| 00:00.15 | LucasAR | Hello, |
| 00:00.18 | epoch | florz: something very strange is going on ;/ |
| 00:00.28 | epoch | but now my GotoIf isn't even working |
| 00:00.29 | LucasAR | N1 playing with Steves R2 chan_unicall ? |
| 00:01.44 | tessier_ | If I ever make much money off of * I'm gonna pay someone to put in better error messages |
| 00:01.59 | tessier_ | Unable to create channel of type 'SIP' |
| 00:02.02 | tessier_ | WHY THE FUCK NOT? |
| 00:02.27 | Moc | tessier, because it is not registred maybe ?> |
| 00:02.32 | epoch | tessier_: that really could be any reason... |
| 00:02.54 | Moc | and I had recently a cvs head that didnt work with some sip peer anymore |
| 00:03.21 | tessier_ | Moc: This was all working 30 minutes ago... |
| 00:03.25 | tessier_ | trying to figure out what I might have changed |
| 00:03.33 | tessier_ | The phones are all configured to register... |
| 00:03.36 | epoch | can someone have a look at this: http://pastebin.ca/1796 |
| 00:03.54 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 00:04.46 | *** join/#asterisk gambolputty (~gambolput@cblmdm204-118-177-213.buckeye-express.com) |
| 00:05.02 | tessier_ | hmmm |
| 00:05.14 | epoch | http://pastebin.ca/1797 <- that's what's happening |
| 00:05.21 | epoch | I don't get it |
| 00:05.37 | *** join/#asterisk ZX81 (~ZX81@222-152-92-158.jetstream.xtra.co.nz) |
| 00:05.49 | epoch | OH WAIT |
| 00:05.58 | epoch | I'm friggen retarded :( |
| 00:06.13 | patdk | that is what I thought |
| 00:06.17 | epoch | haha |
| 00:06.25 | epoch | I've had a long week :( |
| 00:06.29 | patdk | hehe |
| 00:06.34 | epoch | very little sleep, too much work ;/ |
| 00:06.40 | *** join/#asterisk brc_ (~bob@brc.base.supporter.pdpc) |
| 00:06.44 | epoch | I've been at work 12 hours so far today |
| 00:06.46 | patdk | me too, thankfully everything worked correctly the first time |
| 00:06.49 | epoch | (and this is a short day) ;/ |
| 00:06.50 | patdk | better than expected |
| 00:07.06 | epoch | that's always good :) |
| 00:07.35 | ZX81 | If anyone has any questions they want included in an interview with Mark Spencer could they msg me. |
| 00:08.16 | ZX81 | or join #asterisk-smoker |
| 00:08.16 | ZX81 | or join #asterisk-smokers |
| 00:08.23 | *** join/#asterisk rowter (~Drake@201.129.87.106) |
| 00:08.50 | epoch | Now, this issue of GotoIf() taking a long time was weird... it seems it sped up as soon as I changed the extension names to ones that weren't like 30 characters long |
| 00:11.04 | Wangster | Can someone explain to me how VOIP providers gateway to the PSTN in any city? |
| 00:11.05 | patdk | need to reinstall quicken onto this computer, so I can pay my bills |
| 00:11.16 | patdk | wangster, they don't |
| 00:11.39 | Wangster | patdk: so how does it work? |
| 00:11.40 | ZX81 | Wangster: they sort peering agreements |
| 00:12.27 | epoch | Wangster: for the ones who do have presences in other cities, all they need is a net connection, a PSTN connection, a * box, and some rackspace to put it all in :) |
| 00:12.45 | epoch | but peering arrangements are usually much less expensive ;P |
| 00:12.48 | *** join/#asterisk wolfson (hehe@cpe-68-187-186-066.man.nc.charter.com) |
| 00:12.55 | Wangster | I signed up with sixtel for some testing. I dial a number in my area code, how is that connection made? They don't have any local presense. |
| 00:13.02 | Wangster | So who do they peer with? |
| 00:13.20 | epoch | Wangster: ask them :) |
| 00:13.34 | patdk | wangster, they did a long distance phone call |
| 00:13.56 | patdk | they buy bulk min, so long distance doesn't cost them anything or hardly nothing |
| 00:14.16 | Wangster | patdk: so you figure they just gateway to the PSTN in their own locality and then resell the minutes? |
| 00:14.37 | patdk | that is how most people do it, inside a single country |
| 00:14.53 | patdk | when you do inter-country/contanet, they normally have another gateway |
| 00:15.07 | patdk | if they take their stuff seriously |
| 00:15.17 | ZX81 | Also, if you go to the asterisk-biz list you will see people buying and selling minutes and requesting area codes |
| 00:15.27 | ZX81 | for exchange between providers |
| 00:16.17 | Wangster | A PRI/PSTN gateway here is min $800/mo. and that doesn't include any DIDs. |
| 00:16.41 | epoch | eh? |
| 00:16.53 | epoch | you mean, the PRI is $800/mo? |
| 00:16.55 | Wangster | eh! |
| 00:17.05 | Wangster | yes, something in that area. |
| 00:17.17 | epoch | that's what one of our customers pays |
| 00:17.22 | epoch | $2/mo per DID |
| 00:17.38 | epoch | and, 17000mins/mo of LD throughout NA |
| 00:17.45 | Marlow | Wangster : that depends on where your gateway is . |
| 00:17.54 | astoria | recently, i got three quotes in the metro detroit area for PRI and I got 800 from SBC, 700 from LDMI, and 300 from XO |
| 00:18.03 | Marlow | Wangster : if you want the pri to your location, it'll cost ya .. |
| 00:18.10 | epoch | astoria: gee, go with SBC! |
| 00:18.24 | Marlow | Wangster : if the telco and you are in the same datacenter, you get it a lot cheaper |
| 00:18.36 | Wangster | I'm trying to figure out how to gateway to my local area code. |
| 00:18.36 | astoria | I'm worried about why XO is so cheap |
| 00:18.51 | Wangster | Marlow: I can put a box in my telco co-location but its more than the PRI !! |
| 00:18.56 | epoch | astoria: why worry? just get it! ;) |
| 00:18.57 | astoria | oh yea, the installation was 2k on both LDMI and SBC, but XO's installation was free |
| 00:19.17 | epoch | maybe XO is trying to get more customers |
| 00:19.23 | Marlow | Wangster : that's not my case .. |
| 00:19.30 | astoria | epoch: it doesn't hurt to know someone at XO either ;) |
| 00:19.37 | patdk | ya, XO around here is charging like just over 100 |
| 00:19.37 | epoch | astoria: oh, well gee :) |
| 00:19.41 | Marlow | Wangster : i pay something like 100 eur/month for hosting .. (1U) |
| 00:19.43 | patdk | most others wanted 600-800 |
| 00:19.43 | epoch | $100? |
| 00:19.45 | epoch | wtf |
| 00:19.48 | patdk | I know |
| 00:19.50 | epoch | where are you, patdk? |
| 00:19.55 | patdk | washington d.c. |
| 00:19.58 | Marlow | Wangster : and get the pri for nearly free .. |
| 00:20.01 | epoch | that's nutty |
| 00:20.04 | florz | Well, is there _anyone_ successfully using app_[rt]xfax? If so, using which libtiff? |
| 00:20.11 | Wangster | Marlow: ya, thats probably close to here but then the connection would be extra. |
| 00:20.14 | Marlow | Wangster : a PRI to my home would cost me 3000 EUR/year |
| 00:20.14 | maco | anybody connected directly to Interoute in EU? |
| 00:20.35 | ZX81 | 3.5.7 libtiff if i remember properly |
| 00:21.07 | astoria | I'm waiting for some of the big telco's to get into the voip termination business... only one telco does it around here... |
| 00:21.33 | Wangster | Ok, so there is no magic here. If you want to gateway to the local PSTN then you have to do it with a PRI (or FXO). |
| 00:21.36 | Marlow | Wangster : when you are in a hosting facility you often can deal about the trunk price |
| 00:21.42 | Monil | Marlow: you pay 100Euro/month for just 1u? |
| 00:21.49 | Monil | how much bandwidth with that? |
| 00:22.11 | Marlow | Monil : 100 gb .. it's Ireland .. Ireland is Rip off |
| 00:22.17 | Monil | ah, shame |
| 00:22.31 | Wangster | I thought Ireland was a big tech country now? |
| 00:22.37 | florz | ZX81: Vanilla from libtiff.org? |
| 00:22.43 | Marlow | Monil : for my US server it 7000 GB, Dual P4 Xeon server included for 270$ |
| 00:22.59 | Moc | Marlow, where ? |
| 00:23.08 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 00:23.21 | Marlow | Monil : and in germany i pay 20 eur incl. server incl 50gb traffic .. and 0,35 c pr.. GB after that |
| 00:23.24 | Moc | best I found is 2000gig for 200$ with the server |
| 00:23.32 | tessier_ | Phones keep becoming unregistered |
| 00:23.39 | tessier_ | One minute I can call them, the next minute I cannot. |
| 00:23.45 | Marlow | Moc : US ? Houston, Texas .. ev1 servers .. but no PRI trunk .. i use that for hosting |
| 00:23.47 | patdk | I assume that is ide drives? |
| 00:23.48 | tessier_ | It's a firewall/nat thing I'm pretty sure. :( |
| 00:23.56 | ZX81 | florz: yeah |
| 00:23.56 | tessier_ | ah! |
| 00:24.01 | Moc | tessier, same here.. Must be something wrong in head |
| 00:24.03 | tessier_ | The clue has been here in front of me all along I think... |
| 00:24.13 | tessier_ | Moc: I'm running a fairly old asterisk. |
| 00:24.16 | Moc | Marlow, yes |
| 00:24.21 | Moc | ha :/ |
| 00:24.37 | tessier_ | I think their nat implementation has a VERY short memory and times out connections too fast |
| 00:24.48 | tessier_ | My connection to my mail server keeps dropping if I leave the window alone for a few minutes |
| 00:24.58 | Moc | ha |
| 00:25.21 | Moc | Adagio for string... nice song |
| 00:25.30 | *** join/#asterisk Alric (~nbowyer@64.6.45.2) |
| 00:25.57 | *** join/#asterisk lancey (Shady@support.net1.cc) |
| 00:25.59 | lancey | hi guys |
| 00:26.08 | lancey | is there an init.d script shipped with asterisk? |
| 00:26.17 | ManxPower | lancey: Yes. |
| 00:26.21 | Marlow | Moc: hm .. the current deals seem to be 2000 gb |
| 00:26.28 | ManxPower | I'll leave its location as an exersize for the reader. |
| 00:26.37 | lancey | ManxPower : :) |
| 00:26.39 | Moc | ha.. Marlow, theplanet ? |
| 00:26.46 | ManxPower | tessier: Set your mail client to check for new mail every 1 min. |
| 00:27.33 | ZX81 | where did "thanks for all the fish" come from in Manager logofff command |
| 00:27.55 | ManxPower | ZX81: Hirchhiker's Guide to the Galaxy series. |
| 00:27.59 | ZX81 | ah |
| 00:28.01 | ZX81 | k |
| 00:28.16 | ManxPower | It's what the Dolphins said when they left Earth |
| 00:28.18 | robl^ | damn Hirchhikers! |
| 00:28.25 | ZX81 | :-) ok |
| 00:28.32 | lancey | ManxPower: i just copy it ot init.d? |
| 00:28.35 | Moc | Anyone, I got my hosting for free as long I dont take too much gigs |
| 00:28.45 | Moc | I got 4 box heeh |
| 00:29.13 | tessier_ | We are going through NAT and I have qualify=yes to keep the connection open but this nat box seems to drop idle connections REALLY fast. |
| 00:29.22 | ZX81 | mine is free too...but only cos some kind sould likes the daily asterisk news! |
| 00:29.24 | ZX81 | :-) |
| 00:29.40 | ManxPower | tessier: tell the SIP client to register every 60 seconds |
| 00:29.40 | ZX81 | any maybe he wanted quicker responses than going to NZ for page |
| 00:29.42 | ZX81 | :-) |
| 00:29.54 | ManxPower | ARGH! Maybe if I buy a hundred powerstrips I won't run out in a week. |
| 00:29.55 | tessier_ | ManxPower: ah....I forgot I could set that. |
| 00:30.07 | ZX81 | lol |
| 00:30.14 | ZX81 | IEC cables are the one I run out of |
| 00:30.23 | ZX81 | people steal them everytime i do a gig |
| 00:30.29 | Moc | hehe, I offered a P4 1.5,512mb ram with 60gig transfer for free to bkw for Asterisk community, but he didnt want it |
| 00:30.37 | ZX81 | I want it |
| 00:30.38 | ZX81 | :-) |
| 00:30.40 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 00:31.38 | *** join/#asterisk gh0st (~gh0sthead@204.8.140.251) |
| 00:31.58 | Marlow | Moc : he'll more be looking for a development machine with zaptel hardware |
| 00:32.13 | Marlow | Moc : my guess |
| 00:32.36 | Marlow | Monil : did you get that sipgate thingy running ? |
| 00:32.46 | Moc | Well it was open for mirror/dundi peer, whatever |
| 00:32.50 | Moc | conference server |
| 00:33.01 | *** part/#asterisk LucasAR (~mm@lineaAH254.velocom.com.ar) |
| 00:33.14 | Moc | 1 thing I like to offer, is a free voip service, based on donation |
| 00:33.38 | astoria | free voip service? kind of like PBS, but for voip? |
| 00:33.51 | Moc | like if someone donate 10$, I give 1111minutes |
| 00:34.01 | Moc | to us and canada |
| 00:34.06 | Monil | Marlow: sipgate stuff works fine, just cant get my phone to authenticate with asterisk in order to use the sipgate stuff :p |
| 00:35.32 | Moc | that the cheapest I could get |
| 00:36.09 | Moc | my idea was to have this donation service on the dundi network |
| 00:36.52 | lancey | ManxPower |
| 00:36.59 | lancey | i copied the script into /etc/init.d |
| 00:37.03 | lancey | but it doesn't start..... |
| 00:37.09 | lancey | what should i check? |
| 00:37.15 | maco | what kind of VoIP desk phones do you use most ? |
| 00:37.30 | Moc | maco, the best phone I found so far is the polycom |
| 00:37.48 | Moc | IP 500 is excellent and 'cheap' |
| 00:37.54 | maco | Moc..hmmm...price? URL ? (thx) |
| 00:38.06 | Moc | maco, check froogle |
| 00:38.11 | Moc | but about 170$ |
| 00:38.27 | Moc | www.polycom.com |
| 00:38.54 | astoria | I've been looking at getting the ArtDio IPF-3000 |
| 00:39.05 | maco | hmm |
| 00:39.18 | astoria | of course, the ciscos rock the free world. |
| 00:39.28 | Moc | astoria, polycom beat cisco |
| 00:39.37 | astoria | check out www.voipsupply.com |
| 00:40.14 | Moc | voipsupply aint that cheap |
| 00:40.14 | astoria | yeah, but they have a wide range to look at. |
| 00:40.14 | Marlow | rhex : you are sending the calls to localhost :) |
| 00:40.14 | Moc | that I agree |
| 00:40.48 | Moc | Polycom ip 500 are really AMAZING... it configurations options are GREAT |
| 00:41.01 | maco | Moc SIP/H323 ? |
| 00:41.03 | Moc | of course ip 600 is better.. but cost more |
| 00:41.07 | Moc | maco, both |
| 00:41.13 | astoria | Moc: the phone looks kind of ugly though, is it really that ugly in person? |
| 00:41.14 | Moc | depending of the firmware |
| 00:41.34 | astoria | Moc: thats a stupid question, i know, but it does matter to my clients! |
| 00:41.45 | Moc | astoria, it aint that bad. Ofcourse cisco look better, but actually, polycom handfree sound better. And the display is easier to read |
| 00:42.03 | maco | IP500 looks terrible :) |
| 00:42.12 | astoria | maco: do you have one? |
| 00:42.23 | maco | i'm a product designer ant this is the worst IP phone I have ever seen :) |
| 00:42.35 | Moc | maco, but it the bes |
| 00:42.36 | maco | astoria no, I'm checking it on the net :) |
| 00:42.38 | Moc | best |
| 00:42.56 | Moc | I couldn't find a phone that compare to the polycom.. |
| 00:42.56 | maco | why is it so big ? :) |
| 00:43.06 | Moc | it the same size than the cisco |
| 00:43.16 | Moc | alittle larger, but less high |
| 00:43.17 | astoria | Moc: does it have nat pass-thru? |
| 00:43.17 | maco | what about IP300 ? |
| 00:43.27 | Moc | maco, I didnt saw the ip 300 in person |
| 00:43.47 | Moc | astoria, well I use it behing a nat without problems |
| 00:44.01 | astoria | Moc: no, i mean does it have a port to plug your PC into? |
| 00:44.05 | Moc | if you dont want to port forward, you can set the register to 60 second, so it never timeout |
| 00:44.15 | Moc | maybe there is better way, but that one work for me |
| 00:44.35 | maco | astoria switch u mean I guess, not nat pass thru |
| 00:44.53 | astoria | maco: my bad, sorry |
| 00:45.00 | Moc | on incoming call on the polycom, you can ignore it, also you can compose number without the need to pickup the handset, or pressing a new call button |
| 00:45.16 | Moc | it show longer name correctly also compare to the cisco |
| 00:45.44 | Moc | and if it too long, it has nice feature. Like if Im call Jean-Francois Dionne, it will convert to JF Dionne on the display |
| 00:45.57 | maco | We have ciscos in HP and the work pretty fine |
| 00:46.14 | Moc | you can configure the locations of all the button, have different voicemail ext for each extentions |
| 00:46.27 | Moc | have different ringtone for every extentions |
| 00:46.39 | maco | sounds nice |
| 00:46.44 | Moc | and all kind of nice options.. |
| 00:46.57 | Moc | intercom functionallity is great on it |
| 00:47.13 | astoria | Moc: do you work for polycom ? ;) |
| 00:47.16 | Moc | so autoanswer can be configured |
| 00:47.16 | maco | hm, I need something VERY VERY cheap for open-wireless community |
| 00:47.28 | astoria | maco: grandstream? |
| 00:47.30 | maco | astoria :)) |
| 00:47.35 | Moc | astoria, nope, but I wish they give me free phone thought ;) |
| 00:47.37 | maco | grandstream ? |
| 00:47.39 | Moc | astoria, |
| 00:47.57 | astoria | maco: they're under 100 bucks |
| 00:48.19 | Moc | astoria, I bought a cisco phone out of comments in this channel, being said it the best phone. But I once I found out of the price, and tryed out the polycom... Im sold .. |
