOP POSTING LINKS WITH INFO THAT HAVE FOREIGN DOMAINS.. YOURE CONFUSING HIM WITH TLDs |
05:08.39 | maddog01 | drmessano: who still uses analog |
05:08.46 | drmessano^ | maddog01: Thankya |
05:08.56 | SlicerDicer | drmessano^: no.. I just dont usually order international if I can help it |
05:09.06 | SlicerDicer | I dont like customs peering in my shit |
05:09.07 | SlicerDicer | lol |
05:09.14 | drmessano^ | maddog01: I do.. ATA + Cordless analog is a WAYYY better idea than some VoIP DECT phone.. |
05:09.26 | maddog01 | it's a site for information |
05:09.32 | SlicerDicer | ohh |
05:09.47 | maddog01 | you have to find the phones locally |
05:09.54 | drmessano^ | At least with an ATA you have a known working SIP client and can put whatever Cordless you want on it |
05:10.13 | Zippoman | can any of you help me? |
05:10.25 | drmessano^ | Otherwise youre stuck with a niche product that is gonna have less QC in the field |
05:10.29 | mchou | yup. and cordless analog phone are improve very quickly and are inexpensive |
05:10.42 | maddog01 | drmessano: your right but i stopped using analog in 2002 |
05:10.56 | maddog01 | drmessano: i dont want to go back |
05:11.05 | drmessano^ | maddog01: What's wrong with an ATA? |
05:11.14 | drmessano^ | I didnt say "USE ANALOG" |
05:11.19 | SlicerDicer | mchou: vtech are nice no ;-) |
05:11.20 | drmessano^ | I said USE AN ANALOG PHONE |
05:11.33 | maddog01 | i still have to used one for my 10K fax/printer thats what |
05:11.46 | mchou | SlicerDicer: next time get panasonic :) |
05:11.51 | SlicerDicer | haha |
05:11.56 | drmessano^ | Christ, an analog phone on an ATA is hardly going back 20 years in time.. What the hell do you think is inside your dect VoIP phone? |
05:12.03 | drmessano^ | Cordless bits + ATA |
05:12.10 | drmessano^ | Just not in 2 boxes |
05:12.30 | maddog01 | depends on the cordless phone |
05:12.49 | maddog01 | some are pure digital some are hybrids |
05:12.58 | rob0 | Folks, I find myself in the very sad situation of having to block an abusive caller, by caller ID. I know it can be done, just not sure about the "best" way. Suggestions please? Thanks. |
05:12.58 | maddog01 | your right |
05:13.03 | drmessano^ | Well, I dont need a multiline Dect VOIP system.. thats why I have a PBX |
05:13.31 | maddog01 | call then and tell them to stop calling you. lol |
05:13.32 | rob0 | and no, it is not a creditor ;) |
05:13.33 | mchou | actually, are there consumer corldeless analog phones with full duplex speaker? |
05:13.48 | mchou | cordless* |
05:13.53 | SlicerDicer | http://www.cordless-phones.uk.com/img/watermark/008189.jpg thats just lunacy maddog01 |
05:14.05 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:5d00:b8cf:9894:f1a5) |
05:14.05 | maddog01 | a pbx for a house is over kill |
05:14.29 | drmessano^ | So is a multiline VoIP Dect system |
05:14.40 | drmessano^ | I see no justification for it |
05:14.55 | mchou | rob0: google astdb blacklist |
05:15.27 | mchou | rob0: is it an ex-girlfriend? :) |
05:15.31 | maddog01 | SlicerDicer: what phone was that |
05:15.38 | SlicerDicer | http://www.cordless-phones.uk.com/voip-phones/voip-dect-phones/siemens-a580-ip-phone |
05:15.44 | drmessano^ | HA |
05:15.46 | maddog01 | ah |
05:15.46 | Zippoman | can any of you help me configure my asterisk for certain tasks...id be happy to pay for services |
05:16.05 | maddog01 | you can do up to 8 handsets |
05:16.05 | mchou | Zippoman: depends on the task |
05:16.07 | drmessano^ | mchou: Im staying out of this one.. Maybe Panasonic will sell a VoIP dect phone after all |
05:16.08 | maddog01 | nice |
05:16.30 | mchou | drmessano^: panasonic sells consumer voip phones |
05:16.39 | rob0 | mchou: it is a very sad story. I failed as a parent. It is my daughter. :( |
05:16.45 | drmessano | mchou: I know.. scary |
05:16.51 | maddog01 | all: it's on the link i posted |
05:16.51 | rob0 | thanks for the tip BTW |
05:16.55 | mchou | rob0: you serious? |
05:17.14 | rob0 | unfortunately yes, phone is ringing even now. |
05:17.20 | rob0 | ringers turned off |
05:17.21 | mchou | lordy |
05:17.38 | maddog01 | rob0: get a new number |
05:17.44 | mchou | wtf is going on w/ society these days |
05:17.51 | Zippoman | lmao |
05:17.58 | maddog01 | lmao |
05:18.20 | drmessano | You're blocking your daughter? |
05:19.12 | Zippoman | anyone think they can help me out |
05:19.18 | drmessano | puts on his drphil mask |
05:19.20 | denon | Dr Phil |
05:19.26 | drmessano | So, tell me where the problem started |
05:19.28 | maddog01 | lmao |
05:19.33 | denon | no .. not drmessano as dr phil .. the real dr phil |
05:19.35 | drmessano | What brought things to this point |
05:19.35 | denon | drmessano is trouble |
05:19.36 | denon | :) |
05:20.00 | mchou | lol |
05:20.21 | mchou | drmessano is not a REAL doctor |
05:20.22 | drmessano | At what point did things get so bad, that you needed to Asterisk blacklist your own daughter? |
05:20.27 | drmessano | Was it the iPhone? |
05:20.35 | Zippoman | hahaha |
05:21.05 | maddog01 | l don't think i have ever laughted this hard |
05:21.17 | maddog01 | LMFAO LMFAO LMFAO LMFAO LMFAO LMFAO LMFAO LMFAO LMFAO |
05:21.24 | drmessano | Where are the hugs? Where is the love? Where is "I love you, even if you did stab my cat?" |
05:21.28 | mchou | dude, this is no laughing matter |
05:21.29 | drmessano | Can we get a hug here? |
05:21.46 | Zippoman | this sucks for that guy |
05:21.48 | drmessano | When did you know she had a drug problem? |
05:21.58 | drmessano | errr |
05:22.05 | rob0 | Thanks mchou, and guys, we really are broken up about it. She's 19, just married a 52-y-o loser. |
05:22.13 | drmessano | Ouch |
05:22.22 | rob0 | a total wacko |
05:22.23 | drmessano | readjusts his hat |
05:22.26 | maddog01 | sugar daddy |
05:22.30 | Zippoman | im sorry dude |
05:22.30 | rob0 | you think I'm bad, boy! |
05:22.34 | drmessano | rob0: Can we get him on the show |
05:22.52 | rob0 | heck no, he's a total loser, has NEVER had a real job that I can tell |
05:22.53 | drmessano | rob0: I want to know.. what is going through this mans head. She is just a chid |
05:22.56 | drmessano | child* |
05:23.11 | rob0 | I can tell you. He wants someone to be his meal ticket. |
05:23.43 | drmessano | rob0: We don't extend this offer very often, but the Dr Phil show has a team of hired assassins that keep us from having followup shows in tough cases |
05:23.48 | drmessano | rob0: How can WE help YOU? |
05:23.58 | rob0 | haha :) that might help |
05:24.29 | *** join/#asterisk Maliuta_CA (n=biteme@206.47.36.150) |
05:24.49 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-5393d80c55968350) |
05:24.56 | maddog01 | some one should rename this channel the Comedy Channel |
05:24.58 | maddog01 | lol |
05:25.15 | drmessano | You know.. Im not against the whole "marrying someone half your age" thing, but when its because her job at Burger Barn is your way out of the trailer park, TIME FOR A LIFE READJUSTMENT |
05:26.25 | Zippoman | hey guys someone please help me out with this...i call my number and i just get a long silence...im using voicepulse...i think i have a problem with my extensions.conf...i basically want to be able to call my number and then me prompted to enter another number...any help greatly appreciated |
05:26.26 | rob0 | I hate to get anyone's interest piqued, but all I can really say is that it's far worse than it sounds. |
05:26.51 | drmessano | rob0: Perhaps rather than helping block your daughter, maybe we can mend this problem by helping you set up a dialplan to give him a cardiac at 3am on some random Thursday.. I am checking "show applications" now.. |
05:26.52 | rob0 | Zip, verbosity in the console? Are the calls arriving? |
05:27.38 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
05:29.18 | drmessano | rob0: While it's not my daughter, I have a sister, my only sibling, who has two children by some hippie druggie loser who has beat her to the point of being put in the hospital twice, which doesnt count the daily beatings that dont warrant emergency care. I almost never want to answer the phone when she calls |
05:29.42 | drmessano | People are sick |
05:30.11 | maddog01 | i there is more serious matters but im watching Angelina Jolie get her ass handed to her in Mr & Mrs Smith and she is so hot |
05:30.12 | rob0 | Indeed, a screwed up society. At least my other kids are doing better. |
05:31.37 | drmessano | rob0: Her kids have already learned profanity that I have yet to pick up in my day-to-day.. and I work in *IT*.. oh, they are 4 and 2 |
05:32.14 | maddog01 | thats suck drmessano |
05:32.16 | mchou | rob0: I'm curious. did you use asterisk before all this happened? Or are you setting asterisk up just for this? |
05:32.30 | maddog01 | that sucks drmessano |
05:32.59 | rob0 | I've been using * for several years, dating back to pre-1.0 and pulling from cvs. |
05:33.17 | rob0 | 2004 maybe |
05:35.30 | maddog01 | rob0: just setup a inbound route for that Caller ID and tell it to ring busy or hang up or go to voice mail. |
05:35.45 | rob0 | yeah I want to do the VM |
05:35.48 | drmessano | I was JUST gonna say that |
05:36.52 | rob0 | it will take her some time to figure out that she's blocked, and we'll be able to identify those calls by area code. |
05:37.05 | rob0 | (those calls==calls from other numbers) |
05:37.29 | maddog01 | ah |
05:37.54 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
05:38.32 | drmessano | You know, I always store paint thinner in Wild Turkey bottles |
05:38.33 | maddog01 | has anyone setup fax to email on a trixbox install |
05:38.37 | drmessano | Im just sayin |
05:39.09 | drmessano | Oh, world.. ---> /clear |
05:39.24 | maddog01 | the faxes come in and get stored to the tmp folder but don't send |
05:39.28 | maddog01 | any idea |
05:39.56 | drmessano | You said trixbox.. I heard noises and fuzzy light from there |
05:40.10 | maddog01 | they get saved as tif but not converted to pdf |
05:40.24 | maddog01 | lol |
05:40.58 | drmessano | Kerry has a hitman looking for me, I would think |
05:41.07 | maddog01 | i know but i liked the trixbox asterisk scripts |
05:41.11 | drmessano | GPLassassin |
05:41.56 | maddog01 | i ment trixbox aastra scripts |
05:42.06 | drmessano | Trixbox blows harder than an air pump for the Charlie Brown float at the Macy's Thanksgiving Day Parade |
05:43.14 | Zippoman | Ok so I need help...I have an asterisk box and I am going to use this line so my cellphones can dial in to the asterisks DID and then have a voice prompt on it asking for a telephone number to call next...I looked into this one extension type thing that had the same idea but it was for entering caller id so you could spoof.... then asking for the telephone number to dial out...this does not work...when i call the phone I just get t |
05:43.15 | SlicerDicer | trixbox is ET |
05:44.04 | maddog01 | whats ET?? |
05:44.26 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-236-67-190.dsl.pltn13.sbcglobal.net) |
05:44.30 | mchou | Entertainment tonight? |
05:44.44 | maddog01 | us canadian arent down with the mexican lingo |
05:44.46 | maddog01 | lol |
05:45.38 | maddog01 | Zippoman: look into inward system access |
05:46.20 | SlicerDicer | maddog01: ET, Phone Home! |
05:46.26 | maddog01 | lol |
05:46.30 | SlicerDicer | lol |
05:46.55 | maddog01 | SlicerDicer: did you find anyphones you like |
05:47.07 | SlicerDicer | what at the site you linked? |
05:47.16 | SlicerDicer | those dect phones are out of control man.. thats all I can say |
05:47.16 | maddog01 | yes |
05:47.24 | maddog01 | lol |
05:47.31 | SlicerDicer | wtf is with USA not having phones like that? |
05:47.45 | SlicerDicer | I mean the one that vtech offers for USA is like a FORD... |
05:47.45 | mchou | cause USA is sane |
05:47.56 | SlicerDicer | fucker only rolls downhill ;-) |
05:48.03 | maddog01 | North america is always behind |
05:48.05 | SlicerDicer | like the Taurus sorta that garbage car |
05:48.06 | Zippoman | i looked into the inward system access didnt have much info and all pretty much has sip.conf |
05:48.34 | jaytee | peace out! |
05:49.00 | drmessano | mchou: Indeed |
05:49.04 | SlicerDicer | mchou: the dect phones would not be so bad I guess if they were as cheap as normal models |
05:49.22 | mchou | SlicerDicer: what happens when your phone batteries no longer take charge? |
05:49.25 | SlicerDicer | like the set I got from costco if they were cheap like that for all the handsets and shit. I could justify it |
05:49.37 | SlicerDicer | mchou: well with vtech you can buy new batteries |
05:49.39 | SlicerDicer | they are not expensive |
05:49.44 | drmessano | mchou: You need to find a battery for an uncommon niche cordless phone, DUH |
05:49.56 | SlicerDicer | if they are NMIH beat them with a hammer |
05:50.01 | SlicerDicer | short the connections |
05:50.05 | SlicerDicer | will give you a bit more life |
05:50.09 | SlicerDicer | I keep doing that with my drill lol |
05:50.13 | maddog01 | Direct Inward System Access: its part of free pbx |
05:50.14 | mchou | you get a regular dect phone, when batteries die you buy new phone (cause batteries alone cost same as new phone) |
05:50.23 | SlicerDicer | yeah |
05:50.26 | drmessano | mchou: Exactly |
05:50.30 | SlicerDicer | mchou: I was saying if they were sane |
05:50.40 | SlicerDicer | and offered stuff like the analog variants |
05:50.43 | drmessano | SlicerDicer: THATS THE WHOLE POINT.. ITS A DUMB IDEA |
05:50.54 | SlicerDicer | well it came from europe |
05:50.56 | SlicerDicer | what do you expect |
05:51.23 | SlicerDicer | (formerly Digital European Cordless Telephone) |
05:51.35 | drmessano | My 5.8GHZ $25 GE Wal Mart phone on a PAP2 works great when I need cordless |
05:51.45 | maddog01 | Zippoman: DISA for short |
05:51.52 | drmessano | SlicerDicer: I dont mean DECT |
05:52.12 | SlicerDicer | drmessano: thats what I was talking about.. |
05:53.09 | maddog01 | vtech is the lower form of consumer phone there is. what do you want you get what you pay for. |
05:53.19 | drmessano | SlicerDicer: We are talking about this stupid idea of buying a VoIP DECT phone that not only has a SIP UA that will see little real world testing and likely never a firmware update, but a handset that will be nearly impossible to find batteries for, and you're locked into a $100 cordless phone long after than generations tech has died and theres something newer/better that all the other kids are using on now $18 phones |
05:54.21 | mchou | in fact almost all consumer cordless VoiP phones have died on the vine |
05:54.53 | mchou | just too many issues |
05:55.09 | drmessano | Too much lock-in as well.. Which will always keep that market small |
05:56.09 | mchou | plus most consumer phones arent easily hackable :) |
05:56.40 | mchou | firmware hacking is fun |
05:56.51 | maddog01 | the avg. joe is not ready for cordless voip phone but for someone with a tech savy why not |
05:57.03 | drmessano | maddog01: For all the mentioned reasons |
05:57.07 | drmessano | Its a BAD idea |
05:57.34 | maddog01 | that's becuse your hung up on analog |
05:57.38 | maddog01 | let it go |
05:57.40 | drmessano | Has nothing to do with it |
05:57.44 | drmessano | Youre not reading |
05:57.47 | drmessano | Im not hung on analog |
05:57.58 | maddog01 | just like tv signals in feb 19. |
05:58.06 | maddog01 | analog is dying |
05:58.06 | drmessano | Im hung on not putting all my eggs in one basket like a fucking moron |
05:58.14 | mchou | maddog01: how do you overcome the battery and firmware issues? :) |
05:58.21 | drmessano | Buying some $120 phone with a SIP agent that will never see a firmware update |
05:58.30 | drmessano | That less than 1000 people are gonna buy |
05:58.44 | drmessano | that I will get HALF the life out of than I should due to quality issues |
05:58.58 | maddog01 | buy a phone from a company like seimens or panasonic |
05:58.59 | drmessano | Go buy a TV/DVD combo if you like all-in-one so much |
05:59.13 | drmessano | BTW, when the DVD player starts skipping, enjoy the TV |
05:59.25 | drmessano | Seimens or panasonic doesnt mean shit to VoIP |
05:59.26 | maddog01 | lol |
05:59.42 | maddog01 | means more then vtech |
06:00.08 | drmessano | Youre a trixbox user.. why is this convo even relevant? |
06:00.27 | drmessano | Seriously.. youre arguing about Siemens and Panasonic phones and clutching to Trixbox |
06:00.51 | drmessano | Make me stop LOLing |
06:01.23 | drmessano | Buying a DECT VoIP phone is about as lame as buying a Skype phone |
06:01.35 | maddog01 | not clutching im in the process of installing asterisk from scrach on cent os |
06:01.53 | maddog01 | trixbox was a start point |
06:03.34 | drmessano | VoIP DECT phones make no sense economically.. An ATA with your choice of Cordless gives you the ability to swap the phone out whenever you want/need, gives you a PROVEN SIP User Agent that actually HAS firmware updates, that is supported on every platform out there, and is also much cheaper |
06:03.37 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
06:03.39 | drmessano | Its a win, win, win |
06:04.08 | orkid | well, where's that >5.1.7 update for spa-3102! |
06:04.09 | orkid | arg |
06:04.15 | orkid | (for the double hook flash problem :P |
06:04.20 | mchou | lol |
06:04.34 | drmessano | "How can I make my Siemens FUFJGN-8078475 work on Asterisk? Help me config?" |
06:04.38 | drmessano | ^^^^ LAFF |
06:05.23 | mchou | orkid: gotta hack the firmware for that |
06:05.27 | drmessano | "Oh, theres a bug in that firmware thats been well documents on Voxilla. The new Siemens fixed it, but they dont do firmware updates" |
06:05.35 | drmessano | doucmented |
06:05.38 | mchou | orkid: or rever to 3.x |
06:05.47 | mchou | revert* |
06:06.02 | drmessano | Thats what I want.. Lemme go buy a new VoIP DECT phone for a bug fix |
06:06.12 | orkid | mchou: is there a webpage that describes how to hack it up right? |
06:06.41 | mchou | orkid: please. firmware hacking is fine art |
06:06.59 | mchou | orkid: nobody gonna put it up on web page :) |
06:07.05 | drmessano | mchou: So youre saying I need Word instead of Notepad? |
06:07.05 | orkid | why not :P |
06:07.18 | orkid | mchou: are there hacked up firmwares available for the 3102? |
06:07.33 | drmessano | heh |
06:07.41 | mchou | orkid: disassemble |
06:08.16 | orkid | ... |
06:08.19 | drmessano | mchou: Just a link to the hacked firmware please.. I cant code, but I wanna tell people I am using hacked firmware and be cool |
06:08.28 | drmessano | mchou: KTHX |
06:08.32 | denon | ata+cordless is always going to have a crappy interface |
06:08.41 | denon | and you'll be jackin around with flash and # |
06:08.46 | drmessano | So are most DECT VoIP phones |
06:08.57 | orkid | drmessano: not everyone is as 'cool' as you |
06:08.58 | denon | well, I guess I was thinking of a native sip phone |
06:08.58 | mchou | drmessano: spa-3102 has real double hook flash bug |
06:09.00 | drmessano | They're barely more than that |
06:09.08 | denon | wifi sip |
06:09.20 | mchou | lol |
06:09.21 | denon | of course, they're lacking in other ways .. |
06:09.23 | drmessano | This isnt about Wifi SIP phones |
06:09.31 | denon | but they have a consistent interface with other sip phones people are used to |
06:09.33 | mchou | that's even eorse than VOIP DECT |
06:09.39 | mchou | worse* |
06:10.12 | orkid | mchou: have you ever heard of anyone successfully hacking in double-flash into 5.1.7? which is the last firmware that works with double-flash, do you know? |
06:10.43 | mchou | orkid: 3.x, cant remember exact version |
06:10.59 | mchou | not posted on linksys official web site |
06:11.13 | drmessano | Other than IP configuration, my $25 cordless GE phone as a single line phone is not missing one feature my desktop VoIP phones have interface-wise.. except for a VM specific button, but stored memory 1 does that too |
06:11.28 | orkid | only one in the 3.1.x i know of is 3.1.2 |
06:11.45 | mchou | orkid: then that must be it |
06:12.04 | denon | drmessano: attended xfer, blind xfer, sip hold ... |
06:12.19 | mchou | denon: that's ALL supported |
06:12.23 | orkid | ... i was actually going to get a VOIP DECT.. siemens A580IP.. what are you guys' issues with VOIP DECT? |
06:12.24 | drmessano | denon: I use the same feature codes |
06:12.35 | drmessano | denon: No loss there |
06:12.40 | denon | but no dedicated button for em |
06:12.52 | drmessano | My desktop phones dont have it |
06:13.02 | denon | oh, then you have crappy desktop phones :) |
06:13.11 | drmessano | Polycoms are shit, I agree |
06:13.15 | drmessano | As are the Linksys |
06:13.16 | mchou | lol |
06:13.22 | drmessano | Yep, youre right |
06:13.30 | mchou | this ia great |
06:13.31 | drmessano | What a dumb fucking conversation |
06:13.34 | drmessano | Seriously |
06:13.42 | denon | I dont get it .. since when don't your linksys phones have a xfer button? |
06:13.50 | denon | soft buttons count as buttons |
06:14.38 | drmessano | Generally, I dont use the soft buttons |
06:14.58 | denon | well, ok, Hold then .. |
06:15.06 | denon | your linksys has a big huge honkin hold button |
06:15.17 | drmessano | Ok |
06:15.25 | mchou | so does my analog handset |
06:15.37 | mchou | (cordeless) |
06:15.40 | denon | mchou: a hold button that will trigger pbx music .. not just beep-beep |
06:15.55 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-13a2bc6c6cbb9e04) |
06:16.03 | orkid | denon is pro VOIP DECT, mcho/drmessano anti? |
06:16.08 | drmessano | Well I give up.. I guess a Diahatsu VoIP DECT phone is better |
06:16.09 | denon | users always get confused by flash for hold/xfer/etc .. they always lost more calls |
06:16.15 | drmessano | Well worth it |
06:16.38 | orkid | hello? |
06:16.46 | mchou | orkid: I have Desktop sip phones. thaes up way too much desk space |
06:16.46 | drmessano | As is a 2.4GHZ VoIP cordless |
06:16.47 | denon | orkid: I'm not pro anything, I'm just mentioning the fact that native sip phones do have their advantages, such as voip sip, not necessarily dect, unless they extend the sip featureset to the handset |
06:16.53 | mchou | takes* |
06:16.55 | drmessano | Which is still useful 2 years alter |
06:17.01 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
06:17.04 | drmessano | Yep.. |
06:17.29 | drmessano | I bet you can get a good deal on used 2.4GHZ VoIP Cordless phones on eBay |
06:17.33 | orkid | denon: do you know of any 'decent' ones w/o major bugs? or are most still buggy |
06:17.38 | mchou | orkid: the only strike against analog handests are crappy speakerphones |
06:17.43 | drmessano | Real cheap.. Paid $250, probably get em for $125 |
06:17.47 | mchou | handsets* |
06:18.04 | denon | orkid: that seems to vary a lot on user |
06:18.30 | orkid | there are lots of crappy voip 2.4ghz phoens on ebay.. they all strike me as real junky, not nice buttons, numbers that ruboff.. etc. i was thinking about siemens gigaset, but i never had one really. only used a few.. they aren't sold in canada |
06:18.46 | drmessano | orkid: The point went WAY over your head |
06:18.53 | drmessano | Let me use smaller thoughts |
06:19.00 | orkid | maybe.... is it your usual sarcasm? |
06:19.04 | drmessano | No |
06:19.25 | drmessano | Lets go back two years and have this convo about some great 2.4GHZ Siemens DECT VoIP phones |
06:19.37 | orkid | ... |
06:19.41 | drmessano | Now lets go forward two years and look back at my awesome $175 purchase and how its doing |
06:19.46 | drmessano | errr |
06:19.49 | orkid | why not just get to the point? |
06:19.52 | drmessano | Lets go back two years and have this convo about some great 2.4GHZ Siemens Cordles VoIP phones |
06:19.55 | drmessano | I AM |
06:19.57 | drmessano | FUCKING LISTEN |
06:20.02 | orkid | whoa |
06:20.09 | orkid | ok |
06:20.10 | orkid | ... |
06:20.26 | drmessano | Lets go back two years and have this convo about some great 2.4GHZ Siemens Cordless VoIP phones and flash forward to today about how useful they are now |
06:20.36 | drmessano | and how much my $175 purchase was worth |
06:21.12 | drmessano | ^^^^^^ Obsolete waste of money thanks to two years of technological advances |
06:21.34 | drmessano | What made it obsolete? |
06:21.37 | drmessano | Not the SIP agent |
06:21.45 | drmessano | The RF technology |
06:22.06 | orkid | i think i still use a 900MHz analog cordless, we've had it for almost 10 years? |
06:22.10 | orkid | and it's great. |
06:22.28 | drmessano | You're throwing away the $50 part of the $175 phone because a 2.4GHZ phone is useless |
06:22.28 | denon | 900's only great now that everyone is off it ;) |
06:22.43 | drmessano | Point >>> **** You're throwing away the $50 part of the $175 phone because a 2.4GHZ phone is useless **** |
06:22.49 | orkid | i actually was going to think that SIP/VOIP is too mocing too fast/much for the voip dects to keep up.. but i guess that's not what you were trying to say |
06:23.04 | drmessano | IT IS MY POINT |
06:23.05 | orkid | what's wrong with DECT though? are there too many users? |
06:23.08 | drmessano | No |
06:23.43 | orkid | so? |
06:23.45 | drmessano | But in two years, when DECT 8.0 is out and you want that 2 mile coverage and ePenis bragging rights, you gotta toss all that out the door |
06:24.11 | orkid | oh... i'm ok with a phone that does sip/voip well, whatever the rf and distance.. |
06:24.11 | drmessano | All because the $50 part of the device is outdated |
06:24.21 | drmessano | .... |
06:24.24 | drmessano | Forget it |
06:24.27 | denon | drmessano: you're forgetting that to some business users, they'd be replacing in 2 years anyway |
06:24.28 | orkid | the 900 is good enough really, but the double-hook-flash doesn't work, and having multi-handsets would be nice |
06:24.31 | denon | constantly rotating |
06:24.38 | denon | ie: warehouse staff are hard on hardware |
06:24.42 | denon | but need lots of toys, often |
06:25.01 | drmessano | denon: Bullshit.. Do you have any idea how long people keep phones? YEARS |
06:25.08 | orkid | 10 years here :) |
06:25.13 | denon | personal phones, yes |
06:25.16 | drmessano | No |
06:25.19 | *** join/#asterisk jicksta_ (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
06:25.19 | drmessano | BUSINESS PHONES |
06:25.41 | drmessano | People dont buy new PBX's and handsets every two years.. and often those purchases go hand in hand |
06:25.41 | orkid | so... any thoughts on 'good' voip/dect phones available now? is gigaset normally good? :) |
06:25.41 | denon | drmessano: we've all been in this business a long time, each vertical is different |
06:25.50 | drmessano | People go 6, 8, 10, 15 years |
06:26.02 | orkid | .... for home use |
06:26.15 | denon | portable phones get replaced much more often in ruggid environments |
06:26.16 | orkid | or is that ccc dect 'crack' really bad |
06:27.01 | denon | anyway, whatever .. I'm not making a holy war out of this, and at this hour, you usually get into that mood |
06:27.08 | denon | so I think I'll sign out |
06:27.17 | drmessano | denon: All the more reason not to waste $150 on a $25 quality phone because its got a SIP UA built in |
06:27.39 | drmessano | Oh.. Classic |
06:27.50 | orkid | denon: any thoughts before you go? |
06:27.55 | orkid | :) |
06:28.14 | drmessano | If you dont want to get into a holy war, dont be an IRC douchebag and make a personal attack like "Well, we all know how you are" and run off |
06:28.25 | drmessano | I didnt detect any tone here, but whatever |
06:28.54 | drmessano | If you cant take a counterpoint without being a drama queen, dont discuss |
06:29.00 | drmessano | Anwyay |
06:31.04 | orkid | dude really, why not just talk about it w/o any emo/drama getting in the way |
06:31.11 | drmessano | There wasnt any |
06:31.50 | drmessano | "I'm not making a holy war out of this, and at this hour, you usually get into that mood" <-- I didnt make any of this personal.. But denon always gets to a point where if you disagree with him enough, he does |
06:31.52 | drmessano | Oh wait.. |
06:31.55 | *** join/#asterisk thx2000 (n=bob@ip68-101-126-92.oc.oc.cox.net) |
06:32.02 | drmessano | But again.. moving on |
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06:32.53 | orkid | just another way of saying "i dont think this is going anywhere, we're going in circles" i think, to be more polite |
06:32.59 | drmessano | HA |
06:33.08 | orkid | anyway, you have a voip dect phone? |
06:33.19 | drmessano | No, it was quite clear what he said.. as I expect from him.. |
06:33.21 | orkid | which ones from your experience are buggy, and which ones are not? any comments on the gigaset? |
06:33.41 | drmessano | "You usually get in that mood" is definitely not an amicable statement of mutual disagreement |
06:33.44 | drmessano | But whatever |
06:33.45 | [TK]D-Fender | orkid: General word is that Seimens DECT is pretty decent |