| 00:48.30 | Moc | you could get a polycom ip 300 for 120$ |
| 00:48.39 | Moc | I would get than ratter than other cheap phone |
| 00:48.40 | astoria | I'll have to give the ip500 a shot |
| 00:49.00 | astoria | the polycomm's are 100% compatible with *, right? |
| 00:49.05 | Moc | astoria, the configuration (via xml and dl from a FTP...) |
| 00:49.14 | Moc | astoria, well polycom has feature that * dont support ;) |
| 00:49.27 | Moc | it has a instant messaging on it.. |
| 00:49.33 | astoria | Moc: nothing major though right? I mean all the vm buttons work, etc |
| 00:49.33 | Moc | that I disabled since * dont support it.. |
| 00:49.38 | Moc | oh yes |
| 00:49.52 | Moc | everything work, do not disturb button work too.. |
| 00:49.52 | astoria | Moc: the user isn't going to notice a feature missing, right? |
| 00:50.15 | Moc | nope, conference, transfer (blind..) all work |
| 00:50.22 | Moc | call forward work |
| 00:50.25 | astoria | nice |
| 00:50.46 | maco | anybody using those Grandstream phones ? |
| 00:50.53 | Moc | maco, I got one :( |
| 00:50.58 | maco | Moc and ? |
| 00:51.08 | Moc | astoria, check out the Administration pdf for the polycom and look at the features |
| 00:51.19 | Moc | maco, well it a cheapo phone. |
| 00:51.27 | Moc | I dont think they can make it worst |
| 00:51.43 | Moc | it actually work.. but it aint great |
| 00:51.47 | maco | IC |
| 00:51.55 | Moc | it look very cheapo |
| 00:52.06 | maco | hm, may I ask? what is FXO port ? |
| 00:52.07 | Moc | but it has a blue backlight ;) |
| 00:52.19 | Moc | maco, FXO port can connect to your Telco |
| 00:52.37 | Moc | so your telephone line from ... AT&T or whatever... Bell .. is a FXO |
| 00:52.41 | astoria | FXO=telco, FXS=dialtone for conventional phones |
| 00:52.46 | Moc | well it need to connect to a FXO port |
| 00:53.19 | Moc | Think of FXO = a Phone. |
| 00:53.38 | Moc | and it always FXO -> FXS or FXS -> FXO |
| 00:53.43 | *** join/#asterisk scubasteve (~tiffany@rdu88-251-252.nc.rr.com) |
| 00:53.47 | astoria | I've always wondered, whats different between a FXO and a normal voice/data modem? |
| 00:54.03 | scubasteve | Can someone give me a hand with siproxd configuration? |
| 00:54.06 | scubasteve | I think I'm close. |
| 00:54.09 | jsharp | nothing. A voice/data modem has an FXO port on it. |
| 00:54.25 | maco | so 4 example if I want to make telco gateway from VoIP I would need some HW with FXO port into my Asterisk box? |
| 00:54.29 | astoria | then why do i need to buy a digium card, can't i just use a voice/data modem? |
| 00:54.30 | Mavvie | for everybody who bought a TE410P.... did yours come with documentation? |
| 00:54.51 | jsharp | because "FXO" port doesn't necessarily mean "it works with asterisk". |
| 00:55.00 | jsharp | FXO simply means "can take a phone line in". |
| 00:55.10 | astoria | jsharp: figures.. lol thanks |
| 00:55.17 | ManxPower | maco: Things that expect to RECEIVE ring voltage and dialtone (FXO) plug into devices that expect to PROVIDE ring voltage and dialtone (FXS) |
| 00:55.21 | scubasteve | I've got a 7960 here at home behind a NAT and Asterisk at work. Trying to get siproxd running on my home linux firewall.. |
| 00:55.38 | Moc | scubasteve, you dont need that |
| 00:55.40 | ManxPower | scubasteve: Is the asterisk at work behind NAT. |
| 00:55.52 | Moc | well, your * is dirrect on internet ? |
| 00:56.25 | maco | ManxPower that doesn't give me the answer to my question :( |
| 00:56.27 | astoria | aight, i'm out begin the drinking, ya'll have a good night |
| 00:56.27 | scubasteve | ManxPower: No. |
| 00:56.50 | scubasteve | ManxPower: I'm trying to use SIP though.. not confident enough to play MGCP yet. |
| 00:56.58 | maco | so is it possible to plug standart external modem trough COM port to the PC and make telco-voip gateway ? |
| 00:57.01 | jsharp | maco: If you wanted to take a phone line in & have it available by voip, yes...then you need FXO ports. |
| 00:57.05 | jsharp | maco: No. |
| 00:57.10 | Moc | scubasteve, install a asterisk at home ;) |
| 00:57.24 | scubasteve | Moc: IAX back to the office? |
| 00:57.34 | Moc | scubasteve, yes, and SIP to your home asterisk |
| 00:57.47 | maco | jsharp is there any other way to make telco GW than trough dedicated FXO HW ? |
| 00:58.01 | scubasteve | Moc: I know that's possible, but was hoping a proxy would do the trick. |
| 00:58.10 | Moc | maco, PRI/BRI |
| 00:58.11 | flashrom | anyone successfully pulling iax.conf from a db using the realtime engine? |
| 00:58.20 | Moc | scubasteve, make * your proxy |
| 00:58.30 | Moc | or ask help for that proxy problem to the right place |
| 00:58.33 | scubasteve | Moc: I don't follow. |
| 00:58.50 | jsharp | maco: You could use a regular voice/data modem...assuming you're masochistic enough to try to make it work. |
| 00:59.02 | jsharp | But it would have to be a PCI based modem...not through the com port. |
| 00:59.04 | ManxPower | jsharp: And a good programmer too. |
| 00:59.15 | Moc | maco, check for VoIP service for DID.. |
| 00:59.20 | maco | hot my parquette |
| 00:59.26 | maco | s/hot.not |
| 00:59.29 | maco | s/hot/not |
| 00:59.34 | maco | what a typo! :) |
| 00:59.55 | maco | jsharp is this: http://www.voipsupply.com/product_info.php?products_id=186 what we are talking about ? |
| 01:00.09 | ManxPower | God the Polycom phones take a long time after boot to enable their internal HTTP server. |
| 01:00.18 | flashrom | jsharp: your pulling iax.conf from the db arent you? |
| 01:00.23 | scubasteve | Moc: Would a SIP Proxy not do what I'm trying to accomplish? |
| 01:00.44 | jsharp | maco: yes. That will let you take 1 phone line into *. |
| 01:00.49 | ManxPower | scubasteve: Only like 5 people on the planet use an external SIP proxy with Asterisk |
| 01:00.51 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
| 01:00.58 | jsharp | flashrom: sip.conf, not iax.con |
| 01:00.59 | scubasteve | ManxPower: Damn. |
| 01:01.26 | Moc | ManxPower, yes, if you go the the CPU usage monitor, once it booted, it aint really finished to boot |
| 01:01.31 | flashrom | jsharp: whats your table/field setup? var_val,var_name etc? |
| 01:01.43 | scubasteve | ManxPower: Would I need anything open except 5060 and 1024-65535/udp on the firewall at work? |
| 01:01.43 | Moc | it run at 100% until it get down to lower level, then the webpage is active |
| 01:02.18 | jsharp | I have username, secret, mailbox, ipaddress, port, registrationtime, and account code. |
| 01:02.29 | *** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl) |
| 01:03.11 | ManxPower | scubasteve: If asterisk is NOT behind NAT or a firewall then you don't need to do any port forwarding, only nat=yes in sip.conf. |
| 01:03.31 | flashrom | jsharp: hm.. is there a list of required table/fields you found or did u find that in the wiki |
| 01:03.36 | maco | ManxPower what nat=yes actually does ? |
| 01:03.43 | scubasteve | ManxPower: Well, there's a firewall at the office and the NetOps folks have 5060 and high ports open for UDP so I can mess with the phone here. |
| 01:03.51 | ManxPower | maco: Black magic |
| 01:04.02 | ManxPower | scubasteve: But no NAT? |
| 01:04.14 | maco | ManxPower got it ! :) |
| 01:04.27 | scubasteve | ManxPower: At work, I don't believe so. At home, yes. |
| 01:04.31 | jsharp | flashrom: I found it in the wiki under sipfriends |
| 01:04.40 | ManxPower | scubasteve: OK. What SIP client do you have at hime? |
| 01:04.42 | ManxPower | home? |
| 01:04.49 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
| 01:04.58 | scubasteve | ManxPower: I really try to avoid network stuff at the office. It's a god-awful mess... |
| 01:05.21 | scubasteve | ManxPower: Asterisk + TDM stuff is at work, phone is at home. Phone is a Cisco 7960 with SIP code. |
| 01:05.30 | flashrom | jsharp: do you know or have u heard of the iax.conf being pulled from the db? i'm having trouble figuring out what the table/field setup should be |
| 01:05.51 | ManxPower | scubasteve: Good. The Ciscos default to odd numbered ports from 16384 to 32768 UDP |
| 01:05.55 | jsharp | Yes. There's a similar setup for iaxfriends out of a mysql table. |
| 01:06.07 | ManxPower | Now turn off all port forwarding on the home firewall/nat router |
| 01:06.35 | flashrom | jsharp: using res_config right? |
| 01:06.39 | scubasteve | ManxPower: Ok. |
| 01:07.09 | scubasteve | ManxPower: Actually, I don't have any forwarding on the router at home. Just allowing the traffic through, was hoping the proxy would pick it up... |
| 01:07.16 | jsharp | flashrom: Not the one I've seen. The one I saw had chan_iax2 reading directly from mysql. |
| 01:07.27 | jsharp | I'm running older code, though. |
| 01:07.31 | ManxPower | scubasteve: Now point the phone to the ip/name of the asterisk server at work. |
| 01:07.34 | scubasteve | ManxPower: Turned off the proxy and repeatedly verbally assaulted it... |
| 01:07.59 | scubasteve | ManxPower: I had that set up last night that way. I couldn't call out, but could call in. No audio though. |
| 01:08.31 | bkw_ | http://www.mcalesternews.com/articles/2004/10/29/news/top_stories/top01.txt |
| 01:08.36 | bkw_ | see how small minded oklahoma is |
| 01:08.57 | flashrom | jsharp: res_config is supposed to be compatible with any .conf file isnt it |
| 01:08.59 | Nugget | I've only been to oklahoma once and only had feet on the ground for about four hours. |
| 01:10.20 | Nugget | we'd flown to dallas for dinner and decided on a whim to touch down in oklahoma city since none of us had been before. |
| 01:10.30 | ManxPower | scubasteve: If you are getting no or one way audio your network people at the office didn't forward the UDP ports. |
| 01:10.36 | Nugget | I'm in no hurry to go back. |
| 01:10.42 | ManxPower | scubasteve: Can you ping the Asterisk server? |
| 01:11.12 | ManxPower | scubasteve: do you have nat=yes and is the phone registering to asterisk? |
| 01:11.34 | scubasteve | ManxPower: Standby. |
| 01:12.17 | scubasteve | ManxPower: I do now. I think I had it enabled and then turned it off ... last nite. |
| 01:12.35 | ManxPower | not nat=yes could give you one way audio too |
| 01:13.17 | redder86 | I'm not sure why Sam didn't just will his stuff to his partner rather than letting the gov't take care of it. Doesn't everyone already know that if you die without a will that the gov't will not do what you want done with your stuff? |
| 01:13.45 | redder86 | People always assume too much about how things will happen after their passing. |
| 01:14.09 | ManxPower | redder86: After I'm dead I won't care. |
| 01:14.23 | scubasteve | ManxPower: phone2/phone2 (Unspecified) D N 255.255.255.255 0 Unmonitored |
| 01:14.30 | redder86 | ManxPower: well, this Earl guy cared about Sam's stuff |
| 01:15.21 | ManxPower | scubasteve: Your phone is not registering |
| 01:15.22 | redder86 | It really has nothing to do with sexual preference... if you don't plan out stuff for after your death, then you won't have any control over it. |
| 01:15.27 | *** join/#asterisk inv_arp (junya@adsl-10-168-193.mia.bellsouth.net) |
| 01:16.36 | redder86 | There are a ton of heterosexual people who have the same problem... they live together all their lives, never marry, but then when one dies, oops, all the relatives come out of the woodwork to claim what the government says is theirs because the deceased didn't have a will. |
| 01:17.32 | ManxPower | I've always thought a Limited Liability Partnership (LLP) is the way to go. |
| 01:17.39 | redder86 | Then again, maybe asking people in Oklahoma to think ahead is asking too much ;-) |
| 01:18.42 | Nugget | heh |
| 01:19.14 | Nugget | if they could plan effecitvely they wouldn't be living in oklahoma. |
| 01:19.40 | bkw_ | redder86 well doesn't matter if you have a will in oklahoma |
| 01:19.47 | bkw_ | the relatives can challange it |
| 01:19.48 | bkw_ | and win |
| 01:19.50 | bkw_ | thats it |
| 01:20.09 | bkw_ | hell oklahoma tried to slip the sodomy laws back on the books |
| 01:20.15 | bkw_ | but they got bitch smacked |
| 01:20.18 | redder86 | bkw_: then they ought to not own stuff themselves. They can have a private corporation own their stuff. |
| 01:20.28 | bkw_ | redder86 yep :) |
| 01:20.53 | ManxPower | bkw_: That's why I think being an LLC or LLP is the best way. |
| 01:20.59 | redder86 | bkw_: they just didn't plan ahead |
| 01:21.04 | ManxPower | Relatives to not get to context business assets |
| 01:21.06 | bkw_ | most people dont think about that |
| 01:21.13 | ManxPower | contest, that is |
| 01:21.38 | bkw_ | I just have the double tax |
| 01:21.43 | bkw_ | double heatlh care |
| 01:21.53 | bkw_ | its just stupid |
| 01:21.55 | ManxPower | bkw_: There are ways around that too. |
| 01:21.59 | bkw_ | ya |
| 01:22.07 | bkw_ | I have been with Greg for 9 years |
| 01:22.32 | bkw_ | thats more than most heterosexual couples even accomplish in oklahoma |
| 01:23.10 | scubasteve | ManxPower: Ha! This silly Cisco phone had a "NAT ENABLED? NO" .. changed to "Yes" and I'm up and running. |
| 01:24.06 | ManxPower | scubasteve: DON'T DO THAT!!!!!!!!!!!!!!!!!!!!! |
| 01:24.19 | scubasteve | ManxPower: Arg. But it worked...! |
| 01:24.24 | ManxPower | scubasteve: Asterisk's nat=yes removes the need for setting NAT on the phone. |
| 01:24.43 | *** join/#asterisk dan2 (dan@beta3.user) |
| 01:24.43 | scubasteve | ManxPower: I've got nat=yes in the * |
| 01:24.47 | ManxPower | scubasteve: Your phone calls are not matching the sip.conf entry as seen by the fact that it's not registering. |
| 01:25.22 | scubasteve | ManxPower: I take it then if I remove the defaultip address in sip.conf it will go to hell in handbasket then? |
| 01:25.47 | ManxPower | scubasteve: Gads yes. Your phone is not registering. This will make your life a living hell until you fix that. |
| 01:26.12 | ManxPower | scubasteve: Not even 1 of my SIP clients have NAT enabled on the phone. |
| 01:26.32 | scubasteve | ManxPower: Ok, turning it off.... |
| 01:26.34 | *** join/#asterisk usam (~usam@203.147.59.54) |
| 01:26.44 | ManxPower | scubasteve: Before you proceed get the phone registering |
| 01:27.02 | ManxPower | and paste the [blah] entry in sip.conf for the phone to www.pastebin.com |
| 01:27.38 | scubasteve | ManxPower: Standby.. |
| 01:28.30 | ManxPower | scubasteve: There are a zillion ways you can set things up, I'm just telling you the simpliest way. |
| 01:28.40 | usam | is it so that the X101P cannot detect line reversal? It is strange that asterisk announce that the system suuport line reversal, but yesterday i read it from the bugs mailing list saying that the X101P doesnt support this... can some1 confirm? |
| 01:28.54 | scubasteve | ManxPower: Appreciated! Pasted the config for ya. |
| 01:29.04 | ManxPower | scubasteve: URL? |
| 01:29.11 | scubasteve | http://www.pastebin.com/115078 |
| 01:29.27 | ManxPower | usam: The X100P supports disconnect indication/supervision for normal telco PSTN lines in the USA |
| 01:30.16 | ManxPower | http://www.pastebin.com/115079 notice default IP is gone and I fixed your callerid line to be correct. |
| 01:30.26 | usam | ManxPower: and what about X101P ? What I know is that X100P is based on motorola, nad x101p is based on a amibient chipset ... |
| 01:30.28 | ManxPower | Now, who here knows how to make a Cisco phone REGISTER? |
| 01:30.46 | ManxPower | usam: The X100P and X101P are the same card as far as these things are concerned. |
| 01:31.56 | scubasteve | ManxPower: Cool. I'm looking at the phone and I see "Register with proxy: NO" ... Is that right? |
| 01:31.59 | LTG30 | Which model? |
| 01:32.01 | usam | ManxPower: ... OK ... then i will have the busydetection on, and tweak the zonedata to detect dialtone as busytone. |
| 01:32.04 | scubasteve | 7960. |
| 01:32.15 | ManxPower | usam: Are you in the USA? |
| 01:32.19 | Alric | register with proxy, set to yes |
| 01:32.21 | LTG30 | You need a cnf file for SIP mode |
| 01:32.32 | scubasteve | Dangit... Tried to stop * and it's freakin out... "Ouch ... error wile writing audio data: Broken pipe" |
| 01:32.38 | ManxPower | LTG30: It can be set via the interface on the phone, can't it? |
| 01:32.48 | Alric | ManxPower: Yes. |
| 01:32.50 | LTG30 | I have 2 7940 ans 2 7920 running on mine. |