06:34.41 | maddog01 | it;s way to late for another round of this shit. good night. |
06:35.13 | orkid | [TK]D-Fender: thanks. i'll look into it once more, and maybe get one |
06:35.39 | drmessano | I hear Linksys makes some nice SIP Wifi phones too |
06:36.20 | orkid | do they allow for non VOIP handsets to still use the base station through some compatibility mode? since i'm guessing they might not have all the features (i think some of the siemens handsets have a way to choose voip or pstn directly) |
06:36.39 | orkid | i actually heard good things about dlink wifi phones |
06:37.03 | [TK]D-Fender | orkid: WiFi = shit. Avoid unless you have little other practical choice |
06:37.13 | orkid | but am weary of the pwer consumption/emission.. even though i dont know much about the differences between dect/wifi emissions, but i think dect is smaller |
06:37.22 | [TK]D-Fender | orkid: If you are dealing with single site users, never touch Wifi |
06:37.35 | [TK]D-Fender | orkid: Wifi batter life is garbage. |
06:37.37 | drmessano | For $250? HA |
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06:40.19 | drmessano | THose Dlinks are pretty, but I am not a Dlink person.. and the price is nuts |
06:40.30 | [TK]D-Fender | ok, checkout time. |
06:40.30 | drmessano | Nevermind SIP Wifi in general being suck |
06:40.31 | [TK]D-Fender | later all |
06:40.35 | drmessano | later |
06:41.11 | drmessano | Im gonna go too before someone accuses me of being a bastard IRC asshole troll because I dont like Green M&M's |
06:41.19 | drmessano | **poof** |
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06:46.13 | orkid | lets stick to what we know :) |
06:46.34 | Rug | Is there a FAQ I can read to find some basic answers? I've search voip-info.org with no success. |
06:47.39 | Rug | Or, could anybody tell me if an X100P card will work with basci analog phones? I want asterisk to act as a basic answering machine, nothing fancy, no VOIP |
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06:48.03 | rob0 | ~x100p |
06:48.04 | jbot | well, x100p is an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
06:48.20 | Rug | cool, thanks |
06:49.12 | Rug | Will the above card work or shall I research it first? |
06:49.26 | rob0 | It can be done, but it really is a piece of junk. |
06:49.47 | Rug | The TDM01B is junk? |
06:50.02 | rob0 | nono I thought you meant the x100p |
06:50.34 | Rug | you convinced me to ignore the X100P. Moving on, now |
06:50.37 | Rug | =) |
06:50.40 | orkid | there are also some clones, like openvox |
06:51.43 | Rug | I don't really care about cost, I am doing the contracting on this project. |
06:53.05 | rob0 | oh. hmmm ... I don't know what to recommend other than to avoid the x100p |
06:53.16 | Rug | thanks, that is good news. |
06:53.29 | rob0 | I think there's a newer Digium card, newer than the TDM cards. |
06:54.23 | Rug | It looks like the TDM01B requires an FXO digital phone, is that correct? |
06:54.39 | orkid | fxo is analog |
06:54.42 | orkid | fxs is analog |
06:54.47 | orkid | t1/e1 is digital |
06:54.47 | Rug | ack, sorry |
06:55.47 | Rug | POTS -> TDM01B -> Analog desk phone (right?) |
06:56.11 | Rug | if I need more lines, just get a card with more FXO ports? |
06:56.30 | Rug | ** Scratch that |
06:56.48 | rob0 | TDMxyB .... x is number of FXS, y is number of FXO ports |
06:56.49 | Rug | If I need more desk-extensions, get a card with more FXO ports? |
06:57.04 | rob0 | nope, your extensions will be on FXS. |
06:57.35 | rob0 | ~fxo |
06:57.36 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
06:57.39 | rob0 | ~fxs |
06:57.40 | jbot | from memory, fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
06:59.04 | Rug | If the customer wants to upgrade in the future, can I add a second card, or should I get a card with N+1 ports? |
07:00.21 | Rug | Would you suggest Debian or Ubuntu for the server OS? |
07:01.08 | rob0 | I would suggest that you choose the distro you are most comfortable with. |
07:01.54 | Rug | Are the debian dependancies 'new-enough' ? |
07:02.07 | rob0 | <== not a Debian user |
07:02.13 | Rug | ahh |
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07:04.12 | Rug | If the customer wants to upgrade in the future, can I add a second card, or should I get a card with N+1 ports? |
07:04.57 | clone_juno | i have a problem |
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07:06.09 | clone_juno | Now ds3000p is being produced ? |
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07:07.21 | Assid | yo |
07:07.33 | Assid | anyone here tinkered with the likes of cisco 7960 |
07:07.42 | Assid | i cant seem to get it to pick up the sip firmware |
07:08.07 | rob0 | The cards are modular, and you can indeed have more than one, but I'm not sure that's going to be cost-effective. At some point you might want to switch from analog to digital, with IP phones. |
07:09.02 | Rug | Right now there is 1 line and 1 phone. Sometime in the future it might grow? |
07:09.57 | Rug | It's a Doctors office. He _just_ wants a fancy answering machine with "unlimited" storage. |
07:10.02 | clone_juno | sorry, ds300p, is that now continue producing? |
07:10.51 | rob0 | Cheapest (not necessarily best, but it works) way to grow is to add ATA's for additional extensions. |
07:10.58 | rob0 | ~ata |
07:10.58 | jbot | ata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
07:11.08 | Rug | ok thanks |
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07:15.28 | clone_juno | how many signalling link support on asterisk+libss7? |
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07:36.44 | rjared | anybody here set up iax? im trying to connect t two asterisk boxes but keep getting Rejected connect attempt from xxx |
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07:42.59 | DaPrivateer | ztdummy on FreeBSD... anyone tell me why I can't seem to find how to install it / make it work? |
07:44.45 | drmessano | http://tfot.leifmadsen.com/ch03s04.html |
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07:45.02 | Corydon76-dig | DaPrivateer: are you using the FreeBSD port? |
07:45.09 | DaPrivateer | affirmative |
07:45.41 | Corydon76-dig | DaPrivateer: http://www.mercenary.ca/articles/zaptel_asterisk.php |
07:46.02 | DaPrivateer | hrm, ill give that a shot -- thanks |
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09:29.34 | KOCATEPE | hi all |
09:29.43 | KOCATEPE | anybody from Turkey ?? |
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09:31.27 | KOCATEPE | anybody from Turkey ?? |
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09:50.41 | Daejeo | does switchvox Encrypt the Partitions during Installation? |
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10:03.03 | kinnaz | has anyone installed linux on audiocodes mediant 1000 ? |
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10:10.28 | ibm2 | I want to know how to retrieve the status of a call |
10:14.49 | DigitalIrony | Asteriskdocs.org |
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10:36.51 | Daejeo | hello DI |
10:37.03 | Daejeo | are you still awake? |
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10:39.42 | aiksa[LV] | tzafrir_laptop: hay man, found the problem - i had added the option to configuration sections |
10:40.02 | aiksa[LV] | to the wrong configuration section* . my bad |
10:40.10 | DigitalIrony | Daejeo sup |
10:40.14 | tzafrir_laptop | :-) |
10:40.56 | Daejeo | does switchvox Encrypt the Partitions during Installation? |
10:41.34 | DigitalIrony | Daejeo: I am not sure. I don't really work with gui's much |
10:41.52 | DigitalIrony | Daejeo: I would guess not, but Im probably wrong :P |
10:45.44 | aiksa[LV] | btw, I was wondering why even there is jitterbuffer option in dahdi configuration? |
10:46.28 | aiksa[LV] | PRIs shouldnt have any jitter at all. and it is hard to imagine where would jitter show up on analog interfaces |
10:46.36 | aiksa[LV] | or am I missing something here? |
10:47.30 | angryuser | aiksa maybe during transcoding to sip, imagine server overload |
10:47.50 | tzafrir_laptop | the jitter is on voip |
10:47.50 | angryuser | to another codec* |
10:48.04 | aiksa[LV] | tzafrir_laptop: i do understand that |
10:48.10 | tzafrir_laptop | and hence you may need a jitter buffer on the pstn/voip border |
10:48.22 | aiksa[LV] | thats why I was wondering why receiving side of sahdi should have jitterbuffer at all |
10:48.31 | aiksa[LV] | tzafrir_laptop: oh i see |
10:48.46 | aiksa[LV] | if the far side had jitter due to transcoding, network, etc. |
10:49.20 | aiksa[LV] | then to be able to succesfully deal with that, jb should be used. |
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11:59.05 | SHZOL | hi |
11:59.44 | SHZOL | how to implmnt Hot-desking |
11:59.48 | SHZOL | any idea ? |
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12:01.44 | Assid | so anyone here managed to get cisco 7960 working? with sip? |
12:01.55 | Assid | i cant seem to get the firmware to upgrade.. im just lost |
12:03.00 | mort_gib | Assid: Polycom and Snom are nice phones to :-) |
12:03.20 | Assid | mort_gib: i know i mostly deploy polycoms |
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12:03.41 | Assid | but my stupid friend has a cisco 7960 and asked me to get it up on my system |
12:04.02 | mort_gib | You need the SIP firmware from Cisco don't you?? |
12:04.47 | Assid | i got that.. i just cant get this phone to get it |
12:06.19 | mort_gib | tftp server setup and all?? |
12:06.25 | Assid | yes |
12:06.39 | mort_gib | Strange.... |
12:06.51 | Assid | maybe i didnt configure it right.. the files |
12:07.23 | mort_gib | Well you need to load the right firmware first, but I'm not a Cisco phone wizard... |
12:07.53 | Assid | im trying even 6.0->6.3 |
12:07.57 | Assid | doesnt wanna work :( |
12:10.27 | mort_gib | Why is that we don't turn around and tell our "clients" that it's unsupported?? |
12:11.04 | mort_gib | Commercial support is all too prepared to do so! |
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12:16.55 | mort_gib | I have an issue with caller id on a Sangoma A200 card... I get : callerid.c: No start bit found in fsk data. |
12:17.05 | mort_gib | in /var/log/asterisk/messages |
12:17.37 | mort_gib | I have usecallerid=yes in zaptel.conf |
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12:19.55 | Assid | mort_gib: cause we're idiots |
12:20.05 | Assid | and they keep saying hey its supported.. its cisco.. yadda yadda |
12:20.13 | Assid | and we want the business |
12:20.26 | SHZOL | hi |
12:20.30 | SHZOL | i want to do hot-desking |
12:20.31 | SHZOL | how |
12:20.31 | mort_gib | True... |
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12:20.42 | mort_gib | SHZOL: Budget?? |
12:20.55 | SHZOL | wat u mean ? |
12:22.09 | mort_gib | You are talking about call center solutions -right?? |
12:23.11 | mort_gib | SHZOL: you want to have a look at "agents" and call queues |
12:23.44 | mort_gib | -But there are some commercial solutions that builds on Asterisk that are good value for money.... |
12:24.01 | mort_gib | -Mind you, it ALL depends on the size and type of your install |
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12:40.53 | aiksa[LV] | hmm, i sometimes see the CPU % usage in top for asterisk to jump up to 9999% what could cause it? |
12:41.05 | aiksa[LV] | nothing special happens at that time |
12:41.12 | aiksa[LV] | just normal call flow |
12:41.17 | aiksa[LV] | nothing extraordinary |
12:42.00 | aiksa[LV] | aversion 1.4.22 |
12:43.33 | SHZOL | i want to do hot-desking |
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12:56.16 | orTix | Got this problem, i got an cisco ip phone 7940 here (same was 7980) I got some new firmware from cisco, however the phone loads the firmware correctly from my tftp server butt it keeps reloading itself all the time.. ? anyone know howto solve this problem :X |
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13:00.14 | phpboy | SHZOL: Hot agents? |
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13:07.39 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
13:09.57 | SHZOL | +phpboy: what u mean by hot agent |
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13:34.05 | SHZOL | i want to do hot-desking |
13:34.18 | SHZOL | heloo |
13:34.56 | coppice | the hot agents usually find better paying jobs |
13:35.27 | SHZOL | coppice: what u mean |
13:36.00 | beek | SHZOL: Read the book in the section under FUNC_ODBC. There's a whole application written to do hot desking. |
13:36.02 | beek | ~book |
13:36.03 | jbot | i heard book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
13:36.13 | beek | SHZOL: ^^^^^^^^^^^^^^^^^^^^^^ |
13:37.04 | SHZOL | coppice: thanks |
13:37.14 | beek | SHZOL: you're welcome. |
13:39.05 | Rico29 | hi |
13:39.19 | Rico29 | I've got a problem with a perl AGI that I really don't understand : |
13:39.31 | SHZOL | coppice: thereis no sample and eyes way of doing it ? |
13:39.42 | orTix | Got this problem, i got an cisco ip phone 7940 here (same was 7980) I got some new firmware from cisco, however the phone loads the firmware correctly from my tftp server butt it keeps reloading itself all the time.. ? anyone know howto solve this problem :X |
13:40.05 | [TK]D-Fender | SHZOL: Maybe somebody blogged some sample of this, but good luck finding it. |
13:40.14 | Rico29 | this line works : $AGI->stream_file($country . "/custom-enter_pwd"); |
13:40.14 | Rico29 | but this one doesn't : my $SIP_passwd =$AGI->get_data($country . "/custom-enter_pwd"); |
13:40.26 | Rico29 | does anybody knows why ? |
13:40.29 | [TK]D-Fender | SHZOL: And you need to be very clear about your definition of "hot desking" and how it pertains to your dialplan |
13:40.31 | SHZOL | [TK]D-Fender: can u help me doing this project |
13:41.25 | SHZOL | [TK]D-Fender: i have more then 300 ip phone and 300 emp in our office, i want any emp can login from any ip phone, thats what i need. |
13:42.04 | SHZOL | [TK]D-Fender: also he can ably to make outgoing call and incoming calls. |
13:43.30 | [TK]D-Fender | SHZOL: You'll need to make an exten to login toa phone that will set a value based on the emp that is logging in, and to associate to that phone device. Then in every exten that they can diaal out through it will first check who is logged into that device and act accordingly. |
13:43.55 | [TK]D-Fender | SHZOL: And then of course an exten so they can "log out. |
13:45.13 | SHZOL | [TK]D-Fender: i found what i want in this web site, but is not full, ducomantions, http://etel.wiki.oreilly.com/wiki/index.php?title=Simple_Hot-desking&redirect=no |
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13:47.19 | telnettech | good morning. I have another question. I have zaptel workng on a 64bit Linus server. with asterisk 1.2.28. It is requiring me to have ztdummy turned on for sound files to work. But i need to have ztdummy off so that my redfone device works properly.....any suggestions? |
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13:47.46 | [TK]D-Fender | SHZOL: tahts a pretty good sample |
13:48.11 | [TK]D-Fender | SHZOL: You need to master the dialplan. Keep reading and keep trying stuff. |
13:48.34 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:50.37 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
13:51.08 | aiksa[LV] | is there a way to catch transfer event in pre 1.6 asterisk? |
13:51.57 | aiksa[LV] | any transfer - through SIP messages as well as feature codes |
13:52.08 | aiksa[LV] | through AMI of course |
13:52.41 | Rico29 | anyone for my probleme ? please |
13:53.32 | aiksa[LV] | Rico29: sorry, perl is not my field |
13:53.43 | Rico29 | ok |
13:54.02 | phpboy | Lovely, asterisk crashes when I introduce a lot of calls via IAX2 :( |
13:54.13 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
13:54.58 | BBHoss | phpboy: with what error? |
13:55.13 | [TK]D-Fender | Rico29: You don't seem to be showing us the error |
13:56.00 | Rico29 | ok, 2sec [TK]D-Fender , i'm pastebining it all |
13:56.30 | SHZOL | [TK]D-Fender: u sow the link i give it you, how many database is there ? |
13:56.56 | [TK]D-Fender | SHZOL: HUH!? |
13:57.04 | BBHoss | heh |
13:57.14 | aiksa[LV] | Is there a way I could take a closer look at the internals of the asterisk - its the same old CPU%9999 issue i am having. Is there a way to have a more detailed split by modules? |
13:57.20 | timeshell | Anyone know the current build number of asterisk-gui? |
13:57.56 | timeshell | And whether it works better with DAHDI hardware detection? |
13:58.02 | aiksa[LV] | but without introducing profiler to the whole scene :P |
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13:58.44 | SHZOL | [TK]D-Fender: how many databases in this sample http://etel.wiki.oreilly.com/wiki/index.php?title=Simple_Hot-desking&redirect=no |
13:58.54 | SHZOL | [TK]D-Fender: can u tell me pls |
13:59.10 | [TK]D-Fender | SHZOL: What the hell does "how many database" mean? |
13:59.23 | [TK]D-Fender | SHZOL: *1* |
13:59.32 | [TK]D-Fender | SHZOL: This just uses AstDB |
13:59.49 | [TK]D-Fender | SHZOL: "core show function DB" <- go read up on this |
14:00.01 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
14:02.16 | SHZOL | [TK]D-Fender: ok |
14:05.52 | SHZOL | [TK]D-Fender: can u give brief introduction about it pls |
14:06.25 | file | DB is a dialplan function which gives access to the underlying Asterisk database, a berkeley database |
14:06.29 | file | you can store and retrieve values |
14:09.38 | aiksa[LV] | hmm, htop is nice for monitoring processes |
14:09.46 | aiksa[LV] | somehow I had not noticed it before |
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14:18.34 | Rico29 | [TK]D-Fender> http://debian.pastebin.com/m1e58703f |
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14:27.35 | gambler1 | is it possible for dial application to react on sip messages? ie when I get "address incomplete" which dial translates to busy |
14:27.37 | Rico29 | [TK]D-Fender> any idea ? |
14:27.50 | *** join/#asterisk Borai (n=No@S0106001c109e98db.no.shawcable.net) |
14:29.06 | [TK]D-Fender | SHZOL: Go read the WIKI and the instructions for DB like I showed you. |
14:29.31 | Borai | morning |
14:29.48 | [TK]D-Fender | Rico29: Seems to play. Whats the issue? |
14:29.57 | Rico29 | doesn't play... |
14:30.12 | Rico29 | it seems, but it doesn't |
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14:30.56 | Borai | I got the ftp server and all the configs ready now my phone boots from the server. |
14:31.03 | [TK]D-Fender | Rico29: is that a complete call from beginning to end? |
14:31.13 | Rico29 | this is one |
14:31.18 | [TK]D-Fender | Rico29: I don't see CLI output there.... |
14:31.29 | Rico29 | i can pastebin the agi debug |
14:31.49 | [TK]D-Fender | BRB |
14:31.51 | Rico29 | http://debian.pastebin.com/m1e58703f the 5/6th first lines |
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14:32.09 | Rico29 | what is "BRB" ? |
14:32.13 | Rico29 | ah ok |
14:32.15 | Rico29 | be right back |
14:32.17 | Rico29 | :p |
14:32.24 | Borai | lol |
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14:32.48 | tzafrir_laptop | waits for someone to ask what lol is |
14:33.19 | Daejeo | How many user extensions can I create in Switchvox Free Edition ? |
14:34.22 | Borai | OMG! |
14:34.50 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
14:34.51 | Borai | *is |
14:36.04 | Borai | however my polycom tells me line 1 (not registered), where sip show peers tells me the peer is online |
14:36.13 | Rico29 | tzafrir_laptop> huhu |
14:36.44 | troubled | anyway to test that wideband codecs are working in my 1.6? Know of any good clients that would work? |
14:37.16 | troubled | ive tried 2 clients so far with multiple codes that do 16khz, but * doesnt seem to have them, or they arent being used, like speex |
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14:39.13 | Rico29 | [TK]D-Fender> did you see the pastebin ? |
14:39.15 | coppice | most things supporting wideband support G.722, and * has that |
14:39.24 | Rico29 | the CLI log is in the 5th first lines |
14:39.27 | [TK]D-Fender | Rico29: do you see when I came BACK? |
14:39.48 | Rico29 | euh... yes ? |
14:40.00 | SHZOL | [TK]D-Fender: already read, but not understanding, pls give simple world defnations, what is it so i can fllow up with my priject |
14:40.01 | [TK]D-Fender | Rico29: No PB since then <- |
14:40.33 | [TK]D-Fender | SHZOL: read that apps INSTRUCTIONS. AstDB is fairly well described on the WIKI. |
14:40.34 | [TK]D-Fender | ~wikis |
14:40.35 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
14:40.36 | [TK]D-Fender | ^^^^^^^^^ |
14:40.40 | Rico29 | mh, I don't understand everything |
14:40.58 | troubled | coppice: any recommended clients that do 722? |
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14:41.23 | coppice | have you found a wideband client that *doesn't* do G.722? |
14:41.38 | SHZOL | [TK]D-Fender: thanks you. |
14:41.39 | troubled | well, i have xlite and wengo atm |
14:41.51 | coppice | they do G.722 |
14:42.08 | Rico29 | ahahaha |
14:42.37 | Rico29 | $agi->exec(Background...) works, and $agi->get_data(sound) doesn't...... WTF ?! |
14:42.44 | troubled | coppice: well, wengo only has speex/16000 and AMR-WB/1600 listed |
14:43.07 | Borai | does the polycom extension have to be peer or friend? |
14:44.13 | [TK]D-Fender | Borai: peer |
14:44.31 | SHZOL | [TK]D-Fender: got it man, Berkeley DB which works very much like the Windows registry. thats it, Right ?? |
14:44.46 | [TK]D-Fender | SHZOL: a little, yes |
14:45.23 | eppigy | TRABAJO |
14:45.28 | troubled | coppice: well, trying xlite, and I noticed that the sip channel said it was using the .slin sound file, but I can only tell its using speex atm |
14:46.02 | coppice | well, wengo became qutecom, and that certainly supports G.722 |
14:46.09 | [TK]D-Fender | eppigy: YOU go work. |
14:46.11 | Borai | [Jan 6 06:45:39] WARNING[17785]: chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission bb467dac-17c92445-6e9a77fe@192.168.1.3 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. |
14:46.21 | Borai | this is what i get all the time |
14:46.34 | Borai | when i try to dial some extension from the phone |
14:46.40 | eppigy | 8[] |
14:47.18 | troubled | coppice: hmm, im using wengo 2.1.2, guess I grabbed an old fork or something |
14:47.39 | [TK]D-Fender | Borai: last we checked you were running a phone behind a remote NAT. Naturally I don't trust that you set any of this up right. |
14:48.04 | [TK]D-Fender | ~sipnat |
14:48.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
14:48.11 | telnettech | ok guys i really need some help....I have an issue where my zap channels are not loading and cant figure out what it is trying to tell me......i have included both 'ztcfg -vv' and 'load chan_zap.so' outputs on the pastebin.....please help |
14:48.16 | telnettech | http://pastebin.com/d7dc2054d |
14:48.27 | coppice | trouble: oh, yeah, xlite is about the one thing which does not support G.722 |
14:49.01 | troubled | coppice: i noticed it does wideband speex, but it seems asterisk doesnt? |
14:49.02 | Borai | I run the asterisk server on a public ip |
14:49.42 | [TK]D-Fender | telnettech: Sure looks like you've mixed up settings between T1 & E1 |
14:49.42 | Nugget | telnet is eeeeeeevil! |
14:49.44 | troubled | coppice: so is there a way to tell the freq/fidelity of the sip call to verify? |
14:49.54 | [TK]D-Fender | Borai: but your PHONE is behind NAT |
14:50.09 | [TK]D-Fender | Borai: and of course I still wouldn't trust your firewalling |
14:50.13 | [TK]D-Fender | Borai: go read <- |
14:50.27 | telnettech | TK: it is supposed to be an ISDN 30.....i have tried to find everything i can but it just doesnt make sense to me |
14:50.28 | Borai | ok but shouldn't technically DMZ avoid NAT? |
14:50.40 | coppice | troubled: * doesn't really support wideband right now. it just has G.722 fudged in |
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14:51.12 | troubled | coppice: not even in 1.6? |
14:51.31 | Borai | and im assuming that in my case it is Asterisk as a SIP server outside nat, clients on the outside connecting to Asterisk |
14:51.37 | Borai | am i right? |
14:52.33 | [TK]D-Fender | Borai: Your peer should have "nat=yes", "canreinvite=no", "qualify=yes", "host=dynamic" |
14:52.39 | troubled | coppice: seems that even the latest qutecom client only does 8000 g722 |
14:53.14 | coppice | that is actually wideband. the 8000 is because everyone has to follow a typo in an RFC |
14:53.14 | jaytee | [TK]D-Fender, good morning! feelin any better? |
14:53.30 | [TK]D-Fender | jaytee: Yea, I've had this beat since about Sat night |
14:53.36 | troubled | coppice: the only 16000 listed codec is speex |
14:53.47 | jaytee | [TK]D-Fender, good to hear. :-) |
14:53.50 | beek | Good morning [TK]D-Fender and jackson__ |
14:53.55 | troubled | coppice: oh |
14:53.59 | beek | Good morning jaytee |
14:54.01 | Borai | i did not have qualify=yes let me reload |
14:54.07 | jaytee | good morning beek |
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14:54.33 | jaytee | everytime he types something I see Sasha Boren-Cohen's face :-) |
14:54.47 | telnettech | TK do you have a good website that explains how to setup an ISDN30 |
14:55.08 | [TK]D-Fender | telnettech: www.google.com |
14:55.15 | Borai | [TK]D-Fender: thank you very much adding qualify=yes fixed my problem. |
14:55.17 | jaytee | telnettech, morning |
14:55.20 | troubled | coppice: looks like it only wants to use gsm *shrug* thanks for trying |
14:55.26 | telnettech | morning jaytee |
14:55.42 | coppice | what only wants to use GSM? |
14:55.58 | jaytee | usually about this time Nugget chimes in |
14:56.05 | troubled | coppice: when i made a call, the "sip show channels" only shows the call as gsm |
14:56.41 | [TK]D-Fender | troubled: that its what it wanted... thats what it SETTLED ON |
14:56.48 | coppice | have you listed G.722 as your preferred codec? |
14:56.57 | telnettech | i get about 3000 hits there TK.....i have looked at about 20 or so.....i cant seem to find any info at all.....alot of the hits have to do with issues with asteirsk and BT ISDN30 |
14:56.58 | [TK]D-Fender | troubled: Look at SIP debug of the call attempt or you're wasting your time |
14:57.22 | [TK]D-Fender | telnettech: I don't see your configs, do I? |
14:57.40 | telnettech | TK: no but i can show you |
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14:58.17 | troubled | coppice: its just a priority list, i tried with 722 at the top and bottom just in case. still using 722 in asterisk either way |
14:58.54 | coppice | well, if you only allow G.722 in *, that should force it :-) |
14:59.07 | troubled | ya, just gonna try that now. let you know in a sec |
14:59.43 | telnettech | TK: here are the config for zaptel and zapata http://pastebin.com/d909e80d |
15:00.26 | troubled | coppice: ooh, that played the demo-echotest.g722 file, but it was choppy |
15:00.40 | kannan | hello , i am editing the SIP<MAC>.cnf for a cisco 7960 to change the phone extension number, the tftp server is also the same as *. After restart the tftpd and the phone, it is still not updated . any ideas what i am missing |
15:02.06 | troubled | coppice: probably just the client cpu lag |
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15:03.01 | *** mode/#asterisk [+o russellb] by ChanServ |
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15:05.05 | [TK]D-Fender | telnettech: And your fonebridge dies when it hits 24 (T1 D-Chan). Check ITS config |
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15:06.06 | telnettech | the fonebridge programming is fine.....it is the zaptel and zapata......the redfone is just an external T-1 card basically using zaptel to interface with asterisk |
15:06.50 | NoxIn- | Hello all, is it possible to call-limit only calls comming from a peer to asterisk but not the calls from asterisk to the peers ? since the call-limit apply to both, I envisaged to create a type=peer for outgoing calls and a type friend for incoming calls, but type=friend is deprecated so anyone have a clue ? |
15:06.53 | telnettech | I verified the fonebridge programming by putting a loopback jack in the rj45 port and it goes green...that tells me the red fone is working fine |
15:07.27 | [TK]D-Fender | telnettech: Doesn't mean it AGREES with * |
15:07.37 | [TK]D-Fender | telnettech: And doesnt' tell me what it thinks it should be doing. |
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15:07.49 | [TK]D-Fender | telnettech: You have reassured NOTHING. |
15:07.52 | telnettech | right but that is where the zaptel comes into play |
15:08.06 | telnettech | it uses zaptel to communicate |
15:08.12 | telnettech | with asterisk |