| 01:32.58 | usam | ManxPower: nope. I got a gsm line interface, it sends out line reversal, but my x101p cannot detect it |
| 01:33.00 | ManxPower | Alric: Would you tell scubasteve how to do it. |
| 01:33.10 | Alric | .cnf file is only needed for config from TFTP, the phone can be provisioned completely from the Config button though. |
| 01:33.21 | ManxPower | usam: Have you confirmed it's sending reversal? |
| 01:33.25 | Alric | Scubasteve: Turn that "Register with Proxy" to YES. |
| 01:33.27 | LTG30 | Yes, or you can control it through the cnf file from a TFTP server |
| 01:33.40 | Alric | I just said that... |
| 01:33.44 | scubasteve | mpg123's stuff were goin nuts. |
| 01:33.49 | ManxPower | LTG30: He's just configureing it via the phone at this point |
| 01:33.57 | scubasteve | Altric, cool... |
| 01:34.23 | usam | ManxPower: havent done with a voltimeter yet, but the manufacture confirm that they have line reversal |
| 01:34.45 | LTG30 | I think I have a sample file for the 7960 |
| 01:34.50 | ManxPower | usam: In the USA disconnect is indicated by removal of voltage from the line for some x fractions of a second |
| 01:35.15 | ManxPower | scubasteve: Listen to Alric |
| 01:35.17 | scubasteve | Altric + ManxPower: Up and running! |
| 01:35.25 | Alric | Its working? :) |
| 01:35.33 | ManxPower | scubasteve: sip show peers is showing the external IP of the phone? |
| 01:35.43 | scubasteve | phone2/phone2 24.88.251.252 D N 255.255.255.255 5060 Unmonitored |
| 01:35.45 | scubasteve | Yep! |
| 01:35.48 | ManxPower | Cool! |
| 01:35.49 | Alric | Whee. |
| 01:35.55 | scubasteve | Yippie skippy. |
| 01:36.05 | flashrom | anyone loading iax.conf from db using res_config? i have a question about the table/field setup |
| 01:36.05 | usam | ManxPower: ic. I will use a voltimeter after i have cured my hangover |
| 01:36.10 | scubasteve | Now I'm going to hop into the tftp file and try to approximate what we've done here... |
| 01:36.19 | scubasteve | usam: Ouch. |
| 01:36.23 | ManxPower | scubasteve: If you lose connectivity with your phone after X mins, then you need to tell the phone to register every 60 seconds to keep the NAT translations open. |
| 01:36.58 | scubasteve | ManxPower: I think I saw something in there about that in the config. I'll set that. |
| 01:37.01 | fearnor- | ok. retarded question: |
| 01:37.15 | flashrom | anyone see this before |
| 01:37.19 | fearnor- | zhone channel bank+asterisk. fxs lines. patch panel. |
| 01:37.19 | flashrom | Oct 30 02:49:49 WARNING[-150740864]: pbx.c:3103 ast_merge_contexts_and_delete: Requested contexts didn't get merged |
| 01:37.23 | Alric | ManxPower: I can't even get my polycom to register correctly :) |
| 01:37.31 | fearnor- | cannot send a fax even between phones on same channel bank |
| 01:37.34 | scubasteve | ManxPower: We've got a bunch of old Nortel PolyCom speakerphone lookin' things at work.. non IP.. but I suspect they're Poly because they look so similar. Good stuff. |
| 01:37.43 | usam | scubasteve: remember 2 years ago, i had a hangover and start dealing with electricity, ... got 220V for about 5 sec ... not a good feeling to remember .. |
| 01:37.44 | fearnor- | line sounds *clear* |
| 01:37.58 | *** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net) |
| 01:38.01 | ManxPower | scubasteve: If you can plug them into an analog telco line they will work at least in some way with Asterisk |
| 01:38.12 | fearnor- | yet faxes won't negotiate. |
| 01:38.21 | scubasteve | usam: Ouch. Yeah, shocks really suck. I've got a defective breaker (I think) and I'm gonna wait until tomorrow when the Wife Is Home so she can witness me kill myself trying to find it... |
| 01:38.46 | LTG30 | ManxPower: Ever work with the H323 gateway for Asterisk? |
| 01:38.48 | fearnor- | anyone ever ran into that? |
| 01:38.55 | ManxPower | LTG30: No need to get vulgar. |
| 01:39.02 | scubasteve | ManxPower: Yep. Unfortunately, I don't have any Digium hardware yet. We've got a Cisco gateway that I've got * talking to (thanks again Nugget!) |
| 01:39.06 | LTG30 | Not trying to.... |
| 01:39.17 | Alric | LTG30: He's talking about using the "H323" word :) |
| 01:39.26 | usam | scubasteve: VIC cards? |
| 01:39.28 | scubasteve | Digium probably didn't expect such an interest.. hopefully they can ramp up the production... |
| 01:39.55 | ManxPower | One of these days I gotta sell my DSP for 1750 and my 2xFXO and 2xFXS cards for Cisco |
| 01:40.00 | flashrom | Oct 30 02:49:49 WARNING[-150740864]: pbx.c:3103 ast_merge_contexts_and_delete: Requested contexts didn't get merged |
| 01:40.01 | scubasteve | usam: No clue. I've never seen the Cisco gear.. it's in a different facility. I got an IP address though, that's about it. |
| 01:40.07 | fearnor- | C.C.D. Cogent Communications Deutschland GmbH fax : +49-69-299896-40 |
| 01:40.08 | fearnor- | err |
| 01:40.14 | LTG30 | Not to sound dumb but why is h*** so bad? I know it is old but? |
| 01:40.29 | scubasteve | usam: I think it was horribly expensive (10 or 20k) and is 1u. |
| 01:40.39 | usam | scubasteve: ic |
| 01:40.55 | scubasteve | usam: And I've got the boss all over me about getting asterisk up and running so we can ditch the Cisco. |
| 01:41.19 | Alric | Gotta love rushed conversions... |
| 01:41.30 | fearnor- | assterisssk |
| 01:41.35 | scubasteve | usam: And what's really interesting... is the company I work for used to be a division of Nortel. They spun it off a few years back... |
| 01:42.57 | *** join/#asterisk Chuji (Chuji@pcp09929633pcs.tulipgrove.tn.nash.comcast.net) |
| 01:43.35 | scubasteve | ManxPower: I signed up for a stanaphone account earlier and keep getting "noisy feedback tells" and it won't seem to register. Any advice? |
| 01:43.49 | scubasteve | stanaphone/5166 216.128.82.18 N 255.255.255.255 5060 Unmonitored |
| 01:44.24 | *** join/#asterisk adamb18 (~adam@69.17.96.176) |
| 01:44.52 | scubasteve | I kept reading how horrific SIP was with NAT... I just assumed it would take a proxy to make it work... Why is this working with so little fuss?? |
| 01:46.11 | adamb18 | hi, i've been setting up my asterisk box all day, i've got everything working now except dialing out from my cisco 7960, i've routed incoming calls.. can someone just, point me in the right direction, please? :-) |
| 01:46.46 | Moc | adamb18, well your phone register É |
| 01:46.54 | scubasteve | Can't get away from these Cisco phones ;-) |
| 01:46.59 | adamb18 | it's registered |
| 01:47.15 | Moc | adamb18, check your dialplan |
| 01:47.16 | LTG30 | Are you running SIP or SCCP |
| 01:47.21 | adamb18 | i dont wanna paste, but it's listed with sip show peers |
| 01:47.25 | Moc | damn can't compile head !! |
| 01:48.01 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 01:48.01 | adamb18 | yeah, thats what i dont understand, where in the dial plan does picking up my handset initiate? |
| 01:48.01 | adamb18 | does that make sens? i have an outgoing context setup for my voicepulse line |
| 01:48.08 | adamb18 | but, where do i _BEGIN_ to tell it, when my extension picks up.. do..... .... |
| 01:48.39 | adamb18 | im just missing this tiny piece of understanding, heh |
| 01:48.47 | *** join/#asterisk postel (~canonical@host81-152-232-90.range81-152.btcentralplus.com) |
| 01:49.32 | damin | adamb18: What does yout dialplan.xml file look like? |
| 01:49.36 | damin | Here's mine.. |
| 01:49.36 | damin | <DIALTEMPLATE> |
| 01:49.37 | damin | <PROTECTED> |
| 01:49.37 | damin | </DIALTEMPLATE> |
| 01:49.46 | adamb18 | ohhhh |
| 01:49.49 | adamb18 | THAT dialplan! |
| 01:49.59 | damin | Then, you need a context in extensions.conf that looks kinda like: |
| 01:50.02 | scubasteve | ManxPower: Ruh roh.. I called myself on the house line from the Cisco phone and can't hear anything on either... |
| 01:50.42 | damin | ; 10 Digit |
| 01:50.42 | damin | exten => _NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN}) |
| 01:50.42 | damin | exten => _NXXNXXXXXX,2,Congestion |
| 01:50.42 | damin | ; 1+ 10 Digit |
| 01:50.42 | damin | exten => _1NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) |
| 01:50.44 | damin | exten => _1NXXNXXXXXX,2,Congestion |
| 01:51.05 | postel | hi ppl, after 2.5 hours and something like 54374354745 possible confings on the access lists on the cisco side and asterisk on the nat side i *think* it works, i tested the echo test (works) tested the conference number (works), could please somebody give me a direct call thru FWD and we're all done |
| 01:51.23 | damin | Where ${TRUNK} is set to the Zap device or other channel device that you are sending calls too.. |
| 01:51.40 | *** join/#asterisk denon (denon@vast.decay.org) |
| 01:51.40 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:51.44 | adamb18 | damin: yeah, gotcha, or my voicepulse login? |
| 01:52.13 | damin | adamb18: Yep.. |
| 01:52.32 | adamb18 | does it matter the name of the context for that? |
| 01:52.40 | postel | nobody? just a simple call, you dont even have to say hello :/ |
| 01:52.46 | adamb18 | mines just [outgoing] but i dont have that specified in the dialplan.xml or anything.. |
| 01:53.33 | gambolputty | I have compiled * and a box and iax won't laod |
| 01:53.34 | gambolputty | load |
| 01:53.51 | *** join/#asterisk tessier_ (~treed@wsip-68-224-172-77.sd.sd.cox.net) |
| 01:54.07 | LTG30 | postel: what nmber do you want me to cal at FWD? |
| 01:54.13 | damin | adamb18: You define the context in sip.conf, and then you create it in extensions.conf. "context=outgoing" |
| 01:54.20 | postel | LTG30 274818, thanks |
| 01:55.07 | ChulJin | LTG30: I've got <blasphemy>H.323</blasphemy> working fine with *...calls to and from my office's pricey Sony videophone (audio only though) perfectly...Netmeeting a bit less success, but only because of a stingy firewall |
| 01:55.10 | postel | LTG30 i can see the call in the console chris but i get Failed to authenticate user |
| 01:55.16 | pointer-gaim | anyone care to recommend a specific win32 sip client? |
| 01:55.25 | ChulJin | x-lite, hands-down. |
| 01:55.27 | LTG30 | Yea, timed out on me. |
| 01:55.34 | pointer-gaim | ChulJin: hmm...okies |
| 01:55.45 | adamb18 | err... i have my context=incoming in sip.conf, which is my menu that handles all my incoming calls.. is there two menus? if not how do i mix handling incomings with outgoings? :-/ |
| 01:56.02 | Juggie | dont mix |
| 01:56.07 | Juggie | causes problems |
| 01:56.20 | damin | And hangovers.. |
| 01:56.51 | LTG30 | ChulJin: I get a RTP_UDP error with mine. Not sure if it is a setup problem or something not up to dat. |
| 01:57.09 | postel | LTG30 any idea what causes the "Failed to authenticate user "C.. G.." i see on the console? |
| 01:57.44 | adamb18 | Juggie: is that to me? |
| 01:57.47 | Juggie | yes |
| 01:57.51 | LTG30 | How are you connecting with FWD IAX2 or SIP? |
| 01:57.53 | damin | Put all the outbound crap in default. |
| 01:57.55 | Juggie | dont mix your incomming and outgoing context's |
| 01:58.00 | postel | LTG30 sip |
| 01:58.04 | adamb18 | oh |
| 01:58.15 | damin | adamb18: And totally clean out all the crap in your extensions.conf file. |
| 01:58.37 | LTG30 | Let me look at something. |
| 01:58.39 | david | I'm having some issues with asterisk not reading an outgoing spool file correctly |
| 01:58.40 | damin | adamb18: I start with a blank slate, and then cut/paste only the crap I need into it. Makes it 1,000 times easier to deal with.. |
| 01:58.40 | david | Oct 29 21:57:58 WARNING[294930]: pbx_spool.c:307 scan_service: Unable to open /var/spool/asterisk/outgoing/1978.voice.out: Permission denied, deleting |
| 01:58.40 | david | Oct 29 21:57:58 WARNING[294930]: pbx_spool.c:349 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/1978.voice.out' |
| 01:58.45 | adamb18 | i dont have any crap, i downloaded the sample extensions.conf from voicepulse and built from there |
| 01:58.49 | david | anyone have a clue what the heck that means? |
| 01:58.54 | adamb18 | well theres a little crap, but not much :-) |
| 01:59.17 | damin | david: Looks wacked.. |
| 01:59.28 | adamb18 | i'ev got the general, globals, incoming (which calls:) main, and outgoing |
| 02:00.04 | damin | What context does the CLI say your SIP phone is calling when you dial? |
| 02:00.12 | adamb18 | incoming sets musiconhold and ahnds off to main which says some stuff via festical and fwds to my handset exten |
| 02:00.20 | david | damin, yep, it does - Problem is, the file is 666, owned by the 'asterisk' use |
| 02:00.20 | adamb18 | oh, what command to monitor thaT? |
| 02:00.22 | LTG30 | postel: do you have a secret setup for it in the SIP.conf file? |
| 02:00.27 | david | user |
| 02:00.32 | adamb18 | i know no cli yet :-) |
| 02:00.35 | *** join/#asterisk freestyler_netwo (~icechat5@S0106000f6630d841.ed.shawcable.net) |
| 02:01.01 | postel | LTG30 i got secret setup on the cisco ATA definition |
| 02:01.23 | Silik0n | anyone tried to build * on obsd3.6? |
| 02:01.26 | LTG30 | Can you disable it for the moment. |
| 02:01.29 | adamb18 | user and show user do nothing :-/ |
| 02:01.46 | postel | LTG30 sure |
| 02:01.50 | postel | a sec |
| 02:02.17 | damin | david: Your running asterisk as suid "asterisk"? |
| 02:02.28 | flashrom | ok can someone take a look at this http://www.pastebin.com/115083 = i've been workingon this for 4 hours and I cant figure out what is going on (this is an asterisk log) |
| 02:02.38 | david | damin, nfc - let me look |
| 02:03.04 | david | damin, yep |
| 02:03.34 | flashrom | I have the databases setup but its not pulling the info |
| 02:03.51 | damin | david: I'd strace it... |
| 02:03.51 | damin | david: I run as root.. |
| 02:04.23 | postel | LTG30 ready |
| 02:04.57 | postel | aaaaa, now it does Execute ringing |
| 02:05.06 | LTG30 | Timed out |
| 02:05.18 | postel | nice, it comes throu, it just needs a rule on the context |
| 02:05.27 | postel | LTG30 dont worry, i know what to do now |
| 02:05.29 | postel | thanks |
| 02:05.42 | david | damin, I was wondering if he's even supposed to work at all |
| 02:05.43 | LTG30 | Let me know if you need me to call again. |
| 02:05.59 | postel | LTG30 thanks :) |
| 02:06.00 | adamb18 | damin: the context is incoming |
| 02:06.32 | adamb18 | so should i just put the contents of my [outgoing] at the top of my incoming? |
| 02:06.52 | *** join/#asterisk mindCrime (~mindCrime@rrcs-24-106-188-2.se.biz.rr.com) |
| 02:07.44 | flashrom | anyone - I've looked through every config and every mailing list |
| 02:08.17 | *** join/#asterisk Ippimp (Ippimp@68.113.223.18.ts46v-10.otnd2.ftwrth.tx.charter.com) |
| 02:08.34 | Ippimp | wuz up everybody |
| 02:09.29 | Ippimp | Has anyone ever used DID's with asterisk |
| 02:09.44 | LTG30 | ChulJin: What version of Lynux are you running * on with h(word) gateway? |
| 02:10.28 | ChulJin | rh9 |
| 02:10.50 | damin | adamb18: You can try that to get started, but you really should just change the sip.conf to say "ougoing" and reload. |
| 02:11.00 | LTG30 | What version on *? I am trying to find what is wrong with mine. |
| 02:11.10 | adamb18 | damin: but then what will happen when people call me? |
| 02:11.34 | ChulJin | HEAD |
| 02:11.39 | david | damin, hrm, dialing works when I run it as r00t |
| 02:11.41 | ChulJin | well, about 1-day-old HEAD |
| 02:11.48 | damin | adamb18: Then, you can use the incoming context on your specific peers. Like VoicePulse's entry can have "context=voicepulse-incoming" and Free World Dial cna have "context=fwd-incoming" |
| 02:12.18 | flashrom | hey has anyone looked at my pastebin |
| 02:12.20 | LTG30 | Mine is 4 weeks old. |
| 02:12.40 | damin | adamb18: Each peer, user or friend can be assigned it's own context.. get it? |
| 02:12.42 | adamb18 | oooh i think i got it |
| 02:13.01 | adamb18 | yeah |
| 02:13.03 | damin | adamb18: If you get it, you owe me a beer. |
| 02:13.07 | adamb18 | i missed the assignment in sip.conf |
| 02:13.10 | adamb18 | by accident |
| 02:13.27 | damin | david: Are you still running that silly .deb for asterisk? |
| 02:13.37 | LTG30 | Could you email me a copy of the h***.conf file and entries in the extention.conf file so I can see if I have it setup correctly? |
| 02:13.38 | Wangster | Do any of the codecs that come with astersik offer any compression? |