15:08.33 | [TK]D-Fender | telnettech: fonebridge needs to be set up for the proper signalling in its OWN config. |
15:08.51 | telnettech | ok so you want to see the redfone config |
15:09.01 | [TK]D-Fender | telnettech: I have to ask? |
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15:11.20 | telnettech | Tk no....i just thought that i would tell you that the progrmming is correct cause im geting signalling verified thru the loopback.....here is the pastebin for the redfone.conf |
15:11.24 | telnettech | http://pastebin.com/d6bd72898 |
15:11.28 | doolph | hi there |
15:11.57 | [TK]D-Fender | telnettech: Loopback does not tell me it AGREES WITH ASTERISK |
15:13.08 | [TK]D-Fender | telnettech: Ans since when does E1 use ESF, B8ZS? |
15:13.24 | [TK]D-Fender | telnettech: those are T1 typical |
15:13.42 | telnettech | span 2 is not used but redfone requires that you program both all spans even if your not using it |
15:13.52 | telnettech | span 1 is supposed to be the isdn30 |
15:14.28 | [TK]D-Fender | telnettech: Go ask Redfone for support with their device.... |
15:14.46 | telnettech | so you are thinking redfone problem? |
15:15.24 | [TK]D-Fender | telnettech: So far, yes |
15:16.37 | telnettech | so the zaptel and zapata are correct for an isdn30 |
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15:18.47 | kannan | can we config the cisco ip phone thru the TFTP server, or does the .cnf in tftpboot directory get written by the phone? |
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15:20.22 | orTix | My cisco ip phone (7940) keeps loading the new firmware thought TFTP server, does anyone know the problem (google is empty :p) |
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15:40.58 | Nugget | eyes jaytee |
15:41.12 | jaytee | you missed your queue |
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15:41.21 | Nugget | dang! |
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15:41.22 | [TK]D-Fender | hands Nugget a spoon |
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15:53.45 | eppigy | hello |
15:53.48 | eppigy | i am dave |
15:54.42 | Rico29 | hello, I am rico :) |
15:56.06 | orTix | nobody can awnser my question?:X |
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16:00.17 | telnettech | TK: thanks....you wer correct...there were a couple of the settings incorrect on the redfone config |
16:00.50 | telnettech | i apologize if i ruffled your feathers there |
16:04.07 | jaytee | he doesn't have feathers, he has scales and talons and breathes fire |
16:04.58 | jaytee | and to him, you're crunchy and taste good with ketchup |
16:04.58 | jaytee | so beware |
16:05.06 | jaytee | file, you around? gotta real quick question |
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16:08.37 | [TK]D-Fender | jaytee: If humans weren't meant to be eaten then why are they made of meat & treasure? :D |
16:08.59 | jaytee | [TK]D-Fender, good point! |
16:09.41 | file | jaytee: moo |
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16:10.50 | jaytee | file, do you think it's a good idea to include the h extension in each subsection of an IVR tree that calls SpeechDestroy()? |
16:11.10 | file | it doesn't really matter |
16:11.27 | Assid | geta |
16:11.29 | Assid | heya |
16:11.30 | file | the speech structure itself is associated with the channel and when the channel is hung up it also gets destroyed |
16:11.38 | jaytee | ah, excellent |
16:11.39 | Assid | anyone here worked with a cisco 7960 |
16:11.47 | jaytee | file thanks! one more? |
16:11.51 | file | go ahead |
16:12.04 | Assid | i cant seem to get the firmware shifted to sip |
16:12.51 | jaytee | can I have one tree handling spanish and one handling english? as long as I'm not trying to activate two different languages in the grammars for one section? |
16:13.23 | file | you can |
16:14.37 | jaytee | ok, that's what I thought but wanted some confirmation. I read the docs and thought that's what they meant. I can't have both languages in one single instance though. Cool. Thanks for the help!!! |
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16:18.22 | kannan | sorry to repeat, howto effect the changes made in SIP<mac>.cnf in root tftpd dir file. The phone's old number still shows up , even after a restart of the Cisco 7960 phone? |
16:18.49 | *** part/#asterisk hjeffg (n=hjeffg@ool-4573d996.dyn.optonline.net) |
16:18.52 | kannan | i have followed the Cisco Pgone Admin guide |
16:19.23 | Assid | kannan: you got it to do that? mine looks for SEP<mac>.cnf.xml |
16:20.20 | kannan | Assid: the current extensions are working fine |
16:20.28 | kannan | SIP<mac>.cnf |
16:20.57 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
16:21.49 | Assid | nah.. mine looks for sep.. and i cant get it to load the new firmware |
16:23.15 | engaged | if i need 10 channels for CSR queue... what is a good VOIP provider? |
16:26.07 | *** join/#asterisk anonymouz666 (n=anonymou@189.36.179.210) |
16:26.39 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
16:26.44 | [TK]D-Fender | ~itsplist-us |
16:26.45 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
16:27.06 | [TK]D-Fender | engaged: les.net & vitelity seem to be pretty decent |
16:27.17 | [TK]D-Fender | engaged: as I mentioned we nned to rate shop. |
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16:31.16 | nerdygirl_ellie | Hello All. Is it possible to accept a pound/hash key (#) with the Asterisk Read() command in extensions.conf? If so, how do I match it with a GotoIf?? |
16:31.24 | nerdygirl_ellie | Thanks in Advance, Ellie |
16:31.49 | nerdygirl_ellie | (Sorry about the Double-??). |
16:33.05 | [TK]D-Fender | nerdygirl_ellie: No |
16:33.33 | [TK]D-Fender | nerdygirl_ellie: You'll have to make a custom IVR to collect digits |
16:34.20 | nerdygirl_ellie | That is unfortunate. |
16:34.27 | jameswf | whooop there i is |
16:35.04 | nerdygirl_ellie | Does that mean a separate application outside of extensions.conf, or...? Any pointers would be appreciated. |
16:35.45 | nerdygirl_ellie | (I think the lazy solution is to re-record the greeting and use 9 instead of #.. :) ) |
16:37.01 | nerdygirl_ellie | Thank you for the help. |
16:37.07 | [TK]D-Fender | nerdygirl_ellie: No, as I said, a special IVR to collect your input. All dialplan. |
16:38.08 | anonymouz666 | nerdygirl_ellie: better yet, try to use read() and at each digit you read you repeat to the user. ;) |
16:39.48 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
16:40.57 | Assid | okay cisco just made it to my list of DO NOT buy ip phones |
16:41.12 | Assid | i cant configure this thing.. and every guide up there says do this.. do that.. |
16:41.13 | dmz | why? |
16:41.28 | dmz | ah yeah configuring is a pain, but they do work when setup is done :) |
16:41.48 | Assid | so do polycoms.. and they werent that hard to configure |
16:42.03 | Assid | has to thank [TK]D-Fender for suggesting polys.. |
16:42.12 | jameswf | I like AASTRA |
16:42.24 | Assid | i wanna make this stupid thing move to sip.. |
16:42.34 | [TK]D-Fender | Assid: Funny you say that after going against advices and hitting brick walls.. |
16:42.43 | [TK]D-Fender | Assid: Hind-sight is 20/20 |
16:43.00 | Assid | going against advices? me ? |
16:43.36 | Assid | all my other phones are either ATA's or polycoms |
16:43.55 | Rico29 | i think I really have a problem with asterisk::agi |
16:44.08 | Rico29 | I've installed an all clean Asterisk |
16:44.19 | Assid | but i need to get a friend onto my pbx.. so.. trying to whack my brains on this |
16:44.21 | Rico29 | i'v made the simpliest agi I can |
16:44.26 | Rico29 | and it's still not working |
16:44.36 | *** join/#asterisk maddog01 (n=minotaur@d221-91-175.commercial.cgocable.net) |
16:44.36 | RypPn | Assid you need help getting sccp going? |
16:44.57 | Assid | RypPn: i wanna move it to sip.. i like sip.. i want it to run on that instead of sccp if possible |
16:45.03 | dmz | i wish i could try a polycom, i got my cisco for free so i "had" to get it working :) |
16:45.10 | Assid | given the option" exists" |
16:45.30 | RypPn | Assid I use sccp with mine, If you change your mind let me know :) |
16:45.37 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
16:45.46 | Rico29 | I erlly need help |
16:45.48 | Rico29 | really |
16:45.56 | Assid | hrmm i wouldnt mind learning... nevertheless |
16:46.11 | Assid | so sure |
16:46.23 | Assid | one sec |
16:46.25 | Assid | phone |
16:46.50 | RypPn | pm me if you like and we can look at it :) |
16:47.40 | Rico29 | if someone can take a look : http://debian.pastebin.com/m558c0ca3 |
16:48.41 | *** join/#asterisk ta^3 (n=tacvbo@189.146.181.5) |
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16:49.02 | Linuturk | if I call dialplan reload, will that drop any existing calls? |
16:49.16 | [TK]D-Fender | Rico29: ANSWER the damn call first |
16:49.17 | Rico29 | Linuturk> no |
16:49.27 | Rico29 | TK > ok |
16:50.06 | Rico29 | TK > Doesn't change anything |
16:50.13 | kannan | dmz, do you config the phones sip params by tftp? |
16:50.18 | kannan | or on the phone itself |
16:50.21 | dmz | yeah tftp |
16:50.28 | Linuturk | o, well that command doesn't seem to exist in 1.2x thanks anyway Rico29 |
16:50.28 | Rico29 | I've found a solution, but it looks stupid for me |
16:50.30 | Linuturk | :) |
16:50.35 | kannan | i will be glad to learn how to modify a phone setting .cnf in tftp |
16:50.46 | Rico29 | I have to put a "sleep" after the stream_file |
16:50.48 | kannan | i edited the SIP<mac> and thats it? |
16:50.53 | Rico29 | then I can hear the sound |
16:50.58 | kannan | on restart the phone shld take the new params? |
16:51.17 | Rico29 | kannan> depends on the phone |
16:51.21 | Rico29 | of |
16:51.31 | kannan | cisco 7960 |
16:51.42 | kannan | i have 10 working |
16:51.48 | kannan | but cant modify the setting |
16:51.59 | kannan | i want to change the exten numbers |
16:52.00 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
16:52.15 | dmz | i've only setup the lineN_name, lineN_displayname, lineN_shortname, lineN_authname, lineN_password |
16:52.20 | dmz | what other settings do you want to change? |
16:52.22 | kannan | me ttoo |
16:52.39 | kannan | only the lineX_name and passwd |
16:52.41 | dmz | inbound or outbound exten numbers? |
16:52.42 | kannan | authname |
16:53.16 | dmz | setup different sip names/details if you want different outbound (or inbound depending on dialplan config) |
16:53.21 | kannan | dmz, i dont get that? these are phones registering to *, and can do both. the sip.conf handles the registers |
16:53.23 | [TK]D-Fender | Rico29: Show me |
16:53.54 | kannan | if i change the .cnf in the tftp, the phone's extension number like 101, 102 etc doesnt update |
16:53.57 | dmz | kannan, you have 6 line buttons on the right side of the phone, do you want to assign a different extension to each? |
16:54.03 | Rico29 | what do you want to see [TK]D-Fender ? |
16:54.04 | [TK]D-Fender | Rico29: In AGI I suppose you can do things while the audio is playing |
16:54.19 | [TK]D-Fender | Rico29: Therefor you would ahve to wait until end of stream |
16:54.20 | dmz | kannan, the extension info is on how you have your dialplan send calls to the phone |
16:54.20 | kannan | no, i want to change the params for line 1 in the .cnf files |
16:54.25 | [TK]D-Fender | Rico29: Makes sense |
16:54.30 | kannan | dmz, thats not what i meant |
16:54.40 | kannan | sorry i said extension |
16:54.45 | kannan | cisco's phone number |
16:55.04 | kannan | the phone is SIP/101 |
16:55.11 | kannan | i want to change to SIP/102 |
16:55.20 | kannan | for a particul;ar mac |
16:55.23 | Kobaz | 102? that's crazy talk |
16:55.23 | Rico29 | TK > but with get_data (= Background), it seems right to you to dont stream the audio message ? |
16:55.33 | eppigy | D: |
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16:56.01 | kannan | dmz, hope i am clearer now |
16:56.07 | dmz | I just have multiple defined extensions in my dialplan, and each of those has: Dial, SIP/dmz7960 |
16:56.37 | dmz | then i've setup a findme function to either ring all of them or have escallation based on if they are busy for inbound calls going to different lines; not sure what your trying to do |
16:56.46 | kannan | dmz, :( |
16:57.15 | kannan | dmz, i have defines the sip params for a Cisco phone like SIP<mac>.cnf in the tftp root dior |
16:57.36 | kannan | i want to change those params and so the Cisco phone gets a new number and , authname and passwd |
16:58.32 | kannan | but editing the .cnf and restarting the phone doesnt do it |
16:58.38 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
16:59.02 | kannan | dmz, hope i communicate correctly now |
16:59.22 | dmz | hmm, are you sure it's realoading file? |
16:59.37 | dmz | i have same config for each line, i've not tried different ones, just assumed it worked |
16:59.38 | kannan | the phone or * or the tftp? |
16:59.41 | SlicerDicer | well I should have my Aastra 480i in 2 days :) |
16:59.45 | SlicerDicer | x2 |
16:59.55 | dmz | look at your syslog output when booting to see if it finds/loads file |
16:59.57 | Kobaz | polycom! |
17:00.44 | SlicerDicer | Kobaz: they were 1000 pesos or 75$ maddog01 ;-) |
17:00.55 | kannan | when rebooting it takes only the old values |
17:00.56 | Kobaz | hmm |
17:00.57 | Kobaz | not bad |
17:01.01 | SlicerDicer | not at all |
17:02.32 | SlicerDicer | Kobaz: sorry for injecting the pesos was messing with maddog01 as I am in "New" Mexico heh |
17:03.33 | Talkradio | do you see any of the violence happening near the border |
17:04.10 | Talkradio | my girlfriend goes to mexico every other weekend and said it's getting really really bad |
17:04.34 | mog | she a mule? |
17:04.41 | mog | ^_^ |
17:05.13 | Talkradio | i wish then i wouldn't have to support her lol |
17:05.54 | Talkradio | hmm is she a mule? she can take a big load and keep on trucking so maybe heheh |
17:08.55 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
17:09.05 | jameswf | you know you have spent to much time on the internet when you see a post titled "Two People One IP Phone" and your mind goes directly to the gutter |
17:09.47 | jjshoe | haha |
17:10.29 | Talkradio | haha |
17:10.57 | Rico29 | [TK]D-Fender> stream_file : "Returns: -1 on error or hangup, 0 if playback completes without a digit being pressed, or the ASCII numerical value of the digit if a digit was pressed" |
17:11.01 | Rico29 | mine returns 1... |