| 02:13.51 | flashrom | ok can someone take a look at this http://www.pastebin.com/115083 = i've been workingon this for 4 hours and I cant figure out what is going on (this is an asterisk log) trying to pull configs using the realtime engine- everything as far as I can see is setup right - it connects to the database but doesnt pull the info |
| 02:13.53 | david | damin, of course not - I'm using CVS with the dundi patches :-) |
| 02:13.54 | adamb18 | i got it!! |
| 02:13.58 | adamb18 | you're DAMAN :-) |
| 02:19.58 | damin | adamb18: No, I'm DAMIN! |
| 02:24.17 | LTG30 | ChulJin: Do you have a sound card in your * server? |
| 02:24.20 | tzanger | damin... we be damin... |
| 02:24.53 | adamb18 | anyone using a wrt54g (modded?) |
| 02:25.19 | tzanger | I have a wrt54g with sveasoft fw but no hardware mods |
| 02:25.26 | adamb18 | yeah, me too |
| 02:25.30 | adamb18 | are you using QoS? |
| 02:25.32 | adamb18 | alchemy firmware? |
| 02:25.34 | tzanger | yep |
| 02:25.43 | tzanger | works reasonably well too |
| 02:25.46 | adamb18 | hmm |
| 02:25.53 | adamb18 | im getting crappy connection and im on a decent dsl line |
| 02:25.57 | ChulJin | LTG: no I don't |
| 02:26.00 | adamb18 | 6mbit down 768kup |
| 02:26.08 | tzanger | 1040 in each dir |
| 02:27.21 | LTG30 | ChulJin: did you ever get a RTP Received packet with bad UDP checksum? |
| 02:27.51 | adamb18 | hmm.. how did you set it? did you just QoS the protocol or by port/ip? |
| 02:28.14 | tzanger | adamb18: set up iax2 service and then set that to high priority |
| 02:28.57 | adamb18 | IAX2 isnt listed in my services dropdown..where can i update the list |
| 02:29.04 | tzanger | that was the trick |
| 02:29.23 | tzanger | you do it in the lockdown screen (can't think of hte name offhand) -- that is where yo uedit the service list |
| 02:29.29 | znoG | how does g729 licensing work? |
| 02:29.35 | znoG | is it $10/year or something? |
| 02:29.44 | tzanger | znoG: $10/concurrent transcode |
| 02:29.51 | Rez | anybody with a PRI or T1 and an analog card and v.90 or v.92 modem connected to said analog card around? |
| 02:29.53 | adamb18 | ahh okay, nice, thanks :-) |
| 02:29.55 | Moc | i dont think it per year, it 10$ per channels usage |
| 02:30.05 | znoG | tzanger: whats the concurrent transcode mean? |
| 02:30.16 | znoG | per channels hey... so every call is one channel? |
| 02:30.19 | adamb18 | hmm.. lockdown screen.. |
| 02:30.20 | Rez | znoG: meaning if you want to do 4 at a time, pay $40 |
| 02:30.20 | tzanger | znoG: that $10 gets you g729<-->anything_else |
| 02:30.30 | znoG | 4 calls at a time right? |
| 02:30.39 | tzanger | g729<-->g729 doesn't use up a channel unless asterisk has to listen to the audio |
| 02:30.45 | tzanger | er use up a license, not a hcannel |
| 02:30.48 | Moc | znoG, yes, and if you put on hold, and want MOH, it 1 more channel |
| 02:30.49 | Rez | znoG: dict concurrent |
| 02:31.12 | znoG | eek |
| 02:31.15 | LTG30 | But you will need a second license if the call goes to voicemail. |
| 02:31.20 | Moc | also, 7960 doesn't support conference on g729 |
| 02:31.26 | Moc | yes voicemail too |
| 02:31.26 | znoG | thats okay, i don't have a 7960 |
| 02:31.40 | znoG | so what would happen if a call goes to voicemail and i only have one license? |
| 02:31.44 | znoG | will it drop back to some other codec? |
| 02:31.47 | Moc | znoG, call drop |
| 02:31.58 | Rez | so, nobody with a digital connection and v.90+ modem |
| 02:32.35 | Juggie | i use a 7960 but i have it set to g711 |
| 02:32.58 | znoG | ahh.. and for my last question, i hear g729 is great for long, long distance VoIP connections.. i'm currently using GSM for asterisk<->asterisk - will there be that much noticeable difference? |
| 02:33.17 | Juggie | depends on your latency |
| 02:33.26 | znoG | its about 400ms to the other asterisk server |
| 02:33.33 | Juggie | hows the bandwidth? |
| 02:33.41 | tzanger | I use GSM for everything... have not tried g729 |
| 02:33.42 | znoG | both have 128k upload |
| 02:33.58 | Juggie | if u want to avoid licensing use gsm |
| 02:34.01 | Juggie | thats your best bet |
| 02:34.06 | *** join/#asterisk |^Angel^| (~funschrip@port336.ds1-alb.adsl.cybercity.dk) |
| 02:34.08 | znoG | the call quality is good as it is, just wondering how it will help |
| 02:34.15 | adamb18 | tzanger: this is a screen on the admin page? i just looked through all of em i dotn see a services edit list. |
| 02:34.17 | znoG | yeah, so far gsm is working okay.. |
| 02:34.18 | tzanger | it will help your bandwidth use |
| 02:34.20 | LTG30 | will the Cisco 7940 support GSM? |
| 02:34.24 | Juggie | no |
| 02:34.28 | *** join/#asterisk rustyb (~rustyb@68.235.250.116) |
| 02:34.36 | Juggie | cisco only does g711 g729 and maybe one more |
| 02:34.44 | tzanger | it's in the security page where you can say what services/ips get access to the WAN by date/time |
| 02:34.45 | LTG30 | I thought so. |
| 02:34.47 | Juggie | on an internal network use g711 |
| 02:34.52 | Juggie | you really wont get better quality |
| 02:34.57 | scubasteve | Darn, ManxPower is gone. Just realized the 7960 still isn't right - I get no audio when connecting to PSTN. Meetme seems to work though..:( |
| 02:34.57 | Juggie | with anything else |
| 02:35.11 | LTG30 | I want to use the G729 for long range. |
| 02:35.15 | znoG | i use g711 from client to asterisk, and gsm from asterisk to far away asterisk, and g711 again from asterisk to client on the other end |
| 02:35.19 | ManxPower | scubasteve: canreinvite=no in the phone entry in sip.conf |
| 02:35.20 | mishehu | scubasteve: using sccp or sip? |
| 02:35.38 | Juggie | znog, more conversions = more latency |
| 02:35.38 | scubasteve | ManxPower: You the man. |
| 02:35.48 | mishehu | ManxPower: sip firmware of 6.x and higher CAN support canreinvite=yes |
| 02:35.51 | scubasteve | ManxPower: I will read up on all of this stuff you've taught tonight, I promise. |
| 02:36.01 | znoG | Juggie: true, i suppose if i used gsm all around it would help, but the call quality wouldn't be as good |
| 02:36.22 | ManxPower | mishehu: He's having enough trouble without trying to optimize things |
| 02:36.25 | znoG | worth a shot using gsm all around |
| 02:36.48 | Juggie | znog, it would be the same |
| 02:36.54 | Juggie | possibly better |
| 02:37.02 | LTG30 | postel: do you want me to call you at FWD? |
| 02:37.16 | Juggie | u loose all your quality on your g711->gsm conversion |
| 02:37.19 | mishehu | ManxPower: what firmware does he have now? |
| 02:37.20 | scubasteve | ManxPower: Perfect... That did the trick. Gonna go look up that canreinvite thing now! Thanks!!! |
| 02:37.22 | Juggie | when u go gsm->g711 u dont get it back |
| 02:37.25 | Juggie | and it infact may get worse |
| 02:37.42 | mishehu | since he didn't respond to my quesiton. |
| 02:38.07 | scubasteve | mishehu: Trying to find it... |
| 02:38.46 | mishehu | the 7960s can be a pain in the ass the first time you set them up. |
| 02:38.47 | scubasteve | mishehu: Ok.. Found it.. Application Load ID: P0s3-07-2-00 |
| 02:38.58 | mishehu | scubasteve: pretty new... even newer than I have. |
| 02:39.16 | scubasteve | mishehu: Well, I got a stack of IP 110's waiting for me next week. Damn things are crap. |
| 02:39.37 | mishehu | ip110's? |
| 02:39.54 | scubasteve | mishehu: They came with an index card piece of paper that says "Your documentation is online at http://www.ip110.com/" Visit that URL for a good laugh. |
| 02:40.16 | scubasteve | mishehu: Seems to be a SwissTel product, although it's not listed on their website. There's an IP10 listed, but no IP 110. |
| 02:40.39 | mishehu | I just got a blank page... |
| 02:40.56 | scubasteve | mishehu: Trying to enter my PIN from the Cisco phone for MeetMe (presume it would be same for voicemail, but haven't configured it yet)... Doesn't seem to recognize tone. |
| 02:40.59 | ManxPower | Try it with Internet Exploder |
| 02:41.01 | scubasteve | mishehu: Exactly. |
| 02:41.13 | scubasteve | ManxPower: It's a directory index of . |
| 02:41.14 | adamb18 | okay my context is now set to outgoing but dialing seems to timeout then end me up with a fast busy signal, is there a way to monitor the wholl process by peer from the cli? |
| 02:41.26 | znoG | Juggie: is there another name GSM goes by? my sipura only supports g711u/a and a bunch of g726 codecs, as well as g729 and g723.. that's it. If GSM doesn't go by another name then i guess it doesn't support it. |
| 02:41.34 | mishehu | ManxPower: well, I get a link to "parent directory", but that's a blank page for all intents and purposes |
| 02:41.45 | LTG30 | Postel |
| 02:41.50 | mishehu | scubasteve: does voicemail recognize your tones? |
| 02:42.01 | scubasteve | mishehu: Don't have voicemail configured yet. |
| 02:42.11 | mishehu | hmm... |
| 02:42.40 | scubasteve | mishehu: It seems to recognize # ... but nothing else. |
| 02:42.43 | znoG | Juggie: never mind, it doesn't support GSM |
| 02:43.02 | mishehu | scubasteve: what dtmfmode do you have set for it in sip.conf? |
| 02:43.08 | scubasteve | mishehu: With SIP debug on, when I call in from PSTN it seems to notice each keypress... no such luck on the Cisco though. |
| 02:43.26 | scubasteve | mishehu: Tried 'em all.. ; Choices are inband, rfc2833, or info |
| 02:43.36 | mishehu | I use rfc2833 |
| 02:43.53 | scubasteve | mishehu: Interesting. |
| 02:44.22 | scubasteve | mishehu: Want to try my MeetMe? |
| 02:44.35 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 02:44.53 | mishehu | scubasteve: what settings in your SIPDefault.cnf for dtmf... I have "dtmf_inband: 1", "dtmf_outofband: avt", "dtmf_db_level: 3" |
| 02:44.54 | znoG | Juggie: g729 license would be great because a) my voip provider supports it and b) my sipura spa-2000 supports it, meaning no transcoding... |
| 02:45.02 | scubasteve | mishehu: Standby |
| 02:45.21 | znoG | Juggie: just gotta figure out how many i'll need.. only one incoming call at a time from my provider, for now and voicemail. And I do put them on hold occasionally, so thats another license. |
| 02:46.11 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 02:46.22 | scubasteve | mishehu: If sipdefault.cnf is supposed to be in /etc/asterisk, I don't have it.. |
| 02:46.26 | LTG30 | znoG: two should work. |
| 02:47.29 | znoG | LTG30: one for voicemail, one for on hold, and one for the incoming call - right? actually, i wouldn't be putting them on hold and using voicemail at the same time so they can share a license. Am I on the right track to work out how many licenses i'll need? |
| 02:47.45 | scubasteve | mishehu: A grep of dtmf in /etc/asterisk returns hits in mgcp.conf (I use SIP for now) modem.conf and zapdata.conf. The entries listed in sip.conf are the phone-level directives except for relaxdtmf which is yes. |
| 02:47.54 | LTG30 | znoG: Yep. |
| 02:48.10 | znoG | LTG30: great, thanks. |
| 02:48.26 | LTG30 | znoG: I figure 2 for every active user. |
| 02:48.39 | znoG | LTG30: and if i want to receive two concurrent calls and possibly put one of them on hold, thats 3 licenses - right? |
| 02:48.55 | LTG30 | znoG: yep |
| 02:48.59 | mishehu | scubasteve: SIPDefault.cnf is where your firmware for the 7960 is located (ie the tftp dir) |
| 02:49.05 | znoG | LTG30: one for the call plus one incase they put them on hold - thats how you figure it out? |
| 02:49.28 | scubasteve | mishehu: Ah. |
| 02:49.39 | LTG30 | znoG: correct. I have 4 incase I have 2 calls |
| 02:49.41 | mishehu | scubasteve: the phone queries it for basic configurations. |
| 02:50.05 | scubasteve | mishehu: All I have in the tftp is the SIP000xxxx.cnf files for the 3 phones we have. |
| 02:50.25 | LTG30 | znoG: you can add more later if needed. |
| 02:50.40 | scubasteve | mishehu: Which seems to have those directives in it.. |
| 02:51.02 | scubasteve | mishehu: dtmf_inband: 1 outofband none db_level is missing. |
| 02:51.41 | scubasteve | mishehu: Made my config look like yours, and will go back and set the new stuff I did manually in this file as well... |
| 02:52.42 | znoG | LTG30: now, hang on... it says here i dont need a license if im using g729 in pass thru mode. If my sipura supports g729 and my asterisk server too (of course), and my provider uses g729.. thats pass-thru, is it not? |
| 02:52.52 | scubasteve | mishehu: I don't see a Register With Proxy in the .cfg .. I need to turn that and NAT on. |
| 02:53.32 | LTG30 | znoG: Yes, but they can not go to voicemail. |
| 02:53.54 | znoG | LTG30: thats right, so i need one license. Oh, if a call comes in via Zap, and it rings my Sipura (g729), thats transcoding right? |
| 02:54.16 | LTG30 | znoG: I beleave so. |
| 02:54.41 | znoG | LTG30: ok, so if i use gsm for voicemail, and gsm for Zap calls, then i don't need a g729 license - right? :) |
| 02:54.52 | znoG | LTG30: i'm not sure whether an incoming call that goes to voicemail can be changed on the fly to a different codec |
| 02:55.02 | LTG30 | znoG: |
| 02:55.10 | LTG30 | i don't think it can. |
| 02:55.20 | znoG | yeah, i figured. otherwise everyone would do the same |
| 02:55.45 | LTG30 | get 2 now an add more later. |
| 02:55.59 | znoG | if its only $20 once off, why not. |
| 02:56.13 | LTG30 | one time payment. |
| 02:56.15 | znoG | are they handed over by Digium immediately or you have to wait? |
| 02:56.31 | LTG30 | You have to wait. |
| 02:57.04 | LTG30 | mine came the same business day. You would need to wait till Monday. |
| 02:57.19 | znoG | ok, i read' something about the key being attached to your network cards in the Asterisk system.. bummer! |
| 02:57.39 | scubasteve | mishehu: Ok, gonna kick the phone and see if the tftp goodies are right now. |
| 02:58.00 | mishehu | scubasteve: don't kick it, it's too expensive to replace ;-) |
| 02:58.09 | LTG30 | Yea, From what I hear it can not be move to another computer without a call to Digium. |
| 02:58.32 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc10c.dialup.mindspring.com) |
| 02:58.51 | scubasteve | mishehu :) |
| 02:59.09 | Rez | anybody with a PRI or T1 and an analog card and v.90 or v.92 modem connected to said analog card around? |
| 02:59.33 | scubasteve | mishehu: The tftp download ("Requesting Configuration...") seems to take a really long time (minute or more) ... |
| 02:59.49 | scubasteve | mishehu: I'm assuming it's only pulling down a 2k text config file. |
| 03:00.56 | LTG30 | Postel you there? |
| 03:01.22 | znoG | LTG30: and if i read' correctly, you can only change once. I presume thats to avoid people handing the keys out |
| 03:02.26 | LTG30 | Correct, but from what I was told if you call Digium and take to them you can move it more than once. |
| 03:02.45 | znoG | ah, good stuff. :) |
| 03:03.25 | rustyb | Rez: yea but they're not in the same * |
| 03:03.42 | LTG30 | I would give Digium a call Monday and get more information about it. |
| 03:03.48 | scubasteve | mishehu: Well.. it looks like all of my config file changes took, but I still can't seem to enter the PIN for meetme. |
| 03:04.56 | scubasteve | mishehu: Someone else has had this problem.. will see if the advise dispensed helps... http://lists.digium.com/pipermail/asterisk-users/2004-September/063503.html |
| 03:05.01 | Rez | rustyb: I'm looking for a v.90 or v.92 training sequence recording. any way you can make it transport through the other * server and out to a digital dialup modem? |
| 03:06.15 | *** join/#asterisk Odie_flocon (~Odie_floc@S010600095b4c7aaa.lb.shawcable.net) |
| 03:06.27 | wolfson | rez, its not gonna sound normal. v.90 is not gonna train over a compressed connection |
| 03:06.31 | LTG30 | znoG: I think I am going to go. I need to rebuild this box from RH8 to RH9. |
| 03:06.32 | rustyb | i think its not a recording but an interactive negociation between the modems |
| 03:06.56 | Rez | wolfson: I realize this. I was hoping he could keep the interconnect from being compressed. |
| 03:07.16 | ManxPower | scubasteve: Are you using ULAW? |
| 03:07.19 | wolfson | rez, you realise this recording will have no use |
| 03:07.41 | scubasteve | MaxPower: Um... |
| 03:07.44 | rustyb | i think if a v.90 modem "heard" the fsk data it would not sync. its an interactive dialog with the far modem |
| 03:08.02 | wolfson | its incredibly interactive |
| 03:08.05 | rustyb | only the old Bell 202T modems would work like that |
| 03:08.08 | scubasteve | MaxPower: Preferred Codec in the phone is g711alaw. |
| 03:08.18 | Rez | rustyb: v.90's aren't fsk, they're pcm |
| 03:08.30 | wolfson | rez, what do you want to do with the recording? |
| 03:08.40 | ManxPower | scubasteve: And you have disallow=all and allow=ulaw in your [happyphone] entry in sip.conf |
| 03:09.30 | Rez | It would really have to be two recordings, one for each direction on the digital side |
| 03:09.32 | *** join/#asterisk AgiNamu (~zzzs@4.79.150.34) |
| 03:09.39 | wolfson | rez, what do you want to do with the recording? |
| 03:09.42 | AgiNamu | Hey, anyone up for a religious debate over file formats? |
| 03:09.45 | scubasteve | ManxPower: No. |
| 03:10.10 | Rez | I want to see what the training looks like, and possibly the front end of the ppp connection. |
| 03:10.36 | wolfson | its gonna be next to impossible to seperate out each side |
| 03:10.48 | wolfson | and v.92 requires echo supression to be off |
| 03:10.54 | wolfson | it sends a tone to disable it in the switch |
| 03:10.55 | Rez | and yes, I realize that replaying it for another modem won't work, that's not my interest |
| 03:10.59 | Rez | yeah, 2100hz |
| 03:11.17 | jsharp | Not if you use app_monitor. That gives you two files...one for inbound recording, the other for outbound. |
| 03:11.22 | wolfson | you'd be better off reading the v.90 spec |
| 03:11.27 | wolfson | rather than reverse engineeringit |
| 03:11.29 | scubasteve | MaxPower: I put them in, commented out.. until the phone boots and I try the stuff I just read in a thread http://lists.digium.com/pipermail/asterisk-users/2004-September/063503.html |
| 03:11.29 | Rez | got a copy off hand? |
| 03:11.39 | wolfson | let me look |
| 03:13.45 | ManxPower | scubasteve: If the dtmfmode is not rfc2833 on both the phone and Asterisk then DTMF will not work except if you use the ulaw or alaw codecs |
| 03:14.44 | ManxPower | scubasteve: That message only applies if you are using ulaw or alaw codecs. |
| 03:14.52 | scubasteve | ManxPower: Gotcha. |
| 03:16.27 | scubasteve | ManxPower: Ok, did what the thread said.. and set the disallow/allow per your instructions.. still no juice. |
| 03:16.42 | wolfson | http://www.itu.int/rec/recommendation.asp?type=items&lang=e&parent=T-REC-V.92-200011-I |
| 03:18.59 | rustyb | using app_monitor would give you the two independant audio paths. hopefully you would have low transhybrid loss in the modems |
| 03:20.17 | sleepy_one | hi all, does anyone know anything about gafachi? ever used them for VoIP termination? |
| 03:20.28 | Rez | wolfson's correct though, the echo coming back from the digital signal would be on the client side transmission |
| 03:20.37 | bkw_ | jsharp you mean res_monitor :P |
| 03:20.49 | bkw_ | and res_monitor can mux the files if you have sox installed and you pass the m flag |
| 03:20.52 | bkw_ | it will mix on hangup |
| 03:22.01 | Odie_flocon | Hey When Writing AGI scripts in C is there a class that needs to be compiled into the program? |
| 03:22.33 | AgiNamu | Why would you write an AGI script in C?? |
| 03:22.43 | jsharp | Why wouldn't you? |
| 03:22.49 | PatrickDK | why would you write one in perl? |
| 03:23.03 | *** join/#asterisk jdg (~chatzilla@CA03F912.adsl.mana.pf) |
| 03:23.08 | Odie_flocon | because I am writing a Security application to go with my hardware. |
| 03:23.39 | Odie_flocon | and I need to use C to interface with my hardware. |
| 03:23.40 | jsharp | Write them in Intercal. Or Pascal |
| 03:24.17 | Odie_flocon | C is still better. |
| 03:24.51 | sleepy_one | embed perl in your C ;) |
| 03:25.40 | PatrickDK | odie, you don't need anything special |
| 03:25.47 | PatrickDK | it's just normal stdin and stdout |
| 03:25.50 | Chuji | write it in vb |
| 03:25.58 | PatrickDK | vb3 :) |
| 03:26.09 | sleepy_one | basic 1.1 |
| 03:26.17 | bkw_ | no res_perl |
| 03:26.18 | bkw_ | NEXT!!! |
| 03:26.44 | Chuji | biginners all-purpose symbolic instruction code? |
| 03:26.57 | Nugget | LEFT 90 |
| 03:26.58 | Nugget | FORWARD 10 |
| 03:27.04 | Chuji | lol |
| 03:27.18 | *** join/#asterisk ZX81 (~ZX81@222-152-95-57.jetstream.xtra.co.nz) |
| 03:27.23 | PatrickDK | oh no, not logo |
| 03:27.26 | bkw_ | here ya go |
| 03:27.27 | bkw_ | asterisk*CLI> load res_perl.so |
| 03:27.27 | bkw_ | <PROTECTED> |
| 03:27.27 | bkw_ | Oct 29 22:25:55 NOTICE[1458186]: AstAPIBase.c:27 asterisk_log: perl is in the house.Oct 29 22:25:55 NOTICE[1458186]: AstAPIBase.c:27 asterisk_log: Hi! I'm using perl to call ast_log |
| 03:27.28 | bkw_ | Oct 29 22:25:55 NOTICE[1458186]: res_perl.c:1132 _load_module: perl config engine disabled. |
| 03:27.30 | bkw_ | Oct 29 22:25:55 NOTICE[1458186]: res_perl.c:1156 _load_module: Perl CDR Disabled. |
| 03:27.32 | bkw_ | Oct 29 22:25:55 NOTICE[1458186]: res_perl.c:1169 _load_module: Perl Switch Disabled. |
| 03:27.34 | bkw_ | <PROTECTED> |
| 03:27.58 | Chuji | bkw_ : Use pastebin |
| 03:27.59 | Chuji | lol |
| 03:28.01 | Chuji | :P |
| 03:28.13 | bkw_ | no |
| 03:28.35 | bkw_ | this is my fav |
| 03:28.36 | bkw_ | asterisk*CLI> perl labotomy |
| 03:28.36 | bkw_ | OK, One Flew Over The KooKoo's Nest!..... |
| 03:28.49 | sleepy_one | bkw_ is excempt! don't u know that? |
| 03:29.04 | Chuji | bkw_ : Get to documenting that thing so I can convert my agi's |
| 03:29.11 | AgiNamu | um.... no, if you're gonna write in C, why not use the real API? |
| 03:29.40 | ZX81 | Argh!!! No audio when bridged between x100p and tdm400p! |
| 03:29.48 | ZX81 | Fine when I pick it up with IAX |
| 03:29.50 | ZX81 | ?! |
| 03:30.13 | ZX81 | Any ideas? reinstalled asterisk... |
| 03:30.14 | ManxPower | scubasteve: Make a call to voicemail. While the call is happening do a "sip show channels" find the channel for the call (there may be many channels that are registration, etc stuff) then do a "sip show channel <channel ID> and look for the DTMF mode. |
| 03:30.32 | Odie_flocon | explain agiNamu? |
| 03:30.37 | ManxPower | If you need to you can do a "sip debug peer <yourpeername> and look for telephony-events when you do your touchtones. |
| 03:31.03 | AgiNamu | Odie, AGI works by sending strings back and forth |
| 03:31.13 | AgiNamu | so you have a non-typed system |
| 03:31.22 | AgiNamu | with C, you can just compile directly against the functions themselves |
| 03:31.36 | AgiNamu | and you wont find yourselve in a position where you will be limited by agi |
| 03:31.36 | ZX81 | grrr...gonna have to use this channel bank instead and see if that works... |
| 03:31.48 | ZX81 | what am i goin to do with 24 fxs in my house |
| 03:31.49 | ZX81 | lol |
| 03:32.00 | Odie_flocon | cool. |
| 03:32.03 | robl^ | has anyone released res_php yet? |
| 03:32.16 | mikegrb | ZX81: send it to me? |
| 03:32.21 | ManxPower | How about res_erlang |
| 03:32.35 | ManxPower | ERLANG is DESIGNED for telephony stuff |
| 03:32.42 | robl^ | hrmmm |
| 03:32.46 | sleepy_one | res_lisp? |
| 03:32.47 | robl^ | never seen erlang |
| 03:32.51 | AgiNamu | FastAGI... looks as if it still uses strings, doesnt it? |
| 03:33.07 | robl^ | sleepy_one: almost as useful as res_pilot :) |
| 03:33.22 | jsharp | res_fortran |
| 03:33.25 | sleepy_one | I was kidding |
| 03:33.29 | ManxPower | robl^: It's one of those obsecure languages telephony geeks for large projects do stuff in/with. |
| 03:33.46 | AgiNamu | With ast_mono, I'll have a REAL binary protocol for inter-computer communication |
| 03:33.57 | robl^ | ManxPower: hrmm.. I might google and read up on it tonite |
| 03:33.59 | AgiNamu | and the dev cost and configureation will be null |
| 03:34.29 | scubasteve | ManxPower: I am beginning to wonder if the cfg file I've been changing is being read. The phone seems to boot 2 or 3 times and when I look in "Status Messages" I see tftp timeout errors... |
| 03:34.37 | ManxPower | robl^: ERLANG is both the name of a programming language AND a set of formulas for calculating number of trunks required given specific input. |
| 03:34.44 | robl^ | I just now got Asterisk Flash Operator Panel installed and tweaked, works very well |
| 03:34.59 | Odie_flocon | there is an application for Security devices, But I havn't seen any doc's for it anywhere. |
| 03:35.15 | AgiNamu | odie -- whatdo you mean? |
| 03:35.24 | AgiNamu | "application security devices"? |
| 03:35.32 | robl^ | ManxPower: hrmm.. very interesting |
| 03:35.43 | Odie_flocon | I'm not sure if it's what I'm looking for. |
| 03:35.52 | Odie_flocon | I want something to interface with an alarm system. |
| 03:35.55 | Nugget | INTERCAL or bust. |
| 03:36.05 | ManxPower | robl^: I looked at it a little and it seems more of a language for writing PBXs rather than writing config files. |
| 03:36.49 | jsharp | Odie: Do you want to be able to call into * and query the alarm? Or have * call out when the alarm detects a problem? |
| 03:37.14 | ManxPower | Odie_flocon: There is already an app for alarm systems in Asterisk |
| 03:37.14 | scubasteve | ManxPower: DTMF Mode: rfc2833 |
| 03:37.33 | ManxPower | scubasteve: That's Asterisk's DTMF mode, now you need to confirm your phone is set to the same mode. |
| 03:37.39 | Odie_flocon | That's what I was asking about. |
| 03:38.46 | *** join/#asterisk kFuQ (~somedude@c-24-17-173-130.client.comcast.net) |
| 03:39.22 | Odie_flocon | ManxPower, I'm looking for information on that app... |
| 03:41.15 | scubasteve | ManxPower: I don't think any of the dtmf settings I've made in the config are getting pulled by the phone. |
| 03:41.25 | scubasteve | Damn tftp. |
| 03:41.32 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
| 03:41.35 | ManxPower | scubasteve: Try setting it on the phone if you can. |
| 03:41.58 | scubasteve | ManxPower: Yeah, I've got to wait for it to give up the tftp retries.. |
| 03:43.04 | mishehu | bah. |
| 03:43.29 | mishehu | scubasteve: tcpdump/ethereal to find out what it's having a problem with. |
| 03:43.52 | ManxPower | OR just use a tftp client to try to get the file, then figure out why you can't. |
| 03:45.54 | *** join/#asterisk ZX81_AFK (~ZX81@222-152-132-56.jetstream.xtra.co.nz) |
| 03:51.52 | jsharp | blurf |
| 03:54.45 | gambolputty | can a variable and its value be passed from one * box to another? |
| 03:54.56 | ManxPower | gambolputty: no |
| 03:55.02 | gambolputty | why not? |
| 03:55.10 | ManxPower | gambolputty: No support for it. |
| 03:55.22 | ManxPower | A couple of patches to add that feature exist on the bugtracker |
| 03:55.33 | gambolputty | ok |
| 03:57.53 | *** join/#asterisk DubLo7 (~Owner@dsl2.ptskmi.racc2000.gaslightmedia.net) |
| 03:58.32 | *** join/#asterisk ipdeman (~Gary@cpe-maint-port0.netpathway.com) |
| 04:05.46 | *** join/#asterisk flebb (~basin@pc-24-151-28-122.newm2.ct.charter.com) |
| 04:07.56 | redder86 | Is there a way to dial many, many clients/channels/extensions at once other than Dial(this&that&another&yetanother) ? |
| 04:07.59 | fearnor- | yoe |
| 04:08.55 | *** join/#asterisk edguy (~edguy@host-24-225-213-218.patmedia.net) |
| 04:08.59 | flebb | Oct 30 05:21:37 NOTICE[-150257536]: config.c:556 ast_config_register: Registered Config Engine odbc /// == Parsing '/etc/asterisk/extconfig.conf': Found /// == Binding iax.conf to odbc/asterisk/iaxfriends /// I've followed the example from the docs README.extconfig - but it doesnt pull the info from the database. Anyone have this working? |
| 04:10.57 | redder86 | looks like queues would do it |
| 04:11.51 | DubLo7 | Does anyone know if I could use a ~350K DSL line for VoIP? |
| 04:11.57 | DubLo7 | what kind of bandwidth is needed? |
| 04:12.21 | robl^ | around 70K or less per call |
| 04:12.22 | redder86 | DubLo7: yes, 14K minimum |
| 04:13.14 | DubLo7 | ok |
| 04:13.39 | DubLo7 | what do I need to get a regular PTSN line to call my system? |
| 04:13.55 | DubLo7 | I live in Northern Michigan and there is not much around me |
| 04:14.05 | DubLo7 | I do know most people who own the ISPs though |
| 04:14.12 | Moc | DubLo7, you need a FXO to asterisk |
| 04:14.38 | Moc | x100p, TDM400 + FXO module, a Sipura-3000, a T1 card + Channel bank with FXO module |
| 04:14.49 | Moc | those are options you have |
| 04:15.54 | DubLo7 | I'll have to look that up... I don't know what those are at the moment |
| 04:16.08 | Moc | google is your best freidn |
| 04:16.22 | DubLo7 | So I see... just wondering. |
| 04:16.23 | *** join/#asterisk DiveFox (~trillian@adsl-68-253-225-210.dsl.emhril.ameritech.net) |
| 04:16.42 | Moc | www.digium.com is the first place you should look at, they developped asterisk, and gave it to the community, so they sell hardware that work with it |
| 04:16.49 | DubLo7 | I have two phone lines + DSL and I'm trying to decide if I could save money going straight VoIP for two lines |
| 04:16.54 | *** join/#asterisk ZX81 (~ZX81@222-152-134-60.jetstream.xtra.co.nz) |
| 04:17.06 | redder86 | DubLo7: certainly |
| 04:17.10 | Moc | DubLo7, well problem with DSL is you need a line ;) |
| 04:17.12 | redder86 | DubLo7: you'll save money |
| 04:17.22 | Moc | DubLo7, how many minutes of call you make per month |
| 04:17.30 | redder86 | DubLo7: just by dropping your 2nd line |
| 04:17.51 | DubLo7 | not many minutes LD, mostly local |
| 04:17.55 | ZX81 | ok, so now I have changed over to T1 fxs in my house...but now when I try to dial 600 it hangs up because there is no extension 6...how do I make it wait for more digits? |
| 04:17.59 | Moc | well how many locals ? |
| 04:18.00 | DubLo7 | I use 2nd line for business only |
| 04:18.07 | Moc | 100min, 1000min, 5000min |
| 04:18.23 | redder86 | DubLo7: as long as you don't make more than 1000 minutes of calling on your 2nd line each month you'll probably be less expensive going VoIP. |
| 04:18.30 | ZX81 | I thought it was something to do with overlapdial but doesn't appear to be |
| 04:18.30 | DubLo7 | 1000 or so I would imagine |
| 04:18.54 | Moc | DubLo7, for 21$US, you have a 1800 with 1000minutes incoming on it |
| 04:18.55 | DubLo7 | how is VoIP billed? |
| 04:18.59 | DubLo7 | by the minute? |
| 04:19.07 | Moc | second |
| 04:19.09 | Moc | depend of the provider |
| 04:19.17 | redder86 | DubLo7: NuFone charges by the second |
| 04:19.18 | Moc | DubLo7, what area code are you ? |
| 04:19.23 | DubLo7 | 231 |
| 04:19.25 | *** join/#asterisk pfn (500@adsl-69-107-210-254.dsl.pltn13.pacbell.net) |
| 04:19.34 | DubLo7 | I would love to find service in my area |
| 04:19.42 | Moc | let me check if I do |
| 04:19.56 | redder86 | DiveFox: it's a nice basic phone, nothing fancy |
| 04:20.06 | Moc | well I would more direct you to someone who does |
| 04:20.07 | DiveFox | OK, decent sound quality? |
| 04:20.21 | redder86 | DiveFox: get the 102D, though, if you want alpha-text for callerid names |
| 04:20.26 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
| 04:20.27 | redder86 | DiveFox: sound is fine to me |
| 04:20.38 | Moc | sorry no 231 |
| 04:20.41 | DiveFox | OK, where do I get a 102D? I can't find prices for them online |
| 04:21.03 | Moc | 102D ? |
| 04:21.08 | DiveFox | And this is for a Pop-A-Lock franchise, I don't think callerID is important |
| 04:21.08 | redder86 | DiveFox: dunno, I bought a 101 from Pulver |
| 04:21.15 | DubLo7 | I'll talk with my friend who runs the ISP. He's always looking to make money. Maybe he could start VoIP in our area |
| 04:21.20 | DiveFox | Redder: What'd you pay for the 101 there? |
| 04:21.29 | redder86 | DiveFox: $75 after shipping |
| 04:21.33 | DiveFox | OK |
| 04:21.45 | redder86 | DiveFox: that was 6 mos ago |
| 04:21.47 | Moc | want a good cleap phone, get the polycom ip 300, it 120$ US |
| 04:22.01 | DiveFox | Moc: My customer needs it under 100 bucks. |
| 04:22.15 | Moc | DiveFox, too bad, polycom are really good phone |
| 04:22.25 | DiveFox | I know, I have polycom in the office here. |
| 04:22.41 | DiveFox | The lady's running a Pop-A-Lock franchise out of her house |
| 04:22.43 | redder86 | DiveFox: the 101 does not have a 2-port hub, so you'll need a separate hub/switch port for both the PC and the phone |
| 04:22.59 | Moc | DiveFox, get her a Sipura + a normal phone I guess |
| 04:23.09 | Moc | or a IAXy |
| 04:23.11 | DiveFox | Moc: Pricing, and link me |
| 04:23.22 | redder86 | IAXy is more than the BT101 |
| 04:23.26 | Moc | www.digium.com |
| 04:23.46 | DiveFox | I also need speakerphone, conference, and the ability to pick up any of the 4 lines they have coming in |