17:11.07 | jameswf | anyone who caught the reference should log off too |
17:13.25 | Kobaz | term should be rxvt |
17:13.26 | Kobaz | er |
17:13.30 | SlicerDicer | Talkradio: from what I hear the shit has hit the fan and _is_ spilling across the border |
17:13.45 | SlicerDicer | its not borderline or anything civil war... its 100% civil war down there |
17:14.19 | SlicerDicer | drug lords, corrupt cops, bad government, angry citizens all mad... |
17:14.22 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
17:14.28 | Kobaz | fun fun |
17:14.32 | SlicerDicer | I dont think the druglords are as much to blame as the USA imo... |
17:14.39 | SlicerDicer | it is our fault for inviting them |
17:14.42 | Talkradio | it's scary |
17:14.46 | SlicerDicer | indeed |
17:14.48 | russellb | let's play a game called "talk about Asterisk" |
17:14.51 | russellb | it's really fun |
17:14.52 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
17:14.56 | Talkradio | :D |
17:14.57 | jjshoe | russellb agreed |
17:15.03 | SlicerDicer | sorry russellb he asked a question :) |
17:15.21 | russellb | np, not directing it at any specific person ;) |
17:22.08 | angryuser | anyone using res_snmp.conf with nagios/centreon ? |
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17:25.06 | *** join/#asterisk Get_The_Fish (n=IceChat7@75.151.94.189) |
17:26.30 | Get_The_Fish | Anyone running 1.6 in here? Whats the verdict- is it ready for "prime time"? |
17:31.20 | jaytee | for most uses in testing I've found 1.6 to be solid. I understand there are a few minor bugs with SIP TCP though but that'll be worked out quickly enough. |
17:31.34 | Get_The_Fish | well thats good to hear |
17:32.25 | Get_The_Fish | I dont use SIP TCP currently anyways |
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17:47.35 | DarkRift | Simple question, when I want to make a .call file, let's say I want it to come from a "service" to monitor service status, for the channel, can I provide a non-existant one like if there is a built-in one that asterisk provides that would lead nowhere but in memory, or do I need to create a static user for it ? This call doesn't need to have a real connection on the other part cause it will only play a sound/text file |
17:49.16 | DarkRift | Let's say I have a cron job that monitors my apache service, and this cron job creates a .call file to warn a tech that it failed, is there a pseudo-channel I can use for that or I absolutely need to create a static user ? |
17:50.18 | file | a call file calls one destination... and then executes dialplan logic OR executes an application... |
17:50.44 | file | so what do you mean by "create a static user"? |
17:51.36 | DarkRift | The caller that 'originates the call' |
17:51.50 | file | Asterisk is originating the call |
17:52.13 | DarkRift | Ic so Channel refers to the callee and not the caller, I think that was what I had wrong |
17:52.19 | file | yes |
17:52.24 | DarkRift | Alright, thank you |
17:53.16 | file | in this context at least... if you were using a call file for something that called a destination and then called elsewhere then it sort of would be the caller... |
17:53.28 | phpboy | Time for a fresh re install |
17:53.30 | phpboy | YAY! :D |
17:53.52 | DarkRift | yeah I understand that |
17:54.41 | DarkRift | I just thought the Channel initialy referd to the "real" caller that originated the call, thank you for this explanation |
17:54.46 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-236-67-190.dsl.pltn13.sbcglobal.net) |
17:59.22 | markgreene | is anyone in here aware of an RTP timestamp issue in the current branch that affects polycom phones? |
18:04.18 | *** join/#asterisk dieguito84 (n=diego@host118-190-dynamic.12-79-r.retail.telecomitalia.it) |
18:09.46 | *** join/#asterisk l2trace99 (n=jr@p1-bh-mco-1.prismone.net) |
18:11.05 | *** join/#asterisk fukz (n=basti@pD9543FDE.dip.t-dialin.net) |
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18:14.33 | Get_The_Fish | markgreene: no, is this something that you are experiencing? |
18:14.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
18:15.27 | *** join/#asterisk ManxPower (n=manxpowe@router.asteriasgi.com) |
18:17.17 | Qwell | ManxPower: interesting hostname |
18:18.09 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
18:18.34 | *** join/#asterisk Sargun (n=Sargun@66.151.148.225) |
18:18.36 | markgreene | Get_The_Fish, I've got a polycom that has no audio for the first ~4 seconds of conversation |
18:19.13 | Get_The_Fish | interesting. What model and firmware rev? |
18:20.54 | markgreene | Get_The_Fish, asterisk 1.4.16.2 and the model ip650 or ip550, etc. |
18:21.31 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr) |
18:22.23 | russellb | there have been 1156 changes to asterisk 1.4 since 1.4.16 btw ;) |
18:24.04 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr) |
18:25.05 | markgreene | russellb, that helps I guess |
18:25.15 | Get_The_Fish | markgreene, which rev of the polycom SIP application though? |
18:25.49 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
18:26.51 | markgreene | Get_The_Fish, I don't have that on hand. The phone is offsite. Are there known issues with the old application? |
18:27.30 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
18:28.08 | Get_The_Fish | not that I know of, but I was going to look. Have you contacted the vendor? They have access to polycom's support channel, which is pretty good. It's the best place to start if this is only affecting polycom phones. I assume that you dont see the same behavior on a softphone, right? |
18:30.23 | phpboy | Is it safe to install asterisk and dundi off a fresh CentOS 5.2 installation? |
18:30.29 | Get_The_Fish | yes |
18:30.32 | phpboy | I.e. Would i require any extras? |
18:30.46 | [TK]D-Fender | phpboy: no |
18:30.59 | phpboy | [TK]D-Fender: no to which question? |
18:31.00 | Get_The_Fish | phpboy, it depends on what options you chose on the centos install, but using defaults no |
18:31.09 | Get_The_Fish | no extras |
18:31.11 | [TK]D-Fender | phpboy: Install CentOS, install *, done. |
18:31.17 | phpboy | perfect |
18:31.31 | Get_The_Fish | si, did it myself, and it installs flawlessly |
18:31.36 | phpboy | I always went through the process of upgrading everything |
18:31.45 | phpboy | It was very timeconsuming |
18:31.59 | Get_The_Fish | normally a good idea phpboy, but not really necessary |
18:32.27 | phpboy | Well, this server will be replacing our main server which has been giving me hell of late |
18:32.43 | phpboy | I've also been doing live testing and dev on it which has also caused some headaches |
18:33.00 | Get_The_Fish | I have had very very good luck on centos 5.2 and asterisk 1.4.22 |
18:33.14 | phpboy | So this is going to be a 'clean' install and I'm going to reconfigure it from scratch |
18:33.24 | phpboy | Get_The_Fish: Zaptel or Dahdi also? |
18:33.32 | Get_The_Fish | no, SIP trunking only... |
18:33.44 | phpboy | ah, I'm having problems on the PSTN side |
18:33.48 | phpboy | load related issues |
18:34.02 | Get_The_Fish | but I know of several installations using centos 5.2 with zaptel on both sangoma and digium hardware |
18:34.16 | Get_The_Fish | load issues? Such as? |
18:35.32 | phpboy | Get_The_Fish: Well, this box is VERY VERY busy, To give you an idea, we have 110 outbound/inbound IAX2 channels, 2 spans Outbound only E1, and 2 spans inbound and some outbound E1 |
18:35.46 | phpboy | so you can imagine how busy the box must be to justify all that hardware |
18:35.53 | phpboy | or rather all the channels |
18:36.06 | phpboy | anyhoo, at night when it's quite, everything runs just fine |
18:36.28 | ManxPower | Why not just split the one box into 2 boxes? |
18:36.34 | Get_The_Fish | is the the rest of the hardware underpowered- cpu, RAM, etc? |
18:36.54 | phpboy | office hours, after 10 to 20 or so calls I get timing errors, get yellow alarms on my inbound calls. It's a mission |
18:37.07 | phpboy | ManxPower: I'm not sure how I'd lay that out? |
18:37.14 | phpboy | considering how the call centers work, etc |
18:37.14 | ManxPower | do you also get HDLC abort errors? |
18:37.22 | phpboy | ManxPower: yes |
18:37.32 | ManxPower | you must have a very old card and old zaptel |
18:37.36 | Get_The_Fish | glare? |
18:37.44 | Get_The_Fish | what protocol are the T's using? |
18:37.49 | ManxPower | Get_The_Fish: Glare does NOT call yellow alarms |
18:38.25 | phpboy | ManxPower: Zaptel zaptel-1.4.12.1 ? |
18:38.40 | ManxPower | phpboy: you must have an old card then |
18:39.00 | phpboy | It's not THAT old, it was the newest card at the time back in 01-04-2008 |
18:39.22 | phpboy | I've got 2 new TE420's on PCIx in this new box |
18:39.33 | ManxPower | your problem is that the audio data is coming from the card too fast for the computer to process, usually these issues are caused by things like SATA or GigE on the motherboard locking interrupts for a long time. |
18:39.35 | phpboy | I believe in my heart that this will tend to my problems |
18:39.39 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
18:40.01 | ManxPower | the new cards are much better at working around interrupt latency issues. |
18:40.05 | phpboy | ManxPower: I read something about that, I'm using SATA2 based RAID 5? |
18:40.18 | anonymouz666 | phpboy: what card? |
18:40.25 | anonymouz666 | TE110P? |
18:40.31 | phpboy | yes |
18:40.34 | anonymouz666 | RMA it. |
18:40.39 | phpboy | RMA it? |
18:40.50 | ManxPower | also why did you put TWO cards in the box? You are generating twice as many interrrupts as you need to. |
18:41.05 | ManxPower | each card generates 1,000 interrupts/second if I recall correctly. |
18:41.21 | phpboy | ManxPower: the thing is, we have more lines that we're currently not using |
18:41.30 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
18:41.34 | phpboy | although, I will initially only be running one TE420 in the new box |
18:41.37 | ManxPower | phpboy: how does that apply to using one card instead of two cards? |
18:42.12 | ManxPower | or do you mean you need 8 ports? I don't keep up with the way Digium randomly names their cards these days. |
18:42.13 | phpboy | See, We had trouble with this hardware, specifically with the IRQ's |
18:42.25 | phpboy | ManxPower: I need 8 ports |
18:42.30 | phpboy | 2 x 4 port cards |
18:42.41 | ManxPower | Then forget what I said about 2 cards. |
18:42.50 | phpboy | anonymouz666: What's your thoughts on this concidering the TE410P ? |
18:42.58 | phpboy | ManxPower: ok |
18:43.00 | [TK]D-Fender | phpboy: ANCIENT |
18:43.06 | phpboy | :( |
18:43.08 | telnettech | i want to 1st thank everyone on here for the help that they have given me over the last 2 months....i dont think i could have gotten as far as I have with this install. I hve been working on Asterisk for only 4 months and have learned alot from here, the Jason Smith/Leif Madsen book and voip-info.org |
18:43.11 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
18:43.13 | phpboy | [TK]D-Fender: you think this could be my problem? |
18:43.17 | [TK]D-Fender | phpboy: And ditched my 2 fast. |
18:43.41 | ManxPower | phpboy: those old cards have massive issues with interrupts and lost data on many different motherboards. |
18:43.53 | telnettech | with that said, I am needing some more help....I can see on the CLI that sound files are playing but I dont hear them on the phones....can omeone point me in the right direction? |
18:44.07 | ManxPower | Chances are you could have just replaced the cards in your existing system to fix the problem |
18:44.15 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
18:44.39 | phpboy | ManxPower: I can give that a shot. |
18:44.56 | ManxPower | telnettech: either you have an unconfigured zaptel card in the system or you have a system with a kernel HZ not equal to 1000 or you have borked RTC support in your kernel |
18:46.04 | phpboy | ManxPower: I'm going to put one of the new cards into the current server and test it tomorrow durin office hours. |
18:46.20 | ManxPower | remember some of the new cards use new drivers |
18:46.53 | phpboy | ManxPower: this would mean I'd have to switch to dahdi :/ |
18:46.56 | telnettech | ManxP: i have no zaptel cards in my system. we are using a redfone device for the Telco connection. what is kernel HZ and what is borked RTC support...i am using Red Hat Linux 5.1 OS as a 64bit system |
18:47.05 | ManxPower | phpboy: I did not say that! |
18:47.25 | phpboy | ManxPower: Well, I've got the latest drivers and the claim to support this specific card, well kinda |
18:47.28 | ManxPower | telnettech: just make sure ztdummy is not loaded (lsmod) |
18:47.33 | phpboy | TE4XX |
18:48.20 | *** join/#asterisk crevetor (n=crevetor@bureau.ubity.com) |
18:48.24 | phpboy | ManxPower: I shouldn't assume that if it works now (afterhours) with one of the new cards that the drivers are fine? |
18:48.29 | crevetor | hi |
18:48.37 | ManxPower | phpboy: yes |
18:49.02 | telnettech | ManxP: I dont have ztdummy running.....funny thing is that if i do, the sound files play fine but my redfone device doesnt work |
18:49.12 | crevetor | quick question : how bad is it if I have my sip peers in a mysql DB using dynamic realtime but I don't allow asterisk to write to the DB |
18:49.36 | crevetor | from what I see in the debug it only tries to set empty values when my user registers |
18:49.46 | ManxPower | telnettech: TDMoE is not well supported in Asterisk since IAX2 happened. Best of luck with that. |
18:49.48 | telnettech | i have been working on this since yesterday afternoon |
18:50.01 | crevetor | <PROTECTED> |
18:50.09 | ManxPower | crevetor: I don't think anyone has been stupid enough to try using a read only Realtime DB |
18:50.10 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
18:50.24 | crevetor | ManxPower: Well, I am :) |
18:50.54 | phpboy | Ok, let's give it a shot |
18:50.57 | crevetor | ManxPower: and I'd like to know in which way it's so stupid |
18:51.01 | ManxPower | telnettech: Not suprizing. very few people ue TDMoE |
18:51.02 | telnettech | ManxP: this is Asterisk 1.2.28.....we have the same setup at another location that is working...the only difference between the 2 sites is 1 is a 32bit OS and this is a 64bit OS |
18:54.17 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
18:55.04 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
18:55.10 | Assid | stupid stupid net connection |