| 04:23.46 | redder86 | ... but if you need to punch through firewalls, IAXy is the way to go for an ATA device |
| 04:24.26 | DiveFox | It's all LAN to FXOs on the Asterisk server |
| 04:24.30 | redder86 | DiveFox: the BT101 has speakerphone - conference and line support comes from Asterisk |
| 04:24.36 | DiveFox | OK |
| 04:24.41 | Moc | bt101 SUCK.. |
| 04:24.45 | Moc | it a cheapo phone.. |
| 04:24.55 | Moc | you need to have that in mind when you buy it |
| 04:24.57 | redder86 | Moc: I'm okay with mine except for the missing alphatext on caller id |
| 04:25.00 | DiveFox | She has AT&T trimline phones now |
| 04:25.27 | Moc | well mine is under my bed.. |
| 04:25.30 | redder86 | Granted, BT101 are the cheap-o phones, but they're not bad |
| 04:25.31 | DiveFox | They can't sound worse than that |
| 04:25.34 | Moc | unplug and taking dust |
| 04:25.42 | DiveFox | Moc: Wanna sell it? |
| 04:26.23 | DiveFox | I can't afford to lose this customer to someone who's gonna underbid me for shit work |
| 04:26.29 | Moc | I keep it to show people how really cheap they are when they want to buy it.. |
| 04:26.35 | DiveFox | Heh |
| 04:26.55 | Moc | for a kid room, it ok |
| 04:26.56 | DiveFox | So what would you recommend for a $65-$85 price point? |
| 04:27.03 | syslod | Anyone have a working 410 3.3v Quad T1 card? |
| 04:27.24 | Marlow | syslod : whats the problem .. |
| 04:27.43 | Mavvie | syslod: that's a 405, but I have one |
| 04:27.55 | Moc | the best at lowerest IP phone is the polycom ip300 at 120$... under that, you could get a Sipura-1000 maybe or a 2000 and put a analog cordless phone |
| 04:28.03 | Mavvie | oh sorry. |
| 04:28.05 | Moc | so you can also add a cheap headphone too |
| 04:28.06 | Mavvie | 410 is 3.3 |
| 04:28.13 | syslod | :) I can't get the driver to load consistantly. Some IRQ APIC problem within the driver or kernel. |
| 04:28.14 | *** join/#asterisk ChulJin (chuljin@24-205-55-37.gln-eres.charterpipeline.net) |
| 04:28.15 | DiveFox | Moc: The customer has specced desk phones. |
| 04:28.35 | DiveFox | Gave me that price point, said if I can't hit it she'll have her brother get some off ebay |
| 04:28.40 | Mavvie | syslod: which one doesn't load? zaptel or wct4xxp ? |
| 04:28.42 | syslod | I've been working on it for about 10 hours straight now with no results |
| 04:29.13 | Marlow | syslod : sounds odd .. is the card sharing interrupts with something else ? |
| 04:29.27 | syslod | well they both load. Sometimes wct4xxp shows unused and I get a IRQ conflict error. Sometimes it works but after a reboot it goes screwy again. |
| 04:29.32 | Moc | you do what you want. But dont except much from the BT101... |
| 04:29.47 | DiveFox | I want to know if it will always ring and not fall apart. |
| 04:29.49 | Moc | it work, sound aint that bad... but... |
| 04:30.03 | Moc | they finally added ringtone to the phone, witch help it alittle bit |
| 04:30.04 | syslod | Sometimes it loads shared sometimes not. I have a tyan board. It doesn't seem have a way to assign a IRQ to a slot. |
| 04:30.05 | DiveFox | Reliability, not features, is key here. |
| 04:30.16 | redder86 | free soft-phone + $15 headset = $15 |
| 04:30.23 | Moc | well budgetone sometime freeze |
| 04:30.31 | redder86 | my BT101 never freezes |
| 04:30.31 | DiveFox | Redder: Doing that on the main phone |
| 04:30.45 | Moc | I had to unplug it every week |
| 04:30.56 | DiveFox | Old firmware, perhaps? |
| 04:31.03 | redder86 | Moc: I've not had one problem with it. I did upgrade firmware first thing, though. |
| 04:31.04 | Mavvie | wonder if this is the one: 02:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) |
| 04:31.13 | Moc | same here |
| 04:31.22 | redder86 | Mavvie: that's a Digium card, yup |
| 04:31.36 | Moc | look, make a opinion for yourself and get it.. |
| 04:31.36 | Mavvie | aha: it has IRQ 233 here..... |
| 04:31.36 | ManxPower | Mavvie: Must be a TE* card. |
| 04:31.38 | Mavvie | Flags: bus master, medium devsel, latency 32, IRQ 233 |
| 04:31.56 | fearnor- | its not T400P? |
| 04:31.56 | ManxPower | Mavvie: cat /proc/interrupts and see if they match |
| 04:31.57 | syslod | flags for boot loader? |
| 04:31.58 | Mavvie | this is too weird, my life with hardware ended when there were 15 intererupts. |
| 04:32.01 | redder86 | IRQ 233? |
| 04:32.04 | Moc | I bought TDM card, budgetone, sipura-2000 sipura-3000, cisco 7960 and polycom ip 500 |
| 04:32.09 | Mavvie | 233: 36705901 36739678 IO-APIC-level t4xxp |
| 04:32.11 | redder86 | APIC will get you more IRQ |
| 04:32.12 | Mavvie | yay! |
| 04:32.12 | Moc | for fun mainly |
| 04:32.16 | ManxPower | Wow! |
| 04:32.25 | Moc | and polycom is really my best toy |
| 04:32.31 | *** join/#asterisk Styxfan (irc@vp210.dds01.sea.blarg.net) |
| 04:32.34 | syslod | I think APIC is what is screwing up. Looks like it is getting the wrong IRQ. |
| 04:32.35 | Moc | im looking to get the ip 600 now |
| 04:32.41 | ManxPower | Moc: You're sticking with good hardware, I see. |
| 04:32.42 | syslod | Without APIC I can't share. |
| 04:32.47 | DiveFox | syslod: Try a different slot? |
| 04:33.01 | Mavvie | syslod: I have it at the first slot (next to the CPU) |
| 04:33.05 | syslod | Tried different slot and another motherboard with similar results. |
| 04:33.06 | redder86 | APIC always seems to mess stuff up somehow. |
| 04:33.19 | Moc | snom phone might be of interess (the console) |
| 04:33.21 | ZX81 | what is an urgent handler message? |
| 04:33.30 | redder86 | Moc: I've been happy with a Snom 190 |
| 04:33.31 | syslod | Not sure but either the driver or APIC in kernel is messing up. |
| 04:33.33 | ManxPower | ZX81: I don't know, but it's normal. |
| 04:33.38 | ZX81 | ok cool |
| 04:33.42 | ZX81 | ta |
| 04:33.47 | ZX81 | i got lots |
| 04:33.48 | DiveFox | redder or moc: Can you recommend a good (free) Win32 softphone? |
| 04:33.57 | Moc | DiveFox, x-ten |
| 04:33.59 | redder86 | DiveFox: X-ten |
| 04:34.00 | Moc | x-free |
| 04:34.05 | DiveFox | link me |
| 04:34.08 | Mavvie | well, that's unanimous :-) |
| 04:34.11 | Moc | GOOGLE !!! |
| 04:34.12 | redder86 | xten.com |
| 04:34.16 | DiveFox | I'm banned from google |
| 04:34.16 | mikegrb | I played with firefly for the first time last night |
| 04:34.22 | mikegrb | I was amazingly impressed |
| 04:34.26 | Mavvie | banned from google.... what happened? |
| 04:34.27 | ZX81 | DiveFox: lol |
| 04:34.31 | redder86 | firefly is okay, but... |
| 04:34.34 | mikegrb | definatly prefer it's stability to xten's |
| 04:34.40 | mikegrb | doesn't look as professional though |
| 04:34.40 | Moc | DiveFox, altavista, yahoo, MSN.. whatever.. www.voip-info.org |
| 04:34.41 | DubLo7 | anyone have experience with Packet8.net? |
| 04:34.44 | ZX81 | xten has memory leaks |
| 04:34.49 | DiveFox | Also, is the asterisk package that comes with debian testing a decent build? |
| 04:34.52 | mikegrb | that doesn't suprise me |
| 04:34.56 | redder86 | I'm waiting anxiously for the MWI for the xten soft phone. |
| 04:35.04 | mikegrb | just know it is hellof unstable on mac and win32 |
| 04:35.14 | Moc | DiveFox, get stable tar, or head CVS |
| 04:35.17 | redder86 | xten seems okay for me on Win98 |
| 04:35.29 | ZX81 | I;m waiting patiently for xten soft phones to allow calls over an hour without using 256mb of ram |
| 04:35.47 | mikegrb | now that I think of it maybe I haven't had too many problems with it on win32 |
| 04:35.51 | redder86 | ZX81: well, I don't use mine that much, admittedly. |
| 04:35.55 | mikegrb | but lots of crashes on os x |
| 04:35.57 | Marlow | DiveFox : what do you call a decent build ? |
| 04:36.00 | mikegrb | I'd love a firefly for osx |
| 04:36.13 | DiveFox | Marlow: Stable and mostly feature-complete |
| 04:36.15 | ZX81 | lol it was just a problem for the astricon conference call... |
| 04:36.17 | ZX81 | :-) |
| 04:36.18 | Moc | ManxPower, I mean, I thought cisco was a good phone... but damn polycom really BEAT THEM... xml config is so advanced |
| 04:36.19 | DubLo7 | Are they a good VoIP company? It looks like packet8.net may have service in my area. |
| 04:36.19 | redder86 | mostly I use my soft-phone for caller-id notifications since my BT101 doesn't do alpha-text |
| 04:36.33 | Marlow | DiveFox : it's currently at 1.0.1 |
| 04:36.42 | DiveFox | *goes to see what the build in debian testing is* |
| 04:36.47 | Moc | DubLo7, outgoing, nufone is excellent |
| 04:36.50 | Marlow | DiveFox : and it's just the default. |
| 04:36.59 | ChulJin | MOC! |
| 04:37.00 | ChulJin | :) |
| 04:37.01 | Marlow | DiveFox : i've got a bit more in mine .. |
| 04:37.05 | Moc | DubLo7, inbound, you need to search, I know I did.. I found 4 so far |
| 04:37.08 | mikegrb | anyone use one of the ezeeeeeephone net phones? |
| 04:37.09 | Moc | hey ChulJin |
| 04:37.29 | DiveFox | Marlow: I just need it to drive 4 FXOs, and 3 SIP phones |
| 04:37.30 | DubLo7 | Moc: ok, I'll keep it up. What keywords should I use? |
| 04:37.49 | flebb | anyone have res_config working via this example table list. http://www.mail-archive.com/asterisk-users@lists.digium.com/msg55846.html |
| 04:38.06 | redder86 | anyone here use agents.conf ? |
| 04:38.08 | Moc | DubLo7, DID ;) |
| 04:38.16 | Marlow | DiveFox : debian testing is std. 1.0.1 |
| 04:38.30 | DiveFox | OK |
| 04:38.36 | DiveFox | So should work OK for a small setup? |
| 04:38.41 | Moc | DubLo7, I got one of the biggest list of DID available |
| 04:38.48 | Marlow | DiveFox : sure |
| 04:38.55 | Moc | and your aint there, so maybe you will need to find a local that does voio |
| 04:41.05 | DubLo7 | Thanks. BTW, what do you do? How do you know so damn much? :o) |
| 04:41.05 | *** part/#asterisk DiveFox (~trillian@adsl-68-253-225-210.dsl.emhril.ameritech.net) |
| 04:41.05 | *** part/#asterisk Styxfan (irc@vp210.dds01.sea.blarg.net) |
| 04:42.15 | Marlow | flebb : what do you mean by that example ? |
| 04:42.34 | Moc | DubLo7, nearly everything... I like to touch at everything |
| 04:42.56 | Moc | and voip is damn fun to play with |
| 04:43.11 | Mavvie | if it works.... |
| 04:43.32 | jsharp | It turns you into a babbling idiot if it doesn't. |
| 04:43.45 | DubLo7 | Skype works for me and my previous company of about 100 people had packet phones, but that's my level right now. |
| 04:44.00 | DubLo7 | I'm a programmer for 10 years though, just no Telco experience |
| 04:44.13 | flebb | Marlow: www.bkw.org/load.txt |
| 04:44.27 | Moc | Ive programmed for .. 15 year now.. and Im 22 ;) |
| 04:44.39 | Marlow | flebb : that's just a perl tool to import your config .. |
| 04:44.44 | flebb | Marlow: no mater what I do, I cant get res_config to pull the data out |
| 04:45.06 | Marlow | flebb : have you configured everything ? |
| 04:45.10 | Moc | well if you count basic as programming |
| 04:45.15 | Marlow | flebb : like it is on the wiki ? |
| 04:45.19 | DubLo7 | lol |
| 04:45.34 | DubLo7 | I suppose I've got 17 if you count basic |
| 04:45.56 | flebb | Marlow: I know, I'm using var_name, category, var_metric, var_val, filename, cat_metric and in extconfig i have iax.conf => odbc,asterisk,iax |
| 04:45.56 | Moc | of not.. I started at 11 to program in C |
| 04:46.23 | Moc | but im not a programmer really |
| 04:46.33 | Moc | I just know what I need to get what I want to do |
| 04:46.40 | ManxPower | MCI is trying to take over our MIS operations. I need links to recent articles on the web about how terrible they are. |
| 04:46.40 | DubLo7 | well this month it sucks, but I like it |
| 04:47.04 | DubLo7 | no projects right now and new baby means no time to work and great need... |
| 04:47.12 | syslod | Aren't MCI,AT&T, and QWEST doomed???? |
| 04:47.15 | Moc | DubLo7, * is great because you got access to the source, so you can build applications |
| 04:47.24 | Marlow | flebb: that would be "iax.conf => odbc" |
| 04:47.24 | Moc | and add features |
| 04:47.29 | Marlow | flebb: nothing else |
| 04:47.35 | Moc | that get to be shared by everyone |
| 04:48.03 | DubLo7 | Moc, I was thinking of that. Integrating it into a CRM I was building. Not sure about GPL though. |
| 04:48.32 | jsharp | syslod: Doomed? Nah. They'll just file chap 11 bankruptcy, then give the executive team a fat bonus for saving the company millions. |
| 04:48.41 | Moc | DubLo7, well people are affraid of the gpl when they should not be |
| 04:49.08 | Moc | it like your best friend ;) |
| 04:49.12 | syslod | Doomed in the market maybe not as a company. I know here I have about 99% of there old customer base. |
| 04:49.50 | syslod | It'll take them awhile to reinvent themselves. |
| 04:50.05 | DubLo7 | I don't mind it, I'm Linux Certified etc, but as a coder I'm giving away all my work in the hopes of support contracts, or maybe good will. Where as before I could sell and get money direct |
| 04:50.08 | mikegrb | DubLo7: check out asternic.org, the flash operator panel has web based crm integration |
| 04:50.14 | DubLo7 | it just feels risky |
| 04:50.36 | Moc | DubLo7, well you sell a solutions |
| 04:51.05 | *** join/#asterisk linagee (~linagee@netblock-66-245-227-49.dslextreme.com) |
| 04:51.36 | Moc | the thing you need to think about is.. if someone else start it as GPL, it will eventually overrun yours |
| 04:51.52 | flebb | Marlow: if I put it like that what table is it going to look for that in |
| 04:52.00 | Moc | so far, gpl software is slowly overtaking everything |
| 04:52.15 | Nugget | yeah, it's a damn shame. |
| 04:52.29 | Nugget | monocultures are never good |
| 04:52.51 | Moc | Nugget, well the things is that custures is OPEN ;) |
| 04:52.57 | Nugget | GPL isn't open. |
| 04:53.02 | Nugget | it's proprietary |
| 04:53.13 | linagee | are calls dropped when you do a reload? |
| 04:53.18 | jsharp | No |
| 04:53.26 | DubLo7 | Now as someone who makes programs that's what I find disturbing. |
| 04:53.36 | Moc | you can do whatever you wish, as long you give the code |
| 04:53.47 | DubLo7 | If I make something great, but someone else takes it they don't need me |
| 04:53.52 | Nugget | You can do whatever you wish, as long as what you wish is to use the GPL. |
| 04:53.53 | DubLo7 | how do I get $$? |
| 04:53.55 | Nugget | there's a difference. |
| 04:54.03 | Nugget | it's the Henry Ford license. :) |
| 04:54.15 | Marlow | flebb : let me check |
| 04:54.21 | jsharp | The Ebola license. It infects everything it touches. |
| 04:54.51 | Nugget | From the perspective of a coder who prefers to use a license which is more free than the GPL, the GPL is every bit as closed and proprietary as Microsoft's code. |
| 04:54.56 | Corydon76-home | The infectious freedom license |
| 04:54.57 | Moc | DubLo7, well if someone take it to do something else, they have to give you back what they changed |
| 04:55.00 | linagee | DubLo7: what do you mean $$? programmers aren't supposed to make money, just code. :) |
| 04:55.08 | Marlow | flebb: table is defined in res_config_odbc.conf |
| 04:55.09 | Moc | so you gain from their developpement too |
| 04:55.14 | tessier_ | Money -> Programmer -> Code |
| 04:55.29 | Nugget | no. |
| 04:55.38 | Nugget | Caffeine -> Programmer -> Code :) |
| 04:55.40 | Moc | DubLo7, if it were BSD, then your screwed ;) |
| 04:55.42 | tessier_ | GPL as closed and proprietary as MS? Puh-eez. |
| 04:55.45 | Nugget | it is. |
| 04:55.52 | linagee | Nugget: no |
| 04:55.52 | nestAr | :o |
| 04:55.54 | Nugget | GPL code is as untouchable to me as is Microsoft's code. |
| 04:55.54 | tessier_ | You are giving BSD a bad name. |
| 04:56.00 | DubLo7 | :o) Yeah, but now I've got kids to feed. Need more then just stale Pizza and beer now |
| 04:56.06 | linagee | Nugget: Money -> Caffeiene -> Programmer -> Code. :) |
| 04:56.09 | Nugget | hee |
| 04:56.16 | linagee | Nugget: the caffeine costs money. :) |
| 04:56.24 | flebb | Marlow: now I'm totaly confused - so in res_config_odbc.conf i can put something like iax.conf => odbc,asterisk,iax? |
| 04:56.31 | Moc | tessier, well, it a good and bad liscence.. But it work for people |
| 04:56.32 | Corydon76-home | Nobody ever said that you can't charge money for GPL software |
| 04:56.38 | Marlow | flebb : no .. |
| 04:56.48 | tessier_ | Corydon-w: Indeed. The FSF charges money for GPL software. |
| 04:56.48 | Nugget | Corydon76-home: except economics cause GPL licensed code to be worth $0. |
| 04:56.55 | Nugget | so you can sell it once. |
| 04:56.56 | Marlow | flebb : res_config_odbc.conf contains: |
| 04:57.07 | Corydon76-home | Nugget: funny, we sell it multiple times |
| 04:57.08 | syslod | Anyone know where to get source code for NewT used by zttool? |
| 04:57.14 | Marlow | flebb : [settings] |
| 04:57.17 | tessier_ | She turned me into a NewT! |
| 04:57.24 | Marlow | flebb : table = ast_config |
| 04:57.27 | tessier_ | (I got better.) |
| 04:57.28 | Nugget | all you can sell is a promise to keep working, which isn't the same thing as selling the code. |
| 04:57.29 | Moc | DubLo7, 1 thing like digium does is dual liscencing. digium can release * under what ever liscence he want, but there is also GPL |
| 04:57.30 | Corydon76-home | Nugget: note that you're assuming a perfect marketplace, which does not exist. |
| 04:57.36 | Marlow | flebb : connection = mysql |
| 04:57.36 | ManxPower | syslod: http://www.google.com/search?q=slackware+newt&btnG=Search&hl=en&lr= |
| 04:57.38 | Nugget | or you can sell an ugly EULA like RedHat does. |
| 04:57.42 | Marlow | flebb : these 3 lines |
| 04:57.47 | Marlow | flebb : that's it |
| 04:58.06 | ManxPower | newt seems to come with both Slackware 9.2 and 8.1 |
| 04:58.19 | ManxPower | sorry, 9.1 and 8.1 |
| 04:58.25 | DubLo7 | So I start it, sell it, give away source GPL... but count on the fact that not everyone can turn source to programs. Some people would rather just pay for binary and/or support. Others will have to release their fixes to me |
| 04:58.27 | Marlow | flebb : res_odbc.conf has your database, user, password etc. |
| 04:58.27 | Moc | DubLo7, it depend of what you want to do, and where you want to bring what your doing |
| 04:58.28 | Nugget | I think that dual licensing is a good approach. |
| 04:58.49 | Moc | DubLo7, buisness want SUPPORT.. |
| 04:58.52 | Marlow | flebb : and extconfig.conf just tells, which config files are in odbc |
| 04:58.54 | Moc | that what they wish to pay for |
| 04:59.01 | Marlow | flebb : check the wiki .. it's all there |
| 04:59.35 | Moc | * in a buisness without support after installation, aint a solution |
| 04:59.52 | Corydon76-home | Yep |
| 05:00.29 | jsharp | They all want that mythical "Vendor" to call when stuff breaks. |
| 05:00.36 | Moc | yep |
| 05:00.40 | DubLo7 | Moc, so lets say I get my friends who own ISPs to startup VoIP services... They do the hardware / server side, I could setup * PBX and they call me to set everything up |
| 05:00.53 | Corydon76-home | And we're more than happy to take their money and support them... |
| 05:01.07 | jsharp | DubLo7: Yup. |
| 05:01.14 | *** join/#asterisk adamb` (~adam@69.17.96.176) |
| 05:01.25 | *** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net) |
| 05:01.28 | *** join/#asterisk usam (~usam@203.147.59.180) |
| 05:01.30 | DubLo7 | Well your DID list... whatever that is says no-one in 231 area. What is needed and what does DID mean? |
| 05:01.47 | PatrickDK | search did |
| 05:01.51 | linagee | why is my windows laptop suckiong so bad right now? :( |
| 05:01.56 | Corydon76-home | direct inward dial |
| 05:01.59 | Moc | DID direct inward... dial |
| 05:02.06 | tessier_ | linagee: The problem is obvious... |
| 05:02.11 | jsharp | linagee: You've got a virus. Windows. |
| 05:02.15 | Moc | but yes GOOGLE is your FRIEND ;) |
| 05:02.19 | DubLo7 | I searched DID get tons of "I Did it" type links |
| 05:02.20 | tessier_ | And I suspect that it has in fact sucked all along. |
| 05:02.27 | usam | ManxPower: ok i have done the measure of my line interface. |
| 05:02.29 | Moc | whatis.com |
| 05:02.31 | DubLo7 | Direct inward dial... that helps |
| 05:02.51 | linagee | tessier_: no, i mean it's going super slow. i can't even read what i'm typing |
| 05:03.02 | jsharp | reboot! |
| 05:03.06 | tessier_ | linagee: You must be behind on a service pack or something. |
| 05:03.06 | Corydon76-home | DID is the extension sent by the telco... it usually is the last several digits of the number dialled |
| 05:03.27 | Corydon76-home | How many digits depends upon the setup |
| 05:03.36 | usam | idle line: +26V, uner conversation: -8V, after hangup, +26V ,,, would this trigger the X100P to ensure the like is hungup? |
| 05:03.46 | Corydon76-home | I've seen anywhere from 4 to 17 digit DIDs |
| 05:04.17 | usam | s/like/line |
| 05:04.34 | adamb` | hey, is there a way to execute an external script or anything with asterisk? i wanna send the callerid info to a third party program after a call gets auto-answered.. |
| 05:04.43 | jsharp | adamb: AGI. |
| 05:04.52 | Corydon76-home | adamb`: show application system |
| 05:04.59 | adamb` | thanks :-) |
| 05:05.56 | Moc | 17 ? |
| 05:06.06 | Corydon76-home | Yeah, 17 |
| 05:06.09 | adamb` | oh man, PERFECT. |
| 05:06.15 | adamb` | thanks Corydon76-home :-) |
| 05:06.21 | Corydon76-home | ANI + last-7 of tollfree number |
| 05:07.05 | ChulJin | corydon: I think you're thinking of DNIS :P |
| 05:07.21 | Corydon76-home | adamb`: don't forget to try 'show application describing word1 word2 ... ' |
| 05:07.37 | DubLo7 | hmm, so for a full in home 2 line VoIP system. Looks like I would need a regular local line, DSL, VoIP FXO card, computer to run it... |
| 05:07.48 | Corydon76-home | ChulJin: essentially the same thing |
| 05:07.52 | usam | asteisk alternative: http://wandel.ca/homepage/pbx.html |
| 05:07.53 | DubLo7 | sound right? |
| 05:08.12 | *** join/#asterisk rowter (~Drake@201.135.5.90) |
| 05:08.16 | DubLo7 | oh yeah special phones too |
| 05:08.22 | Moc | usam, ouch hehe |
| 05:09.03 | usam | Moc: ;) |
| 05:09.20 | Moc | but the guy got a very nice project there |
| 05:10.25 | Moc | he could probably interface his thing with Asterisk ;) |
| 05:11.42 | linagee | DubLo7: you don't have to have a local line. :) |
| 05:13.15 | DubLo7 | <PROTECTED> |
| 05:13.28 | DubLo7 | I don't get that. |
| 05:13.42 | linagee | DubLo7: use the internet and a voip provider. :) |
| 05:13.57 | Moc | linagee, he might have none in his area |
| 05:14.20 | DubLo7 | Doesn't internet go over a local phone company line? I've never heard of internet w/o phone |
| 05:14.25 | DubLo7 | My town is about 5,000 people |
| 05:14.28 | jsharp | Cable modem? |
| 05:14.28 | linagee | Moc: use a voip provider in a different area. hehehe. of course everyone would have to call you long distance... |
| 05:14.30 | DubLo7 | Petoskey, MI |
| 05:14.57 | Moc | DubLo7, get a 1800 number |
| 05:15.05 | DubLo7 | that's an idea |
| 05:15.20 | DubLo7 | no broadband cable here |
| 05:15.24 | DubLo7 | not enough people on my road |
| 05:15.34 | tessier_ | well, 800 numbers aren't free. Someone has to pay for it. |
| 05:15.34 | jsharp | Wireless? |
| 05:15.40 | linagee | Moc: how is 1800 international number different from a 1800 national number? |
| 05:15.50 | DubLo7 | lol, nope... I live in a valley over the river and through the woods |
| 05:16.07 | linagee | DubLo7: satellite. that'll sound good over voip. :) |
| 05:16.31 | Moc | linagee, it a question of routing restriction... kind of piss me off.. |
| 05:16.34 | DubLo7 | I talked someone into doing DSL, but technically I'm outside of range |
| 05:16.43 | DubLo7 | I get about 60% of what I pay for |
| 05:17.04 | jsharp | Satellite wouldn't be bad. it would be a bit latent, but the latency would be consistent. |
| 05:17.07 | file | who wants to help me test something? |
| 05:18.20 | adamb` | hmm, my calls seem to be breaking up a bit on my side, i've tried calls with about 5 people and noone has had any issues hearing me, it's on my side only, it seems kind of "jittery", as in, people will say a word and it will maybe repeat that word two or three times in total, also words will just get chopped up like poor cell phone transmissions... This is extra confusing as my dsl is 6mbit down and 768k up |
| 05:18.31 | adamb` | any thoughts? |
| 05:18.35 | ChulJin | small world again. |
| 05:18.54 | ChulJin | dublo: my paternal grandmother's family is from petoskey |
| 05:19.25 | linagee | Moc: kind of piss you off? |
| 05:19.31 | DubLo7 | cool |
| 05:19.36 | DubLo7 | I'm semi-local |
| 05:19.42 | DubLo7 | from Mackinac Island oriiginally |
| 05:19.57 | ManxPower | Asterisk seems to attract people from MI |
| 05:19.59 | *** join/#asterisk naturalvoice (joao@node-40247a6a.ewr.onnet.us.uu.net) |
| 05:20.10 | Moc | because im in canada, and most voip provider of 1800 can't route to canada |
| 05:20.10 | DubLo7 | ChuJin, probably a road named after you then |
| 05:20.14 | ChulJin | haha |
| 05:20.16 | ChulJin | hmm |
| 05:20.19 | ChulJin | I suppose |
| 05:20.21 | linagee | Moc: ah |
| 05:20.32 | ChulJin | except I remember visiting petoskey when I was a kid |
| 05:20.39 | ChulJin | and there are like 5 roads. |
| 05:20.49 | ManxPower | <-- originally from Holland, MI |
| 05:20.55 | ManxPower | I left as soon as I could, of course. |
| 05:21.20 | ChulJin | <originally from laporte, in, just across the border from michigan |
| 05:21.24 | linagee | adamb`: are you using QoS? |
| 05:21.51 | *** join/#asterisk AgiNamu (~zzzs@4.79.150.34) |
| 05:21.53 | AgiNamu | Hey |
| 05:22.00 | linagee | ChulJin: what is that? |
| 05:22.03 | AgiNamu | Does anyone remember the name of the DnD wizard spell to stop time? was it Time Stop? |
| 05:22.18 | adamb` | linagee: yeah i setup qos on sip and iax2 |
| 05:22.26 | ChulJin | it's a hamburger restaurant in extreme southwest michigan, just across the border from where I grew up in IN |
| 05:22.30 | adamb` | im using voicepulse |
| 05:22.33 | ChulJin | still the best burgers I've ever had. |
| 05:23.51 | linagee | adamb`: did you try it with no other data going over your DSL? |
| 05:24.19 | DubLo7 | gotta go to bed |
| 05:24.21 | DubLo7 | night all |
| 05:24.30 | DubLo7 | thanks for info |
| 05:24.32 | *** part/#asterisk DubLo7 (~Owner@dsl2.ptskmi.racc2000.gaslightmedia.net) |
| 05:24.53 | adamb` | linagee: yeah, im not doing anything right now other than aim and a couple ssh sessions |
| 05:25.11 | adamb` | weird thing is it sounds clear on the remote (pots) ends |
| 05:25.18 | adamb` | even when the remote end is a cell phone etc |
| 05:25.26 | adamb` | the choppiness is only on my side |
| 05:26.03 | linagee | adamb`: have you tried different voip clients? is your asterisk server powerful enough? |
| 05:29.01 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
| 05:29.03 | Moc | * support sip presence ? |
| 05:30.58 | linagee | i don't get it. why can SER support so much more call volume? reading docs, a dual CPU running SER powering a city of 7.2 million at peak hour??? |
| 05:31.28 | syslod | Anyone seen this? |
| 05:31.31 | syslod | Oct 30 01:33:34 WARNING[16384]: chan_zap.c:757 zt_open: Unable to open '/dev/zap |
| 05:31.31 | syslod | /channel': No such file or directory |
| 05:31.31 | syslod | Oct 30 01:33:34 ERROR[16384]: chan_zap.c:6195 mkintf: Unable to open channel 1: |
| 05:31.31 | syslod | No such file or directory |
| 05:31.31 | syslod | here = 0, tmp->channel = 1, channel = 1 |
| 05:31.32 | syslod | Oct 30 01:33:34 ERROR[16384]: |
| 05:32.21 | naturalvoice | Onyone knows how to put * working with caller-ID in Brazil ? |
| 05:33.20 | AgiNamu | does SER just redirect? |
| 05:33.20 | jsharp | syslod: Do you have zaptel cards? did you modprobe them correct? |
| 05:33.34 | sleepy_one | gnite y'all |
| 05:34.02 | syslod | I have zaptel and I think it finally worked. I see that it registered the device and created things in /dev |
| 05:34.24 | AgiNamu | linagee -- Can't asterisk handle tons of registrations? |
| 05:34.44 | linagee | AgiNamu: simultaneous calls? not according to the wiki |
| 05:35.09 | AgiNamu | well, what are we counting? I think the wiki refers to people calling into the asterisk |
| 05:35.09 | AgiNamu | and then temrinating to TDM |
| 05:35.10 | linagee | AgiNamu: the wiki says like 30-120 calls. |
| 05:35.17 | AgiNamu | the max I've heard is 400 calls |
| 05:35.19 | AgiNamu | ULAW |
| 05:35.22 | AgiNamu | Dual Xeons |
| 05:35.31 | AgiNamu | 30-120 is if you're using compression |
| 05:35.35 | AgiNamu | and actually handling the audio |
| 05:35.41 | AgiNamu | SER doesnt do that. |
| 05:35.47 | AgiNamu | at all |
| 05:35.51 | AgiNamu | so it's not comparable. |
| 05:35.56 | syslod | jsharp: Looks like its creating the DEV files just not the ones I get errors on in *. |
| 05:36.08 | AgiNamu | SER just handles SIP itself |
| 05:36.27 | linagee | oic. so the codecs have to be the same or something? |
| 05:36.39 | linagee | or phone A has to support phone B's codecs |
| 05:36.54 | AgiNamu | Well, that's besides the point |
| 05:36.59 | AgiNamu | SER just connects them |
| 05:37.02 | AgiNamu | Asterisk can do that too |
| 05:37.12 | AgiNamu | and I'm sureit will hit MUCH more than 400 "calls" |
| 05:37.24 | linagee | so.. there really is no advantage to using SER then... |
| 05:37.27 | AgiNamu | But, if you actually TERMINATE the traffic with ASterisk, i.e., asterisk touches the audio |
| 05:37.30 | AgiNamu | then you're gonna pay |
| 05:37.37 | AgiNamu | well, it depends what you're trying to do :) |
| 05:37.55 | AgiNamu | some people use SER as the front end SIP proxy/registry whatever |
| 05:38.01 | AgiNamu | and Asterisk to do the heavy work |
| 05:38.01 | linagee | what if it doesn't have to transcode and the codecs are all the same? |
| 05:38.10 | AgiNamu | its not about that. |
| 05:38.15 | AgiNamu | If you call Phone A to Phone B using Asterisk |
| 05:38.17 | linagee | with asterisk |
| 05:38.22 | AgiNamu | and it SIP redirects/reinvites whatever |
| 05:38.32 | AgiNamu | then if you dont have matching codecs, you'll just fail. |
| 05:38.42 | AgiNamu | i.e., in that situation , Asterisk doesnt get involved. |
| 05:38.59 | linagee | oic |
| 05:39.03 | AgiNamu | If however, you connect to asterisk and asterisk is doing stuff with the audio, and bridging the calls, then you're using the CPU |
| 05:39.10 | linagee | that's what canreinvite or whatever is for? |
| 05:39.11 | AgiNamu | so as long as the audio stays clear |
| 05:39.25 | AgiNamu | yea, i think so. im not really that good on SIP yet. i need to go read the RFC. |
| 05:39.40 | linagee | i mean using asterisk, sip.conf |
| 05:39.51 | AgiNamu | yea, i dont know much about sip, even in asteirsk :) |
| 05:39.53 | linagee | there's some sort of canreinvite variable |
| 05:39.59 | linagee | ah |
| 05:40.02 | AgiNamu | but effectively saying SER is faster is saying "processing ASCII messages is faster than processing audio" :) |
| 05:40.14 | AgiNamu | linagee, but i think it's related to the reinvite thing |
| 05:40.30 | AgiNamu | someweek ill figure it all out |
| 05:40.38 | linagee | AgiNamu: i guess you can always use SIPp and find out. :) |
| 05:40.47 | linagee | (it's like a SIP benchmarking thing) |
| 05:40.48 | AgiNamu | yea i could |
| 05:40.54 | AgiNamu | i just want to actually LEARN what's going on |
| 05:41.18 | linagee | AgiNamu: go out and buy more voip books. :) |
| 05:41.27 | AgiNamu | well, I'll just read the RFC |
| 05:41.34 | AgiNamu | I bought the book on asteirsk |
| 05:41.38 | linagee | AgiNamu: better yet, just live in the bookstore for a few months :) |
| 05:41.39 | AgiNamu | and well, it leaves much to be desired. |
| 05:41.49 | AgiNamu | half the damn book is just copy and paste from the website |
| 05:41.52 | linagee | set up a little shelter in there. :) |
| 05:42.02 | AgiNamu | it even says stuff like "click here to read more" |
| 05:42.07 | linagee | LOL! |
| 05:42.10 | AgiNamu | yea ,well, im in guatemala right now |
| 05:42.17 | AgiNamu | so i dont know of any decent bookstores. |
| 05:42.21 | linagee | oic |
| 05:42.29 | AgiNamu | unless I want a 20 year old primer on ADA , in spanish :P |
| 05:42.45 | AgiNamu | anyways, back to ast_mono |
| 05:42.58 | AgiNamu | which is gonna be the bomb :) |
| 05:43.03 | AgiNamu | whenever we finish |
| 05:43.28 | AgiNamu | I'm trying to figure out how to patch asterisk |
| 05:43.30 | AgiNamu | while running |
| 05:43.43 | linagee | what's it for? using a monochrome screen w/ *? :) |
| 05:43.52 | AgiNamu | haha |
| 05:43.54 | AgiNamu | writing in .NET |
| 05:44.01 | AgiNamu | while retaining all the power of the C API |
| 05:44.14 | AgiNamu | and getting all the benefits of .NET (GC, remoting, etc. etc. etc.) |
| 05:44.41 | linagee | oic. *that* mono. :) |
| 05:45.03 | AgiNamu | :D |
| 05:45.35 | AgiNamu | http://www.atrevido.net/blog/CategoryView.aspx?category=ast_mono |