18:58.14 | telnettech | ManxP: any other suggestions? |
19:00.43 | ManxPower | telnettech: nope. never used redfone, never used Asterisk on a 64 bit OS. |
19:00.54 | ManxPower | Can't imagine why anyone would want to do either of those things. |
19:01.12 | phpboy | ManxPower: I'm running my asterisk on a 64bit OS |
19:01.23 | telnettech | im not the developer....im just a technician told to make it work |
19:01.38 | phpboy | ManxPower: Just put in the new card still getting Yellow alarms every now and then :( |
19:01.52 | ManxPower | phpboy: but are you getting HDLC aborts? |
19:02.08 | phpboy | ManxPower: no |
19:02.34 | phpboy | I can see no reason for these yellow alarms :/ |
19:03.31 | phpboy | [Jan 6 21:01:04] WARNING[6967] chan_zap.c: Received NOTIFY on unconfigured channel 255/255 span 3 <--- now that's new |
19:04.30 | ManxPower | phpboy: especially because span 3 is channels 49 - 73 |
19:05.03 | phpboy | yes and 255 is not in zaptel.conf or zapata.conf |
19:05.21 | phpboy | It seems this new card is dropping the calls even more then the old card :( |
19:05.43 | ManxPower | phpboy: what verison of zaptel did you say you are using? |
19:06.27 | phpboy | I just put it onto another span and this lovely error poped up |
19:06.31 | phpboy | [Jan 6 21:03:58] NOTICE[8699] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 <----- |
19:07.04 | ManxPower | phpboy: What version of Zaptel did you say you are using? |
19:07.05 | phpboy | Zaptel Version: 1.4.12.1 |
19:07.24 | ManxPower | at least you have the latest. |
19:07.43 | ManxPower | for those errors you can contact Digium support, it's covered under the card support. |
19:07.51 | ManxPower | since there is nothing you can do in ASTERISK to fix them. |
19:08.04 | ManxPower | IT's all zaptel/dahdi, the card, the cables, or the telco |
19:08.13 | telnettech | ManxP: thanks.....i convinced the devlopment lead that the trouble is that the 64bit is not the correct version..they are going to do more development but I have been told to get customer working |
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19:08.31 | ManxPower | ~mailinglist |
19:08.32 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
19:09.39 | dandate | my asterisk machine cannot connect to the internet, do i just need to plug the cable modem directly to the machine or does it have to be connected through a router? |
19:09.45 | phpboy | ManxPower: This is even stranger, I plug in another E1 and the errors become a lot more |
19:09.59 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
19:10.08 | phpboy | if it makes any diffs, the one E1 gets here through fiber and the other through copper from the telco |
19:11.20 | dandate | im trying to install from flashpbx, it wants to download an update but it cannot connect to the internet with this default centos install |
19:17.06 | riddlebox | dandate: plug the machine into the switch |
19:17.13 | riddlebox | or router |
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19:19.15 | phpboy | ManxPower: to further your theory on my problems, it is posible that this would only affect one PRI? |
19:19.31 | phpboy | As the other two are running just fine on the same card |
19:23.11 | *** join/#asterisk LemensTS (n=customgt@70.238.154.243) |
19:24.00 | LemensTS | <PROTECTED> |
19:24.34 | rob0 | Try "password". |
19:25.00 | Qwell | LemensTS: Asterisk does not create an 'asterisk' user. |
19:25.01 | LemensTS | nope |
19:26.05 | rob0 | (I wasn't being serious.) |
19:26.46 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
19:27.14 | rob0 | Ask whoever has root on that box to reset it for you. If that's you, see "man passwd" and your distro docs. |
19:27.19 | SlicerDicer | how do I make sure asterisk is acting as a media proxy? |
19:27.57 | SlicerDicer | rob0: take a lesson from macrumors eh ;-) |
19:28.07 | [TK]D-Fender | SlicerDicer: "canreinvite=no" <- under [general] & all peers |
19:29.25 | SlicerDicer | ok [TK]D-Fender thanks |
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19:37.13 | Assid | okay, until i figure out how to move this phone to sip, im gonna try SCCP |
19:37.26 | Assid | so, i got sccp up, but i cant manage to make my calls |
19:37.34 | Assid | http://assid.pastebin.com/d39f37288 -- a log of when i try to make it |
19:38.28 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
19:40.42 | Assid | oh it takes time |
19:41.38 | Assid | okay voice from cisco reaches polycom.. not the other way around |
19:42.54 | phpboy | libpri is obviously a requirement when using E1 on dahdi? |
19:43.12 | eppigy | YEAH SON |
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19:54.22 | Assid | okay im having 1 way audio for a sccp <-> sip call |
19:58.47 | mmlj4 | are you behind NAT? |
20:00.42 | phpboy | There's actually quite a bit you've got to do before you can install asterisk/MySQL/dahdi/Libpri |
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20:03.05 | telnettech | hello |
20:03.32 | jaytee | hi |
20:03.46 | telnettech | i was checking to see if i was still connected |
20:04.20 | jaytee | nope, you're offline |
20:04.28 | jaytee | completely out of touch |
20:04.30 | *** join/#asterisk mindCrime (n=chatzill@74.213.159.129) |
20:05.05 | jaytee | adrift in limbo |
20:05.08 | *** join/#asterisk jtodd (n=jtodd@nat/digium/x-854a1f4b87c7e729) |
20:06.36 | telnettech | i wish i was adrift with either a pina colada or tequila sunrise on the beach instead of this hot telephone room |
20:09.08 | denon | if the telephone room is hot, you'll probably be spending even more time in there |
20:09.24 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) |
20:09.25 | denon | likes ~60 degrees F |
20:09.27 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
20:09.51 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
20:10.30 | telnettech | they have AC but i have to door open.....it has a small Mitel but the humidity is bad here in Aruba today |
20:11.00 | telnettech | not too hot for the equipent.....yet |
20:11.39 | telnettech | but im sweating and would rather be sweating sitting on the beach looking and chasing women |
20:14.21 | Assid | mmlj4: yes |
20:14.45 | phpboy | This new server is going to be BEAUTIFUL when I'm done with it |
20:14.48 | Assid | mmlj4: sorry for the delay, trying to figure this out |
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20:17.47 | Assid | cant even get it to play back voicemailmain |
20:18.14 | scottgutman | hello all |
20:19.01 | phpboy | hi |
20:20.15 | scottgutman | i was hoping i could find some help. i am a noob at asterisk |
20:20.42 | phpboy | scottgutman: We cannot help if we don't know what the problem is? |
20:24.24 | scottgutman | i am using a kiax soft phone, asterisk 1.2.24/slackware 12, connecting to binfone IAX termination. When i make a call the Connection is not transferring correctly and route through asterisk instead of making a direct connection. I have transfer=yes in the iax.conf |
20:25.59 | edoceo | how do I get a local number in Thailand to ring my server in Seattle? |
20:26.27 | phpboy | edoceo: Over the internet? |
20:30.42 | *** join/#asterisk c0ldk1ll3r (i=be348af8@gateway/web/ajax/mibbit.com/x-a866f85c853cf93f) |
20:31.00 | SuPrSluG | scottgutman: if your a noob. why use asterisk 1.2.x they're about to release 1.6 already. |
20:32.06 | scottgutman | i followed a step by step guide, and i did not want version conflicts. after this i plan to install freepbx and vicidial |
20:32.11 | SuPrSluG | anyhow are the phone and your phones registered |
20:32.52 | SuPrSluG | err provider |
20:33.36 | scottgutman | should i paste from the iax.conf? |
20:34.28 | edoceo | phpboy: right - but do I need a server in Thailand to have the local number terminate to? |
20:35.00 | SuPrSluG | go to the cli > iax2 show registry |
20:35.49 | scottgutman | Host Username Perceived Refresh State |
20:35.49 | scottgutman | 208.72.186.132:4569 103067 76.18.0.243:4569 60 Registered |
20:37.28 | SuPrSluG | use pastebin when doing that |
20:37.30 | phpboy | dahdi_genconf automatically generates configs? |
20:37.34 | SuPrSluG | ~pb |
20:37.35 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
20:37.51 | scottgutman | got it |
20:38.11 | phpboy | edoceo: yes, you would need a server in thailand |
20:38.32 | SuPrSluG | couple lines ok. more than that some folks get ornery |
20:38.49 | scottgutman | makes sense |
20:40.22 | SuPrSluG | so the phone and provider are connected. what happens when you make a call? do a cli> set verbose 7 and make the same call |
20:41.04 | SuPrSluG | look for errors |
20:42.35 | scottgutman | i have a message log open and verbose is set to 88. here is the last call. http://pastebin.com/d6cc00ec1 |
20:43.57 | jaytee | beek, are you here? |
20:44.13 | beek | jaytee: yes sir |
20:44.24 | phpboy | This sucks |
20:44.35 | jaytee | beek, embedded silence in a text source? ever done it? |
20:44.37 | phpboy | dahdi_genconf doesn't wanna auto generate config files :( |
20:45.00 | beek | jaytee: no. I could try it though. Are you having issues with it? |
20:45.51 | jaytee | was just wondering if there was a special character set for Cepstral to recognize. Didn't find anything on their site yet. I'll keep searchin and googling. |
20:46.11 | beek | jaytee:IIRC, you can get swift to embed other audio files so you should be able to do it using the wav files from the Asterisk distribution. |
20:46.13 | jaytee | if all else fails I'll just pad silence with audacity |
20:46.54 | beek | jaytee: Look in their docs for the link to the embedded TTS codes from another web site. It had that which you seek. Let me see if I can find it again. |
20:47.33 | jaytee | beek, never mind, I'll do the diggin. just thought I'd ask if you knew off the top of your head. |
20:47.36 | riddlebox | Katty: you around? |
20:48.06 | beek | jaytee: start here http://www.cepstral.com/cgi-bin/support?page=faq&type=ssml |
20:48.16 | jaytee | beek, thanks. will do |
20:48.30 | beek | jaytee: Then here: http://www.w3.org/TR/speech-synthesis/ |
20:49.10 | *** join/#asterisk ManxPower (n=manxpowe@router.asteriasgi.com) |
20:50.51 | scottgutman | SuPrSluG: did i put too much? |
20:51.29 | jaytee | beek, yup! found it. <break time='4500ms' /> inserting that into the text where I want silence gives me a 4.5 second pause. |
20:52.13 | beek | jaytee: cool. |
20:55.15 | *** join/#asterisk joesuffceren2 (n=chatzill@srv.fgp.com) |
20:55.35 | joesuffceren2 | can anyone recommend an asterisk compatible tapi driver that will work on a 64-bit terminal server? I really like activatsp (installed on my workstation),but they don't have a 64-bit version, and I don't have visual studio to compile my own |
20:57.01 | Assid | anyone know why my cisco phone cant hear anything? i mean outgoing audio works, incoming nope |
20:57.10 | Assid | the box is remote, phone is behind a nat |
20:57.39 | phpboy | I wish I could find a nice doc on how to configure dahdi properly for E1 :/ |
20:57.53 | Assid | calls between cisco and polycom polycom can hear cisco, but not the other way around |
20:58.11 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
20:59.32 | Qwell | <Assid> the box is remote, phone is behind a nat |
20:59.37 | Qwell | Assid: sounds like you answered your own question |
20:59.52 | jjshoe | sipnat~ |
20:59.55 | jjshoe | or something like that |
20:59.59 | jjshoe | however that works in here |
21:00.01 | Assid | ccspnat |
21:00.04 | [TK]D-Fender | Qwell: Worse yet... he's using SCCP |
21:00.12 | Assid | sccp even |
21:00.14 | jjshoe | ~sipnat |
21:00.14 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
21:00.31 | Assid | my other sip devices are able to talk to each other |
21:00.36 | Assid | its only this stupid thing |
21:01.28 | Kobaz | is nat=yes |
21:01.31 | Assid | mainly cause i cant figure out how to push this to sip, and if i get it working, theres a chance i will have a job/project i can work with |
21:01.37 | Assid | Kobaz: on sccp? |
21:01.45 | Kobaz | you said sip |
21:01.51 | Kobaz | oh, other sip devices |
21:02.21 | Assid | and i normally use nat=route :P |
21:04.13 | SuPrSluG | scottgutman: host=dynamic ? |
21:04.56 | scottgutman | yes |
21:06.49 | SuPrSluG | have you tried using sip termination? this is an odd problem. |
21:07.18 | DaPrivateer | trying to use Page() on 1.4.22 w/ FreeBSD 6.2 and getting error app_meetme.c:1620 conf_run: Unable to set flags: Inappropriate ioctl for device -- anyone have any suggestions? |
21:07.37 | phpboy | Dahdi is a mission to get working :/ |
21:07.38 | SuPrSluG | that or try newer verison of asterisk |
21:07.49 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
21:07.58 | scottgutman | the sections from the iax.conf: http://pastebin.com/m1ca33f |
21:08.56 | scottgutman | do you know about vicidial? do you think there will be a conflict? |
21:09.47 | SuPrSluG | yes I played around with it a while back. they too are using 1.4 now |
21:09.48 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-221-124.phlapa.east.verizon.net) |
21:10.07 | SuPrSluG | <PROTECTED> |
21:11.00 | SuPrSluG | i had a 3 man call center with it. works as advertised |
21:11.06 | scottgutman | to upgrade, just download the tarballs, make, make install and poof, upgraded? |
21:11.18 | SuPrSluG | yeah |
21:11.31 | scottgutman | show i do 1.6 or 1.4 |
21:12.18 | SuPrSluG | 1.4 if your going towards vicidial they'll be months behind when 1.6 is released. |
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21:12.51 | scottgutman | any particular ver of 1.4 or the lastest one will do? |
21:13.05 | SuPrSluG | there's a lot to test and rewrite |
21:13.38 | *** join/#asterisk qdk_ (n=qdk@79.138.251.161.bredband.3.dk) |
21:13.46 | SuPrSluG | check what they're using in their forums i'd guess 1.4.22 should be good |
21:15.44 | scottgutman | thanks for your help |
21:15.52 | scottgutman | and advice |
21:16.12 | Assid | okay i wnna shift this to sip |
21:16.16 | Assid | im tired of sccp |
21:16.24 | Assid | i want to make calls and be able to hear people |
21:17.10 | srd | if you want to hear people you'll have to buy the hearing people package |
21:19.54 | kannan | Assis, try to change rtp.conf to match the cisco's RTP map, its some like 16000+ to 3000o i think |
21:20.04 | kannan | Assid |
21:20.19 | kannan | it may not be the NAT issue |
21:21.21 | Assid | hrmm |
21:21.21 | Assid | k |
21:23.54 | Assid | yeah it came up once |
21:26.00 | Assid | now i cant get audio again |