| 05:45.39 | *** join/#asterisk ST-3 (ser@dipsy.tch.org) |
| 05:53.12 | Wangster | anyone have any advice on setting up asterisk server behind NAT and accepting incoming SIP calls? |
| 05:53.17 | Wangster | is this possible? |
| 05:55.01 | PatrickDK | easily |
| 05:55.17 | PatrickDK | check out voip-info.org |
| 05:55.44 | Wangster | i have been... to no avail so far |
| 05:56.02 | Wangster | I don't get sound on the external side. |
| 05:56.16 | Wangster | er.. sorry, other way around. |
| 05:56.45 | Wangster | outside SIP call does not transmit sound to asterisk inside the NAT. |
| 05:57.13 | PatrickDK | you didn't portforward your rtp ports |
| 05:57.18 | PatrickDK | rtp = audio udp ports |
| 05:57.43 | Wangster | that is correct. I haven't |
| 05:57.50 | Wangster | because I can't seem to find out which ports they are. |
| 05:58.04 | Wangster | voip-wiki says something like "they aren't well defined" |
| 05:58.07 | PatrickDK | rtp.conf is a good place to try |
| 05:58.30 | Wangster | ohh.. |
| 05:58.41 | Wangster | thank you. Never heard of that file before. |
| 05:58.50 | Wangster | well.. never noticed it before. |
| 06:01.45 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 06:01.55 | *** join/#asterisk edguy3 (~edguy@host-24-225-213-218.patmedia.net) |
| 06:05.42 | Wangster | damnit. I've forwarded 10000-20000 for RTP but still nothing. |
| 06:05.56 | BoRIS | Hey Wangster! |
| 06:06.05 | Wangster | yo boris! |
| 06:06.24 | *** join/#asterisk WellMaluedo (~none@host104-131.pool8249.interbusiness.it) |
| 06:06.25 | BoRiS | Did you get enough sleep last night? |
| 06:06.46 | Wangster | So now that you are here, tell me why I can't do incoming SIP to my asterisk box behind a NAT?!!?! |
| 06:06.54 | Wangster | no.. i'm grumpy |
| 06:06.56 | Wangster | heh |
| 06:06.59 | BoRiS | lol |
| 06:07.35 | cyanoacry | has anyone had a problem with asterisk suddenly dropping out on them? |
| 06:07.54 | cyanoacry | like, you try and get your voicemail, but about half way through the menu, it just cuts out |
| 06:08.05 | Nugget | NAT is enough to make the best of us grumpy. |
| 06:08.41 | brc_ | http://www.anandtech.com/showimage.html?u=http://images.anandtech.com/reviews/video/nvidia/SLIpreview/twocardsinstalled.jpg |
| 06:09.34 | robl^ | NAT = weapons of mass destruction -- oh.. and this WMD really exists |
| 06:10.33 | *** join/#asterisk MyNick (~some@host-69-144-65-139.gdj-co.client.bresnan.net) |
| 06:12.38 | *** join/#asterisk Fpl (~Fpl@200.93.34.189) |
| 06:14.37 | adamb` | linagee: what do you mean clients? protocols? powerful enough in what way? |
| 06:14.38 | adamb` | sorry for the delayed response :-) |
| 06:15.16 | linagee | adamb`: i'm not sure why your audio would skip |
| 06:16.21 | *** join/#asterisk iway (~ariel@ariel-gw.wlcom.com.mx) |
| 06:16.27 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 06:16.53 | ManxPower | Wangster: A message posted to the mailing list less than an hour ago has sip.conf settings for Asterisk behind NAT. |
| 06:17.05 | ManxPower | He's having a different problem, his sip.conf is OK |
| 06:18.24 | Wangster | ManxPower: thanks, will look in the archive |
| 06:19.03 | ManxPower | Wangster: Subscribe to the mailing list. |
| 06:20.51 | Wangster | I may. but I'm already subscribed to so many lists.... I don't ever have time to read any of them so not much point. |
| 06:20.56 | *** join/#asterisk dapper_yapper (T@c-67-166-254-124.client.comcast.net) |
| 06:21.08 | dapper_yapper | Hey foks |
| 06:21.11 | dapper_yapper | folks |
| 06:21.13 | ManxPower | Wangster: *shrug* If you don't want to be serious about Asterisk.... |
| 06:21.23 | dapper_yapper | Question |
| 06:22.01 | dapper_yapper | I am ordering an outsourced dedicated server to put up another * box strictly for VOIP use |
| 06:22.42 | Nugget | I did that. |
| 06:22.46 | Nugget | I went with 800hosting.com |
| 06:22.51 | Nugget | a nice freebsd5 box. |
| 06:22.55 | dapper_yapper | Do you guys think better ping times to VOIP termination providers is more important than a better faster piece of hardware? |
| 06:23.19 | Nugget | I think that a good hosting provider is more important than either of those. |
| 06:23.22 | Moc | being closerto the voip term is always good |
| 06:23.25 | Nugget | there's some real crap out there. |
| 06:23.28 | dapper_yapper | I have my own bandwidth and machines on my T1 but I want to go faster in a datacenter environment |
| 06:23.43 | Moc | btw close for me = low ping, low hops, and high reliability |
| 06:23.43 | ManxPower | I wish VoIP companies would provide co-location |
| 06:23.59 | Nugget | that would be great, ManxPower. |
| 06:24.07 | ManxPower | Nugget: Or better yet CLECs |
| 06:24.12 | Nugget | WORKSFORM |
| 06:24.14 | Nugget | +E |
| 06:24.28 | ManxPower | co-locate an Asterisk box at a CLEC, put in a quad PRI card or two..... |
| 06:24.35 | dapper_yapper | Well here is the thing |
| 06:24.53 | ManxPower | Hmmm..I gotta ask our CLEC/ISP about that. |
| 06:24.59 | ChulJin | do PRI's typically have unlimited outgoing local calls? |
| 06:25.02 | dapper_yapper | <PROTECTED> |
| 06:25.17 | ChulJin | rocks? |
| 06:25.26 | ManxPower | ChulJin: Most of the time, not always. Ask the *LEC |
| 06:25.33 | Moc | is MSN down ? |
| 06:25.34 | dapper_yapper | Anyhow, the ping times to a few different VOIP providers are decent and prices are great |
| 06:25.35 | ManxPower | PRIs from IXCs never do |
| 06:25.40 | *** join/#asterisk jama (abdi@CPE-65-30-252-159.mn.rr.com) |
| 06:25.54 | dapper_yapper | I can get a pretty nice machine for cheap |
| 06:26.11 | ChulJin | oh! |
| 06:26.21 | ManxPower | Anyone know of co-lo companies in Atlanta or Houston? |
| 06:26.23 | ChulJin | see, you learn something new every day |
| 06:26.24 | dapper_yapper | Or I can pay a LOT more to go with ServerCentral where my ping times to VOIP termination are VERY good |
| 06:27.03 | ManxPower | I prefer locations where I can hop on a plane and be there in an hour if something horrid happens. |
| 06:27.08 | dapper_yapper | ChulJin ***** will have to remain a mystery but it does not = anything good |
| 06:27.14 | *** join/#asterisk habakuk (~chatzilla@adsl-64-172-34-40.dsl.snfc21.pacbell.net) |
| 06:27.19 | Moc | oh well, ennuf for me I guess.. |
| 06:27.28 | dapper_yapper | Ping times to Nufone are like 3 ms |
| 06:27.29 | dapper_yapper | :) |
| 06:27.44 | dapper_yapper | 30ms to another one I am considering |
| 06:27.44 | Moc | I have updated my patch, msn is down, * is running.. |
| 06:27.52 | dapper_yapper | 40 back to my T1 in Dallas |
| 06:28.25 | Moc | 3ms to nufone is REALLY good.. |
| 06:28.31 | dapper_yapper | Yah.. :) |
| 06:28.33 | BoRiS | 3ms is *very* good :) |
| 06:28.34 | Moc | you are like in the same city |
| 06:28.41 | dapper_yapper | Exactly |
| 06:28.55 | dapper_yapper | that's why I am having a hard time not going that route |
| 06:29.03 | Moc | ? |
| 06:29.08 | dapper_yapper | but I can't live with a Celeron |
| 06:29.22 | dapper_yapper | so I have to go with a $219/mo box |
| 06:29.27 | dapper_yapper | with Server Central |
| 06:29.34 | Moc | I got 21ms with nufone, 30 with my DID provider |
| 06:29.52 | dapper_yapper | But hell for $219 with The Planet I can get a dual Xeon |
| 06:29.55 | Moc | theplanet resell nice powerfull machine |
| 06:30.01 | Moc | yep |
| 06:30.12 | dapper_yapper | but with much higher ping times |
| 06:30.14 | dapper_yapper | So.. |
| 06:30.18 | dapper_yapper | Thoughts? |
| 06:30.23 | Moc | depend where.. |
| 06:30.32 | Moc | from Montreal Quebec to theplanet, yes |
| 06:30.40 | Moc | but maybe not that much with nufone still |
| 06:30.52 | Moc | let me check |
| 06:31.02 | dapper_yapper | If I was willing to spend the $219, you think I would get better performance out of the faster ping times with less of a machine or more of a machine with slower pings? |
| 06:31.33 | Moc | 30ms to nufone |
| 06:31.36 | dapper_yapper | Dual Xeon with say 40-60ms ping times |
| 06:31.59 | Moc | 26 packets transmitted, 26 received, 0% packet loss, time 26120ms |
| 06:31.59 | Moc | rtt min/avg/max/mdev = 30.106/30.312/30.653/0.175 ms, pipe 2 |
| 06:32.01 | dapper_yapper | vs. single processor 2.4 with 3-40 ms pings |
| 06:32.05 | Moc | from theplanet to nufone |
| 06:32.15 | Moc | well a 2.4 is still very good |
| 06:32.22 | Moc | depend on what you gonna do with it |
| 06:32.56 | ChulJin | I get quite acceptable performance with an old P3-500 :) |
| 06:33.11 | dapper_yapper | http://www.servercentral.net/support/traceroute.php |
| 06:33.11 | Moc | you have to think like this. Me --- 30 ---> you --30---> Nufone = 60ms |
| 06:33.31 | dapper_yapper | Yep, I am thinking like that |
| 06:33.46 | Moc | so if you can take 30ms off.. it better |
| 06:34.01 | dapper_yapper | I am trying to set myself up with the most central location to offer the best in performance from all areas of the country |
| 06:34.04 | Moc | well that within the same server room dapper_yapper |
| 06:34.08 | usam | any1 know a diagram to detect line reversal and generate busy tone ? |
| 06:34.20 | dapper_yapper | until we start putting more boxes up in other parts of the country |
| 06:34.30 | Moc | you might beable to interconnect with Nufone in TDMoE ;) |
| 06:35.31 | dapper_yapper | So it sounds like the better ping times are the way to go |
| 06:35.33 | dapper_yapper | ya? |
| 06:36.08 | dapper_yapper | Sorry, I was mistaken |
| 06:36.14 | dapper_yapper | it's not 3 ms to Nufone |
| 06:36.18 | dapper_yapper | it's less than one |
| 06:36.22 | dapper_yapper | <PROTECTED> |
| 06:36.22 | dapper_yapper | <PROTECTED> |
| 06:36.22 | dapper_yapper | <PROTECTED> |
| 06:36.36 | dapper_yapper | Sick.. :) |
| 06:36.49 | dapper_yapper | 1 hop between the two locations |
| 06:37.43 | dapper_yapper | Then like 37 back to my office in Dallas where a bunch of calls will originate from |
| 06:40.25 | sixTel | We're at SC too, sc is great. |
| 06:44.34 | Moc | damn microsoft doing windows update on his msn server .. |
| 06:46.17 | ChulJin | hmmm...does microsoft have a DUNDi node? :P |
| 06:46.57 | BoRiS | sixTel!! :-| |
| 06:50.33 | linagee | why is my [default] in extensions.conf not being inherited? (shouldn't it be?) |
| 06:50.51 | linagee | oic |
| 06:50.57 | linagee | include => "default" |
| 06:52.02 | linagee | (no quotes) |
| 06:53.06 | *** join/#asterisk PilotPTK-Home (~trillian@pcp02587377pcs.shlb1201.mi.comcast.net) |
| 06:54.31 | *** join/#asterisk tsetane (~tsetane@pppoecl68091.minlos.no) |
| 06:54.38 | *** join/#asterisk dj-ohki (~dj-ohki@nr24-66-42-158-191.fuse.net) |
| 06:56.42 | PilotPTK-Home | Anyone need a 4 Port T1 or E1 Card? VERY Special Late-Night #asterisk Pricing |
| 06:58.22 | *** join/#asterisk sprior (~sprior@dsl1.geekster.com) |
| 06:59.33 | sprior | I am very confused with the docs from asteriskdoc.org |
| 07:00.41 | sprior | anyone out there willing to explain a few things? |
| 07:00.55 | PilotPTK-Home | whats your questins sprior? |
| 07:01.58 | AgiNamu | damn |
| 07:01.59 | WellMaluedo | PilotPTK-Home: 4E1 Card? interesting |
| 07:02.04 | *** join/#asterisk dapper_yapper (T@c-67-166-254-124.client.comcast.net) |
| 07:02.14 | AgiNamu | after extensive research (heh), I've decided that patching asterisk at runtime is too much work |
| 07:02.21 | sprior | Thanks. I've installed my new TDM11B with FXS in slot 1 and FXO in slot 4 as shipped from Digium, and the modules are compiled so that modprobe ecfxs returns the right info. |
| 07:02.36 | AgiNamu | since appparently I'll need PLT Infection, and i have not the slightest clue on how to do tht. |
| 07:03.05 | sprior | I'm trying to follow the info in the asteriskdocs.org and don't understand where the [incoming] context comes from in their extensions.conf. |
| 07:04.18 | sprior | I also notice that they list the channels in zapata.conf as being 1 & 2, but shouldn't they be 1 & 4? |
| 07:07.57 | sprior | any clues? |
| 07:10.22 | sprior | you still here Pilot? |
| 07:18.49 | *** join/#asterisk mbranca (~matteo@ppp-217-133-231-93.cust-adsl.tiscali.it) |
| 07:20.03 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:25.54 | bkw_ | STUPID STUPID STUPID PEOPLE |
| 07:25.58 | bkw_ | STAY AWAY FROM ME |
| 07:26.11 | bkw_ | before I got crazy and pop someone's head off... |
| 07:26.14 | bkw_ | </rant> |
| 07:26.20 | bkw_ | ok all better now |
| 07:27.22 | BoRiS | lol |
| 07:27.24 | BoRiS | hey bkw :) |
| 07:29.36 | bkw_ | well shit the list is full of dumb fucks that think they know it all and would argue with a fence post if they had the chance |
| 07:29.58 | bkw_ | I have to fight back the urge to just get the clue-by-6 out and beat the fuck out of each and every one of them |
| 07:30.25 | bkw_ | Hell I may not know it all but I sure as hell don't open my mouth and remove all doubt. |
| 07:30.51 | ManxPower | bkw_: You're just burning out. Go do something else for a while. |
| 07:31.00 | bkw_ | nope not at all |
| 07:31.05 | ManxPower | Then come back. |
| 07:31.06 | bkw_ | I'm far from burned out |
| 07:31.08 | bkw_ | haha |
| 07:31.20 | bkw_ | I get sick of stupid people slowing down progress :P |
| 07:31.21 | bkw_ | thats all |
| 07:31.23 | ManxPower | Even a few days helps |
| 07:32.25 | bkw_ | na |
| 07:32.37 | bkw_ | i'm ok.. just giving my view of things :P |
| 07:32.49 | *** join/#asterisk without (~dean_davi@CPE-203-45-234-131.qld.bigpond.net.au) |
| 07:32.58 | bkw_ | I should really unsub from the users list |
| 07:33.52 | ManxPower | bkw_: I'd have to put you on /ignore then *tease* |
| 07:34.00 | bkw_ | haha |
| 07:34.03 | bkw_ | na |
| 07:34.24 | bkw_ | i'm a happier person without the users list |
| 07:34.28 | bkw_ | all I need is the CVS list |
| 07:34.36 | ManxPower | bkw_: Make a FAQ |
| 07:34.45 | ManxPower | That's what I did the first time I got Fed Up. |
| 07:35.03 | bkw_ | 1. Is your IQ larger than your Shoe Size? Yes Or No? |
| 07:35.03 | bkw_ | haha |
| 07:35.08 | bkw_ | just kidding |
| 07:35.08 | BoRiS | lol |
| 07:35.14 | bkw_ | I honestly will work on a faq next week |
| 07:35.20 | bkw_ | its late |
| 07:35.23 | bkw_ | I just watched mean girls |
| 07:35.24 | BoRiS | bkw, it does sound like you need to get off that user list. |
| 07:35.33 | BoRiS | I really liked that movie :() |
| 07:35.33 | ManxPower | bkw_: you are welcome to take my long abandonesd FAQ as a starting point. |
| 07:35.54 | bkw_ | BoRiS me too |
| 07:35.56 | bkw_ | it was great |
| 07:36.01 | ManxPower | It was mostly pulled from the channel logs and some from the mailing list. |
| 07:36.11 | without | any one here use a cisco 7960 with freshtel ?? |
| 07:36.21 | bkw_ | no |
| 07:36.23 | bkw_ | they are iax |
| 07:36.24 | bkw_ | NEXT!!! |
| 07:36.31 | bkw_ | bed time |
| 07:37.35 | BoRiS | nighty |
| 07:37.45 | without | my 7960 gets realy jitter with freshtel but my gs101 does not when connected through freshtel |
| 07:37.54 | without | night bkw |
| 07:38.24 | *** part/#asterisk sprior (~sprior@dsl1.geekster.com) |
| 07:39.31 | ManxPower | Wow! http://slashdot.org/pollBooth.pl?qid=1200§ion=mainpage&aid=1 Well at least it shows where the USA geeks are. |
| 07:43.07 | *** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net) |
| 07:49.36 | *** join/#asterisk serdiehard (~serdiehar@202.65.128.18) |
| 07:54.59 | *** join/#asterisk bofh42 (~bofh42@pD95ED3B3.dip.t-dialin.net) |
| 07:57.42 | *** join/#asterisk inspired (mikael@a217-118-63-4.bluecom.no) |
| 07:59.45 | serdiehard | iam uisng soxmix with monitor option but the problem |
| 08:00.09 | twisted | woot |
| 08:00.15 | twisted | my iPod ownz my ballz now |
| 08:00.22 | *** join/#asterisk shout (rcsw@modem-2345.gorilla.dialup.pol.co.uk) |
| 08:01.35 | serdiehard | it s not storing the audio |
| 08:02.15 | WilliamK | morning twist |
| 08:10.03 | *** join/#asterisk aspworld (~richard@209.91.159.221) |
| 08:10.15 | *** join/#asterisk speedwagon (~Ariel@fl-nked-ubr2-c6a-125.miamfl.adelphia.net) |
| 08:10.15 | *** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc) |
| 08:14.01 | Manipura | morning everyone |
| 08:16.15 | jsharp | Heya. |
| 08:16.26 | sarumont | morning |
| 08:17.22 | serdiehard | hi manipura |
| 08:17.37 | Manipura | Its 2 am, boy do I feel like a slave to the system |
| 08:17.54 | Manipura | system, meaning 'Computer" |
| 08:18.35 | Manipura | does this thing every become boring? |