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21:27.05 | [TK]D-Fender | Checkout time, later all |
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21:29.34 | *** mode/#asterisk [+o bkruse] by ChanServ |
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21:30.09 | *** mode/#asterisk [+o bkruse] by ChanServ |
21:37.38 | NovceGuru | Anybody know anything about PTCRB certs? If a gsm/cdma module is certified, does the final product using that module have to be certified? |
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21:38.23 | khronos | <PROTECTED> |
21:38.31 | NovceGuru | oh? |
21:43.10 | phpboy | DAMNIT! |
21:43.23 | phpboy | Now I get seg faults when using IAX2 :/ |
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21:52.35 | gambler1 | Hi, is there any method that we can handle sip messages in dialplan? (like addres incomplete) |
21:53.07 | gambler1 | ax.. my fingers... is there any method that we can use to handle sip messages in dialplan? (like addres incomplete) |
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21:54.12 | dandate | I'm trying to install flash pbx and it has been stuck at *now compiling zaptel for over an hour, is this usual? |
21:54.39 | dandate | i'm sorry thats pbx in a flashj |
21:56.24 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
21:58.01 | *** join/#asterisk jantman (n=jantman@nat02-hill-ext.rutgers.edu) |
21:58.32 | jantman | can anyone recommend a good forum for general VoIP questions (perhaps asterisk-specific, but more general advice than tech stuff)? |
22:00.09 | *** part/#asterisk jantman (n=jantman@nat02-hill-ext.rutgers.edu) |
22:02.16 | x86 | hey guys... trying to think of the best way to do night mode... |
22:02.49 | x86 | was thinking about setting up an extension that would simply touch a file, and if the file exists, the main IVR would know it was in night mode |
22:03.01 | x86 | and then just dial that extension again to have it remove that file |
22:03.22 | x86 | is that a decent idea, or is there a better practice |
22:03.59 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
22:05.27 | carrar | x86, why not use a db? |
22:05.32 | carrar | thats what I use |
22:05.51 | carrar | then they can do it via a call or via the web |
22:06.16 | *** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) |
22:07.40 | thedonvaughn | If I assign a mailbox to a sip user in users.conf via mailbox = , where is the vmpass and other info stored? It's not writing anything to /etc/asterisk/voicemail.conf and I do not see anything stored in astdb. However, my voicemail is working and my pin is set. I'm just asking because I forsee someone forgetting their vm pin sometime in the future after setting it through the menu and i just want to know how i can a) reset it or b) retriev |
22:09.01 | [TK]D-Fender | thedonvaughn: voicemail.conf <- |
22:09.17 | thedonvaughn | none of my voicemails are there |
22:09.49 | [TK]D-Fender | thedonvaughn: thats where the PIN is. VM's are under /var/spool/asterisk/voicemail normally |
22:10.20 | thedonvaughn | [TK]D-Fender: no i mean no entries for any of my voicemail boxes in /etc/asterisk/voicemail.conf. That's why i'm asking :) |
22:10.30 | *** join/#asterisk sevard (n=sev@multimedia.dvc.edu) |
22:10.36 | jtodd | x86: If you're looking for bleeding-edge, you could also look at the calendar stuff that twilson is working on. Then just create a calendar with events that span "night" times (or holidays, or weekends, or whatever.) |
22:10.38 | thedonvaughn | I only set them up in users.conf, and voicemail.conf is never written to. I can even do voicemail show users in console and see all my users. |
22:10.52 | thedonvaughn | everyone's working fine, they can change their pins via the menu system and all is good. Just need to know _WHERE_ that info is stored. |
22:10.56 | thedonvaughn | it's definitely not in voicemail.conf for me |
22:11.14 | [TK]D-Fender | thedonvaughn: Check the VM folders then |
22:11.23 | thedonvaughn | yah let me start there |
22:11.24 | thedonvaughn | thanks |
22:11.24 | [TK]D-Fender | thedonvaughn: perhaps it is in users.conf |
22:11.31 | thedonvaughn | [TK]D-Fender: yah thought that to, it's not heh |
22:11.39 | [TK]D-Fender | ~users.conf |
22:11.39 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
22:11.41 | [TK]D-Fender | ^^^^^^^^^^ |
22:12.05 | thedonvaughn | yah i'm starting to realize this |
22:12.50 | DaPrivateer | trying to use Page() on 1.4.22 w/ FreeBSD 6.2 and getting error app_meetme.c:1620 conf_run: Unable to set flags: Inappropriate ioctl for device -- anyone have any suggestions? |
22:14.35 | [TK]D-Fender | DaPrivateer: http://groups.google.com/group/mailing.freebsd.ports-bugs/browse_thread/thread/0b6d0fbfa8d85255 |
22:15.21 | DaPrivateer | yes, i saw this, but commenting out an fdset slightly worries me |
22:15.59 | *** join/#asterisk dandate2 (n=dandate2@adsl-99-183-242-53.dsl.pltn13.sbcglobal.net) |
22:16.11 | dandate2 | I'm trying to install pbx in a flash and it has been stuck at *now compiling zaptel for all time, is this usual? |
22:16.31 | voxter | Hey FYI whoever out there using polycoms, and potentially has left the polycom default username and password for FTP login for provisioning purposes (PlcmSpIp/PlcmSpIp) - There are bots going around lately trying to log into servers with those credentials and exploit them to join botnets. |
22:19.37 | [TK]D-Fender | voxter: Anyone leaving defaults like that deserve what they get. Just think of all the Trixbox idiots getting pwned out there ;) |
22:19.50 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
22:20.56 | sevard | [TK]D-Fender: I agree. Leave your linksys wrt54g open and your neighbor will leech and not even invite you over for his bbq. |
22:21.09 | voxter | [TK]D-Fender: I've posted stuff about trixbox on their forums like this before too. Just saw it happen to a couple people who thought "nobody will ever guess that anyways" - thought it'd be useful to point out to people that yes, even though its obscure, it does happen to people. |
22:21.49 | [TK]D-Fender | voxter: 2 words : NATURAL SELECTION |
22:22.14 | [TK]D-Fender | voxter: Let'em whine |
22:22.42 | beek | sevard: http://ars.userfriendly.org/cartoons/?id=20090105 |
22:23.05 | sevard | beek: hahahaha. |
22:26.02 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
22:32.26 | Katty | blergh |
22:34.03 | [TK]D-Fender | Katty: Mew. |
22:34.44 | Katty | hey |
22:34.47 | Katty | jbot: who is peter grace? |
22:34.49 | jbot | I think you lost me on that one, Katty |
22:35.19 | [TK]D-Fender | jbot: Who killed J.R.? |
22:35.35 | [TK]D-Fender | jbot: USELESS |
22:35.36 | jbot | ACTION starts crying and hides from [tk]d-fender in the darkest corner of the room. :( |
22:35.46 | [TK]D-Fender | :D |
22:35.52 | Corydon76-dig | ~drumkilla |
22:35.52 | jbot | methinks drumkilla is Russell Bryant, the Asterisk release branch maintainer <russelb@clemson.edu>, or someone who should be ph33r3d, or russellb |
22:35.56 | Katty | wonders how these people know her ^_- |
22:36.03 | Corydon76-dig | ~katty |
22:36.04 | jbot | hmm... katty is the only girl in the channel, so be nice to her |
22:36.14 | seanbright | pffft |
22:36.28 | Katty | seanbright: who is peter grace? :< |
22:36.28 | Corydon76-dig | That's right; seanbright is also a girl... |
22:36.33 | Corydon76-dig | Well, a girly-man |
22:36.39 | seanbright | Katty: no clue |
22:37.51 | Corydon76-dig | Katty: not sure why, but the letter 'N' comes to mind |
22:37.52 | Katty | what about a david mcnett |
22:37.56 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
22:37.57 | seanbright | nope |
22:38.00 | seanbright | nothing |
22:38.11 | Katty | yet they all claim to know me |
22:38.13 | Katty | this is madness |
22:38.16 | russellb | o.O |
22:38.25 | russellb | wonders how he got brought into this |
22:38.25 | Corydon76-dig | I remember Peter Grace, but I don't remember his nick |
22:38.39 | seanbright | i remember big league chew |
22:38.44 | Corydon76-dig | russellb: d-fender made the bot cry |
22:38.54 | russellb | bad [TK]D-Fender ! |
22:38.59 | [TK]D-Fender | Katty: If you believe that.... then you may already have won the Publisher's Clearing House Sweepstakes!!!!!! |
22:39.34 | [TK]D-Fender | ~jbot |
22:39.34 | jbot | it has been said that jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
22:39.36 | Katty | [TK]D-Fender: yeah, but they have 'mutual' friends which i recognize :/ |
22:39.49 | [TK]D-Fender | Corydon76-dig: MY bitch... get your own! :p |
22:40.17 | [TK]D-Fender | unfUNFunfUNFunfUNFunfUNFunfUNFunfUNF |
22:40.53 | Corydon76-dig | Katty: remember "km-" on here? |
22:42.33 | Katty | Corydon76-dig: sound familiar |
22:42.57 | Corydon76-dig | Katty: that's Peter Grace |
22:43.15 | Katty | Ooooo |
22:43.52 | Katty | wow. |
22:43.56 | Katty | 120 friends down to 21 |
22:44.06 | Katty | how about that for almost-spring cleaning |
22:44.08 | Corydon76-dig | Facebook? |
22:44.14 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
22:44.21 | Katty | nods |
22:44.27 | russellb | I bet you unfriended me, didn't you! |
22:44.27 | Corydon76-dig | Katty: add me! |
22:46.05 | file | Katty: pssssssst, David McNett is Nugget over there |
22:57.31 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
23:01.54 | jets | Has any one found a good way of demonstrating G722 inside of asterisk trunk? |
23:02.28 | jets | Two phones that call one another sound great but I was hoping to have an extension with a gsm audo file played |
23:02.36 | jets | and an extension with an mp3 played to show the call quality difference |
23:03.21 | drmessano | Set up a conference |
23:03.35 | drmessano | Put the 2 G722 phones, one set to GSM, maybe one set to G711 in it |
23:03.38 | [TK]D-Fender | drmessano: Meetme still mixes to 8khz IIRC |
23:03.54 | drmessano | hmm |
23:03.58 | drmessano | Didnt realize that |
23:04.12 | drmessano | Ok |
23:04.12 | jets | It seems like everything inside asterisk does :( |
23:04.17 | drmessano | 3 way calling |
23:04.22 | [TK]D-Fender | drmessano: thats a point FreeSWITCH pimps is that their core is higher |
23:04.38 | drmessano | Get a GSM and a G722 phone, call the G722 phone on call waiting |
23:04.41 | drmessano | Click back and forth |
23:04.58 | [TK]D-Fender | ok, off to martial arts, BBIAB |
23:05.15 | jets | I need something automated as if i were in a sound booth. I was hoping I could set extension 100 to be a high bit rate g722 audio file or prompt. |
23:05.20 | *** join/#asterisk murdock_ut (n=chatzill@64-42-64-98.atgi.net) |
23:05.22 | jets | and 101 to be a g711/gsm prompt played |
23:06.03 | _ShrikE | jets: this demo does pretty much exactly what you are asking for. sip uri: wbdemo@conf.zipdx.com |
23:06.07 | murdock_ut | I've ran into a bit of a problem. Using 1.6, when I use one touch parking and I pick back up the parked call before it rings back I cannot repark the call. |
23:06.24 | _ShrikE | assuming it is still up. |
23:07.38 | murdock_ut | If the parked call rings back I can repark it just fine. |
23:11.23 | jets | Any tips to calling this sip uri from asterisk? |
23:11.30 | jets | It keeps telling me I am called in at narrow band :( |
23:12.41 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
23:13.02 | rob0 | "core show application dial"? |
23:20.28 | Nugget | sits down next to Katty |
23:20.43 | _ShrikE | jets: You may need to create a peer and only allow g722 |
23:21.06 | jets | _ShrikE: I did that and a show on the channel showed g722, but ulaw being used |
23:24.14 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
23:26.37 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
23:27.19 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
23:28.25 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
23:39.29 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
23:49.44 | *** join/#asterisk Zippoman (n=bobperry@cpe-76-95-113-203.socal.res.rr.com) |
23:49.47 | Zippoman | Hey guys |
23:49.51 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
23:50.01 | Dovid | hi Zippoman |
23:50.39 | Zippoman | small issue...when i dial out from asterisk it always says my number is blocked when right before it i set the callerid |
23:51.00 | *** join/#asterisk `paul (n=temp_acc@122.55.36.3) |
23:51.17 | *** join/#asterisk jmacz (n=jmacz@190.159.100.152) |
23:51.23 | Dovid | ~pb |
23:51.23 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
23:51.28 | `paul | will dhadi work on older versions of asterisk??? |
23:51.31 | Dovid | post ur extensions.comf |
23:51.47 | Dovid | `paul: What version ? I dont think it will for real old versions |
23:52.09 | `paul | 1.4.... |
23:52.10 | beek | `paul: 1.4.xx (where xx = I'm not quite sure) and above |
23:52.19 | Dovid | extensions.conf* |
23:52.31 | `paul | cause i cant seem to get ztdummy working for meeme |
23:52.46 | `paul | but i remember making it work on dhadi |
23:53.07 | beek | `paul: Is it the newest version of 1.4? |
23:53.08 | Dovid | not sure if all of 1.4.X works |
23:53.16 | Dovid | do a make menuselect and see if dahdi is there |
23:53.53 | *** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net) |
23:54.29 | jaytee | dahdi will only work on 1.4.22 or any higher rc version, not on 1.4.21 or earlier |
23:55.02 | `paul | i did menuselect zaptel is required for meetme |
23:55.11 | Dovid | then ther is ur answer ;) |
23:55.19 | Dovid | u can optomize the kernel to work better |
23:55.23 | `paul | well actually i got zaptel installed but i get a psuedo device warning when using meetme |
23:55.54 | jaytee | or to be more precise as I understand, 1.4.22 was the first release version with dahdi, you might get it working on some slightly earlier versions with some tweaking |
23:56.13 | `paul | app_meetme.c:772 build_conf: Unable to open pseudo device <---all i get is this when using meetme |
23:56.27 | `paul | pls help :( |
23:56.47 | jaytee | do a ps aux | grep ztdummy |
23:57.07 | jaytee | see if it's loading ztdummy |
23:57.33 | `paul | its not |
23:57.50 | `paul | actually it fails when i do a /etc/init.d/zaptel start |
23:58.06 | `paul | Loading zaptel framework: FATAL: Module zaptel not found |
23:58.14 | `paul | Waiting for zap to come online...Error: missing /dev/zap! |
23:58.23 | `paul | those are the error msg |
23:59.09 | jaytee | what distro? |
23:59.30 | `paul | centos 5 |
23:59.58 | jaytee | did you do a make install when you compiled zaptel? |